- sr-users - kamailio.org

[OpenSER-Users] Disconnect A SIP User, how ?
by Ali Jawad 30 Jan '08

30 Jan '08

Dear All I need a way to detect if a user is logged in and disconnect him if possible. A better solution would be to detect if he is in a call and disconnect him. My questions would be 1. How to detect if a user is connected ? 2. How to detect if a user is currently in a call ? 3. How to disconnect a user 4. How to stop the call for a user (If possible) 5. If dialog is needed is my install below sufficient ? I was told in the IRC channel with regard to Q 3 that I can use the dialog module to achieve that through sending bye using dlg_end_dlg mi command. I checked the documentation at http://www.openser.org/docs/modules/1.3.x/dialog.html#AEN578 but I do not have modparam("dialog" // and so on lines in my openser.cfg file I have installed 1.3 with MySQL, Jabber and Presence support using the following First I edited the MakeFile and deleted the modules I want from the exclude_modules entry, it showed up as follows exclude_modules?= cpl-c postgres osp unixodbc \ avp_radius auth_radius group_radius uri_radius xmpp \ pua pua_bla pua_mi \ pua_usrloc pua_xmpp rls mi_xmlrpc perl perlvdb \ ldap carrierroute h350 xcap_client db_berkeley seas NICER=1 make all make prefix=/usr/local install Does that mean that the dialog module is installed? I could not find a loadmodule "dialog" line!! With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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Re: [OpenSER-Users] CFG problem with 3rd party tool SOLVED(ish)
by Andy Smith 30 Jan '08

30 Jan '08

Hi, ok, I found and solved one issue. The acc radius_extra entries seem to have a maximum lenght that openser can parse so when I break the configuration into two "lines" like this: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat)") modparam("acc", "radius_extra", "User-Agent=$hdr(user-agent); \ Event=$hdr(event); \ Contact=$hdr(contact); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") #Billing-Party=$avp(billing_party); \ It all works, I hope OpenSER isnt ignoring the first entry in the cfg file!!!! :S. But its only working if I remove the entry "Billing-Party=$avp(billing_party)", if I change this to "Billing-Party=$avp(s:billing_party)", then all starts fine. Just hope thats how its meant to be! :P Andy.

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Re: [OpenSER-Users] CFG problem with 3rd party tool
by Andy Smith 30 Jan '08

30 Jan '08

If I was guessing (without fully understanding the syntax!) Id say the problem was Billing-Party=$avp(billing_party); should be Billing-Party=$avp(s:billing_party); Although in theory both of the above config entries seem to be valid, though different, openser.cfg entries. According to this: http://www.openser.org/docs/modules/1.2.x/avpops.html :S Also, I tried with the second format (with s:) and that then gives me another error when I start openser: 0(0) INFO:xl_parse_name: using hdr type (27) instead of <user-agent> 0(0) ERROR:acc:parse_acc_extra: parse failed in <User-Name=$Au; Calling-Station-Id=$from; Called-Station-Id=$to; Sip-Translated-Request-URI=$ruri; Sip-RPid=$avp(s:rpid); Source-IP=$si; Source-Port=$sp; Canonical-URI=$avp(s:can_uri); Billing-Party=$avp(s:billing_party); Divert-Reason=$avp(s:divert_reason); X-RTP-Stat=$hdr(X-RTP-Stat); User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); Event=$hdr(event); SIP-Proxy-IP=$avp(s:sip_proxy_ip); ENUM-TLD=$avp(s:enum_tld)> around position 319 0(0) acc:parse_acc_extra: error 0(0) ERROR:acc:mod_init: failed to parse rad_extra param 0(0) init_mod(): Error while initializing module acc ERROR: error while initializing modules

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Re: [OpenSER-Users] Does CDRTool require Radius server?
by Andy Smith 30 Jan '08

30 Jan '08

Hi Iñaki, ah yeah, I hadnt noticed this before, and diameter. However have you tested this to see if it writes data in exactly the same format to a DB as Freeradius would? For sure CDRTool has been written to work with freeradius and will therefore need the data in a freeradius format/schema... Andy. ----- Original Message ----- @Andy: > but Radius will do all other updates to DB accounting data. But the point here is that accounting is done by OpenSer ACC module, which can use radius or mysql to fill "radacct" table.

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[Serusers] Question on Accounting
by Nhadie 30 Jan '08

30 Jan '08

Hi, I'm recently added voip on our services using ser. i bill my users using the time of INVITE and BYE. but got a complain from one of my users, why even if the call was not answered they still get billed, what can i use on ser to check whether a call has been answered or not, then i will bill only base on the answered call. Thanks in advanced. Regards, Ron

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[OpenSER-Users] Does CDRTool require Radius server?
by Iñaki Baz Castillo 30 Jan '08

30 Jan '08

Hi, I'm starting reading about CDRTool and as AFAIK: - MediaProxy can update call media info directly into database (without using radius). - OpenSer can log accounting in database (with "exrta-accounting"). So my question is: does CDRTool require a radius server? why exactly? Thanks for any tip ;) -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Serusers] ser.cfg (forward to ser-sems)
by dim 30 Jan '08

30 Jan '08

hi.. I have attached my "running" ser.cfg...in which i want to include something like : if (uri =~ "sip:1.*@") { log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); break; } ... in order to uris start from 1 to be forwarded to my other ser (stack for sems).. Where exactly in my ser.cfg, should i put this ??? thank you very much... dimitris

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[Serusers] Check on-connected status on SER
by roayu 30 Jan '08

30 Jan '08

Hi, I would like to know whether SER can know the following status? For instance, (1) UA1(1000) is on conversation with UA2(1005) ? (2) UA1(1000) is on conversation with UA2(1005), then UA1 transfer to UA(1006)? Is there any logs that we can keep track? Thanks Roayu

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[Serusers] SER Config
by Rami Abualhasan 30 Jan '08

30 Jan '08

Hi, I'm trying to configure ser to be able to act as a sip registerer and proxy sip simple message between the sip UAC and UAS. I'd like also to enable mediaproxy and natping. It will be really appreciated if some one can give the correct version of ser or openser that can fulfil my requirement with a config file which will load the modules I need. Regards

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[OpenSER-Users] CFG problem with 3rd party tool
by Andy Smith 29 Jan '08

29 Jan '08

Hi list, can someone more familiar with openser CFG syntax tell me why follong the instructions for CDRTool, here: http://download.dns-hosting.info/CDRTool/doc/INSTALL.txt , breaks OpenSER?? With the following acc module settings OpenSER won't start up: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") I get this when starting OpenSER: 0(0) xl_parse_vname: error - bad avp name [billing_party] 0(0) ERROR:acc:parse_acc_extra: parse failed in <User-Name=$Au; Calling-Station-Id=$from; Called-Station-Id=$to; Sip-Translated-Request-URI=$ruri; Sip-RPid=$avp(s:rpid); Source-IP=$si; Source-Port=$sp; Canonical-URI=$avp(s:can_uri); Billing-Party=$avp(billing_party); Divert-Reason=$avp(s:divert_reason); X-RTP-Stat=$hdr(X-RTP-Stat); User-Agent=$hdr(user-agent); \Contact=$hdr(contact); Event=$hdr(event); SIP-Proxy-IP=$avp(s:sip_proxy_ip); ENUM-TLD=$avp(s:enum_tld)> around position 198 0(0) acc:parse_acc_extra: error Thanks for any help! Andy.

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Re: [OpenSER-Users] Does CDRTool require Radius server?
by Andy Smith 29 Jan '08

29 Jan '08

Hi Iñaki, my understanding of the documentation is that mediaproxy can update the (Radius) DB in theevent of a missing SIP BYE, but Radius will do all other updates to DB accounting data. Doesthat sound right to you? Do you have any documentation to the contrary etc?cheers Andy.Hi, I'm starting reading about CDRTool and as AFAIK: - MediaProxy can update call media info directly into database (without using radius). - OpenSer can log accounting in database (with "exrta-accounting"). So my question is: does CDRTool require a radius server? why exactly? Thanks for any tip ;)

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[Serusers] SER/OpenSER Consultant
by Brian Peters 29 Jan '08

29 Jan '08

Hi, We are looking for someone for offer consultancy services for a SER/OpenSER deployment. Please advise if you offer those services or know of anyone that does. Thank you, Brian Peters mailto: brian(a)proximosystems.com

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[OpenSER-Users] ABout Diameter
by dilip.m＠renovau.com 29 Jan '08

29 Jan '08

hello everyone, Anyone has used opendiameter with openser. i can run diameter client and diameter server but about the users i dont know.. How can we add users in the diameter? SO please anyone have any idea then tell me. Regards, Dilip

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[OpenSER-Users] OpenSER XMPP Gateway Feature
by Ali Jawad 29 Jan '08

29 Jan '08

Dear OpenSER Users I would like some info about the XMPP gateway feature, I do have a running XMPP server ejabbard, and I would like to use a client that uses both XMPP and SIP. So I am setting up the OpenSER server and I noticed the XMPP gateway feature. What does it do ? I what scenarios can it be applied ? Thanks With Regards Ali Jawad

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[OpenSER-Users] how to implement handoff?
by Aaash shah 29 Jan '08

29 Jan '08

hi i want to implement soft handoff using SIP, so please give me some information regarding it. And also i dont know how to use openSER and how to register it! i am trying to do register by kphone but registration failed. so please give me some information regarding this problems. thanks shital -- "Never Ignore Anyone Who Cares 4 u, Becoz one day u will Realize that u have lots a diamond while u were buzy in collecting stones...."

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[OpenSER-Users] Strip Digits
by Jonathan Creasy 29 Jan '08

29 Jan '08

I am sending an INVITE to Box A for "F+18007622265@boxa" Part of GW Table: gw_name grp_id ip_addr port uri_scheme transport strip prefix toll-free failover 5 XXXXXXXX /NULL/ /NULL/ /NULL/ 1 Part of LCR Table: id prefix from_uri grp_id priority 170 F+1800 5 1 This sends the call to box B XXXXXXXX per the GW table: Jan 29 04:03:16 boxb-sip /usr/local/sbin/openser[28118]: Req 41070 INVITE From <sip:asterisk@XXX.XXX.XXX.XXX> To <sip:F+18007622265@boxa> RURI <sip:+18007622265@boxb> It strips the F off in the RURI field but not in the TO field so boxb then fails the call as it doesn't know how to route a call to F+1800NXXXXXX. What can I do to make it re-format the to: field? The original problem I am trying to solve is that toll-free calls are routed to a system. If that system cannot handle the call I want that system to then place the call back to the OpenSER box (boxa) but instead of using the grp_id of 2 that +1800 calls would go to (and route to the system in question), I want to indicate that boxa should use group 5 and send the calls to boxb. There is probably a very easy way of doing this which doesn't involve a mangled dial string but I'm missing it so if anyone would like to assist me with this that would be great.

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[OpenSER-Users] [OT] Any UAC sending INVITE with empty SDP?
by Iñaki Baz Castillo 28 Jan '08

28 Jan '08

Hi, in order to try managing NAT when the SDP negoziation is done via "200 OK" - "ACK" (instead of "INVITE - "200 OK") I need a UAC that can send INVITE with empty SDP in order to force the called to generate the SDP offer in the reply. Is there any UAC allowing it? Thanks a lot. -- Iñaki Baz Castillo

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[OpenSER-Users] Increased heap size.
by Sergio Gutierrez 28 Jan '08

28 Jan '08

Hello everybody. Currently, I am watching a non normal behaviour in OpenSER 1.2.1, no tls. My operating System is Solaris 10, running on Sparc The process which appears as timer, is consuming almost the 50% of the whole physical memory of the machine. After a deep inspection of memory distribution, appears that the most of memory consumption is at process heap, as presented with pmap(1M) command: 1978: /toip/sbin/openser -n 9 -m 256 -f /toip/etc/openser/prueba1.cfg Address Kbytes RSS Anon Locked Mode Mapped File 00010000 760 752 - - r-x-- openser 000DC000 24 24 24 - rwx-- openser 000E2000 56 32 24 - rwx-- [ anon ] 000F0000 192 128 - - rwx-- openser 00120000 3906432 3906432 3627200 - rwx-- [ heap ] EE800000 262144 3624 - - rw-s- [ anon ] ... Thanks in advance for any hint about it. Best regards. Sergio Gutiérrez. EPM Telecomunicaciones S.A. E.S.P. Medellin, Colombia.

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[OpenSER-Users] dual nic at single machine openser
by flatworm 28 Jan '08

28 Jan '08

I got some routing problem configuring two NICs at debian. Openser is listening all possible network interface. I've configured one for static global IP address and one for internal private IP address. The reason why I've configured like this is for maintenance and development purpose. The static global IP line is for the service and the private IP line is for monitoring and administrations. But when I try to connect to openser of static global IP address, it was failed to register. 1) My machine was using private IP address of same subnet of openser connected to our private network. 2) My machine is trying to connect openser of static global IP address. In case of configuring dual NICs on Openser machine, is there any routing problem ? So I restricted the openser only listening static global IP address then it allows connection and succeeds in registration.

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[OpenSER-Users] Implementing the Ringing Groups
by raviprakash sunkara 28 Jan '08

28 Jan '08

Hello Users Like Asterisk PBX I implemented the Ringing Group like Ring all, Round Robin and Memory Hunt , RingAll and Round Robin Ringing is working fine with the following UserAgent LinkSys 941 and X-lite, GrandStream, And SJphones, with these is working, When comes to Memory Hunt Ring feature only with Sjphone are working fine, but with User Agent not working the problem is Previous User-Agent number is Not ringing. Can any one tell me solution Why these user-Agents are not working ........ Here is my Memory Hunt logic First A rings Second time A and B rings ( in Failure route Ist time ) Third Time A, B and C ( in Failure route 2 nd time ) Fourth time A, B,C and D rings ( in Failure route 3 time ) -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

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[OpenSER-Users] Are there are any special settings to run openser over SCTP?
by rajat.dudeja＠tcs.com 28 Jan '08

28 Jan '08

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Re: [OpenSER-Users] Is possible to store various values in an AVP?
by Iñaki Baz Castillo 28 Jan '08

28 Jan '08

El Sábado, 26 de Enero de 2008, escribió: > Hello, > > Why not just make allowed_from_numbers a regex, and feed it directly? > I.e. > > $avp(allowed_from_numbers) = "(100|200|300|400)"; > > if($avp(allowed_from_numbers) =~ $fU){ > # allowed > } else { > # not allowed > } > > > Hope this helps! Yes, sure, it's a good idea ;) Thanks. -- Iñaki Baz Castillo

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[OpenSER-Users] Fw: Are there are any special settings to run openser over SCTP?
by rajat.dudeja＠tcs.com 28 Jan '08

28 Jan '08

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[OpenSER-Users] rtpproxy seems to be idle
by Kovács Zoltán 26 Jan '08

26 Jan '08

Hi, my plan is to give the possibility for multiple VoIP users within the same LAN to call each other using OpenSER as the SIP server. I found that if there is a NATed LAN, two different VoIP users inside the NATed LAN cannot talk to each other properly, because IP parts of the RTP packets are of the form of the local network and it is not acceptable by the VoIP software which is prepared for the IP of the router instead. (I hope I understand clearly what is going on here.) As a workaround, I found a suggestion in the internet: using rtpproxy with OpenSER. As I can understand, all RTP packets are sent to the rtpproxy process via OpenSER (nathelper module). rtpproxy will behave as a bridge between the two NATed VoIP clients lying in the same LAN. Both clients will initiate their RTP/UDP traffic on their own side and answers will come from the rtpproxy process just for the same UDP pipe. Using this method the clients do not need to allow incoming UDP traffic, only outgoing traffic should be enabled (which is usually allowed by default in most LANs). Here comes my question. I successfully set up OpenSER (version 1.1.x, but I will migrate to 1.2.x soon). I think I also successfully set up rtpproxy, but it seems that there is no real traffic between OpenSER and rtpproxy. Here I attach my command line for running rtpproxy ( 200.200.200.200 is my public IP of the OpenSER server host; I run rtpproxy on the same server). -------------------------8X snip here------------------------------- # rtpproxy -f -l 200.200.200.200 -s unix:/var/run/rtpproxy.sock -r /home/rtpproxy -S /home/rtpproxy.work rtpproxy started, pid 24568 received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "U 3331228473621-29341876517920(a)160.114.144.65 160.114.144.65 10004 196215176;1" new session 3331228473621-29341876517920(a)160.114.144.65 , tag 196215176;1 requested, type strong new session on a port 35000 created, tag 196215176;1 pre-filling caller's address with 160.114.144.65:10004 sending reply "35000 212.92.3.218 " session timeout RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped session on ports 35000/0 is cleaned up -------------------------8X snip here------------------------------- My debug=7 output for OpenSER, filtered for the string "rtp" comes here (I had to chown openser.openser /var/run/rtpproxy* for OpenSER to make it possible to communicate with rtpproxy at all): -------------------------8X snip here------------------------------- # tail -f daemon.log | grep rtp Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_sock> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_disable> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_disable_tout> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_tout> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_retr> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: find_cmd_export_t: found <force_rtp_proxy>(0) in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: f<force_rtp_proxy> Jan 15 00:27:39 firewall /usr/sbin/openser[25093]: fixing force_rtp_proxy, line 443 Jan 15 00:27:39 firewall /usr/sbin/openser[25095]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25096]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25098]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25100]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25106]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25105]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25117]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25103]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25108]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25104]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25112]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25094]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25113]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25093]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25111]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25122]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25118]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25123]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25097]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25107]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25125]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25120]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25115]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25110]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25119]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25116]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25124]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25114]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25121]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25099]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25102]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25109]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25101]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:28:21 firewall /usr/sbin/openser[25112]: force_rtp_proxy2: proxy reply: 35000 212.92.3.218 Jan 15 00:28:27 firewall /usr/sbin/openser[25114]: build_res_from_sip_res: copied size: orig:77, new: 16, rest: 780 msg= SIP/2.0 200 Ok^M Via: SIP/2.0/UDP 160.114.144.65:1024;branch=z9hG4bK27994379851422672^M From: 06212527755 <sip:06212527755@sip.ephone.hu:5060>;tag=196215176^M To: 06212527722 < sip:06212527722@sip.ephone.hu:5060>;tag=7816428^M Contact: < sip:06212527722@160.114.144.65:5060>^M Record-Route: <sip:200.200.200.200;lr;ftag=196215176>^M Call-ID: 3331228473621-29341876517920(a)160.114.144.65^M CSeq: 1 INVITE^M Content-Type: application/sdp^M Server: X-Lite release 1105d^M Content-Length: 320^M ^M v=0^M o=06212527722 2057424946 2057430520 IN IP4 160.114.144.65^M s=X-Lite^M c=IN IP4 160.114.144.65^M t=0 0^M m=audio 8000 RTP/AVP 0 8 3 98 97 101^M a=rtpmap:0 pcmu/8000^M a=rtpmap:8 pcma/8000^M a=rtpmap:3 gsm/8000^M a=rtpmap:98 iLBC/8000^M a=rtpmap:97 speex/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv^M -------------------------8X snip here------------------------------- My nathelper related part of openser.cfg comes here: -------------------------8X snip here------------------------------- loadmodule " nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "received_avp", 42) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") -------------------------8X snip here------------------------------- Of course, directories /home/rtpproxy* are empty. Could somebody please help me why there is no media stream between OpenSER and rtpproxy? Your help is greatly appreciated in advance. I am using Debian Etch. Best regards, Zoltan

2 4

[OpenSER-Users] Is possible to store various values in an AVP?
by Iñaki Baz Castillo 25 Jan '08

25 Jan '08

Hi, I'd like to store in an AVP various posible values and load it with "load_credentials" parameter of "auth" module. The DB column would be: "allowed_from_numbers" AndI'd like to store there the valid values a subscribe can use as From username. For example: allowed_from_numbers = "200;201;200" So in OpenSer I do: if ($avp(allowed_from_numbers) == $fU) route(FROM_VALID); But unfortunatelly I think this is not possible, is it? which would be the required syntax to store various values? Thanks. -- Iñaki Baz Castillo

1 0

[OpenSER-Users] Revealing Number Dialled
by Jon Farmer 25 Jan '08

25 Jan '08

Hi I have some users successfully using openser and they are able to make and receive calls with any problems. I now want to provide inbound service from other telephone numbers to these registed users and want to be able to put the number dialled into the SIP header. However I am unsure how to represent this in the SIP message. Can anyone help me? TIA Jon __________________________________________________________ Sent from Yahoo! Mail - a smarter inbox http://uk.mail.yahoo.com

2 1

Re: [OpenSER-Users] Schema for OpenSER LDAP module
by Schumann Sebastian 25 Jan '08

25 Jan '08

Thanks Christian! I will use this module then in addition, didn't actually realize it is available for OpenSER :-( So far, I just checked the LDAP module, and on http://www.openser.org/dokuwiki/doku.php/tutorials:openser-auth-ldap was no hint on the H.350.4 module when showing the sample LDAP tree. I will try to implement the H.350.4 into my LDAP server and authenticate against it with OpenSER and LDAP+H.350 module. Thanks so far! Best regards Sebastian -----Original Message----- From: Christian Schlatter [mailto:cs@unc.edu] Sent: Friday, 25. January 2008 4:57 PM To: Schumann Sebastian Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] Schema for OpenSER LDAP module Sebastian, There is an ITU-T Recommendation called H.350 that standardizes LDAP schemas for storing SIP (and other session signaling protocols) account data. The documentation of the OpenSER 1.3 H350 module (http://www.openser.org/docs/modules/1.3.x/h350.html) further describes this standard. The "H.350 Cookbook" includes LDAP schema files ready for use with different LDAP server implementations: http://metric.it.uab.edu/vnet/cookbook/v2.1/node143.html /Christian Schumann Sebastian wrote: > Dear all > > I am wondering whether there is a schema file for OpenSER LDAP module, > to use at the LDAP server for data storage? > > On the tutorial from OpenSER Wiki, e.g. the attributes SIPUserName and > SIPPassword are used, but no standard schema actually defines them acc. > my research. > > Any help with this is as always appreciated! > > Thanks and a nice weekend to all of you! > > Regards > Sebastian > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users

1 0

[OpenSER-Users] Schema for OpenSER LDAP module
by Schumann Sebastian 25 Jan '08

25 Jan '08

Dear all I am wondering whether there is a schema file for OpenSER LDAP module, to use at the LDAP server for data storage? On the tutorial from OpenSER Wiki, e.g. the attributes SIPUserName and SIPPassword are used, but no standard schema actually defines them acc. my research. Any help with this is as always appreciated! Thanks and a nice weekend to all of you! Regards Sebastian

2 1

[OpenSER-Users] OpenSUSE packages required
by Dowling, Dan (Com US) 25 Jan '08

25 Jan '08

Hi, Can anyone tell me how to best determine which OpenSUSE packages are required for openser when used with asterisk? Best regards, Dan

1 0

[OpenSER-Users] presence: difference between watchers and active_watchers
by Klaus Darilion 25 Jan '08

25 Jan '08

Hi! Can someone explain me please the difference between between watchers and active_watchers? If I understand correctly, active_watchers are the SIP clients which have SUBSCRIBEd and thus will receive NOTIFYs. But what are "watchers"? regards klaus

2 1

[OpenSER-Users] Intial INVITES without SDP
by Patrick Miccio 25 Jan '08

25 Jan '08

Hi @ all, I am fighting a pretty annoying problem at the moment. As described in rfc3261 / 13.2.1 Creating the Initial INVITE o The initial offer MUST be in either an INVITE or, if not there, in the first reliable non-failure message from the UAS back to the UAC. In this specification, that is the final 2xx response. My problem is the following: I receive a initial INVITE without SDP, forward it to a PSTN Gateway. >From the Gateway I receive the 200 OK with SDP offer. In the onreply_route I make a call to the mediaproxy use_media_proxy(); Unfortunately use_media_proxy() seems to make a lookup command when used in a reply route. I would need to make a request command which would create the media session and then do a lookup with the ACK from the UAC. Is there a way around this? cheers, Patrick. This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

3 6

[OpenSER-Users] test
by NP南頻電信黃明杰網路電話部 jeff 25 Jan '08

25 Jan '08

南頻電信股份有限公司 NP Telecom Co., Ltd. 網路電話部 VOIP Dept 黃明杰 Jeff Huang E-Mail：jeff(a)np.net.tw 公司電話(Tel)：+886-2-8226-5008 Ext：867 公司網路代表號：070-288-1122 公司傳真(FAX)：+886-2-8226-5558 公司地址：23562台北縣中和市建八路十六號七樓之一 7F-1, NO. 16, Jian-ba Rd., Chung-ho City 23562, Taipei County, Taiwan, R.O.C. 公司網址：www.np.net.tw 統一編號：89459253 匯款銀行：彰化銀行雙和分行匯款帳號：5678-01-00999-1-00

1 0

[OpenSER-Users] WeSIP: Not SER connector available - Check the Configuration, new user issue help!
by rahul kadian 25 Jan '08

25 Jan '08

Hi, I am trying to configure WESIP and trying to run demo application "Click2dial" Everything seems to ve connected properly according to logs But when I call the manager Web-GUI I get as a status of OpenSER: "Not SER connector available - Check the Configuration" What might be the reason for this. Also while trying to run Click2dial , i get the error code "Call failed with code 2" or sometimes with code 3. Have checked the confi and server.xml according to the docs and seem to be running fine. If someone can let me know any solution to this , it would be of great help. -- Regards, Rahul.

1 0

[OpenSER-Users] SVN checkout of 1.2.3?
by Tobias Lindgren 24 Jan '08

24 Jan '08

Hi, I must be getting really tired, but checking out 1.2 from svn only brings me up to 1.2.2, not 1.2.3? Same with latest snapshot from 1.2. # $Id: ChangeLog 2609 2007-08-16 14:02:25Z miconda $ Br, /Tobias

2 2

Re: [OpenSER-Users] Carrierroute help..
by Henning Westerholt 24 Jan '08

24 Jan '08

On Tuesday 22 January 2008, Ricardo Martinez wrote: > Thanks Henning.... > Please see my answers below : > > [..] > According to this,in the "failure route" is checked if receives from the > gateway the messages "408" or "5XX", but since the gateway is offline there > is no way to receive any response from it. So, what is the behaviour of > the module here?. It keeps sending the calls to the de-2.carrier1 and > de-3.carrier1 gateways? or keeps sending the calls between the 3 gateways? Hi, this has nothing to do with the behaviour of carrierroute.. OpenSER will try to resend the invite, and then generate a local 408 after the timer of the tm module run out. And then you can enter a failure route. > > 2.- Now, suppose that "de-1.carrier1" gateway is only full, so i can > > probably have a reply from the gateway (maybe a "480" message"), so the > > carrierroute module now tries with a failover route (defa.carrier1?) or > > tries with any of the other two gateways still not full? > > I suppose you mean that 'de-1' is overloaded. No, the next domain is not > automatically entered. This deficiency is known, and will be addressed in > the future. But this code is not ready yet for a release. > > Yes sorry.. i meant "de-1" is overloaded, so in this case, i can receive a > "408" Message from the gateway indicating maybe "overload". So i handle > the answer with the failure route[2], so all the calls answered with the > "408" are re-routed to the "domain 1" route isn't?. Wouldn't be more > accurate to keep sending calls to the other gateways (de-2 and de-3)? > insted of make a failover to a default route? since the other two gateways > are not overloaded...... Well, 408 is not the right error for overloading, more appropriate would be 503. Carrierroute has no knowledge of the meaning and your policy regarding a 408/ 503, let alone that the two other gws are not overloaded.. A hardcoded logic in this module would not fit to everybody. At the moment you must go with the failure_routes, or use the lcr module instead. In the future there will be probably some better solution, as i wrote in my first mail. Cheers, Henning

1 0

[OpenSER-Users] Openser 1.3 How to rewrite transport protocol from udp to tcp after redirect
by Peter P GMX 24 Jan '08

24 Jan '08

Hello, I have the following scenario: SIP UA <=UDP=> OpenSER <=TCP> Microsoft Unified Messaging 2007 When I invite an account on the UM 2007 server the following happens 1 SIP UA ==Invite UDP=> OpenSER 2 SIP UA <== 407 Proxy Authentication Required UDP= OpenSER 3 SIP UA ==ACK UDP=> OpenSER 4 SIP UA ==Invite UDP=> OpenSER 5 OpenSER ==Invite TCP=> UM2007 6 OpenSER <=Trying TCP== UM2007 7 OpenSER <= 302 Moved Temporarily TCP== UM2007 8 OpenSER == ACK TCP=> UM2007 9 OpenSER == INVITE UDP=> UM2007 .....no answer from UM2007 (I left out some steps between Openser and UA) In Step 5 OpenSER invites "sip:11111@10.10.2.13;transport=UDP>" with TCP protocol (checked with ngrep) and UM2007 answers. In Step 7 UM2007 answers with CONTACT: <sip:11111@10.10.2.13:5065;transport=UDP> but I need to answer on TCP to UM2007. (I can change the message with add_uri_param("transport=tcp") in step 5 and I get a <sip:11111@10.10.2.13:5065;transport=TCP> back from UA2007, but in fact the message is corrupted and OpenSER replies to the UA with Error 500 then, see below) The point is: - I can invite UM2007 with TCP (although I was not able to produce correct headers as there is still "transport=udp;>" mentioned in the INVITE part) - but in the failure route when I force OpenSER to reinvite with get_redirects("*") and t_relay() I am not able to swith to TCP I tried several ways as described in discussions which I found in http://osdir.com/ml/voip.openser.user/2005-12/msg00256.html - add_uri_param("transport=tcp"); ==> added transport=tcp additionally to transport=udp ==> transport=udp;transport=tcp ==> 500 Internal Server error - $avp(s:foo)="sip:10.10.2.13;transport=tcp"; avp_pushto("$du","$avp(s:foo)"); ==>had actually no effect - replace_all("transport=udp", "transport=tcp"); ==> produced invalid headers as it simply replaced the headers wrongly as described in http://openser.org/pipermail/devel/2007-May/007702.html Questions: 1. How can I change the initial INVITe message (Step5) in order to have the right headers. (transport=TCP instead of transport=UDP)? 2. How can I answer to a redirect and change to a different transport protocol (Step 9)? Your help is very much appreciated. Best regards Peter

4 10

[OpenSER-Users] Lookinfg for an SCTP compliant SIP phone
by Rajat Dudeja 24 Jan '08

24 Jan '08

Hi folks, I'm new to openser. I'm using the sctp feature of it. I'm looking to test the openser's sctp functionality. For this I have used the sctp compliant kphone. However, it does not work. The phone does not even register. Please suggest if there is any other tool to it's functionality. Thanks and regards, Rajat =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[Serusers] presence: difference between watchers and active_watchers
by Klaus Darilion 24 Jan '08

24 Jan '08

Hi! Can someone explain me please the difference between between watchers and active_watchers? If I understand correctly, active_watchers are the SIP clients which have SUBSCRIBEd and thus will receive NOTIFYs. But what are "watchers"? regards klaus

3 3

[OpenSER-Users] User Agent (Client)
by J Bhanot 24 Jan '08

24 Jan '08

Hi, I am running openser Server... I want to know that do we have any inbuit user agent (client) in openser which we can use to send messages to this running server.. Thanks, jb ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[Serusers] ser , rtpproxy and pstn gateway
by mashudi 24 Jan '08

24 Jan '08

I have two separate network, for public and vpn , i have install ser and rtpproxy on different machine, all user from public network will register to proxy server on ip 10.44.9.18 and the rtp will be handle by another machine using ip 10.44.9.17 , all call comming from client on public ip will route to pstn gateway using cantata media gateway on ip 10.177.107.90 (vpn ip). When client call to pstn gateway, the signalling reach the callee, but after callee pickup the phone, the client using x-lite still in calling status, and rtpproxy not receive rtp from pstn gateway. the rtpproxy have two nic using two ip for public and vpn, and ser as proxy using two nic public ip and vpn ip. any one can help? here is the log for rtpproxy and ser. the rtpproxy log : Starting rtpproxy: rtpproxy started, pid 25745 received command "2573_0 V" sending reply "2573_0 20040107 " received command "2574_0 V" sending reply "2574_0 20040107 " received command "2572_0 V" sending reply "2572_0 20040107 " received command "2572_6 U NjQzNmQzNzFjZTBmMzM5MDUxYjQwODFlOTFmMTdhMzI. 10.11.11.152 30126 9d08dc55" new session NjQzNmQzNzFjZTBmMzM5MDUxYjQwODFlOTFmMTdhMzI., tag 9d08dc55 requested, type strong new session on a port 35010 created, tag 9d08dc55 pre-filling caller's address with 10.11.11.152:30126 sending reply "2572_6 35010 10.44.9.17 " the ser log: 0(2572) SIP Request: 0(2572) method: <NOTIFY> 0(2572) uri: <sip:02170743627@10.11.11.152:65100> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-9533d56650536c16-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) DEBUG: add_param: tag=59420d70 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) is_preloaded: No 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) check_self: host != me 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) after_loose: Topmost route URI: 'sip:10.44.9.18;lr;ftag=9e3f287c' is me 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <13> <NOTIFY> 0(2572) DEBUG: get_hdr_body : content_length=461 0(2572) found end of header 0(2572) find_next_route: No next Route HF found 0(2572) after_loose: No next URI found 0(2572) DEBUG: t_newtran: msg id=55 , global msg id=54 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=24011, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Reply (status): 0(2572) version: <SIP/2.0> 0(2572) status: <200> 0(2572) reason: <OK> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bKbcd5.79134365.0>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) forward_reply: found module tm, passing reply to it 0(2572) DEBUG: t_check: msg id=56 global id=55 T start=0xffffffffffffffff 0(2572) parse_headers: flags=17 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-9533d56650536c16-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <65100>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=17 0(2572) parse_headers: this is the second via 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <13> <NOTIFY> 0(2572) DEBUG: t_reply_matching: hash 24011 label 1446261143 branch 0 0(2572) DEBUG: t_reply_matching: reply matched (T=0x2ab57bba4030)! 0(2572) DEBUG: t_check: msg id=56 global id=56 T end=0x2ab57bba4030 0(2572) DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=64 0(2572) ->>>>>>>>> T_code=0, new_code=200 0(2572) DEBUG: relay_reply: branch=0, save=0, relay=0 0(2572) old size: 531, new size: 472 0(2572) build_res_from_sip_res: copied size: orig:267, new: 208, rest: 264 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.11.152:65100;branch=z9hG4bK-d87543-9533d56650536c16-1--d87543-;rport=65100 Record-Route: <sip:10.44.9.18;lr;ftag=59420d70> Contact: <sip:02170743627@10.11.11.152:65100> To: "02170743627"<sip:02170743627@10.44.9.18>;tag=9e3f287c From: "02170743627"<sip:02170743627@10.44.9.18>;tag=59420d70 Call-ID: MWNiZWE5ODMyN2U0MTJmMWMyNjQzMTNiNGJjZmRjYjA. CSeq: 13 NOTIFY User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG: reply relayed. buf=0x6c9930: SIP/2.0 2..., shmem=0x2ab57bba6b10: SIP/2.0 2 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: timer routine:2,tl=0x2ab57bba40b0 next=(nil) 1(2573) DEBUG: wait_handler : removing 0x2ab57bba4030 from table 1(2573) DEBUG: delete transaction 0x2ab57bba4030 1(2573) DEBUG: wait_handler : done 0(2572) SIP Request: 0(2572) method: <INVITE> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-2623ee4d6908a453-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) end of header reached, state=9 0(2572) DEBUG: get_hdr_field: <To> [43]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18> ] 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2572) DEBUG: get_hdr_body : content_length=420 0(2572) found end of header 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) parse_headers: flags=16384 0(2572) pre_auth(): Credentials with given realm not found 0(2572) build_auth_hf(): 'Proxy-Authenticate: Digest realm="10.44.9.18", nonce="4798817c850ce0e3c233a35804d9be1afe7fe2a3" ' 0(2572) parse_headers: flags=-1 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Request: 0(2572) method: <ACK> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-2623ee4d6908a453-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=4 0(2572) DEBUG: add_param: tag=a3f1a545ddce5205bb83afef24613e34.78aa 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [85]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18>] 0(2572) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Request: 0(2572) method: <INVITE> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-363a1c17b37ac509-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) end of header reached, state=9 0(2572) DEBUG: get_hdr_field: <To> [43]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18> ] 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <2> <INVITE> 0(2572) DEBUG: get_hdr_body : content_length=420 0(2572) found end of header 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) check_nonce(): comparing [4798817c850ce0e3c233a35804d9be1afe7fe2a3] and [4798817c850ce0e3c233a35804d9be1afe7fe2a3] 0(2572) HA1 string calculated: 0f7e2f302e87dfe46c5ca6566ad32b47 0(2572) check_response(): Our result = '82c47bef0adf71299b8fb157a15e0131' 0(2572) check_response(): Authorization is OK 0(2572) save_rpid(): rpid value is '' 0(2572) check_username(): Digest username and URI username match 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) parse_headers: flags=64 0(2572) lookup(): '02170890010' Not found in usrloc 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=-1 0(2572) DEBUG:check_content_type: type <application/sdp> found valid 0(2572) parse_headers: flags=32 0(2572) ERROR: extract_mediaip: no `c=' in SDP 0(2572) DEBUG: t_newtran: msg id=59 , global msg id=56 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=48512, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) SER: new INVITE 0(2572) parse_headers: flags=-1 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: reply sent out. buf=0x6c9930: SIP/2.0 1..., shmem=0x2ab57bba3ad8: SIP/2.0 1 0(2572) DEBUG: _reply_light: finished 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6bebb8 0(2572) WARNING: get_send_socket: not listening on the requested socket, no fork mode? 0(2572) parse_headers: flags=2048 0(2572) parse_headers: flags=-1 0(2572) clen_builder: content-length: 436 (436) 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[5]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:5,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[6]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:6,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) udp_rcv_loop: probing packet received from 10.11.11.152 19710 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:0,tl=0x2ab57bba4228 next=(nil) 1(2573) DEBUG: final_response_handler:stop retr. and send CANCEL (0x2ab57bba4030) 1(2573) ->>>>>>>>> T_code=100, new_code=408 1(2573) DEBUG: relay_reply: branch=0, save=0, relay=0 1(2573) parse_headers: flags=-1 1(2573) check_via_address(10.11.11.152, 10.11.11.152, 0) 1(2573) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4150 1(2573) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4170 1(2573) DEBUG: reply relayed. buf=0x6c84d0: SIP/2.0 4..., shmem=0x2ab57bba3ad8: SIP/2.0 4 1(2573) DEBUG: final_response_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 0(2572) SIP Request: 0(2572) method: <ACK> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-363a1c17b37ac509-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(2572) parse_headers: flags=128 0(2572) DEBUG: add_param: tag=d548cf07be509c4dc64b42f06e6fb880-a911 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [85]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <2> <ACK> 0(2572) DEBUG: get_hdr_body : content_length=0 0(2572) found end of header 0(2572) DEBUG: is_maxfwd_present: max_forwards header not found! 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) DEBUG: t_newtran: msg id=60 , global msg id=59 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=48512, isACK=1 0(2572) DEBUG: RFC3261 transaction matched, tid=-d87543-363a1c17b37ac509-1--d87543- 0(2572) DEBUG: t_lookup_request: transaction found (T=0x2ab57bba4030) 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4150 next=(nil) 1(2573) DEBUG: timer routine:2,tl=0x2ab57bba40b0 next=(nil) 1(2573) DEBUG: wait_handler : removing 0x2ab57bba4030 from table 1(2573) DEBUG: delete transaction 0x2ab57bba4030 1(2573) DEBUG: wait_handler : done 0(2572) SIP Request: 0(2572) method: <NOTIFY> 0(2572) uri: <sip:02170743627@10.11.11.152:65100> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-3217dc739e312401-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) DEBUG: add_param: tag=59420d70 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) is_preloaded: No 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) check_self: host != me 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) after_loose: Topmost route URI: 'sip:10.44.9.18;lr;ftag=9e3f287c' is me 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <14> <NOTIFY> 0(2572) DEBUG: get_hdr_body : content_length=425 0(2572) found end of header 0(2572) find_next_route: No next Route HF found 0(2572) after_loose: No next URI found 0(2572) DEBUG: t_newtran: msg id=61 , global msg id=60 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=24006, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, NOTIFY si ... ) 1(2573) DEBUG: add_to_tail_of_timer[5]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 0(2572) SIP Reply (status): 0(2572) version: <SIP/2.0> 0(2572) status: <200> 0(2572) reason: <OK> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK6cd5.c487f5f5.0>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) forward_reply: found module tm, passing reply to it 0(2572) DEBUG: t_check: msg id=62 global id=61 T start=0xffffffffffffffff 0(2572) parse_headers: flags=17 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-3217dc739e312401-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <65100>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=17 0(2572) parse_headers: this is the second via 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <14> <NOTIFY> 0(2572) DEBUG: t_reply_matching: hash 24006 label 1600092236 branch 0 0(2572) DEBUG: t_reply_matching: reply matched (T=0x2ab57bba4030)! 0(2572) DEBUG: t_check: msg id=62 global id=62 T end=0x2ab57bba4030 0(2572) DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=64 0(2572) ->>>>>>>>> T_code=0, new_code=200 0(2572) DEBUG: relay_reply: branch=0, save=0, relay=0 0(2572) old size: 531, new size: 472 0(2572) build_res_from_sip_res: copied size: orig:267, new: 208, rest: 264 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.11.152:65100;branch=z9hG4bK-d87543-3217dc739e312401-1--d87543-;rport=65100 Record-Route: <sip:10.44.9.18;lr;ftag=59420d70> Contact: <sip:02170743627@10.11.11.152:65100> To: "02170743627"<sip:02170743627@10.44.9.18>;tag=9e3f287c From: "02170743627"<sip:02170743627@10.44.9.18>;tag=59420d70 Call-ID: MWNiZWE5ODMyN2U0MTJmMWMyNjQzMTNiNGJjZmRjYjA. CSeq: 14 NOTIFY User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG: reply relayed. buf=0x6c9930: SIP/2.0 2..., shmem=0x2ab57bba6b10: SIP/2.0 2 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up # $Id: nat-rtpproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=9 fork=no log_stderror=yes mhomed=yes listen=10.44.9.18 # INSERT YOUR IP ADDRESS HERE listen=10.177.107.124 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/permissions.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "udp:10.44.9.17:12345") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:0217[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { rewritehost("10.177.107.90"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

1 0

[Serusers] SIP Express Bundle - let's get started
by Mike Trest - Personal 23 Jan '08

23 Jan '08

Hi, The holidays and the new year work load seems to have taken all my time. Anyway, next week, I should be clear of regulatory filings and paperwork. That should give me time to resume building stand alone SER to test prototype of the SER Express Bundle. Any first-draft materials from anyone yet? ..mike..

4 5

[OpenSER-Users] UAC module (openser 1.2.1)
by Antonio Reale 23 Jan '08

23 Jan '08

Hi all, I'm trying to use UAC module and I'd like to know if it is possible to configure openser to register on another proxy on start up and wait for incoming calls, instead of use UAC module to terminate calls on an authenticated trunk. Thanks. Regards, Antonio

4 4

[Serusers] forking error - branch response sent to uac
by Tomasz Zieleniewski 23 Jan '08

23 Jan '08

hi, I encountered following error. I have two user agents register under the same uid. When there is an invite request both ua's ring. When one of them UA1 answers the call, by sending 200 response some strange error occurs. UA1's 200 response is immediately sent to UAC who initiated a call. after this ser sends cancel message to other UA2. The other UA2 responses with 200 to the cancel message and sends response to the initial invite: 487 Request Terminated. After this ser should response with ACK to the second UA2 but it isn't instead it forwards 487 response to UAC who sent the initial invite. Please tell why ser behaves this way? What could be the reason for such behavior? Regards Tomasz

1 0

[OpenSER-Users] Performance of Openser as Presence Server
by Noble Antony T 23 Jan '08

23 Jan '08

Hi, Do any one has an idea about the Performance benchmark of Openser as Presence Server..? Like how many Subscribers, how many subscribe/publish/notify messages, it can handle.. [ I want to know how much Subscribers/Subscriptions & how many Subscribe/Notify/Publish messages, Openser can handle. I am using Openser as only Presence Server ( not proxy/registrar )] If anybody have any figures about this performance, please let me know. Thanks -- Noble Antony Thattil.

1 0

[OpenSER-Users] Reg. Proxy authentication
by Padmaja 23 Jan '08

23 Jan '08

Hi all, Please see the call flow below: GW1 Proxy GW2 |-----invite----------->| | |<------407-----------| | |-----invite w/cred--->| | | |-----Invite------->| |<------------200 Ok---------------------| |----------------Ack--------------------->| | | | |<----ReInvite -----| | When the proxy receives the Reinvite below from GW 2, should it again challenge it for proxy authentication or simply forward the call to the GW1? The openser proxy in our lab simply forwards the Invite without asking for authentication this time. However, I need to test the situation, where the proxy asks for authentication for the Reinvite. Is this possible? Please let me know. Thanks, Padmaja

2 2

[OpenSER-Users] Forwarding to SRV records not working
by James Holden 23 Jan '08

23 Jan '08

Hi, I'm having problems with OpenSER forwarding calls to a domain where there are only SRV records. OpenSER seems not to perform any balancing or failover when a target server is unavailable or gives an error. I wish to forward calls to a number of onward gateways that at times may not respond at all, or may respond with something like "503 Service Unavailable". The actual destination address is derived from an enum lookup, but it may be for example 123456(a)example.com There are SRV records as follows: _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. The behaviour I'd expect/like would be that OpenSER will distribute the calls between the two targets specified in the SRV records, and will fail over to the other target if one fails to respond or gives a fatal error such as a 503. What actually happens, is that the calls all go to the first one regardless. Are my expectations reasonable? Does OpenSER's SRV record implementation work like this or am I misled? Kind regards, James

3 3

Re: [OpenSER-Users] Users Digest, Vol 32, Issue 70
by Padmaja 23 Jan '08

23 Jan '08

----- Original Message ----- From: <users-request(a)lists.openser.org> To: <users(a)lists.openser.org> Sent: Wednesday, January 23, 2008 3:28 AM Subject: Users Digest, Vol 32, Issue 70 > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Forwarding to SRV records not working (James Holden) > 2. Re: Serial fork terminated after 5 call (Bogdan-Andrei Iancu) > 3. How to handle INVITES with different To: header (Peter P GMX) > 4. Re: How to handle INVITES with different To: header > (Bogdan-Andrei Iancu) > 5. Re: Forwarding to SRV records not working (Bogdan-Andrei Iancu) > 6. OpenSER: no NOTIFY when updating PUBLISH if integrating with > OpenXCAP? (KevinKinnan) > 7. OpenSER: no NOTIFY when updating PUBLISH if integrating with > OpenXCAP? (KevinKinnan) > 8. Re: Problems trying to build an Active/Active OSer 1.3 > cluster (Stefan Sayer) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 22 Jan 2008 16:41:42 +0000 > From: James Holden <james.holden(a)uk.aql.com> > Subject: [OpenSER-Users] Forwarding to SRV records not working > To: users(a)lists.openser.org > Message-ID: <20080122164142.GD4315@james-desktop> > Content-Type: text/plain; charset=us-ascii > > Hi, > > I'm having problems with OpenSER forwarding calls to a domain where > there are only SRV records. > > OpenSER seems not to perform any balancing or failover when a target > server is unavailable or gives an error. > > I wish to forward calls to a number of onward gateways that at times may > not respond at all, or may respond with something like "503 Service > Unavailable". > > The actual destination address is derived from an enum lookup, but it > may be for example 123456(a)example.com > > There are SRV records as follows: > > _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. > _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. > > The behaviour I'd expect/like would be that OpenSER will distribute the > calls between the two targets specified in the SRV records, and will > fail over to the other target if one fails to respond or gives a fatal > error such as a 503. > > What actually happens, is that the calls all go to the first one > regardless. > > Are my expectations reasonable? > > Does OpenSER's SRV record implementation work like this or am I misled? > > Kind regards, > > James > > > > ------------------------------ > > Message: 2 > Date: Tue, 22 Jan 2008 18:49:21 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] Serial fork terminated after 5 call > To: Sergey Babichev <zaikini(a)yandex.ru> > Cc: users(a)lists.openser.org > Message-ID: <47961E91.7080003(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Sergey, > > the most important thing when searching is to know where to search ;)... > See: > http://www.openser.org/docs/modules/1.2.x/nathelper.html#AEN370 > > Regards, > Bogdan > > Sergey Babichev wrote: >> Okey:)) And last question about documentation, I searched documentation >> about function nat_uac_test from nathelper module for version 1.2.x and >> I'm find it in documentation for version 0.9. Openser 1.2.x not support >> function nat_uac_test or documentation 1.2.x not full? >> >> >> >> >>> Hi Sergey, >>> >> >> >>> I'm afraid there is no docs about the errors - the idea is to have >>> >> >> >>> sef-explaining error messages and to avoid documenting them ;) >>> >> >> >>> Regards, >>> >> >> >>> bogdan >>> >> >> >>> Sergey Babichev wrote: >>> >> >> >>>> Thanks, where i can read about this error (discarding fwd for a >>>> cancelled transaction) and others? >>>> >> >> >> >> >> >> >> >> >>>> >>>> >> >> >>>>> Hi Sergey, >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> the "discarding fwd for a cancelled transaction" error is generated >>>>> when >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> you try to send a request for a transaction that was already cancelled >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> from the client side. So probably the PSTN GW cancels the call while >>>>> you >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> still want to fork a new branch. >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> regards, >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> Bogdan >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> Sergey Babichev wrote: >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Problem is appear when i calling from PSTN. It's work fine when i >>>>>> calling from sip account to huntgroup. I think this is problem whith >>>>>> timer in my PSTN gateway. >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Hi all! >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> I want enable hunting for my extensions. It's work fine, but if my >>>>>>> hunt group consists of more then 5 extensions, call terminated and I >>>>>>> see in syslog error: >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> ERROR:tm:t_forward_nonack: discarding fwd for a cancelled >>>>>>> transaction >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> ERROR:tm:w_t_relay: t_forward_nonack failed >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> DEBUG:tm:relay_reply: branch=5, save=0, relay=5 >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> What is problem may be? >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> -- >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Best regards, >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Sergey Babichev >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> _______________________________________________ >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Users mailing list >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Users(a)lists.openser.org >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> -- >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Best regards, >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Sergey Babichev >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> _______________________________________________ >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Users mailing list >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Users(a)lists.openser.org >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >> >> >> >> >>>> -- >>>> >> >> >> >> >>>> Best regards, >>>> >> >> >> >> >>>> Sergey Babichev >>>> >> >> >> >> >>>> >>>> >> >> >> >> -- >> >> Best regards, >> >> Sergey Babichev >> >> > > > > > ------------------------------ > > Message: 3 > Date: Tue, 22 Jan 2008 19:20:44 +0100 > From: Peter P GMX <Prometheus001(a)gmx.net> > Subject: [OpenSER-Users] How to handle INVITES with different To: > header > To: users(a)lists.openser.org > Message-ID: <479633FC.2090506(a)gmx.net> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > I have the following scenario: > > I receive a request with PSTN_number(a)my.ip.add.res from my VoIP > In alias_db I have defined a rule that points to yyy(a)myip.com which is > a registered account > > After alias_db_lookup OpenSER invites as follows: > Invite yyy(a)myip.com > To: xxxxxxx(a)my.ip.add.res > > The UA (actually a PBX with VoIP support) answers with 404 not found. > > When I invite as follows: > Invite yyy(a)myip.com > To: yyy(a)myip.com > > everything works fine. > > My question: How do you handle these cases? > Rewriting the To: part seems dificult and will mess up a lot - hein? > > Kind regards > Peter > > > > > ------------------------------ > > Message: 4 > Date: Tue, 22 Jan 2008 21:41:32 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] How to handle INVITES with different To: > header > To: Prometheus001(a)gmx.net > Cc: users(a)lists.openser.org > Message-ID: <479646EC.5090301(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Peter, > > According to RFC3261, in SIP, routing is exclusively done based on RURI > (and eventually Route hdrs). TO header has 0 implication in routing. > So, your PBX is deprecated and out of standards and I suggest either > upgrade, either replacement. > > Regards, > Bogdan > > Peter P GMX wrote: >> I have the following scenario: >> >> I receive a request with PSTN_number(a)my.ip.add.res from my VoIP >> In alias_db I have defined a rule that points to yyy(a)myip.com which is >> a registered account >> >> After alias_db_lookup OpenSER invites as follows: >> Invite yyy(a)myip.com >> To: xxxxxxx(a)my.ip.add.res >> >> The UA (actually a PBX with VoIP support) answers with 404 not found. >> >> When I invite as follows: >> Invite yyy(a)myip.com >> To: yyy(a)myip.com >> >> everything works fine. >> >> My question: How do you handle these cases? >> Rewriting the To: part seems dificult and will mess up a lot - hein? >> >> Kind regards >> Peter >> > > > > ------------------------------ > > Message: 5 > Date: Tue, 22 Jan 2008 23:00:28 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] Forwarding to SRV records not working > To: James Holden <james.holden(a)uk.aql.com> > Cc: users(a)lists.openser.org > Message-ID: <4796596C.7000004(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi James, > > Openser does not do balancing over records with same priority - it will > just use the first (as returned by DNS server). > > But failover based on NAPTR and SRV records is in place. > > Regards, > Bogdan > > > James Holden wrote: >> Hi, >> >> I'm having problems with OpenSER forwarding calls to a domain where >> there are only SRV records. >> >> OpenSER seems not to perform any balancing or failover when a target >> server is unavailable or gives an error. >> >> I wish to forward calls to a number of onward gateways that at times may >> not respond at all, or may respond with something like "503 Service >> Unavailable". >> >> The actual destination address is derived from an enum lookup, but it >> may be for example 123456(a)example.com >> >> There are SRV records as follows: >> >> _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. >> _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. >> >> The behaviour I'd expect/like would be that OpenSER will distribute the >> calls between the two targets specified in the SRV records, and will >> fail over to the other target if one fails to respond or gives a fatal >> error such as a 503. >> >> What actually happens, is that the calls all go to the first one >> regardless. >> >> Are my expectations reasonable? >> >> Does OpenSER's SRV record implementation work like this or am I misled? >> >> Kind regards, >> >> James >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > > ------------------------------ > > Message: 6 > Date: Wed, 23 Jan 2008 05:49:18 +0800 > From: KevinKinnan <kinnan2224(a)hotmail.com> > Subject: [OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if > integrating with OpenXCAP? > To: <users(a)lists.openser.org> > Message-ID: <BAY126-W641A1FEB850E880B28F56C43E0(a)phx.gbl> > Content-Type: text/plain; charset="gb2312" > > > Thank you so much, Franz! > > Indeed, the problem is coming from xml. It is well-formed but seems the > server can't understand it. > The whole xml document I used is the similar with yours except the root > element with namespace which > is showing below: > > <cr: ruleset xmlns="urn:ietf:params:xml:ns:common-policy" > xmlns:pr="urn:ietf:params:xml:ns:pres-rules" > xmlns:cr="urn:ietf:params:xml:ns:common-policy"> > > > I got no error sending the xml doc with this as beginning. Only no Notify > message after updating Publish, > and sometimes the first Subscribe can't get through that is no Notify > sending back from server. It's really > weired! Anyway, I finally fix it with your advise, Thanks Indeed. > > Kevin, > > _________________________________________________________________ > ????? MSN ?????????? > http://mobile.msn.com.cn/ >

1 0

Re: [OpenSER-Users] Serial fork terminated after 5 call SOLVED
by Sergey Babichev 23 Jan '08

23 Jan '08

Thanks ))) > Hi Sergey, > the most important thing when searching is to know where to search ;)... > See: > http://www.openser.org/docs/modules/1.2.x/nathelper.html#AEN370 > Regards, > Bogdan > Sergey Babichev wrote: > > Okey:)) And last question about documentation, I searched documentation about function nat_uac_test from nathelper module for version 1.2.x and I'm find it in documentation for version 0.9. Openser 1.2.x not support function nat_uac_test or documentation 1.2.x not full? > > > > > > > > > >> Hi Sergey, > >> > > > > > >> I'm afraid there is no docs about the errors - the idea is to have > >> > > > > > >> sef-explaining error messages and to avoid documenting them ;) > >> > > > > > >> Regards, > >> > > > > > >> bogdan > >> > > > > > >> Sergey Babichev wrote: > >> > > > > > >>> Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > >>> > > > > > > > > > > > > > > > > > >>> > >>> > > > > > >>>> Hi Sergey, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> the "discarding fwd for a cancelled transaction" error is generated when > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> you try to send a request for a transaction that was already cancelled > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> from the client side. So probably the PSTN GW cancels the call while you > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> still want to fork a new branch. > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> regards, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Bogdan > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Sergey Babichev wrote: > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Hi all! > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> What is problem may be? > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> -- > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Best regards, > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Sergey Babichev > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> _______________________________________________ > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Users mailing list > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Users(a)lists.openser.org > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> -- > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> Best regards, > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> Sergey Babichev > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> _______________________________________________ > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Users mailing list > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Users(a)lists.openser.org > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > > > > > > > > > >>> -- > >>> > > > > > > > > > >>> Best regards, > >>> > > > > > > > > > >>> Sergey Babichev > >>> > > > > > > > > > >>> > >>> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > -- Best regards, Sergey Babichev

1 0

[OpenSER-Users] Problems trying to build an Active/Active OSer 1.3 cluster
by Raúl Alexis Betancor Santana 22 Jan '08

22 Jan '08

Hi all, I'm trying to build an Active/Active OpenSer 1.3 cluster with 2 nodes. All it's working ok for the REGISTER messages throught the use of t_replicate and add_path, but I've problems with the INVITE messages. Both servers share the same location table on a PostgreSQL Multimaster cluster (but that's irrelevant, the PostgreSQL cluster I mean...), using db_mode = 2 An that's the logic I used: - On REGISTER ->> - Auth Request - NAT ? - add_path - t_replicate(oser2) - On INVITE ->> - Auth Request - lookup - Load prefs from DB - relay The problem I'm having is that the lookup function changes the RURI of the message on node1 BEFORE taking into account the "Path:" element present inside the AoR, so when the INVITE gets forwarded to node2 (where the NATed UAC it's registered), node2 deny the relaying of the message because $rD of the RURI it's not local (remember that the RURI whas changed on node1). Is there any way to know on node1 that the INVITE request MUST obey the "Path:" so I could to something like: $old_ru = $ru lookup("location") if(path_exists()) { $ru = $old_ru } ... t_relay() ?? Any other aproach ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual S.L.

3 3

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 22 Jan '08

22 Jan '08

Franz, Have you tested 'validity' rule and 'sphere' rule under Openser + Openxcap? Regards, Kevin _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 22 Jan '08

22 Jan '08

Thank you so much, Franz! Indeed, the problem is coming from xml. It is well-formed but seems the server can't understand it. The whole xml document I used is the similar with yours except the root element with namespace which is showing below: <cr: ruleset xmlns="urn:ietf:params:xml:ns:common-policy" xmlns:pr="urn:ietf:params:xml:ns:pres-rules" xmlns:cr="urn:ietf:params:xml:ns:common-policy"> I got no error sending the xml doc with this as beginning. Only no Notify message after updating Publish, and sometimes the first Subscribe can't get through that is no Notify sending back from server. It's really weired! Anyway, I finally fix it with your advise, Thanks Indeed. Kevin, _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

1 0

[OpenSER-Users] How to handle INVITES with different To: header
by Peter P GMX 22 Jan '08

22 Jan '08

I have the following scenario: I receive a request with PSTN_number(a)my.ip.add.res from my VoIP In alias_db I have defined a rule that points to yyy(a)myip.com which is a registered account After alias_db_lookup OpenSER invites as follows: Invite yyy(a)myip.com To: xxxxxxx(a)my.ip.add.res The UA (actually a PBX with VoIP support) answers with 404 not found. When I invite as follows: Invite yyy(a)myip.com To: yyy(a)myip.com everything works fine. My question: How do you handle these cases? Rewriting the To: part seems dificult and will mess up a lot - hein? Kind regards Peter

2 1

Re: [OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Okey:)) And last question about documentation, I searched documentation about function nat_uac_test from nathelper module for version 1.2.x and I'm find it in documentation for version 0.9. Openser 1.2.x not support function nat_uac_test or documentation 1.2.x not full? > Hi Sergey, > I'm afraid there is no docs about the errors - the idea is to have > sef-explaining error messages and to avoid documenting them ;) > Regards, > bogdan > Sergey Babichev wrote: > > Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > > > > > > > > > >> Hi Sergey, > >> > > > > > >> the "discarding fwd for a cancelled transaction" error is generated when > >> > > > > > >> you try to send a request for a transaction that was already cancelled > >> > > > > > >> from the client side. So probably the PSTN GW cancels the call while you > >> > > > > > >> still want to fork a new branch. > >> > > > > > >> regards, > >> > > > > > >> Bogdan > >> > > > > > >> Sergey Babichev wrote: > >> > > > > > >>> Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > >>> > > > > > > > > > > > > > > > > > >>> > >>> > > > > > >>>> Hi all! > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> What is problem may be? > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> -- > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Best regards, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Sergey Babichev > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> _______________________________________________ > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Users mailing list > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Users(a)lists.openser.org > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>> > > > > > >>>> > >>>> > > > > > > > > > > > > > > > > > >>> -- > >>> > > > > > > > > > >>> Best regards, > >>> > > > > > > > > > >>> Sergey Babichev > >>> > > > > > > > > > >>> _______________________________________________ > >>> > > > > > >>> Users mailing list > >>> > > > > > >>> Users(a)lists.openser.org > >>> > > > > > >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>> > > > > > > > > > >>> > >>> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > -- Best regards, Sergey Babichev

2 1

[OpenSER-Users] Strating up with openser
by J Bhanot 22 Jan '08

22 Jan '08

Hi, I am just starting up with this stack.. Could you kindly let me know what strategy i should follow so as to use this stack for building applications rapidly. Thanking in anticipation. Regards, Jagmohan Bhanot Tata Consultancy Services Mailto: j.bhanot(a)tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[OpenSER-Users] cdr records in openser - how to get them
by tloginbr-openserbr＠yahoo.com.br 22 Jan '08

22 Jan '08

Hi all, I'm pretty new to OpenSER, and I'm trying to get it working here. I have one OpenSER machine and a few asterisk machines behind this one. All my call go the OpenSER that does load balancing and fail over between my asterisk servers. Up to now I was getting the CDR from all my asterisk, but that is not going very well since I have to search in all asterisk machines to get the cdr for one extension. So I want to get the cdr directly from the openser thats where all calls have to pass. I found some people talking about CDRTool, but I think it does a little too much for me. I'm just looking for some tool that saves the information I need inside mysql so I can build my cdr report with some simple sql queries. Does anyone in here get the cdr out of the openser or is that not even possible? All help is very welcome here. Thanks in advance, Thiago Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

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Re: [OpenSER-Users] Carrierroute help..
by Henning Westerholt 22 Jan '08

22 Jan '08

On Tuesday 22 January 2008, Ricardo Martinez wrote: > Hello. > I would like to make a question about the carrierroute module. > How does it work when one of the "level 0" gw's are down?. For example , > this is my carrierroute table : > > +----+---------+-------------+--------+------+-------+---------------+ > | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | > +----+---------+-------------+--------+------+-------+---------------+ > | 1 | 1 | 49 | 0 | 0.3 | 0 | de-1.carrier1 | > | 2 | 1 | 49 | 0 | 0.3 | 0 | de-2.carrier1 | > | 3 | 1 | 49 | 0 | 0.3 | 0 | de-3.carrier1 | > | 4 | 1 | | 1 | 1 | 0 | defa.carrier1 | > +----+---------+-------------+--------+------+-------+---------------+ > > > > 1- Let's suppose "de-1.carrier1" gateway is down. So.. any INVITE to > that gateway will not have any answer (for the failure_route failover), how > the module handle this?, It tries with the others+.? Hi Ricardo, no, at the moment there is no automatically failover done from the carrierroute module. So if you want to route your calls to another gw, e.g. 'de-2', you must manual re-route it to another gw/ domain in the failure_route block. E.g. route to domain 0 in route[..], route to domain 1 in failure_route. > 2.- Now, suppose that "de-1.carrier1" gateway is only full, so i can > probably have a reply from the gateway (maybe a "480" message"), so the > carrierroute module now tries with a failover route (defa.carrier1?) or > tries with any of the other two gateways still not full? I suppose you mean that 'de-1' is overloaded. No, the next domain is not automatically entered. This deficiency is known, and will be addressed in the future. But this code is not ready yet for a release. You could use the lcr module, there exist a 'next_gw' function, if you don't want to manually specify the failure_routes. Cheers, Henning

1 0

Re: [OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > Hi Sergey, > the "discarding fwd for a cancelled transaction" error is generated when > you try to send a request for a transaction that was already cancelled > from the client side. So probably the PSTN GW cancels the call while you > still want to fork a new branch. > regards, > Bogdan > Sergey Babichev wrote: > > Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > > > > > > > > > >> Hi all! > >> > > > > > >> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >> > > > > > >> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >> > > > > > >> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >> > > > > > >> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >> > > > > > >> What is problem may be? > >> > > > > > >> -- > >> > > > > > >> Best regards, > >> > > > > > >> Sergey Babichev > >> > > > > > >> _______________________________________________ > >> > > > > > >> Users mailing list > >> > > > > > >> Users(a)lists.openser.org > >> > > > > > >> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > -- Best regards, Sergey Babichev

2 1

[OpenSER-Users] SCTP enabled stack
by J Bhanot 22 Jan '08

22 Jan '08

Hi, I am trying to build the stack with SCTP enabled... I am doing following SCTP=1 make all but I am getting error - In file included from forward.h:49, from action.c:49: sctp_server.h:32:26: error: netinet/sctp.h: No such file or directory make: *** [action.o] Error 1 I am using FC8 - have inserted the SCTP module using /sbin/modprobe sctp without any error.. To overcome this error.... I chaged the absolute path of sctp.h in sctp_server.h to #include "/usr/src/kernels/2.6.23.1-42.fc8-i686/include/net/sctp/sctp.h"*/ instead of #include <netinet/sctp.h> but even then i am getting many erros of redeclration of many variables and files not being found in sctp.h Kindly suggest what should be done so that server can work with SCTP. Thanks, jb __________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

4 7

[OpenSER-Users] register two AOR
by Victor Gamov 22 Jan '08

22 Jan '08

Hi gentlemens! Is it possible to set modparam("registrar", "aor_avp", "$avp(s:sip-aor)") and call save("location") to save AOR from $avp(s:sip-aor) If not, how I can register two AOR when REGISTER message processed? -- CU, Victor Gamov

2 2

[OpenSER-Users] Launch of Soap-Simple-Proxy version 1.0.2
by Adrian Georgescu 22 Jan '08

22 Jan '08

Hello, This software exposes a simple to use SOAP interface for the OpenXCAP server available at http://openxcap.org It is a transparent proxy for SOAP to XCAP requests relying on XCAP server for authenticating requests. Applications ------------ You may use the provided functions to: 1. Publish presence information (using pidf-manipulation application) 2. Set the presence policy for a given presentity (pres-rules application) 3. Retrieve the OpenSER watchers for a given presentity (watchers application) The software can be used as: - Stand-alone data publisher for persistent Presence documents. - Policy manager. You can build a web interface to manage the pres-rules documetn and use it in combination with Eyebeam-lite as a complete SIP SIMPLE client. - To retreive and display the active watchers from OpenSER in a web interface. Usage ----- To use the software you must use a SOAP client. First you must generate the client library from the supplied WSDL file using the SOAP tools available in your programming language. The SOAP functions usage and syntax are self explanatory. Available functions are described in the wsdl/ssp.wsdl file. An example of PHP client library is provided with CDRTool version 6 or higher, the soap client functions are available in the file: /var/www/CDRTool/provisioning/ngnpro_soap_library.phtml in the class WebService_SoapSIMPLEProxy_PresencePort Support ------- The support is provided by AG Projects on a best effort basis. Report problems via the ticketing system from openxcap.org or openser users mailing lists. License ------- Soap-Simple-Proxy is licensed under GNU PUBLIC LICENSE version 2.0. The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

1 0

[OpenSER-Users] New OpenXCAP release 0.9.9
by Adrian Georgescu 22 Jan '08

22 Jan '08

Hello, There is a new OpenXCAP release available. Changes in version 0.9.9 ------------------------ * Fixed node selector decoding * Save presence rules documents under the same file name ("index.xml") * Moved trusted_peers to [Authentication] config section * Default xcap_table to "xcap" in config.ini.sample * Improved URI logging * DELETE of a full document doesn't produce an Etag in the 200 response * Added SIP Thor backend Changes in version 0.9.8 ------------------------ * Support subdirectories by using doc_uri column in xcap table * Updated MySQL xcap table structure The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

1 0

[Serusers] SER service: registration problem (load balancing quirk?)
by Tom van der Geer 22 Jan '08

22 Jan '08

Dear SER users, I'm experiencing occasional problems with my SIP clients registration to the (free) iptel.org SER service. I'm not sure if this is the place to ask questions regarding this service. If not, please give me a pointer who/what I should contact. I've made 2 tcpdumps. 1 from a failing registration request and from a successful registration request. The setup for both situations is the same. These captures were made within 40 minutes from each other. In both cases the first registration attempt is replied with a 401 Unauthorized (obviously) which request for authentication. In both cases my SIP client (SJPhone) replies correctly (I assume) with a new registration and the requested account credentials. In the case of a successful registration the SER replies with a 200 OK. In the case of a failing registration my SIP client receives another 401 Unauthorized message. What struck me is that this second 401 is sent from another SER system. (The "Server:" header from the first registration attempt is different from the second registration attempt.) The first header has "Server: Sip EXpress router (2.1.0-dev16-tcp (i386/linux))" while the second 401 has "Server: Sip EXpress router (2.0.0-rc5 (i386/linux))". (note the difference in SER versions) I've verified that in the case of a successful registration the Server header in the first (401) and second (200) response are the same: "Server: Sip EXpress router (2.1.0-dev16-tcp (i386/linux))" Is this some kind of weird load balancing quirk? From a security point of view I don't want to send my TCP Dumps (with my authentication) to the mailing list, but I can send it to a personal email address when required. Thanks, Tom

2 1

Re: [OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > Hi all! > I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > What is problem may be? > -- > Best regards, > Sergey Babichev > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users -- Best regards, Sergey Babichev

2 1

[OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Hi all! I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 What is problem may be? -- Best regards, Sergey Babichev

1 0

[OpenSER-Users] Why RPID related functions instead of PAI?
by Iñaki Baz Castillo 21 Jan '08

21 Jan '08

Hi, "Remote-Party-ID" is a draft ("draft-ietf-sip-privacy-04") [1] expired in 2002 while "P-Asserted-Identity" is a official RFC (3325) [2]. I'd like to know why OpenSer manages RPID in some functions ("auth" module) instead of PAI. I assume that some gateways just implement RPID and so, but can't understand why PAI is ignored (the use of PAI is extended in gateways). As a curiosity, both draft and RFC have really **common** parts, as chapter "Introduction" (4 in draft, 3 in RFC): "3/4 Introduction Various providers offering a telephony service over IP networks have selected SIP as a call establishment protocol. Their environments require a way for trusted network elements operated by the service providers (for example SIP proxy servers) to communicate the identity of the subscribers to such a service, yet also need to withhold this information from entities that are not trusted when necessary. Such networks typically assume some level of transitive trust amongst providers and the devices they operate. These networks need to support certain traditional telephony services and meet basic regulatory and public safety requirements. These include Calling Identity Delivery services, Calling Identity Delivery Blocking, and the ability to trace the originator of a call ... " So my question is again:why OpenSer supports RPID related functions and not PAI functions? Thanks for any explanation. [1] RPID: http://tools.ietf.org/html/draft-ietf-sip-privacy-04 [2] PAI: http://www.faqs.org/rfcs/rfc3325.html -- Iñaki Baz Castillo

3 3

[OpenSER-Users] Openser + Mediaproxy not working with 2 domains
by Andreti 21 Jan '08

21 Jan '08

Hi everybody, I'm working with Openser + Mediaproxy 1.9.0 and it seems that everything is working when the calls are establised between users attached to the same proxy server, even with different kind of NATs. However It doen't work in 2 different scenarios, and the result is exactly the same , the video and audio is only sent in one way. Scenario 1 ======== User A attached to the SIP proxy xxx.xxx.xxx.13 (Public IP) calls to a GW xxx.xxx.xxx.11 (Public IP) with several users internally associated. In this case the user A can see the video and audio sent by the GW, but the GW doesn't receive any RTSP stream. It seems that the mediaproxy doesn't do anything, why? maybe because the GW blongs to other domain (xxx.xxx.xxx.11) ? What can I do? If the GW calls to user A, it works fine (I can see the session in the mediaproxy with sessions.py) Scenario 2 ======== In this case, I have another GW with Public IP address xxx.xxx.xxx.14, but it doesn't include in the INVITE message the SDP body. The GW calls to the same user attached to the SIP proxy xxx.xxx.xxx.13 , and the behaviour is exactly the same as scenario 1, the calling site can sse the video and audio but the called can't. Unlike the previous scenario, the signalling is: INVITE without SDP --> 200 OK (SDP) -- > ACK (SDP) In theory, Mediaproxy 1.9.0 should support this procedure since it's a SIP standard mechanism, however the called party doesn't receive RTP stream. In my opinion, the problem could be related to scenario 1, I mean , the calling party is not attached to the SIP proxy (belongs to other domain) and when the 200 OK (SDP) message arrives to the SIP proxy, the mediaproxy doesn't do anything Sorry for the complex explanation. I've waste a lot of time trying to solve this solution and honestly I don't know what to do. Please, could somebody help?? I attach my openser.conf. I hope it helps. Andreti debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes listen=xxx.xxx.10.12 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- mpath = "/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "uri_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" loadmodule "nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 20) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) #cambio 16_04_07 modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") #puesto para el CDRTool modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) #modparam("acc", "radius_config", "/etc/openser/radius/client.conf") modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp;\ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp($can_uri); \ Billing-Party=$avp($billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$avp(s:rtp_statistics); \ From-Header=$hdr(from); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event)") # SIP-Proxy-IP=$avp(s:sip_proxy_ip)") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") modparam("avpops","avp_aliases","can_uri=i:34") modparam("avpops","avp_aliases","billing_party=i:1") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # setflag(ACCOUNTING_FLAG); # avp_write("SER_IP","$avp(s:sip-proxy)"); # avp_write("$ru", "$avp(can_uri)"); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); setflag(1); } route(1); }; if (src_ip==193.36.177.227) { fix_nated_sdp("2"); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (method=="MESSAGE") { setflag(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("sip.com", "subscriber")) { # www_challenge("sip.com", "0"); # exit; # }; if (!radius_www_authorize("")) { www_challenge("","1"); exit; } if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; }; if (method=="INVITE") { t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri=="sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; t_relay(); return; }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri=="sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; append_hf("P-hint: usrloc applied\r\n"); # route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } -- View this message in context: http://www.nabble.com/Openser-%2B-Mediaproxy-not-working-with-2-domains-tp1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] Calls disconnect automatically
by VoIP Forums www.Go4Calls.com 21 Jan '08

21 Jan '08

Hi Friends, I start getting one problem, the calls disconnect automatically in 30 and 32 sec. I am using openser + rtpproxy before with the same openser.cfg it was running smoothly and once traffic increased this problem appeared. Could you please help me to solve this issue because i put openser in production and now no one can make long call. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

Re: [OpenSER-Users] xlite with openser / hello world posts SOLVED
by A.smith 21 Jan '08

21 Jan '08

Hi Moujane, In case you haven´t worked it out yet, I managed to get an openser.cfg to work for simple testing between XLite or other SIP client. Its a shame there isnt a working config easily findable on the openser site or other internet resource for those people new to SIP and OpenSER.... :S Try this, for me it works on OpenSER 1.2.2 # SECTION 1 - GLOBAL DEFINTIONS debug=3 fork=yes log_stderror=no #fork=no #log_stderror=yes listen=10.10.10.10 #put your server IP here port=5060 disable_dns_blacklist=yes dns=no rev_dns=no children=4 # SECTION 2 - MODULE DEFINTION mpath="/usr/local/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "registrar.so" # SECTION 3 - MODULE CONFIGURATION modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # main routing logic FOR RADIUS ACC ACCOUNTING route{ if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (method=="REGISTER") { record_route(); if (!save("location")) { sl_reply_error(); }; exit; }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # SECTION 5 - SECONDARY ROUTE BLOCKS route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } ________________________________________________ Message sent using UK Grid Webmail 2.7.9

3 3

[OpenSER-Users] Adding "Reason: SIP ; cause=200" header when CANCEL in forking scenarios
by Iñaki Baz Castillo 21 Jan '08

21 Jan '08

Hi, according to RFC 3326 [1] (Reason Header): 3.1 Call Completed Elsewhere A proxy forks an INVITE request and one of the branches returns a 200 (OK). The forking proxy includes this status code in a Reason header field in the CANCEL request that it sends to the rest of the branches. Reason: SIP ;cause=200 ;text="Call completed elsewhere" With this behaviour, if an AoR is registered in two phones and one of them answers a call, that call will not appear as MISSED in the other phone (if the phone supports "Reason" headers). Is it possible OpenSer to add this header when cancelling the rest of ringing branches? If not, do you think it could be a feature request? [1] http://www.faqs.org/rfcs/rfc3326.html -- Iñaki Baz Castillo

3 2

[OpenSER-Users] Routing logic Check
by SP GLE 21 Jan '08

21 Jan '08

Hi, Is there any tool to validate an OpenSer Routing Configuration Logic ? We would like to validate a configuration to avoid regressions on routing behaviour. Regards. _____________________________________________________________________________ Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail http://mail.yahoo.fr

2 1

Re: [OpenSER-Users] Calls disconnect automatically
by VoIP Forums www.Go4Calls.com 21 Jan '08

21 Jan '08

Hi sorry, I forget to give my openser.cfg, there is one more point if i am using STUN server in our linksys device the call goes normal for long time till user finish the call. It seems something wrong in NAT configuration. # # sample config file to be used with nathelper/rtpproxy # # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=212.XXX.XXX.XXX:5064 #port=5064 children=4 disable_dns_blacklist = yes # --- module loading mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("usrloc|auth_db","db_url","mysql://proxy:MoHaKa21@192.168.1.50/emafone") # -- usrloc params -- modparam("usrloc", "nat_bflag", 6) # -- registrar params -- modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # -- auth params -- # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") #modparam("nathelper", "rtpproxy_sock", "udp:212.100.235.229:22222") #modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") #modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@openser.org") # --- main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { #append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("212.XXX.XXX.XXX", "subscriber")) { www_challenge("212.XXX.XXX.XXX", "0"); exit; }; if (isflagset(5)) { setbflag(6); # if you want OPTIONS natpings uncomment next # setbflag(7); }; save("location"); exit; }; #########PSTTN CALL ############################################# if (uri=~"sip:00[1-9][0-9]+@.*") { strip(2); rewritehostport("212.XXX.XXX.XXX:5060"); route(1); exit; }; ################################################################### if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(6); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; } Regards, www.Go4Calls.Com VoIP Forums From: go4calls(a)hotmail.com To: users(a)lists.openser.org; users(a)openser.org Subject: Calls disconnect automatically Date: Sun, 20 Jan 2008 18:01:48 +0800 Hi Friends, I start getting one problem, the calls disconnect automatically in 30 and 32 sec. I am using openser + rtpproxy before with the same openser.cfg it was running smoothly and once traffic increased this problem appeared. Could you please help me to solve this issue because i put openser in production and now no one can make long call. Regards, www.Go4Calls.Com VoIP Forums Express yourself instantly with MSN Messenger! MSN Messenger _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[OpenSER-Users] hor to register ?
by Aaash shah 21 Jan '08

21 Jan '08

hi i am making project on SIP handoffs i already install openSER on fedora i dont know how to register it so please give me reply thanks -- "Never Ignore Anyone Who Cares 4 u, Becoz one day u will Realize that u have lots a diamond while u were buzy in collecting stones...."

1 0

[Serusers] lookup_contacts for natted clients returns local ip
by Tomasz Zieleniewski 21 Jan '08

21 Jan '08

Hi, I have the followning natted contact in my location table: mysql> select * from location where uid ='tw(a)sip.touk.pl'; +--------------------+-------+-----------------------------------------------------------+ | contact | flags | received | +--------------------+-------+-----------------------------------------------------------+ | sip:tw@192.168.7.2 | 1 | sip:80.53.110.142:61066;dstip=192.168.129.74;dstport=5060 | +--------------------+-------+-----------------------------------------------------------+ when I invoke lookup_contacts the ruri has local address 192.168.7.2instead of public one. Is this correct behaviour? Please point me is this should work as I think. Thank You for any help Tomasz

2 3

[OpenSER-Users] Obtaining the domain of "Refer-to" URI
by Iñaki Baz Castillo 21 Jan '08

21 Jan '08

Hi, in order to match the domain of "Refer-to" header I must do a dirty string substitution: $avp(s:rt) = $rt; avp_subst("$avp(s:rt)", "/(.*(a)|[;?].*)//gi"); If for example: Refer-To: <sip:bob@biloxi.example.net?Accept-Contact=sip:bobsdesk. biloxi.example.net&Call-ID%3D55432%40alicepc.atlanta.example.com> Then $avp(s:rt) is "biloxi.example.net". Couldn't be possible to have generic string transformations to get part of a URI in any header/variable? Is there any other way I don't know to get it easily? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] [OT] Annoying empty lines when using "M4 Macro Processor"
by Iñaki Baz Castillo 20 Jan '08

20 Jan '08

Hi, as suggested here: http://www.openser.org/docs/scripting I'm using M4 to declare "define" in my OpenSer script. I've an annoying issue since M4 adds some empty lines at the top of the script. If my script has any error, the logs will show me "ERROR in line 235 in openser.cfg" but that line correspondes to line 187 in my openser.cfg.m4 file, so makes it very difficult to debug. I've read the entire doc and have no idea of how to fix it. For example, consider this simple case: ---- local.m4 ---- define(`IP',`127.0.0.1') define(`PORT' ,`5061') ----------------------- ---- file.txt.m4 ---- My ip is IP and my port is PORT. -------------------------- ~# m4 local.m4 file.txt.m4 > file.txt ---- file.txt ---- <-- Empty line <-- Empty line <-- Empty line <-- Empty line My ip is 127.0.0.1 and my port is 5061. --------------------- Does someone know how to solve it? Thanks a lot. -- Iñaki Baz Castillo

2 5

[OpenSER-Users] mysql error shuts down openser 1.3
by Christian Schlatter 20 Jan '08

20 Jan '08

Hi all, Is it normal that a simple mysql error causes the worker process to die which in turn shuts down all other openser processes? If I remember correctly, this never happened to me with the 1.2 version. The following log comes from an 1.3 installation: [30812]: ERROR:mysql:db_mysql_submit_query: driver error: Can't lock file (errno: 4009) [30812]: ERROR:mysql:db_mysql_raw_query: error while submitting query [30812]: ERROR:usrloc:get_all_db_ucontacts: raw_query failed [30792]: INFO:core:handle_sigs: child process 30812 exited by a signal 11 [30792]: INFO:core:handle_sigs: core was not generated [30792]: INFO:core:handle_sigs: terminating due to SIGCHLD [30797]: INFO:core:sig_usr: signal 15 received [30814]: INFO:core:sig_usr: signal 15 received [30798]: INFO:core:sig_usr: signal 15 received [30800]: INFO:core:sig_usr: signal 15 received [30802]: INFO:core:sig_usr: signal 15 received [30804]: INFO:core:sig_usr: signal 15 received [30806]: INFO:core:sig_usr: signal 15 received [30808]: INFO:core:sig_usr: signal 15 received Jan 14 12:54:45 proxy-2 ser[30810]: INFO:core:sig_usr: signal 15 received /Christian

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[OpenSER-Users] [OT] Nortel gateway allows "tel:1234@host" but not "tel:1234" ¿?¿?¿
by Iñaki Baz Castillo 20 Jan '08

20 Jan '08

Hi, I'm trying "P-Asserted-Identity" header against a Nortel CS2000_NGSS/8.0. It works (the number in PAI header is shown as PSTN callerid number) in those cases: P-Asserted-Identity: <sip:+34666555444@host/IP> P-Asserted-Identity: <tel:+34666555444@host/IP> As you can see, the second case is wrong since a "tel" uri has no host part according to RFC 3966 [1]. And the worst thing is that Nortel disallows or doesn't understand a correctly formed "tel" uri in PAI header as this: P-Asserted-Identity: <tel:+34666555444> My experience with gateways is very limited and new, but I should assume that it is a sad bug in the Nortel SIP stack, is it? Thanks for any help. [1] RFC 3966: http://www.faqs.org/rfcs/rfc3966.html -- Iñaki Baz Castillo

1 1

[Serusers] opensource prepaid system
by Nhadie 19 Jan '08

19 Jan '08

Hi, Has anyone tested an opensource prepaid system to work on a ser/freeradius/asterisk-b2bua/asterisk setup? TIA Regards, Nhadie

1 0

[OpenSER-Users] unixodbc (ARRGH!)
by Douglas Garstang 19 Jan '08

19 Jan '08

Can someone tell me what the correct syntax is for the avp_url command? I've been at this for hours and it's driving me CRAZY! this: modparam("avpops","avp_url","unixodbc://user:password@127.0.0.1/database") gives: unixodbc:extract_error: unixodbc:SQLDriverConnect=IM002:1:0:[unixODBC][Driver Manager]Data source name not found, and no default driver specified this: modparam("avpops","avp_url","unixodbc://dsn" ) gives: core:new_db_id: error while parsing database URL: unixodbc://dsn and I've tried about a HUNDRED other combinations. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[OpenSER-Users] Problems with NAPTR records and failover
by Robert McGilvray 19 Jan '08

19 Jan '08

Greetings list, I'm trying to setup uri rewriting and failover using NAPTR records. I've gone wrong somewhere since instead of trying the first uri and only failing over if there's a transactional and/or communications problem it's creating two branches and doing parallel forking on the records. Example: >From my test SIP phone I dial 3759. Openser receives this INVITE and does a private ENUM lookup on it which returns two NAPTR records: 9.5.7.3 NAPTR 10 10 "u" "E2U+sip" "!^.*$!sip:3759@mygateway1.whoever.com!" . NAPTR 20 10 "u" "E2U+sip" "!^.*$!sip:7100@myfailovergateway1.whoever.com!" . >From what I understand this should first try to contact my gateway (a cisco box) using a uri of sip:3759@mygateway1.whoever.com. If this fails it should then take the second higher order and rewrite the URI to sip:7100@myfailovergateway1.whoever.com. Instead what I end up with is two parallel calls to gateway and failover, the second NAPTR record is a conference bridge so the cisco immediately sends back a 200 OK and openser cancels the first branch. First gateway barely gets off a ring to my phone. I haven't adjusted any of the SIP timers so the defaults should allow plenty of time for call setup etc. Debug output from the call is in http://www.idled.net/~bobm/openser-debug.txt . Am I handling the INVITE incorrectly in my config? Any assistance would be appreciated. route[1] { if (!t_relay()) { xlog("L_NOTICE", "sipprx-wpf1.globeop.com: $mi route[$rm][0] $fu -> $ru -> Error with t_relay\n"); sl_reply_error(); } exit; } INVITE routing block route[6] { sl_send_reply("100", "Trying"); if (isflagset(1)) { xlog("L_NOTICE", "sipprx-wpf1.globeop.com: $mi route[$rm][0] $fu -> $ru Processing INVITE\n"); } if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); if ((uri =~ "sip:[0-9]{4}@.*") || (uri =~ "sip:[0-9]{10}@.*")) { prefix ("+"); if (enum_query("e164.globeop.com")) { xlog("L_INFO", "sipprx-wpf1.globeop.com: Number found in ENUM, new uri is $ru\n"); } else { xlog("L_INFO", "sipprx-wpf1.globeop.com: Number not found in ENUM\n"); revert_uri(); } } if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); route(1); } else if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); route(1); } else { sl_send_reply("404","Not found"); exit; } exit; } -- Bob This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediatley delete this email in its entirety and any attachments thereto.

1 1

[OpenSER-Users] Sippy b2bua
by TCB 18 Jan '08

18 Jan '08

Anyone managed to setup sippy b2bua . Pointers most welcome. I have tried setting it up and this is as far as i have gone. I have done the necessary radius config in my /etc/radiusclient-ng/{ radiusclient.conf,servers}. When i send a call from my proxy i get 403 (Auth Failed) Find attached the log file . regards -- TC

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[OpenSER-Users] Sippy b2bua
by TCB 18 Jan '08

18 Jan '08

Anyone suceeded in implementing sippy b2bua. Pointers most welcome. -- TC

1 0

[OpenSER-Users] ERROR: presence_xml:get_final_notify_body?
by KevinKinnan 18 Jan '08

18 Jan '08

Did I miss something on configuration? or it simply does not send NOTIFY message when changes occur in the published data after integrating OpenXCAP? Thx in advance! Kevin _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

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[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 18 Jan '08

18 Jan '08

Did I miss something on configuration? or it simply does not send NOTIFY message when changes occur in the published data after integrating OpenXCAP? Thx in advance! Kevin _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

1 0

[OpenSER-Users] OpenSER: Table "subscriber", only one new user can be added?
by KevinKinnan 18 Jan '08

18 Jan '08

Only one new entry can be added in the "subscriber" table. Or, "Duplicate entry '' for key 3" error responses. Is there any way to create more than one user in this table? Kevin, _________________________________________________________________ MSN圣诞礼物火热登场，免费发放中，快来领取吧！ http://im.live.cn/emoticons/?ID=18

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[OpenSER-Users] OpenXCAP application support
by Richard Good 18 Jan '08

18 Jan '08

Hi, I am using the OpenXCAP implementation in co-ordination with OpenSER. Is there any way to support any (or at least a few more) applications than those currently supported by OpenXCAP? I previously used XCAP-Lite and stored various XML documents ranging from address books and TV guides to privacy and domain policies. I've found that with OpenXCAP if you use any application not supported ( anything besides resource-lists, pidf-manipulation, pres-rules, etc.) the Document Selector is not recognised and the documents can not be uploaded or downloaded. Is there a way to bypass this? Apologies if this is not the correct mailing list - the OpenXCAP page points to this list. Thanks and Regards, Richard

1 0

[OpenSER-Users] Help with this mediaproxy + Openser nat problem... thankssss
by Mario Bedialauneta 17 Jan '08

17 Jan '08

Hi everybody, I'm working with Openser + Mediaproxy 1.9.0 and it seems that everything is working when the calls are establised between users attached to the same proxy server, even with different kind of NATs. However It doen't work in 2 different scenarios, and the result is exactly the same , the video and audio is only sent in one way. Scenario 1 ======== User A attached to the SIP proxy xxx.xxx.xxx.13 (Public IP) calls to a GW xxx.xxx.xxx.11 (Public IP) with several users internally associated. In this case the user A can see the video and audio sent by the GW, but the GW doesn't receive any RTSP stream. It seems that the mediaproxy doesn't do anything, why? maybe because the GW blongs to other domain (xxx.xxx.xxx.11) ? What can I do? If the GW calls to user A, it works fine (I can see the session in the mediaproxy with sessions.py) Scenario 2 ======== In this case, I have another GW with Public IP address xxx.xxx.xxx.14, but it doesn't include in the INVITE message the SDP body. The GW calls to the same user attached to the SIP proxy xxx.xxx.xxx.13 , and the behaviour is exactly the same as scenario 1, the calling site can sse the video and audio but the called can't. Unlike the previous scenario, the signalling is: INVITE without SDP --> 200 OK (SDP) -- > ACK (SDP) In theory, Mediaproxy 1.9.0 should support this procedure since it's a SIP standard mechanism, however the called party doesn't receive RTP stream. In my opinion, the problem could be related to scenario 1, I mean , the calling party is not attached to the SIP proxy (belongs to other domain) and when the 200 OK (SDP) message arrives to the SIP proxy, the mediaproxy doesn't do anything Sorry for the complex explanation. I've waste a lot of time trying to solve this solution and honestly I don't know what to do. Please, could somebody help?? I attach my openser.conf. I hope it helps. Andreti debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes listen=xxx.xxx.10.12 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- mpath = "/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "uri_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" loadmodule "nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 20) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) #cambio 16_04_07 modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") #puesto para el CDRTool modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) #modparam("acc", "radius_config", "/etc/openser/radius/client.conf") modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp;\ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp($can_uri); \ Billing-Party=$avp($billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$avp(s:rtp_statistics); \ From-Header=$hdr(from); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event)") # SIP-Proxy-IP=$avp(s:sip_proxy_ip)") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") modparam("avpops","avp_aliases","can_uri=i:34") modparam("avpops","avp_aliases","billing_party=i:1") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # setflag(ACCOUNTING_FLAG); # avp_write("SER_IP","$avp(s:sip-proxy)"); # avp_write("$ru", "$avp(can_uri)"); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); setflag(1); } route(1); }; if (src_ip==193.36.177.227) { fix_nated_sdp("2"); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (method=="MESSAGE") { setflag(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("sip.com", "subscriber")) { # www_challenge("sip.com", "0"); # exit; # }; if (!radius_www_authorize("")) { www_challenge("","1"); exit; } if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; }; if (method=="INVITE") { t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri==" sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; t_relay(); return; }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri==" sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; append_hf("P-hint: usrloc applied\r\n"); # route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

1 0

[OpenSER-Users] ERROR: presence_xml:get_final_notify_body?
by KevinKinnan 17 Jan '08

17 Jan '08

The debug log in 9 level is attached. K, _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

Re: [OpenSER-Users] openSER Call Route Logic
by Patrick Baker 17 Jan '08

17 Jan '08

Sipwise.com might help ----- Original Message ----- From: users-bounces(a)lists.openser.org <users-bounces(a)lists.openser.org> To: users(a)lists.openser.org <users(a)lists.openser.org> Sent: Wed Jan 16 23:48:27 2008 Subject: [OpenSER-Users] openSER Call Route Logic Hi, I recently installed openser-1.3.0 and able to connect my cisco sip phones and everything is working. Now i need to integrate this with asterisk box for call forwarding to other locations & voicemails. i read the getting started guide and other information, but seems i need more help from the group. can any one give me a helping hand. i am sure i will do my home work before bug you again. I need a starting point to over come the complexity of the routing plan and logic. thanks -asterisk user ________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. <http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/c…>

3 3

Re: [OpenSER-Users] MySQL Alternative (help!)
by Douglas Garstang 17 Jan '08

17 Jan '08

Henning, Let me try asking my question another way. Is it possible to use avp_db_query() with UnixODBC? If so, how? Douglas. ----- Original Message ---- From: Henning Westerholt <henning.westerholt(a)1und1.de> To: users(a)lists.openser.org Cc: Douglas Garstang <dougmig33(a)yahoo.com> Sent: Thursday, January 17, 2008 1:15:09 AM Subject: Re: [OpenSER-Users] MySQL Alternative On Thursday 17 January 2008, Douglas Garstang wrote: > What other options do I have for generic database connectivity besides > MySQL/avp_db_query() in OpenSER? > > Can I use SQLite? Can I use unixodbc? (and I'm not talking about the > limited built in usrloc unixodbc stuff) Hi Douglas, don't know what you mean with the builtin ODBC stuff in usrloc.. I assume that you talking about connectivity options for OpenSER. Available database modules that can be used with most modules: MySQL PostgreSQL UnixODBC (should work with Oracle, MS-SQL, SQLite and other) dbtext db_berkeley (Berkeley DB) Interfaces roughly sorted in terms of maturity, stability.. Most people uses MySQL, so this is the best tested option. Cheers, Henning ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[OpenSER-Users] Duplicated entries in "location" table when using "db_mode 3"
by Iñaki Baz Castillo 17 Jan '08

17 Jan '08

Hi, I use ("usrloc", "db_mode", 3) since some months and I've realized that many times I get duplicated entries in location table. Those entries have **exactly** the same values in all the columns. Note that is a testing serer which I restart 10 or 20 times each hour XD and that I've upgraded from 1.2 to 1.3 getting always this issue. If I send a new REGISTER (refresh) for an AoR with duplicated entries both entries are updated according to the new "expires" and CSEQ. Is it normal? -- Iñaki Baz Castillo

1 0

Re: [OpenSER-Users] best response seems wrong.
by Aymeric Moizard 17 Jan '08

17 Jan '08

(sorry: just repost to follow correct thread/subjet) I just found another example from rfc3261 where an endpoint will use 5xx class answer: 12.2.2 UAS Behavior Page 76: If the remote sequence number is empty, it MUST be set to the value of the sequence number in the CSeq header field value in the request. If the remote sequence number was not empty, but the sequence number of the request is lower than the remote sequence number, the request is out of order and MUST be rejected with a 500 (Server Internal Error) response. Hope I convinced you! tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ On Wed, 16 Jan 2008, Iñaki Baz Castillo wrote: > On Wednesday 16 January 2008 13:44:10 A.smith wrote: > >> If I try and use the whole example cfg (not just the route) I am recieving >> an error like: >> >> 0(23448) find_cmd_export_t: <lookup> not found >> 0(23448) parse error (105,23-24): unknown command, missing loadmodule? > > "lookup()" is defined in "registrar" module that you are not loading: > http://www.openser.org/docs/modules/1.2.x/registrar.html#AEN342 > > -- > Iñaki Baz Castillo > ibc(a)in.ilimit.es > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] xlite with openser / hello world posts
by A.smith 17 Jan '08

17 Jan '08

1 0

[OpenSER-Users] Presence + XCAP issue
by Leonard Broman 17 Jan '08

17 Jan '08

Hi! I have an issue with the presence/xml/xcap modules. I'm trying to run the presence server without an "integrated" xcap server, since I have not managed to get openXCAP running on CentOS. However, for the moment I'd prefer not to be depending on it. I suppose this might be the root of my problem, as follows. Publish is working fine. But Subscribe is always answered with a 481 (which shouldn't occur normally) and I get this log: Jan 17 10:57:16 m2lin3 ./openser[7097]: ERROR:presence:get_stored_info: record not found in hash_table Jan 17 10:57:16 m2lin3 ./openser[7097]: ERROR:presence:handle_subscribe: getting stored info My primary question is wheather the this error is caused by the absence of documents in the xcap table? And if so can I do something to avoid this dependence? /Leo

1 0

[OpenSER-Users] openser 1.4 blocks
by Helmut Kuper 17 Jan '08

17 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, when I send the following SIP-Message on my openser, openser blocks and eats nearly all CPU power it can get: BYE sip:80.228.245.138:5070 SIP/2.0 Via: SIP/2.0/UDP 212.6.95.88:1028;branch=z9hG4bK-7rhl6qnns4ej;rport Route: <sip:212.6.123.118;lr=on;ftag=7w4sllx8cc> Route: <sip:212.6.123.118:5090;lr;ftag=7w4sllx8cc;vsf=ZlRqaD8wOzEbMTUlTiwzCnRNJ2F0eDBzMTZ0HVdlUg--> Route: <sip:Gk-int@80.228.245.176:5050;lr> From: "HK SoftSwitch" <sip:v101@sip0.en.ewetel.de>;tag=7w4sllx8cc To: <sip:12345@sip0.en.ewetel.de;user=phone>;tag=0082-00000054-0103i0 Call-ID: 3c26c9bec396-loo61fn1yu42 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:v101@212.6.95.88:1028;line=nopsgn9d>;flow-id=1 User-Agent: snom370/7.1.30 RTP-RxStat: Total_Rx_Pkts=120,Rx_Pkts=120,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=2263,Tx_Pkts=2263,Remote_Tx_Pkts=160 Content-Length: 0 What openser does is authentication, avp db loading an then a simple t_relay(). opensers debuglog (level 8) is this: Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:check_via_address: params 212.6.95.88, 212.6.95.88, 0 Jan 16 21:30:45 [8406] DBG:sl:run_sl_callbacks: callback id 0 entered Jan 16 21:30:45 [8406] DBG:siptrace:trace_sl_onreply_out: trace off... Jan 16 21:30:45 [8406] DBG:core:destroy_avp_list: destroying list 0xb5b00e50 Jan 16 21:30:45 [8406] DBG:core:receive_msg: cleaning up Jan 16 21:30:45 [8406] DBG:core:parse_msg: SIP Request: Jan 16 21:30:45 [8406] DBG:core:parse_msg: method: <BYE> Jan 16 21:30:45 [8406] DBG:core:parse_msg: uri: <sip:80.228.245.138:5080> Jan 16 21:30:45 [8406] DBG:core:parse_msg: version: <SIP/2.0> Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=2 Jan 16 21:30:45 [8406] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-goxekiv4ggfm>; state=6 Jan 16 21:30:45 [8406] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jan 16 21:30:45 [8406] DBG:core:parse_via: end of header reached, state=5 Jan 16 21:30:45 [8406] DBG:core:parse_headers: via found, flags=2 Jan 16 21:30:45 [8406] DBG:core:parse_headers: this is the first via Jan 16 21:30:45 [8406] DBG:core:receive_msg: After parse_msg... Jan 16 21:30:45 [8406] DBG:core:receive_msg: preparing to run routing scripts... BYE--------------------------------------------------------------------- [Wed Jan 16 21:30:45 2008] - NEW TRANSACTION! Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=78 Jan 16 21:30:45 [8406] DBG:core:parse_to_param: tag=0082-0000004f-00e7i0 Jan 16 21:30:45 [8406] DBG:core:parse_to: end of header reached, state=29 Jan 16 21:30:45 [8406] DBG:core:parse_to: display={}, ruri={sip:044180002850@sip0.en.ewetel.de;user=phone} Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: <To> [74]; uri=[sip:044180002850@sip0.en.ewetel.de;user=phone] Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: to body [<sip:044180002850@sip0.en.ewetel.de;user=phone>] Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: cseq <CSeq>: <4> <BYE> Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: start searching: hash=36300, isACK=0 Jan 16 21:30:45 [8406] DBG:tm:matching_3261: RFC3261 transaction matching failed Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: no transaction found Jan 16 21:30:45 [8406] DBG:core:parse_to_param: tag=bpl4kg8f57 Jan 16 21:30:45 [8406] DBG:core:parse_to: end of header reached, state=29 Jan 16 21:30:45 [8406] DBG:core:parse_to: display={"HK SoftSwitch"}, ruri={sip:v101@sip0.en.ewetel.de} MAIN : New Message <BYE> From: sip:v101@sip0.en.ewetel.de, To: sip:80.228.245.138:5080 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=100 Jan 16 21:30:45 [8406] DBG:maxfwd:is_maxfwd_present: value = 70 Route 22: Check type of source ... Jan 16 21:30:45 [8406] DBG:permissions:allow_source_address: looking for <1, 585f06d4, 1028> Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=800000000 Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: content_length=0 Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: found end of header Jan 16 21:30:45 [8406] DBG:core:pv_get_ppi_attr: no P-Preferred-Identity header Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 17==13 && [sip0.en.ewetel.de] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 17==13 && [sip0.en.ewetel.de] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:avpops:ops_dbquery_avps: query [SELECT uri_user FROM uri WHERE username='v101' AND domain='sip0.en.ewetel.de' LIMIT 1] Jan 16 21:30:45 [8406] DBG:avpops:db_query_avp: rows [1] Jan 16 21:30:45 [8406] DBG:avpops:db_query_avp: row [0] Jan 16 21:30:45 [8406] DBG:avpops:db_close_query: close avp query Jan 16 21:30:45 [8406] DBG:core:db_free_rows: freeing 1 rows Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[0]=0x81dde58 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: 0x81dde58=pkg_free() RES_ROWS From-URI belongs to xxxxxxxxxxx Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:auth:check_nonce: comparing [478e6aa112f6919eb75ced3104f1774c59a1ce01] and [478e6aa112f6919eb75ced3104f1774c59a1ce01] Jan 16 21:30:45 [8406] DBG:auth_db:get_ha1: HA1 string calculated: 98eb7bda25d62322c1ff8261c216e8ad Jan 16 21:30:45 [8406] DBG:auth:check_response: our result = '43c9a3529f44aa0e6dda2f6f0894632c' Jan 16 21:30:45 [8406] DBG:auth:check_response: authorization is OK Jan 16 21:30:45 [8406] DBG:core:db_free_rows: freeing 1 rows Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[0]=0x81dde88 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: 0x81dde88=pkg_free() RES_ROWS Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:comp_scriptvar: str 29 : xxxxxxxxx Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Route 04: <BYE>-Authentication OK Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Route 07: <BYE>: Loading A-Profile ... Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:db_close_query: close avp query Jan 16 21:30:45 [8406] DBG:core:db_free_rows: freeing 15 rows Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[0]=0x81de068 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[1]=0x81de070 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[2]=0x81de078 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[3]=0x81de080 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[4]=0x81de088 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[5]=0x81de090 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[6]=0x81de098 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[7]=0x81de0a0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[8]=0x81de0a8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[9]=0x81de0b0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[10]=0x81de0b8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[11]=0x81de0c0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[12]=0x81de0c8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[13]=0x81de0d0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[14]=0x81de0d8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: 0x81de068=pkg_free() RES_ROWS Jan 16 21:30:45 [8406] DBG:avpops:ops_dbload_avps: loaded avps = 15 OK Jan 16 21:30:45 [8406] DBG:core:comp_scriptvar: int 20 : 1 / 1 MAIN : <BYE>: From-Data is: User='v101', Domain='sip0.en.ewetel.de', URI='sip:v101@sip0.en.ewetel.de' Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=200 Jan 16 21:30:45 [8406] DBG:rr:is_preloaded: is_preloaded: No Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:check_self: host != me Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 13==13 && [212.6.123.118] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:rr:after_loose: Topmost route URI: 'sip:212.6.123.118;lr=on;ftag=bpl4kg8f57' is me Jan 16 21:30:45 [8406] DBG:rr:after_loose: URI to be processed: 'sip:212.6.123.118:5090;lr;ftag=bpl4kg8f57;vsf=ZlRqaD8wOzEbMTUlTiwzCnRNJ2F0eDBzMTZ0HVdlUg--' Jan 16 21:30:45 [8406] DBG:rr:after_loose: Next URI is a loose router Jan 16 21:30:45 [8406] DBG:rr:run_rr_callbacks: callback id 0 entered with <lr=on;ftag=bpl4kg8f57> DEBUG:uac:restore_from: getting 'vsf' Route param DEBUG:uac:restore_from: Route param 'vsf' not found Jan 16 21:30:45 [8406] DBG:avpops:ops_check_avp: check <17> against <8> as int /256 Jan 16 21:30:45 [8406] DBG:avpops:ops_check_avp: no match Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=200 Jan 16 21:30:45 [8406] DBG:rr:is_preloaded: is_preloaded: No Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:check_self: host != me Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 13==13 && [212.6.123.118] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:rr:after_loose: Topmost route URI: 'sip:212.6.123.118;lr=on;ftag=bpl4kg8f57' is me Jan 16 21:30:45 [8406] DBG:rr:after_loose: URI to be processed: 'sip:212.6.123.118:5090;lr;ftag=bpl4kg8f57;vsf=ZlRqaD8wOzEbMTUlTiwzCnRNJ2F0eDBzMTZ0HVdlUg--' Jan 16 21:30:45 [8406] DBG:rr:after_loose: Next URI is a loose router Jan 16 21:30:45 [8406] DBG:rr:run_rr_callbacks: callback id 0 entered with <lr=on;ftag=bpl4kg8f57> DEBUG:uac:restore_from: getting 'vsf' Route param DEBUG:uac:restore_from: Route param 'vsf' not found MAIN : Loose-Route Jan 16 21:30:45 [8406] DBG:uri:has_totag: totag found Route 08: <BYE>: Set Accounting for this call! Route 11: Removing secret Headers Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:registrar:lookup: '80.228.245.138' Not found in usrloc Route 01: Relaying BYE: RURI=sip:80.228.245.138:5080 F=sip:v101@sip0.en.ewetel.de IP=212.6.95.88; Calltimeout=<null> [Wed Jan 16 21:30:45 2008] Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Route 01: Next Hop DST-Set Jan 16 21:30:45 [8406] DBG:tm:t_newtran: transaction on entrance=(nil) Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=78 Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: start searching: hash=36300, isACK=0 Jan 16 21:30:45 [8406] DBG:tm:matching_3261: RFC3261 transaction matching failed Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: no transaction found Jan 16 21:30:45 [8406] DBG:tm:run_reqin_callbacks: trans=0xb5b01048, callback type 1, id 1 entered Jan 16 21:30:45 [8406] DBG:siptrace:trace_onreq_in: trace off... Jan 16 21:30:45 [8406] DBG:tm:run_reqin_callbacks: trans=0xb5b01048, callback type 1, id 0 entered Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=78 Any ideas, why it hangs eating all my cpu power ? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHjmra4tZeNddg3dwRAvu4AJ4rK5jCbKF9UDGu8FP1suEyRHkwigCeM58p K5a16nleRI+zZujWTI0Ntfs= =XJqP -----END PGP SIGNATURE-----

2 3

[OpenSER-Users] Compilation Problem - please help me.
by Noble Antony T 17 Jan '08

17 Jan '08

Hi, I was trying to install Openser with RLS support, I got an error message, *In file included from notify.h:31, from hash.c:37: presentity.h:36:27: libxml/parser.h: No such file or directory make[1]: *** [hash.o] Error 1 make[1]: Leaving directory `/home/nthattil/openser-1.3.x/modules/presence' make: *** [modules] Error 2 * I had installed libxml2 already, * libxml2.so.2 libxml2.so.2.6.27* the above modules are present in folder /usr/lib/ I am using the machine, *Linux version 2.6.18 (gcc version 3.3.5 (Debian 1:3.3.5-13)) #1 SMP Thu Jul 26 01:27:32 EDT 2007 * I edited the Makefile and updated the line as follows, *exclude_modules?= jabber cpl-c postgres osp unixodbc \ avp_radius auth_radius group_radius uri_radius xmpp \ presence_xml presence_mwi pua_bla pua_mi \ pua_usrloc pua_xmpp mi_xmlrpc perl snmpstats perlvdb \ ldap carrierroute h350 xcap_client db_berkeley seas * I gave command prompt$ *make all* for compiling. I am getting the error message as shown above. Please mail me if I am missing something.. -- Noble Antony Thattil, Bangalore.

3 2

[OpenSER-Users] openSER Call Route Logic
by asterisk User 17 Jan '08

17 Jan '08

Hi, I recently installed openser-1.3.0 and able to connect my cisco sip phones and everything is working. Now i need to integrate this with asterisk box for call forwarding to other locations & voicemails. i read the getting started guide and other information, but seems i need more help from the group. can any one give me a helping hand. i am sure i will do my home work before bug you again. I need a starting point to over come the complexity of the routing plan and logic. thanks -asterisk user --------------------------------- Looking for last minute shopping deals? Find them fast with Yahoo! Search.

2 1

[OpenSER-Users] How to interacte X-lite and openser
by hammou moujane 17 Jan '08

17 Jan '08

Hello All, I have installed openser, is it possible to make voice call between two x-lite softphone using this SIP server? how can I start this operation if it's possible? Thankyou very much for your help. Regards -- Hammou MOUJANE Tel: +212 62 46 33 89 INPT

2 1

[OpenSER-Users] MySQL Alternative
by Douglas Garstang 17 Jan '08

17 Jan '08

What other options do I have for generic database connectivity besides MySQL/avp_db_query() in OpenSER? Can I use SQLite? Can I use unixodbc? (and I'm not talking about the limited built in usrloc unixodbc stuff) Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [OpenSER-Users] {Spam?} Hello World Config problem
by Andy Smith 16 Jan '08

16 Jan '08

Hi! Ok, thanks that problem solved and now I have another! I get the error "Registration error: 404 - Not Found" in Xlite. If I comment out this section of the cfg file I then get an error "Too Many Hops" :S :SMy laptop is directly (virtually on VPN) connected to the same LAN as the openser server.Any ideas? Or anyone have an already working hello world config??thanks in advance, Andy.On Wednesday 16 January 2008 15:51:58 Andy Smith wrote: > Thanks Iñaki, ok now my openser starts up ok, but I get the same "filtered > destination" Try adding: disable_dns_blacklist=yes at the top of your script.

1 0

[OpenSER-Users] Problems with unixodbc driver
by Alcindo Schleder 16 Jan '08

16 Jan '08

Hi.. I´m using unixodbc driver to connect with firebird SGDB. The problem is that the unixodbc driver not connect... ERROR:unixodbc:new_connection: failed to connect Jan 15 14:19:06 xsip openser[4577]: unixodbc:SQLDriverConnect=08004:1:-904:[unixODBC][ODBC Firebird Driver]Unable to connect to data source: library 'libfbclient.so' failed to load Jan 15 14:19:06 xsip openser[4577]: ERROR: group_db_bind: unable to connect to the database Jan 15 14:19:06 xsip openser[4577]: ERROR:group:mod_init: unable to open database connection

2 4

Re: [OpenSER-Users] {Spam?} Hello World Config problem
by Andy Smith 16 Jan '08

16 Jan '08

Thanks Iñaki, ok now my openser starts up ok, but I get the same "filtered destination" error from Xlite as I was getting with my original openser cfg with the hello world route. I was hoping the hello world config would allow me to connect/register whatever SIP device without any specific password :S thanks Andy.On Wednesday 16 January 2008 13:44:10 A.smith wrote: > If I try and use the whole example cfg (not just the route) I am recieving > an error like: > > 0(23448) find_cmd_export_t: <lookup> not found > 0(23448) parse error (105,23-24): unknown command, missing loadmodule? "lookup()" is defined in "registrar" module that you are not loading: http://www.openser.org/docs/modules/1.2.x/registrar.html#AEN342 -- Iñaki Baz Castillo ibc at in.ilimit.es

2 1

[OpenSER-Users] Hello World Config problem
by A.smith 16 Jan '08

16 Jan '08

HI, As I dont currently have a working PSTN I wanted to get the hello world config working so I can generate calls from one IP phone to another just across my LAN. However with the route config (put into my previous working registrar test cfg) from this example: http://openser.oralnet.co.uk/#AEN424 I am recieving a "filtered destination" error from XLite when it attempts to register. I can see the packets hitting my server from tcpdump... If I try and use the whole example cfg (not just the route) I am recieving an error like: 0(23448) find_cmd_export_t: <lookup> not found 0(23448) parse error (105,23-24): unknown command, missing loadmodule? Any help or tips on diagnosing this would be greatly appreciated! I am using OpenSER 1.2.2 on CentOS Linux... thanks Andy. ________________________________________________ Message sent using UK Grid Webmail 2.7.9

3 2

[OpenSER-Users] Openser + Wesip Click2dial
by [ACTIVENDY] NOC 16 Jan '08

16 Jan '08

Hi, I've installed my Openser with Wesip Application Server. I've installed Wesip with Click2dial application from Wesip web site. The application is running, but when I try to call to a softphone sip address trough my browser I get an error code -2. In Wesip logs I get the following: CommonsLoggingOutput [HttpProcessor[7080][4]] - Exec: Click2Call.c2cDWR() CommonsLoggingOutput [HttpProcessor[7080][4]] - --Object created, not stored. id=0 MapFactory [HttpProcessor[7080][4]] - Unable to create class java.util.Hashtable returning HashTable instead StandardSipSession [HttpProcessor[7080][4]] - Transport: udp SipRequest [HttpProcessor[7080][4]] - ERROR sending message javax.sip.TransactionUnavailableException: no route! at com.voztele.javax.sip.SipProviderImpl.getNewClientTransaction(SipProviderImpl.java:264) at com.voztele.sipservlet.core.SipRequest.send(SipRequest.java:196) at com.voztele.wesip.apps.click2call.C2C.c2c(C2C.java:134) at com.voztele.wesip.apps.click2call.C2C.c2cDWR(C2C.java:99) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.directwebremoting.impl.ExecuteAjaxFilter.doFilter(ExecuteAjaxFilter.java:34) at org.directwebremoting.impl.DefaultRemoter$1.doFilter(DefaultRemoter.java:428) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:431) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:283) at org.directwebremoting.servlet.PlainCallHandler.handle(PlainCallHandler.java:52) at org.directwebremoting.servlet.UrlProcessor.handle(UrlProcessor.java:101) at org.directwebremoting.servlet.DwrServlet.doPost(DwrServlet.java:146) at javax.servlet.http.HttpServlet.service(HttpServlet.java:760) at javax.servlet.http.HttpServlet.service(HttpServlet.java:853) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:200) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:146) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:206) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedContextValve.invokeHTTP(ConvergedContextValve.java:144) at com.voztele.sipservlet.valves.ConvergedContextValve.invoke(ConvergedContextValve.java:64) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.core.StandardHttpContext.invoke(StandardHttpContext.java:2377) at com.voztele.sipservlet.core.ConvergedContext.invoke(ConvergedContext.java:98) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeHTTP(ConvergedHostValve.java:336) at com.voztele.sipservlet.valves.ConvergedHostValve.invoke(ConvergedHostValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.valves.ErrorDispatcherValve.invoke(ErrorDispatcherValve.java:124) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:118) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedEngineValve.invoke(ConvergedEngineValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.connector.http.HttpProcessor.process(HttpProcessor.java:994) at org.apache.catalina.connector.http.HttpProcessor.run(HttpProcessor.java:1105) at java.lang.Thread.run(Thread.java:619) java.lang.RuntimeException: javax.sip.TransactionUnavailableException: no route! at com.voztele.sipservlet.core.SipRequest.send(SipRequest.java:222) at com.voztele.wesip.apps.click2call.C2C.c2c(C2C.java:134) at com.voztele.wesip.apps.click2call.C2C.c2cDWR(C2C.java:99) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.directwebremoting.impl.ExecuteAjaxFilter.doFilter(ExecuteAjaxFilter.java:34) at org.directwebremoting.impl.DefaultRemoter$1.doFilter(DefaultRemoter.java:428) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:431) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:283) at org.directwebremoting.servlet.PlainCallHandler.handle(PlainCallHandler.java:52) at org.directwebremoting.servlet.UrlProcessor.handle(UrlProcessor.java:101) at org.directwebremoting.servlet.DwrServlet.doPost(DwrServlet.java:146) at javax.servlet.http.HttpServlet.service(HttpServlet.java:760) at javax.servlet.http.HttpServlet.service(HttpServlet.java:853) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:200) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:146) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:206) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedContextValve.invokeHTTP(ConvergedContextValve.java:144) at com.voztele.sipservlet.valves.ConvergedContextValve.invoke(ConvergedContextValve.java:64) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.core.StandardHttpContext.invoke(StandardHttpContext.java:2377) at com.voztele.sipservlet.core.ConvergedContext.invoke(ConvergedContext.java:98) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeHTTP(ConvergedHostValve.java:336) at com.voztele.sipservlet.valves.ConvergedHostValve.invoke(ConvergedHostValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.valves.ErrorDispatcherValve.invoke(ErrorDispatcherValve.java:124) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:118) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedEngineValve.invoke(ConvergedEngineValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.connector.http.HttpProcessor.process(HttpProcessor.java:994) at org.apache.catalina.connector.http.HttpProcessor.run(HttpProcessor.java:1105) at java.lang.Thread.run(Thread.java:619) Caused by: javax.sip.TransactionUnavailableException: no route! at com.voztele.javax.sip.SipProviderImpl.getNewClientTransaction(SipProviderImpl.java:264) at com.voztele.sipservlet.core.SipRequest.send(SipRequest.java:196) ... 44 more My configuration is: - openser.cfg debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=MY_IP_ADDRESS mpath="/lib/openser/modules" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "seas.so" modparam("mi_fifo","fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("seas", "listen_sockets", "MY_IP_ADDRESS:5080") route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if(!as_relay_t("wesip")){ t_reply("500","Application Server error"); } } - server.xml <Server port="5005" shutdown="SHUTDOWN" debug="0"> <Service name="WeSIP-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="7080" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" addresses="217.172.74.8:5060" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="MY_IP_ADDRESS" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5080" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="javax.sip.STACK_NAME" value="wesip" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="217.172.74.8" debug="0" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="MY_IP_ADDRESS" nameSip="MY_IP_ADDRESS_5060" debug="0" appBase="wesipapps" unpackWARs="true" autoDeploy="true" className="com.voztele.sipservlet.core.ConvergedHost">  <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/>  </Host> </Engine> </Service> </Server> Thanks in advance, Sergi

2 2

[OpenSER-Users] How to convert 183 to 180 message
by Tung Tran 16 Jan '08

16 Jan '08

3 3

[OpenSER-Users] OpenSER config Wiz - What does it do? :P
by A.smith 16 Jan '08

16 Jan '08

Hi, Slightly stupid question I know, what I mean is that the config contains no rewrite host so Im just wondering whether the configs that the wizard generate are actually useable for something without modification. I ask because I want to understand what the wizard is trying to produce so I know if its appropriate for me to try and modify the config to match my requirements or Im better to start from scratch. And yes, I know I dont know much about OpenSER! :P Im trying to learn! thanks Andy. ________________________________________________ Message sent using UK Grid Webmail 2.7.9

1 0

[OpenSER-Users] Variable $rc
by Douglas Garstang 16 Jan '08

16 Jan '08

What's the point of the pseudo variable $rc? If I use avp_db_query() with: invalid SQL, then rc = -1. database down: then rc = -1. Everything fine... then rc = -1!!! What's the point? Doug. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

4 3

[OpenSER-Users] Presence Problem on v1.3.0
by fengbin 16 Jan '08

16 Jan '08

Hi,All When I tried to start openser with presence module.I met an error of "ERROR:core:set_mod_param_regex: parameter <force_active> not found in module <presence>" But I used the presence before with openser of v1.2.0 and it works fine. Can you take a look at it? THX BR The encolsed is my config file. -- Fengbin

2 1

[OpenSER-Users] Openser and RTPProxy
by Patrick Baker 16 Jan '08

16 Jan '08

Is is possible to have a SIP UA behind nat and communicate to an OpenSER/RTP Proxy server behind nat behind nat and have everything transverse appropriately? Or will I need something additional like STUN? Example: OpenSER + RTP Proxy + Asterisk will all reside on the same machine. SIP UA -> Cisco PIX ( Public IP to Localnet NAT ) -> OpenSER/RTP -> Asterisk Thank you for your time, Patrick

4 7

[OpenSER-Users] OpenSer 1.3 Presence Problem
by Ross Beer 15 Jan '08

15 Jan '08

Hi All, I am trying to setup an OpenSER server with support for registration, messaging & presence however I am having a problem with the presence server. I have had presence working correctly before, however for some reason it is no longer working. I have spent about two days trying everything! I would really appreciate some advice with this. I have included my config below. Thank you in advance, Ross ------------------------------------------------------------------------------ listen = udp:my.ip.address:5060 mpath = "/usr/local/lib64/openser/modules" children = 8 debug = 3 fork = yes disable_tcp = no #log_facility = LOG_DAEMON log_facility = LOG_LOCAL0 log_stderror = yes tcp_children = 4 mhomed = no server_signature = yes #sock_group = "openser" #sock_mode = 0600 #sock_user = "openser" #unix_sock = "/tmp/openser.sock" #unix_sock_children = 1 #reply_to_via = no #sip_warning = no #check_via = no #dns = no #rev_dns = no #disable_core_dump = no #dns_try_ipv6 = yes #dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://username:password@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 1) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") #modparam("mi_fif0", "fifo_name", "FIFO") modparam("mi_fifo", "fifo_mode", 0660) #modparam("mi_fifo", "fifo_group", "openser") #modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://username:password@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "") modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "textops.so" loadmodule "mediaproxy.so" #modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so" loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://username:password@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") #modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=id") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://username:password@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "avpops.so" modparam("avpops", "db_url", "mysql://username:password@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 0) modparam("avpops", "uuid_column", "id") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") loadmodule "enum.so" modparam("enum", "domain_suffix", "e164.org.") loadmodule "permissions.so" #modparam("permissions", "default_allow_file", "permissions.allow") #modparam("permissions", "default_deny_file", "permissions.deny") #modparam("permissions", "check_all_branches", 1) #modparam("permissions", "allow_suffix", ".allow") #modparam("permissions", "deny_suffix", ".deny") modparam("permissions", "db_url", "mysql://username:password@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col", "src_ip") modparam("permissions", "proto_col", "proto") modparam("permissions", "from_col", "from_pattern") modparam("permissions", "tag_col", "tag") modparam("permissions", "peer_tag_avp", "$avp(i:707)") loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://username:password@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "contact_avp", "$avp(i:711)") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "dm_flag", 25) loadmodule "alias_db.so" modparam("alias_db", "db_url", "mysql://username:password@localhost/openser") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) # ----- presence params ----- loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("presence|presence_xml", "db_url","mysql://username:password@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:my.ip.address:5060") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "max_expires", 3600) modparam("presence", "fallback2db", 1) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "integrated_xcap_server", 1) modparam("pua_usrloc", "default_domain", "my.domain.com") modparam("presence", "to_tag_pref", 'pres') ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER|MESSAGE")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(7); } if(is_method("REGISTER")) { route(6); } if(is_method("INVITE")) { route(8); } if(is_method("PUBLISH|SUBSCRIBE")) { route(17); } if(is_method("CANCEL") || is_method("ACK")) { route(14); } ############################ # MESSAGES # ############################ if(is_method("MESSAGE")) { route(18); } route(15); } ######################################################################## # Request route 'clear-usr-preferences-caller' ######################################################################## route[1] { xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } ######################################################################## # Request route 'clear-usr-preferences-callee' ######################################################################## route[2] { xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } ######################################################################## # Request route 'usr-preferences-caller' ######################################################################## route[3] { route(1); xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load caller avps avp_db_load("$avp(s:caller_uuid)", "*"); } ######################################################################## # Request route 'usr-preferences-callee' ######################################################################## route[4] { xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load callee avps avp_db_load("$avp(s:callee_uuid)", "*"); } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[5] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[6] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[7] { xlog("L_INFO", "ROUTE 7 RUN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); use_media_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[8] { sl_send_reply("100", "Trying"); if(from_gw()) { $avp(s:caller_uuid) = "0"; xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(allow_trusted()) { if(is_avp_set("$avp(s:peer_uuid)/s")) { # use tag-column from trusted-table as uuid for this caller avp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d"); } else { # if no uuid is set, use "0" as default uuid $avp(s:caller_uuid) = "0"; } xlog("L_INFO", "Call from trusted peer with uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!is_domain_local("$rd")) { xlog("L_INFO", "Rejecting peering attempt with non-local request domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Relaying Denied"); exit; } setflag(23); } else { # if(!proxy_authorize("", "subscriber")) # { # # xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # proxy_challenge("", "0"); # exit; # } # if(!check_from()) # { # # xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sl_send_reply("403", "Spoofed From-URI Detected"); # exit; # } } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(9); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[9] { if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(11); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select id from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[10] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[11] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); exit; } route(13); if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(7); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'normalize-e164' ######################################################################## route[12] { # European numbering plans look like this: # CC = country code (i.e. 43 for Austria) # NDC = national destination code (i.e. 1 for Vienna) # SN = subscriber number (i.e. 4001234) # # CC + NDC + SN, i.e. 4314001234 # # Within the same CC+NDC, it can be omitted, so if # +4314001234 wants to call +4315002345, one can dial # just 5002345. # # Within the same CC, CC can be ommitted and a "0" is prefixed # to NDC, so if +4314001234 wants to call +4326003456, # one can dial 026003456. # # For international calls, either "00" or + is prefixed, like # +49123456780 or 0049123456789. # avp_delete("$avp(s:orig_callee_user)/g"); $avp(s:orig_callee_user) = $rU; if(uri =~ "^sip:(\+[1-9])?[0-9]+@") { # looks like a PSTN number if(uri =~ "^sip:0[1-9][0-9]+@") { # we have format 0+NDC+SN strip(1); prefix("+44"); } else if(uri =~ "^sip:00[1-9]+@") { # we have format 00 + CC + NDC + SN strip(2); prefix("+"); } else if(!uri =~ "^sip:\+[1-9][0-9]+@") { # unknown format, maybe NDC wasn't added before? xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); return(-1); } # else we have "+" + CC + NDC + SN xlog("L_INFO", "Normalized callee '$avp(s:orig_callee_user)' to E.164 format '$rU' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); return(1); } else { xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); return(-1); } } ######################################################################## # Request route 'lookup-enum' ######################################################################## route[13] { route(12); if($rc == 1) { if(enum_query("e164.org.")) { # TODO: do GW fallback (load gws, set failure-route)? xlog("L_INFO", "ENUM query succeeded - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); exit; } else { xlog("L_INFO", "ENUM query failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # ENUM query failed, revert $rU avp_pushto("$ru/username", "$avp(s:orig_callee_user)"); } } } ######################################################################## # Request route 'base-route-local' ######################################################################## route[14] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[15] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[16] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); exit; } } ######################################################################### # Presence ######################################################################### route[17] { # if(nat_uac_test("19") || search("^Route:.*;nat=yes")) # { # fix_nated_contact(); # setbflag(6); # } if (!t_newtran()) { sl_reply_error(); exit; }; #if(is_method("PUBLISH")) #{ # xlog("L_INFO", "PUBLISH - Process Publish Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # handle_publish(); # t_release(); #} if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } if(is_method("SUBSCRIBE")) { handle_subscribe(); xlog("L_INFO", "SUBSCRIBE - Process Subscribe Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_release(); } exit; } ######################################################################## # SEND MESSAGES ######################################################################## route[18] { if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } if (!t_relay()) { sl_reply_error(); }; exit; } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(16); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); exit; } t_on_failure("1"); route(7); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(16); route(5); } _________________________________________________________________ Share what Santa brought you https://www.mycooluncool.com

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[OpenSER-Users] OpenSER + OpenXCAP: NOTIFY response to unsubscription includes xml document?
by KevinKinnan 15 Jan '08

15 Jan '08

Openser: NOTIFY response to unsubscription doesn't have message body. Openser + OpenXCAP: NOTIFY response to unsubscription includes xml document which is the same as one in response to subscription, but the subscription_state in unsub NOTIFY message is "terminated". Can anyone give me a hint? Regards, Kevin _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

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[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 15 Jan '08

15 Jan '08

1. Only Openser, no OpenXCAP, force_active=1 NOTIFY with new info is sent from Openser to client when updating PUBLISH message everytime. 2. Openser+OpenXCAP force_active=0 NO NOTIFY is sent from Openser if integrating with OpenXCAP. But PUBLISH message updates properly. I can't figure out why, any clue? Thank you! Kevin, _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

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Re: [OpenSER-Users] avp_db_query() timeout
by Douglas Garstang 15 Jan '08

15 Jan '08

Inaki, I am doing updates with avp_db_query, so there is no avp I can check the value of . As I said in a subsequent post, I wrote a simple server that listened on port 3306, like mysql does, and pointed OpenSER at it. The avp_db_query commands() basically blocked forever and openser seemed to completely lose all ability to process new calls. That's really bad. Doug. ----- Original Message ---- From: Iñaki Baz Castillo <ibc(a)aliax.net> To: users(a)lists.openser.org Sent: Monday, January 14, 2008 1:46:00 PM Subject: Re: [OpenSER-Users] avp_db_query() timeout El Lunes, 14 de Enero de 2008, Douglas Garstang escribió: > I haven't had much luck getting replies to my questions lately, hopefully > I'll have more luck with this one. > > avp_db_query().... does it have a timeout? What is it? Can I set it? > > If I use it, and there is a MySQL database problem, I don't want > avp_db_query() to wait forever trying to connect to MySQL. If it can't > connect after a few seconds, I'd like it to just give up. I have multiple > calls to avp_db_query() in a typical call flow. It would be even better if > I could set a flag on the first timeout and not try again during that call. > > Can that be done? Hi, by the test I've done: avp_db_query("SQL_QUERY", "$avp(s:result)"); xlog("L_INFO", "retcode = $rc \n"); Cases: - If query doesn't return a row/value then $avp(s:result)=null and $rc=(-1). - If query returns a row/value the $avp(s_result) stores that value and $rc=1. - If there is a syntax error in query then $avp(s:result)=null and $rc=(-1). - If MySQL is not running then $avp(s:result)=null and $rc=(-1). I don't know what occurs if MySQL is running but because overload or any other reason it doesn't reply, but AFAIK the "avp_db_query" uses the same code of other OpenSer functions to access DB, so I'm sure it has a correct behaviour in case of failure. BTW you can "play" with the $rc code after "avp_db_query". -- Iñaki Baz Castillo _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] CPL and $var issue when different process
by Iñaki Baz Castillo 15 Jan '08

15 Jan '08

Hi, maybe someone could confirm this: At the begining of my script I set a variable: $var(gateway) = "123.123.123.123"; Later, I run CPL for calls to internal users and there could be forwardings and so. I set CPL "recurse" parameter so convert 302 from UAS into a new branch into CPL. At the end of CPL I run "route[20]" in which I use the variable $var(gateway), but sometimes it appears empty. I think this could occur because CPL can involve various OpenSer process. I think that when CPL receives a 302, maybe sometimes it's a new OpenSer process and since it's a reply the variable at the begining of my script is not read, so it appears empty. Am I right? A solution is to use AVP instead od variables since AVP's remains along al lthe transaction. Am I right? Thanks for any confirmation. Best regards. -- Iñaki Baz Castillo

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[OpenSER-Users] ¿Se puede enviar DTMF durante el Early Media?
by Iñaki Baz Castillo 15 Jan '08

15 Jan '08

Hola, algo que fastidia a todo el mundo es el tema de que te cobren una llamada nada más salir el buzón de voz. Lo lógico y deseable (al margen de la actitud avariciosa de las grandes operadoras) es que al llamar a un número y salir el buzón de voz no se cobre la llamada y se ofrezca la posibilidad de dejar un mensaje pulsando algún dígito. El impedimento técnico que veo es que parece ser no se permite el envío de DTMF durante el EarlyMedia. Pero no encuentro documentación sobre ello. El caso es que durante el EarlyMedia el audio es bidireccional, o sea, el SDP del "183 Session Progress" dice: a=sendrecv (de hecho esto lo he verificado también a nivel de gateway PSTN: una llamada desde la PSTN a un gateway y éste la dirige a un SIP UAS que responde con 183, y el gateway envía y recibe audio desde el UAS). A todo esto una duda que tengo es precisamente esto de poder enviar y recibir audio durante un EarlyMedia, ¿se puede o no se puede? ¿se hace? Por otra parte, mi Twinkle no me permite enviar DTMF durante el EarlyMedia, pero vamos, que ni lo intenta por lo que no sé qué pasaría si sí me dejase. Y por las pruebas que he hecho llamando desde la PSTN (fijo y móvil) y respondiendo con un EarlyMedia, tampoco parece que se permita el envío de DTMF. En fin, que la única opción honesta que he encontrado es la de reproducir el mensaje del buzón en EarlyMedia, y sólo tras el mensaje responder la llamada para que el llamante deje su mensaje (y sólo en ese caso provocar que tenga que pagar su llamada). Con lo fácil que es... lo cab***as que son las compañías. Bueno, sin más, que no sé exactamente cuál es mi pregunta pero tal vez alguien me ayude a definirla XD Saludos. -- Iñaki Baz Castillo

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[OpenSER-Users] Giving more power to CPL language: Addressbook Categories feature
by Iñaki Baz Castillo 15 Jan '08

15 Jan '08

Hi, in my opinion CPL language is not enough powerful, the absence of regular expression or any header reading makes it poor (IMHO). It would become really cool if it could match AVP's but it can :( I'm trying to give some power to CPL logic. I want to implement "addressbook categories" for each user, this is: each user has its own addressbook in which he can set a category for each contact: - Work - Friend - Family - Custom And of course what I need it a way to match that category **inside** the CPL logic. This is impossible with CPL itself so I'm thinking in a very very dirty trick: ** Addressbook Categories ** - In an incoming call for "bob(a)domain.org" OpenSer will match the callerid against Bob's addressbook. This will return a variable or AVP: $avp(s:caller_category) - CPL module just can match "Organization", "Subject", and "User-Agent", so OpenSer will do: append_hf("Bak_Organization: $hdr(Organization)\r\n"); # To preserve original Organization remove_hf("Organization"); # Remove original Organization append_hf("Organization: $avp(s:caller_category)\r\n"); # Set category as Organization - But since OpenSer appends/removes headers when the message leaves the proxy I can't run CPL now (it will not read the new "Organization" header), so I need to do a dirty loop in order to append the header. - When the message comes back into OpenSer the headers are updated so I do: remove_hf("Organization"); # Remove current Organization (that is a hack) append_hf("Organization: $hdr(Backup_Organization)\r\n"); # Restore original Organization (those changes will be applied again when message leaves OpenSer, not now). - And I run CPL: cpl_run_script("incoming","FORCE_STATEFUL"); - Into CPL I match "Organization" header by: <string-switch field="Organization"> <string is="Friends"> <reject status="reject" reason="I have no friends" /> </string> </string-switch> Is it enough dirty or could I make it more? Any suggestion? how do you people implement things like this? Thank a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] OpenSER logo
by Reinhold Buchinger 15 Jan '08

15 Jan '08

Hi, Where can I find the OpenSER logo in a high resolution? Thanks for your help. Regards, Reinhold

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[OpenSER-Users] CRL/OCSP in OpenSer
by Jan Ruzicka 15 Jan '08

15 Jan '08

Hi, is it possible to use CRL (OCSP)? I found an old message from Klaus (http://www.nabble.com/-Users--CRL-in-OpenSER-to12712663.html#a12712666). As far I don't see mentioned new cfg parameter, I'm afraid its not possible. Are there any plan to use these features? Thanks Jan

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[OpenSER-Users] OpenSER Configuration via files?
by Douglas Garstang 15 Jan '08

15 Jan '08

Asked this question earlier in the week, and didn't get an answer. I'm using avp_db_query to log calls to MySQL. I'd like to be able to turn this and other functionality on and off in the openser.cfg file without having to comment lines. Obviously putting the settings into MySQL and reading them with avp_db_query() doesn't make much sense. Does OpenSER have some sort of simple key/value pair configuration it read from a file so that I can control the basic configuration with? The db_text and db_berkeley modules are unfortunately only designed to be used for userloc data. Why is that? Thanks, Doug. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] OpenXCAP: sphere and validity?
by KevinKinnan 15 Jan '08

15 Jan '08

I can't make it work when testing the effect of validity element in conditions element in Openser+OpenXCAP, also happen on sphere element and transformations. The presence document is based on rpid and it is quite simple. The only one I got it work is like below(without validity and sphere elements): <condition> <one id=""> </condition> <actions> <sub-handling>allow</sub-handling> </actions> The following two(adding validity and sphere) are not working: <condition> <one id=""> <validity> <from>...</from> <until>...</until> </validity> </condition> <actions> <sub-handling>allow</sub-handling> </actions> <condition> <shpere value="home"> <one id=""> </condition> <actions> <sub-handling>allow</sub-handling> </actions> Does OpenXCAP support these policies metioned above? Or, Openser is not supporting sphere and validity auth, it only knows sub-handling(xcap_auth.c in openser/modules/presence_xml)? Thank you! Kevin, _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

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[OpenSER-Users] avp_db_query() timeout
by Douglas Garstang 14 Jan '08

14 Jan '08

I haven't had much luck getting replies to my questions lately, hopefully I'll have more luck with this one. avp_db_query().... does it have a timeout? What is it? Can I set it? If I use it, and there is a MySQL database problem, I don't want avp_db_query() to wait forever trying to connect to MySQL. If it can't connect after a few seconds, I'd like it to just give up. I have multiple calls to avp_db_query() in a typical call flow. It would be even better if I could set a flag on the first timeout and not try again during that call. Can that be done? Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[Serusers] xml rpc documentation
by Srinivas Kotamarti 14 Jan '08

14 Jan '08

Hi, Does anybody know where can I find more documentation on the new XML-RPC module of SER2. I want to write a management plugin for the application I am writing on top of SEMS using the XML-RPC interface. The documentation in the SER2.0 documentation page (http://www.iptel.org/rpc_xml_rpc) does not explain its full functionality and is not enough for writing a plugin. Thanks --Srinivas

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[OpenSER-Users] strange error logs in syslog
by Cseke Tamas 14 Jan '08

14 Jan '08

Hello, We have a lots of error log like this, what fills syslog ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction ERROR:tm:w_t_relay: t_forward_nonack failed ERROR: t_check_status: cannot check status for a reply which has no T-state established Could you tell me how could i avoid this? I think these are not errors. Shall i just change the debug level for this logs? Thanks any help, Tamas

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[OpenSER-Users] radius_does_uri_exist
by Marc LEURENT 14 Jan '08

14 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good afternoon, I'm trying to use the radius_does_uri_exist() function to check if the user exist before forwarding it to voicemail when it's not enable. I'm using radius AAA and the radius_does_uri_exist() seems to fail.. Any idea how to solve it? Thanks # -- uri_radius paramaters -- modparam("uri_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("uri_radius", "service_type", 11) # Radius service type used in radius_does_uri_exist check. ....... if (!lookup("location")) { if (radius_does_uri_exist()) { # Will Redirect to voicemail when fixed sl_send_reply("480", "Temporarily Unaivailable - Will Redirect To Voicemail"); exit; }; xlog("L_DBG", "ACC RADIUS: 404 NOT FOUND FOR $si \r\n"); sl_send_reply("404", "Not Found"); exit; ..... When I'm trying to call aeron which is not connected but exists, I have a 404 not found whereas I should have a 480... root@krusty:[~L/freeradius]# tail -f radius.log Mon Jan 14 16:08:18 2008 : Auth: Login incorrect: [aeron(a)sip.bob.net/<no User-Password attribute>] (from client private port 0) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHi311N4+o+2LtdFwRAgGNAKC4xB0R5mHjgZ0zNKCaEEcqCgCUaQCdG+p0 4KoP/SuAUHJ8u6GFnaOTLYw= =OXcY -----END PGP SIGNATURE-----

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[OpenSER-Users] OpenXCAP problem
by Franz Edler 14 Jan '08

14 Jan '08

Hi OpenXCAP experts, I am trying to setup a demo-configuration for XCAP with Eyebeam- and Bria-client. OpenXCAP and OpenSER are running. When I start-up the Bria-client (data storage via XCAP enabled) I get the following errors from OpenXCAP - log (e.g. for resource-list): Jan 2 13:15:38 sip openxcap[10367]: [HTTPChannel,2,192.168.0.2] error: Invalid Document Selector xcap-root/resource-lists/users//resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') Jan 2 13:15:38 sip openxcap[10367]: [-] error: Invalid Document Selector xcap-root/resource-lists/users//resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') The HTTP dialog between the client and the XCAP server is the following: GET /xcap-root/resource-lists/users//resource-list.xml HTTP/1.1 Host: xcap.net1.test:8000 Keep-Alive: Connection: TE, Keep-Alive TE: trailers HTTP/1.1 404 Not Found Date: Wed, 02 Jan 2008 12:15:38 GMT Content-Length: 75 Server: OpenXCAP/0.9.7 the Document Selector context must be 'users' or 'global': 'resource-lists' PUT /xcap-root/resource-lists/users//resource-list.xml HTTP/1.1 Host: xcap.net1.test:8000 Connection: TE TE: trailers Content-Type: application/resource-lists+xml Content-Length: 323 <?xml version="1.0" encoding="UTF-8"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists" xmlns:cp="counterpath:properties"> <list name="Contact List"> <cp:prop name="category" value="Friends" /> <cp:prop name="category#2" value="Home" /> <cp:prop name="category#3" value="Work" /> </list></resource-lists> HTTP/1.1 404 Not Found Date: Wed, 02 Jan 2008 12:15:38 GMT Content-Length: 75 Server: OpenXCAP/0.9.7 the Document Selector context must be 'users' or 'global': 'resource-lists' Any hint highly appreciated! Franz

2 2

[Serusers] Registration expires timer
by Ranganath B 14 Jan '08

14 Jan '08

Hi, Need help in setting the registration expiry timer at SER. I would like to know the way we can enforce the registration expiry timer from server end. Please update me on the configuration details. Thanks in advance, Ranganath B

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Re: [OpenSER-Users] TLS problem.
by Klaus Darilion 14 Jan '08

14 Jan '08

Hi Fengbin! Cc'ed to the openser list ... fengbin schrieb: > Hi,Klaus, > > How to use NULL cipher? Only setting in Openser is ok? I mean do I need > to set NULL cipher at client site? Usually the NULL cipher is not enabled (for security reasons). You have to enable it on both sides, the server and the client. But if you use the following approach you do not need it. > And where to put xlog("L_ERR","message buffer: $mb"); anywhere in > openser.cfg ? Put it just in the beginning of the route block. regards klaus > THX > BR > > > On 1/11/08, *Klaus Darilion* <klaus.mailinglists(a)pernau.at > <mailto:klaus.mailinglists@pernau.at>> wrote: > > The capture file is not helpful, as it is encrypted. You could use NULL > cipher to have plaintext inside the TLS connection to inspect the > incoming SIP message, or add xlog("L_ERR","message buffer: $mb"); to see > the whole incoming SIP request. > > regards > klaus > > fengbin schrieb: > > Hi,Klaus > > Thank you for your reply. > > The enclosed is the config file ,the pcap between client and > server and > > the log on the openser 's console. > > Could you please take a look at them for me? > > > > THX > > BR > > > > > > On 1/10/08, *Klaus Darilion* <klaus.mailinglists(a)pernau.at > <mailto:klaus.mailinglists@pernau.at> > > <mailto:klaus.mailinglists@pernau.at > <mailto:klaus.mailinglists@pernau.at> >> wrote: > > > > Can you show us the REGISTER request? (both, port 5060 and > port 5061). > > > > Further show use your openser config > > > > regards > > klaus > > > > fengbin schrieb: > > > > > > Hi,all > > > I met a strange problem while I am testing TLS connection > between > > > minisip and openser. > > > The following is my openser.cfg (part of that) > > > > > > ......... > > > fork=no > > > log_stderror=yes > > > > > > # Uncomment this to prevent the blacklisting of > temporary not > > > available destinations > > > #disable_dns_blacklist=yes > > > > > > # # Uncomment this to prevent the IPv6 lookup after v4 > dns lookup > > > failures > > > #dns_try_ipv6=no > > > > > > # uncomment the following lines for TLS support > > > disable_tls = 0 > > > listen = tls: 10.11.57.197:5060 > <http://10.11.57.197:5060> <http://10.11.57.197:5060> > > <http://10.11.57.197:5060> > > > > > > > > > tls_verify_client = 1 > > > tls_method = TLSv1 > > > tls_certificate = "/usr/local/etc/openser//tls/user/user- > > cert.pem" > > > tls_private_key = > > "/usr/local/etc/openser//tls/user/user- privkey.pem" > > > tls_ca_list = "/usr/local/etc/openser//tls/user/user- > calist.pem" > > > tls_ciphers_list="NULL-SHA:NULL-MD5:AES256-SHA:AES128-SHA" > > > ...... > > > > > > When I set "tls:10.11.57.197:5061 > <http://10.11.57.197:5061> <http://10.11.57.197:5061> < > > http://10.11.57.197:5061>" the > > > registration never succeed. But if I set it to 5060 the > registration > > > over TLS is OK. > > > I compared the log of two scenarioes and found the TLS > session > > both are > > > OK,but the difference is that: > > > when the port is 5061 there is an error of forwarding. but the > > > forwarding is because openser think it's not the > destination of > > > the registration request. See bellow: > > > > > > Jan 10 16:46:56 [9199] DBG:rr:after_loose: No next URI > found > > > Jan 10 16:46:56 [9199] DBG:core:grep_sock_info: > checking if > > > host==us: 12==12 && [ 10.11.57.197 > <http://10.11.57.197> <http://10.11.57.197> > > <http://10.11.57.197 <http://10.11.57.197>>] == > > > [10.11.57.197 <http://10.11.57.197> > <http://10.11.57.197> <http://10.11.57.197>] > > > Jan 10 16:46:56 [9199] DBG:core:grep_sock_info: > checking if port > > > 5061 matches port 5060 > > > Jan 10 16:46:56 [9199] DBG:core:check_self: host != me > > > Jan 10 16:46:56 [9199] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 16:46:56 [9199] DBG:tm:t_newtran: T on > > entrance=0xffffffff > > > Jan 10 16:46:56 [9199] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 16:46:56 [9199] DBG:core:parse_headers: flags=78 > > > Jan 10 16:46:56 [9199] DBG:tm:t_lookup_request: start > searching: > > > hash=58073, isACK=0 > > > Jan 10 16:46:56 [9199] DBG:tm:matching_3261: RFC3261 > transaction > > > matching failed > > > Jan 10 16:46:56 [9199] DBG:tm:t_lookup_request: no > > transaction found > > > Jan 10 16:46:56 [9199] DBG:core:mk_proxy: doing DNS > lookup... > > > Jan 10 16:46:56 [9199] ERROR:tm:update_uac_dst: failed > to fwd > > to af > > > 2, proto 1 (no corresponding listening socket) > > > Jan 10 16:46:56 [9199] ERROR:tm:t_forward_nonack: > failure to add > > > branches > > > > > > > > > > > > With comparition to that when the port is set to 5060 the > trace is : > > > > > > Jan 10 17:07:59 [9410] DBG:rr:find_next_route: No next > Route > > HF found > > > Jan 10 17:07:59 [9410] DBG:rr:after_loose: No next URI > found > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if > > > host==us: 12==12 && [ 10.11.57.197 > <http://10.11.57.197> <http://10.11.57.197> > > <http://10.11.57.197>] == > > > [ 10.11.57.197 <http://10.11.57.197> > <http://10.11.57.197> <http://10.11.57.197>] > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if port > > > 5060 matches port 5060 > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if > > > host==us: 12==12 && [10.11.57.197 > <http://10.11.57.197> <http://10.11.57.197> > > <http://10.11.57.197>] == > > > [10.11.57.197 <http://10.11.57.197> < > http://10.11.57.197> <http://10.11.57.197>] > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if port > > > 5060 matches port 5060 > > > Jan 10 17:07:59 [9410] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 17:07:59 [9410] DBG:core:parse_headers: > flags=8000000 > > > Jan 10 17:07:59 [9410] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 17:07:59 [9410] DBG:registrar:build_contact: > created > > Contact > > > HF: Contact: > > <sip:888@10.11.57.192:5061;transport=TLS>;expires=1000 > > > > > > > > > > > > And there is no fwd needed then.So the error didnt occur. > > > > > > Its a little bit strange that when I set the port to > 5061,why did > > > openser check the port 5060????? > > > Can anyone help me to figure it out? > > > THX > > > BR > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > Fengbin > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > <mailto:Users@lists.openser.org <mailto:Users@lists.openser.org>> > > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > -- > > Fengbin > > > > > > > -- > Fengbin

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[OpenSER-Users] OpenSER 1.3 Presence Server
by Ross Beer 13 Jan '08

13 Jan '08

Hi All, I am trying to setup an OpenSER server with support for registration, messaging & presence however I am having a problem with the presence server. I have had presence working correctly before, however for some reason it is no longer working. I have spent about two days trying everything! I would really appreciate some advice with this. I have included my config below. Thank you in advance, Ross ------------------------------------------------------------------------------listen = udp:my.ip.address:5060mpath = "/usr/local/lib64/openser/modules"children = 8debug = 3fork = yesdisable_tcp = no#log_facility = LOG_DAEMONlog_facility = LOG_LOCAL0log_stderror = yestcp_children = 4mhomed = noserver_signature = yes#sock_group = "openser"#sock_mode = 0600#sock_user = "openser"#unix_sock = "/tmp/openser.sock"#unix_sock_children = 1#reply_to_via = no#sip_warning = no#check_via = no#dns = no#rev_dns = no#disable_core_dump = no#dns_try_ipv6 = yes#dns_use_search_list = yesloadmodule "usrloc.so"modparam("usrloc", "user_column", "username")modparam("usrloc", "domain_column", "domain")modparam("usrloc", "contact_column", "contact")modparam("usrloc", "expires_column", "expires")modparam("usrloc", "q_column", "q")modparam("usrloc", "callid_column", "callid")modparam("usrloc", "cseq_column", "cseq")modparam("usrloc", "methods_column", "methods")modparam("usrloc", "flags_column", "flags")modparam("usrloc", "user_agent_column", "user_agent")modparam("usrloc", "received_column", "received")modparam("usrloc", "socket_column", "socket")modparam("usrloc", "use_domain", 0)modparam("usrloc", "desc_time_order", 0)modparam("usrloc", "timer_interval", 60)modparam("usrloc", "db_url", "mysql://username:password@localhost/openser")modparam("usrloc", "db_mode", 1)modparam("usrloc", "matching_mode", 0)modparam("usrloc", "cseq_delay", 20)modparam("usrloc", "nat_bflag", 6)loadmodule "registrar.so"modparam("registrar", "default_expires", 3600)modparam("registrar", "min_expires", 60)modparam("registrar", "max_expires", 0)modparam("registrar", "default_q", 0)modparam("registrar", "append_branches", 1)modparam("registrar", "case_sensitive", 0)modparam("registrar", "received_param", "received")modparam("registrar", "max_contacts", 0)modparam("registrar", "retry_after", 0)modparam("registrar", "method_filtering", 0)modparam("registrar", "path_mode", 2)modparam("registrar", "path_use_received", 0)modparam("registrar", "received_avp", "$avp(i:801)")loadmodule "rr.so"modparam("rr", "enable_full_lr", 1)modparam("rr", "append_fromtag", 1)modparam("rr", "enable_double_rr", 1)modparam("rr", "add_username", 0)loadmodule "tm.so"modparam("tm", "fr_timer", 30)modparam("tm", "fr_inv_timer", 120)modparam("tm", "wt_timer", 5)modparam("tm", "delete_timer", 2)modparam("tm", "noisy_ctimer", 0)modparam("tm", "ruri_matching", 1)modparam("tm", "via1_matching", 1)modparam("tm", "unix_tx_timeout", 2)modparam("tm", "restart_fr_on_each_reply", 1)modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so"modparam("xlog", "buf_size", 4096)modparam("xlog", "force_color", 1)loadmodule "mi_fifo.so"modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")#modparam("mi_fif0", "fifo_name", "FIFO")modparam("mi_fifo", "fifo_mode", 0660)#modparam("mi_fifo", "fifo_group", "openser")#modparam("mi_fifo", "fifo_user", "openser")modparam("mi_fifo", "reply_dir", "/tmp/")modparam("mi_fifo", "reply_indent", "\t")loadmodule "domain.so"modparam("domain", "db_url", "mysql://username:password@localhost/openser")modparam("domain", "db_mode", 1)modparam("domain", "domain_table", "domain")modparam("domain", "domain_col", "domain")loadmodule "nathelper.so"modparam("nathelper", "natping_interval", 0)modparam("nathelper", "ping_nated_only", 1)modparam("nathelper", "rtpproxy_sock", "")modparam("nathelper", "rtpproxy_disable_tout", 60)modparam("nathelper", "rtpproxy_tout", 1)modparam("nathelper", "rtpproxy_retr", 5)modparam("nathelper", "sipping_method", "OPTIONS")modparam("nathelper", "received_avp", "$avp(i:801)")loadmodule "textops.so"loadmodule "mediaproxy.so"#modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock")modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock")modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients")modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients")modparam("mediaproxy", "natping_interval", 60)loadmodule "uri.so"loadmodule "sl.so"modparam("sl", "enable_stats", 1)loadmodule "maxfwd.so"modparam("maxfwd", "max_limit", 256)loadmodule "mysql.so"modparam("mysql", "ping_interval", 300)modparam("mysql", "auto_reconnect", 1)loadmodule "auth.so"modparam("auth", "nonce_expire", 300)modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes")modparam("auth", "rpid_avp", "$avp(s:rpid)")loadmodule "auth_db.so"modparam("auth_db", "db_url", "mysql://username:password@localhost/openser")modparam("auth_db", "user_column", "username")modparam("auth_db", "domain_column", "domain")modparam("auth_db", "password_column", "password")#modparam("auth_db", "password_column_2", "ha1b")modparam("auth_db", "calculate_ha1", 1)modparam("auth_db", "use_domain", 0)modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=id")loadmodule "uri_db.so"modparam("uri_db", "db_url", "mysql://username:password@localhost/openser")modparam("uri_db", "uri_table", "uri")modparam("uri_db", "uri_user_column", "username")modparam("uri_db", "uri_domain_column", "domain")modparam("uri_db", "uri_uriuser_column", "uri_user")modparam("uri_db", "subscriber_table", "subscriber")modparam("uri_db", "subscriber_user_column", "username")modparam("uri_db", "subscriber_domain_column", "domain")modparam("uri_db", "use_uri_table", 0)modparam("uri_db", "use_domain", 0)loadmodule "avpops.so"modparam("avpops", "db_url", "mysql://username:password@localhost/openser")modparam("avpops", "avp_table", "usr_preferences")modparam("avpops", "use_domain", 0)modparam("avpops", "uuid_column", "id")modparam("avpops", "username_column", "username")modparam("avpops", "domain_column", "domain")modparam("avpops", "attribute_column", "attribute")modparam("avpops", "value_column", "value")modparam("avpops", "type_column", "type")loadmodule "enum.so"modparam("enum", "domain_suffix", "e164.org.")loadmodule "permissions.so"#modparam("permissions", "default_allow_file", "permissions.allow")#modparam("permissions", "default_deny_file", "permissions.deny")#modparam("permissions", "check_all_branches", 1)#modparam("permissions", "allow_suffix", ".allow")#modparam("permissions", "deny_suffix", ".deny")modparam("permissions", "db_url", "mysql://username:password@localhost/openser")modparam("permissions", "db_mode", 1)modparam("permissions", "trusted_table", "trusted")modparam("permissions", "source_col", "src_ip")modparam("permissions", "proto_col", "proto")modparam("permissions", "from_col", "from_pattern")modparam("permissions", "tag_col", "tag")modparam("permissions", "peer_tag_avp", "$avp(i:707)")loadmodule "lcr.so"modparam("lcr", "db_url", "mysql://username:password@localhost/openser")modparam("lcr", "gw_table", "gw")modparam("lcr", "gw_name_column", "gw_name")modparam("lcr", "ip_addr_column", "ip_addr")modparam("lcr", "port_column", "port")modparam("lcr", "uri_scheme_column", "uri_scheme")modparam("lcr", "transport_column", "transport")modparam("lcr", "grp_id_column", "grp_id")modparam("lcr", "lcr_table", "lcr")modparam("lcr", "strip_column", "strip")modparam("lcr", "prefix_column", "prefix")modparam("lcr", "from_uri_column", "from_uri")modparam("lcr", "priority_column", "priority")modparam("lcr", "gw_uri_avp", "$avp(i:709)")modparam("lcr", "ruri_user_avp", "$avp(i:500)")modparam("lcr", "contact_avp", "$avp(i:711)")modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)")modparam("lcr", "fr_inv_timer", 90)modparam("lcr", "fr_inv_timer_next", 30)modparam("lcr", "rpid_avp", "$avp(i:302)")modparam("lcr", "dm_flag", 25) loadmodule "alias_db.so"modparam("alias_db", "db_url", "mysql://username:password@localhost/openser")modparam("alias_db", "user_column", "username")modparam("alias_db", "domain_column", "domain")modparam("alias_db", "alias_user_column", "alias_username")modparam("alias_db", "alias_domain_column", "alias_domain")modparam("alias_db", "use_domain", 0) # ----- presence params -----loadmodule "presence.so"loadmodule "presence_xml.so"loadmodule "presence_mwi.so"loadmodule "pua.so"loadmodule "pua_mi.so"loadmodule "pua_usrloc.so"modparam("presence|presence_xml", "db_url","mysql://username:password@localhost/openser")modparam("presence_xml", "force_active", 1)modparam("presence", "server_address", "sip:my.ip.address:5060")modparam("presence", "presentity_table", "presentity")modparam("presence", "active_watchers_table", "active_watchers")modparam("presence", "watchers_table", "watchers")modparam("presence", "clean_period", 100)modparam("presence", "max_expires", 3600)modparam("presence", "fallback2db", 1)modparam("presence_xml", "xcap_table", "xcap")modparam("presence_xml", "pidf_manipulation", 1)modparam("presence_xml", "integrated_xcap_server", 1)modparam("pua_usrloc", "default_domain", "my.domain.com")modparam("presence", "to_tag_pref", 'pres') ######################################################################### Request route 'main'########################################################################route[0]{xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");force_rport();if(msg:len > max_len){xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("513", "Message Too Big");exit;}if (!mf_process_maxfwd_header("10")){xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("483", "Too Many Hops");exit;}if(!is_method("REGISTER|MESSAGE")){if(nat_uac_test("19")){record_route(";nat=yes");}else{record_route();}}if(is_method("CANCEL") || is_method("BYE")){end_media_session();}if(loose_route()){if(!has_totag()){xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("403", "Initial Loose-Routing Rejected");exit;}if(nat_uac_test("19") || search("^Route:.*;nat=yes")){fix_nated_contact();setbflag(6);}route(7);}if(is_method("REGISTER")){route(6);}if(is_method("INVITE")){route(8);}if(is_method("PUBLISH|SUBSCRIBE")){route(17);}if(is_method("CANCEL") || is_method("ACK")){route(14);}############################# MESSAGES #############################if(is_method("MESSAGE")){route(18);}route(15);}######################################################################### Request route 'clear-usr-preferences-caller'########################################################################route[1]{xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");}######################################################################### Request route 'clear-usr-preferences-callee'########################################################################route[2]{xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");}######################################################################### Request route 'usr-preferences-caller'########################################################################route[3]{route(1);xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# load caller avpsavp_db_load("$avp(s:caller_uuid)", "*");}######################################################################### Request route 'usr-preferences-callee'########################################################################route[4]{xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# load callee avpsavp_db_load("$avp(s:callee_uuid)", "*");}######################################################################### Request route 'stop-media-proxy'########################################################################route[5]{if(isflagset(22)){end_media_session();}}######################################################################### Request route 'base-route-register'########################################################################route[6]{sl_send_reply("100", "Trying");if(!www_authorize("", "subscriber")) {xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");www_challenge("", "0");exit;}if(!check_to()) {xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("403", "Spoofed To-URI Detected");exit;}consume_credentials();if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) {fix_nated_register();setbflag(6);}if(!save("location")) {xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_reply_error();exit;}xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");exit;}######################################################################### Request route 'base-outbound'########################################################################route[7]{xlog("L_INFO", "ROUTE 7 RUN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");if(isbflagset(6)){if(!isflagset(22) && !search("^Content-Length:[ ]*0")){setflag(22);use_media_proxy();}t_on_reply("2");}else{t_on_reply("1");}if(!isflagset(21)){t_on_failure("2");}if(isflagset(29)){append_branch();}if(is_present_hf("Proxy-Authorization")){consume_credentials();}xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# no 100 (we already sent it) and no DNS blacklistingif(!t_relay("0x05")){sl_reply_error();if(is_method("INVITE") && isbflagset(6)){end_media_session();}}exit;}######################################################################### Request route 'base-route-invite'########################################################################route[8]{sl_send_reply("100", "Trying");if(from_gw()){$avp(s:caller_uuid) = "0";xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");setflag(23);}else{if(allow_trusted()){if(is_avp_set("$avp(s:peer_uuid)/s")){# use tag-column from trusted-table as uuid for this calleravp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d");}else{# if no uuid is set, use "0" as default uuid$avp(s:caller_uuid) = "0";}xlog("L_INFO", "Call from trusted peer with uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");if(!is_domain_local("$rd")){xlog("L_INFO", "Rejecting peering attempt with non-local request domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("403", "Relaying Denied");exit;}setflag(23);}else{# if(!proxy_authorize("", "subscriber")) # {# # xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# proxy_challenge("", "0");# exit;# }# if(!check_from()) # {# # xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# sl_send_reply("403", "Spoofed From-URI Detected");# exit;# }}}if(nat_uac_test("19")) {fix_nated_contact();setbflag(6);}route(9);}######################################################################### Request route 'invite-find-callee'########################################################################route[9]{if(alias_db_lookup("dbaliases")){xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");}route(2);if(!is_domain_local("$rd")){setflag(20);$avp(s:callee_uuid) = "0";route(11);}avp_delete("$avp(s:callee_uuid)");avp_db_query("select id from subscriber where username = '$rU'", "$avp(s:callee_uuid)");if(is_avp_set("$avp(s:callee_uuid)")){xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(10);}else{$avp(s:callee_uuid) = "0";xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(11);}exit;}######################################################################### Request route 'invite-to-internal'########################################################################route[10]{route(4);if(!lookup("location")) {xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("404", "User Offline");}else{xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(7);}exit;}######################################################################### Request route 'invite-to-external'########################################################################route[11]{if(isflagset(20)){xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(7);exit;}route(13);if(!isflagset(23)){# don't allow calls relaying from PSTN to PSTN, if not explicitely forwardedif(uri =~ "^sip:[0-9]+@"){# only route numeric users to PSTNif(!load_gws()){xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("503", "PSTN Termination Currently Unavailable");exit;}if(!next_gw()){xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("503", "PSTN Termination Currently Unavailable");exit;}setflag(21);t_on_failure("1");route(7);}}xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("404", "User Not Found");exit;}######################################################################### Request route 'normalize-e164'########################################################################route[12]{# European numbering plans look like this:# CC = country code (i.e. 43 for Austria)# NDC = national destination code (i.e. 1 for Vienna)# SN = subscriber number (i.e. 4001234)# # CC + NDC + SN, i.e. 4314001234## Within the same CC+NDC, it can be omitted, so if# +4314001234 wants to call +4315002345, one can dial# just 5002345.## Within the same CC, CC can be ommitted and a "0" is prefixed# to NDC, so if +4314001234 wants to call +4326003456, # one can dial 026003456.## For international calls, either "00" or + is prefixed, like# +49123456780 or 0049123456789.#avp_delete("$avp(s:orig_callee_user)/g");$avp(s:orig_callee_user) = $rU;if(uri =~ "^sip:(\+[1-9])?[0-9]+@"){# looks like a PSTN numberif(uri =~ "^sip:0[1-9][0-9]+@"){# we have format 0+NDC+SNstrip(1);prefix("+44");}else if(uri =~ "^sip:00[1-9]+@"){# we have format 00 + CC + NDC + SNstrip(2);prefix("+");}else if(!uri =~ "^sip:\+[1-9][0-9]+@"){# unknown format, maybe NDC wasn't added before?xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");return(-1);}# else we have "+" + CC + NDC + SNxlog("L_INFO", "Normalized callee '$avp(s:orig_callee_user)' to E.164 format '$rU' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");return(1);}else{xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");return(-1);}}######################################################################### Request route 'lookup-enum'########################################################################route[13]{route(12);if($rc == 1){if(enum_query("e164.org.")){# TODO: do GW fallback (load gws, set failure-route)?xlog("L_INFO", "ENUM query succeeded - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(7);exit;}else{xlog("L_INFO", "ENUM query failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# ENUM query failed, revert $rUavp_pushto("$ru/username", "$avp(s:orig_callee_user)");}}}######################################################################### Request route 'base-route-local'########################################################################route[14]{t_on_reply("1");if(t_check_trans()){xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");if(!t_relay()){sl_reply_error();}}else{xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");}exit;}######################################################################### Request route 'base-route-generic'########################################################################route[15]{xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("501", "Method Not Supported Here");exit;}######################################################################### Request route 'base-filter-failover'########################################################################route[16]{if(!t_check_status("408|500|503")){xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(5);exit;}}########################################################################## Presence#########################################################################route[17]{# if(nat_uac_test("19") || search("^Route:.*;nat=yes"))# {# fix_nated_contact();# setbflag(6);# } if (!t_newtran()){sl_reply_error();exit;};#if(is_method("PUBLISH"))#{# xlog("L_INFO", "PUBLISH - Process Publish Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# handle_publish();# t_release();#}if(is_method("PUBLISH")){if($hdr(Sender)!= NULL)handle_publish("$hdr(Sender)");elsehandle_publish();t_release();}if(is_method("SUBSCRIBE")){handle_subscribe();xlog("L_INFO", "SUBSCRIBE - Process Subscribe Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");t_release();}exit;}######################################################################### SEND MESSAGES########################################################################route[18]{if (!lookup("location")) {switch ($retcode) {case -1:case -3:t_newtran();t_reply("404", "Not Found");exit;case -2:sl_send_reply("405", "Method Not Allowed");exit;}}if (!t_relay()) {sl_reply_error();};exit;}######################################################################### Reply route 'base-standard-reply'########################################################################onreply_route[1]{xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n");exit;}######################################################################### Reply route 'base-nat-reply'########################################################################onreply_route[2]{xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n");if(nat_uac_test("1")){fix_nated_contact();}if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {if(!search("^Content-Length:[ ]*0")){use_media_proxy();}}exit;}######################################################################### Failure route 'pstn-failover'########################################################################failure_route[1]{xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(16);if(!next_gw()){xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(5);exit;}t_on_failure("1");route(7);}######################################################################### Failure route 'base-standard-failure'########################################################################failure_route[2]{route(16);route(5);} _________________________________________________________________ Get Hotmail on your mobile, text MSN to 63463! http://mobile.uk.msn.com/pc/mail.aspx

1 0

[OpenSER-Users] Strange OpenXCAP behaviour in virtualizedenvironment
by Adrian Georgescu 13 Jan '08

13 Jan '08

>>> I would _really_ recommend to add this to the installations instructions on openxcap.org. Have a nice weekend! Sebastian Done. We will setup a read/write page for OpenXCAP deployment experiences. Regards, Adrian

1 0

[OpenSER-Users] Script variable duration is unexpected
by Iñaki Baz Castillo 13 Jan '08

13 Jan '08

Hi, I've been suffering an extrange issue and now I've realized why. It's related to script variables and their duration. I'm 90% sure that the behaviour is unexpected according to the doc so I've open a bug containing an easy example: https://sourceforge.net/tracker/index.php?func=detail&aid=1870301&group_id=… My conclusion is that using script variables instead of AVP's can be risky in certain cases. Opinions are welcome. Best regards. -- Iñaki Baz Castillo

1 0

Re: [OpenSER-Users] Strange OpenXCAP behaviour in virtualizedenvironment
by Schumann Sebastian 12 Jan '08

12 Jan '08

Dear all I finally found the mistake: My VMWare ESX server for virtualization is not connected to the internet, my local PC was. It didn't come to my mind ever before. Although - I never saw it documented anywhere, that OpenXCAP needs internet access for downloading the namespace files. I would _really_ recommend to add this to the installations instructions on openxcap.org. Have a nice weekend! Sebastian ________________________________ From: Schumann Sebastian Sent: Friday, 11. January 2008 11:54 AM To: users(a)openser.org Subject: [OpenSER-Users] Strange OpenXCAP behaviour in virtualizedenvironment Dear all This week, I tried again to bring up finally the OpenXCAP server. I tried it first acc. the instruction on OpenXCAP.org on my local virtual VMWare Server 1.0 (before, I always used virtualized ESX server). It worked without major issues, meaning I could install lenny, all required files even via apt-get and the OpenXCAP was running. Surprising for me though, because it never worked yet. Now, when I take _the same_ virtual disk and export it to VMWare ESX server, I get the previously discussed errors: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema How can this be possible? I tried three times with three different machines. They work in VMware Server but not in ESX server. I don't find any answer at all neither can I imagine anyhow, how an application error like this occurs running the program on the very same, identical virtualized system with just two different virtualization platforms around. Anybody else has experiences in that, encountered somehow same strange behaviour with the application or has any other advice? I would really welcome every hint in somehow a direction to solve this. I didn't plan to extend my virtual OpenSER testing platform with a local PC, but it seems I have no other possibility so far... Thanks for your help! A nice weekend to all of you. Best regards Sebastian

2 1

[OpenSER-Users] DB aliases for outgoing calls
by Marc LEURENT 11 Jan '08

11 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Evening, I have a little question: I have users with usernames and 10 phone numbers from my SIP/PSTN provider. I have a dbaliases table for incoming calls, in order to redirect the 0xxxxxxxxx number to an user. 0xxxxxxxx1 => bob I would like for outgoing calls to replace the user by the telephone number in the From: Header..... bob => 0xxxxxxxx1 Any idea how to do it? Thanks, Marc Leurent -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHh6ssN4+o+2LtdFwRApxIAKCxqefV7OLejg1Xw6mwre5Yy3jYTQCeM66R bAjagfV/6aoN43hPQcMeKMM= =DQe8 -----END PGP SIGNATURE-----

2 2

[OpenSER-Users] Fix up source address.
by Alex Balashov 11 Jan '08

11 Jan '08

Greetings, I have a strange problem using OpenSER 1.3.x with nathelper. Two ethernet interfaces: eth0 = 192.168.0.0/24 eth1 = outside.ip/29 For some reason, no matter what I do to mangle the requests with nathelper's functions, the packet is *always* sent out of eth1 with the *source address* of the machine's eth0. Obviously, the response from the far-end SIP peer never gets back. The packet does physically go over eth1, I know that much from packet captures. I don't even see how this is possible; when OpenSER issues a packet, shouldn't it originate according to the machine's routing table, take the most specific route, and consequently, adopt the right source address? What gives? I have OpenSER 'listening' on both interfaces, and have tried both on and off with this setting. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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[OpenSER-Users] RTP-Proxy
by Helmut Kuper 11 Jan '08

11 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, does anybody know, if openser+rtpproxy are able to detect rtp-timeouts and react on them? It would be good for billing. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhjWa4tZeNddg3dwRAmZcAJkBmJGh6b2qt/6NbFri0mBZU9BwPQCeNYNS Pb+u2IWB4Xp2K8G9fHkQKyo= =MxAh -----END PGP SIGNATURE-----

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[Serusers] how to replace ruri host part with an avp value?
by Tomasz Zieleniewski 11 Jan '08

11 Jan '08

Hi, How can one replace the ruri host part with the usage of attr2uri method? One can change the domain part with the usage of "domain" param but this leaves the port not changed. I would like to replace the whole host part together with port value. Maybe this can be done other way round? Thanks in advance for any help. Bests -tomasz

2 2

[OpenSER-Users] OpenXCAP: rls-services 404 Not found,and no log?
by KevinKinnan 11 Jan '08

11 Jan '08

I tried example of rfc 4825. Creating resource-lists is ok, but not rls-services, 404 not found response is received. And, there is not any new log appearing in the syslog, I can't debug where I am wrong. HTTP URI is: http://xcap-root/rls-services/users/sip:alice@open-ims.test/index XML is: <?xml version=\"1.0\" encoding=\"UTF-8\"?> <rls-services xmlns=\"urn:ietf:params:xml:ns:rls-services\"> <service uri=\"sip:myfriends@open-ims.test\"> <resource-list>http://xcap-root/resource-lists/users/sip:alice@open-ims.test/index/~~/reso…</resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> Regards, Kevin _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

2 1

[OpenSER-Users] Strange OpenXCAP behaviour in virtualized environment
by Schumann Sebastian 11 Jan '08

11 Jan '08

Dear all This week, I tried again to bring up finally the OpenXCAP server. I tried it first acc. the instruction on OpenXCAP.org on my local virtual VMWare Server 1.0 (before, I always used virtualized ESX server). It worked without major issues, meaning I could install lenny, all required files even via apt-get and the OpenXCAP was running. Surprising for me though, because it never worked yet. Now, when I take _the same_ virtual disk and export it to VMWare ESX server, I get the previously discussed errors: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema How can this be possible? I tried three times with three different machines. They work in VMware Server but not in ESX server. I don't find any answer at all neither can I imagine anyhow, how an application error like this occurs running the program on the very same, identical virtualized system with just two different virtualization platforms around. Anybody else has experiences in that, encountered somehow same strange behaviour with the application or has any other advice? I would really welcome every hint in somehow a direction to solve this. I didn't plan to extend my virtual OpenSER testing platform with a local PC, but it seems I have no other possibility so far... Thanks for your help! A nice weekend to all of you. Best regards Sebastian

1 0

[OpenSER-Users] Testing.... List Working?
by Douglas Garstang 11 Jan '08

11 Jan '08

Is the list working? Haven't seen much traffic on list today. First post was ok. Second went to list and archive, but I never received it via email. Third has gone nowhere! This is my 4th... ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[OpenSER-Users] TLS problem.
by fengbin 11 Jan '08

11 Jan '08

Hi,all I met a strange problem while I am testing TLS connection between minisip and openser. The following is my openser.cfg (part of that) ......... fork=no log_stderror=yes # Uncomment this to prevent the blacklisting of temporary not available destinations #disable_dns_blacklist=yes # # Uncomment this to prevent the IPv6 lookup after v4 dns lookup failures #dns_try_ipv6=no # uncomment the following lines for TLS support disable_tls = 0 listen = tls:10.11.57.197:5060 tls_verify_client = 1 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser//tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser//tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser//tls/user/user-calist.pem" tls_ciphers_list="NULL-SHA:NULL-MD5:AES256-SHA:AES128-SHA" ...... When I set "tls:10.11.57.197:5061" the registration never succeed. But if I set it to 5060 the registration over TLS is OK. I compared the log of two scenarioes and found the TLS session both are OK,but the difference is that: when the port is 5061 there is an error of forwarding. but the forwarding is because openser think it's not the destination of the registration request. See bellow: Jan 10 16:46:56 [9199] DBG:rr:after_loose: No next URI found Jan 10 16:46:56 [9199] DBG:core:grep_sock_info: checking if host==us: 12==12 && [10.11.57.197] == [10.11.57.197] Jan 10 16:46:56 [9199] DBG:core:grep_sock_info: checking if port 5061 matches port 5060 Jan 10 16:46:56 [9199] DBG:core:check_self: host != me Jan 10 16:46:56 [9199] DBG:core:parse_headers: flags=ffffffffffffffff Jan 10 16:46:56 [9199] DBG:tm:t_newtran: T on entrance=0xffffffff Jan 10 16:46:56 [9199] DBG:core:parse_headers: flags=ffffffffffffffff Jan 10 16:46:56 [9199] DBG:core:parse_headers: flags=78 Jan 10 16:46:56 [9199] DBG:tm:t_lookup_request: start searching: hash=58073, isACK=0 Jan 10 16:46:56 [9199] DBG:tm:matching_3261: RFC3261 transaction matching failed Jan 10 16:46:56 [9199] DBG:tm:t_lookup_request: no transaction found Jan 10 16:46:56 [9199] DBG:core:mk_proxy: doing DNS lookup... Jan 10 16:46:56 [9199] ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no corresponding listening socket) Jan 10 16:46:56 [9199] ERROR:tm:t_forward_nonack: failure to add branches With comparition to that when the port is set to 5060 the trace is : Jan 10 17:07:59 [9410] DBG:rr:find_next_route: No next Route HF found Jan 10 17:07:59 [9410] DBG:rr:after_loose: No next URI found Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: checking if host==us: 12==12 && [10.11.57.197] == [10.11.57.197] Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: checking if host==us: 12==12 && [10.11.57.197] == [10.11.57.197] Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 10 17:07:59 [9410] DBG:core:parse_headers: flags=ffffffffffffffff Jan 10 17:07:59 [9410] DBG:core:parse_headers: flags=8000000 Jan 10 17:07:59 [9410] DBG:core:parse_headers: flags=ffffffffffffffff Jan 10 17:07:59 [9410] DBG:registrar:build_contact: created Contact HF: Contact: <sip:888@10.11.57.192:5061;transport=TLS>;expires=1000 And there is no fwd needed then.So the error didnt occur. Its a little bit strange that when I set the port to 5061,why did openser check the port 5060????? Can anyone help me to figure it out? THX BR -- Fengbin

3 5

[OpenSER-Users] OpenSER Question
by ListAcct 11 Jan '08

11 Jan '08

Hello, I wanted someones honest opinion. I have ran across and read a little on SER and OpenSER have not yet had much time. But would you recommend OpenSER for a service provider who is wanting or thinking about deploying VoIP service for its customers? What other products would you recommend to work along with OpenSER such as billing and accounting interfacing. We have used Asterisk for our internal PBX for about two years and love it. Or would you recommend staying with a commercial product like Cisco or Sonus? We prefer open source since our base will be small to start and not large enough to justify a Sonus which is the preferred commercial product. Any advice is definitely appreciated. Otis.

2 3

[OpenSER-Users] Configuration Data....?
by Douglas Garstang 11 Jan '08

11 Jan '08

I need a way in OpenSER to store some simple configuration data. For example, if a parameter is set, I might perform database updates with avp_db_query(). I looked at the dbtext, db_berkeley and flatstore modules and realise that they cannot be used for storing generic information. They are designed to be used as a db store for userloc info etc. So, simple question, how can I store global persistent (on disk is ok) information and retrieve it with OpenSER? Thanks, Doug. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[OpenSER-Users] Forward to voicemail when not registered
by CSB 10 Jan '08

10 Jan '08

I want to provide for forwarding of calls to voicemail in various circumstances, one of which is when the user is not registered with the registrar. I've looked at various example configuration files that include forwarding (e.g. http://www.sipwise.com/wizard, onsip.org) and have not seen an implementation of this. Could anyone provide any tips regarding this or is it as simple as: route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- <snip> if (!lookup("location")) { if (avp_db_load("$ruri/username", "$avp(i:20116)")) { avp_pushto("$ruri", "$avp(i:20116)"); t_relay; return; } else { sl_send_reply("404", "User Not Found"); return; }; <snip> Am I missing something obvious or is this really quite simple? Regards Cameron

2 1

[OpenSER-Users] openser 1.3 and Serweb
by Daniel 10 Jan '08

10 Jan '08

Hi there, What about openser 1.3 and serweb 0.9.6 ? Do they work fine together ? any experiences ? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

1 0

[Serusers] SER 2_0_0 ERROR: slow timer too slow
by tzieleniewski 10 Jan '08

10 Jan '08

Hi, I encountered the following error: Jun 15 08:57:21 rd ser[12680]: ERROR: slow timer too slow: overflow (7997 - 6974 = 1023) Jun 15 08:57:24 rd last message repeated 46 times Could you please point me what could be the reason? Thanks in advance. Tomasz

2 1

[OpenSER-Users] cdrtool datasources
by TCB 10 Jan '08

10 Jan '08

How do i create a datasource for Cisco RAS. -- TC

1 0

[OpenSER-Users] OpenSER & RADIUS accounting with radiusclient-ng
by Antun Mihalicek 10 Jan '08

10 Jan '08

Hi! I'm trying to enable RADIUS accounting on OpenSER with the use of radiusclient-ng, and using FreeRADIUS as radius server. When I initiate or terminate a session on OpenSER, following error occurs: Jan 10 12:21:58 [30155] ERROR:acc:acc_rad_request: radius-ing failed I've been googling for this issue, and I've found out that some people also had this error, but in their case RADIUS packet was sent to RADIUS server but was rejected , but in my case it wasn't even sent to the RADIUS server, as I saw from debug output on FreeRADIUS. As I've seen on forums, people said that this kind of error means that something's wrong with radiusclient-ng. I've used radiusclient-ng 0.5.5 & 0.5.6, both with same results. Did anyone had similar problem? Does radiusclient-ng have some kind of log or something? Thanks for your answer! openser.conf: #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ #loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth_radius.so" loadmodule "auth.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" loadmodule "avpops.so" ..... # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) ....

3 3

[OpenSER-Users] Parallel Forking
by Gomtesh Jain 10 Jan '08

10 Jan '08

Hi All, I am facing some problem in parallel forking . how should I handle the scenario where I get 200 ok response from both the branch . Gomtesh

2 3

[OpenSER-Users] Openxcap modification - setting config and runtime dir
by Tomasz Zieleniewski 10 Jan '08

10 Jan '08

Hi, I made a small modification in openxcap version 0.9.7 in file openxcap it enables now to specify the configuration directory and runtime directory. This is very useful when You run openxcap in deamon mode. xcap@venom:~/src/openxcap-0.9.7$ ./openxcap -h usage: openxcap [options] options: --version show program's version number and exit -h, --help show this help message and exit --no-fork run the process in the foreground (for debugging) --pid=File pid file (/var/run/openxcap/openxcap.pid) --conf-dir=Dir config directory (/etc/openxcap) --run-dir=Dir pid file (/var/run/openxcap) Cheers tomasz

1 0

[Serusers] alias as regular expression
by Sandrinho 09 Jan '08

09 Jan '08

Hi all. Is it possible to configure aliases as regular expressions? I mean: I have to manage some accounts on a SER: each account must receive calls on an arc of numbers. so i have an account "foo" with (for example) 200 aliases foo000,foo001,...,foo199. Is it possible to configure only one alias in form of regexp? "foo[01][0-9][0-9]" in this case I don't want to touch ser.cfg. Thanks a lot. Sandro.

1 0

Re: [OpenSER-Users] Carrirroute
by Henning Westerholt 09 Jan '08

09 Jan '08

On Wednesday 09 January 2008, TCB wrote: > Got it, you are right , This is set as the default for > rewrite_prefix_column in the > online manual( This might be a typo). > > Got it working 100% now , thanks > > modparam("carrierroute", "rewrite_prefix_column", "rewrite_host" Good. Yes, there is a typo in the documentation. I'll fix this. Cheers, Henning

1 0

[OpenSER-Users] uac_replace_from
by CSB 09 Jan '08

09 Jan '08

I wish to replace the from address in the following manner Received From: "05551233" sip:05551233@203.89.001.001 Replaced From: "5551233" sip:18585551233@203.89.001.001 In other words I want to do what the strip and prefix commands do but on a from header. Could anyone advise how this can be achieved in OpenSER 1.1? Regards Cameron

2 2

Re: [OpenSER-Users] Carrirroute
by Henning Westerholt 09 Jan '08

09 Jan '08

On Wednesday 09 January 2008, TCB wrote: > Thanks Henning , i have just realised i had not set the prob, > now the routing works but i'm afraid i have created another > problem. For some reason the ip address is being prefixed to the > called number. > See below. > > Jan 9 14:56:02 ss-za /usr/sbin/openser[20127]: > INFO:carrierroute:get_route_tree_by_id: searching in carrier default > Jan 9 14:56:02 ss-za /usr/sbin/openser[20127]: > DBG:carrierroute:get_route_tree_by_id: tree default, domain 0 : 0 > Jan 9 14:56:02 ss-za /usr/sbin/openser[20127]: DBG:core:hash_func: hash: > 2298 % 1000 = 298 > Jan 9 14:56:02 ss-za /usr/sbin/openser[20127]: > INFO:carrierroute:carrier_rewrite_msg: uri 0727785841 was rewritten to sip: > 196.15.164.13727785841(a)196.15.164.13 > > My carrierroute table looks like this: > > id | carrier | scan_prefix | domain | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix > ----+---------+-------------+--------+------+-------+---------------+----------------+---------------- > 2 | 0 | 072 | 0 | 1 | 1 | 19.1.164.13 | 27 | > 1 | 0 | 011 | 0 | 1 | 1 | 19.1.164.13 | 27 | Hi, your table is correct. I've seens this before, there was some user who copied the content of rewrite_host into rewrite_prefix, i think. Could you check if the RAM-cache on the module has also the correct data? (with 'openserctl fifo cr_dump_routes') Do you do other RURI rewriting before the route[22] where the cr method is called? Cheers, Henning

1 0

[Serusers] parallel forking
by Steven C. Blair 09 Jan '08

09 Jan '08

I seem to be running into problems using avps and parallel forking. Is it possible to insure the invite message sent to each contact address has the same "avp checks" performed for the username portion of the URI? Thanks,Steve

1 0

[OpenSER-Users] OpenSER in front of Asterisk
by lists＠infoway.net 09 Jan '08

09 Jan '08

Hi all, I'm relatively new to OpenSER (and SER for that matter). I have the following need and can't quite figure out how to do it. We have several Asterisk servers being used strictly for IVR purposes (IVR cluster). We also have several Asterisk servers used for standard telephone services and Agent/Queue services (telephone Cluster). We would like to setup one or two OpenSER machines in front of all that to serve the following purposes: 1) Route incoming calls appropriately to either the IVR cluster or the telephone cluster based on the DNIS 2) Be used as LCR servers so that the telephone cluster will send all outbound calls to and the OpenSER will determine if it needs to reach one of our termination carriers or simply just handle it internally (e.g. extension-to-extension calling) 3) When a caller in the IVR presses certain options, the caller may be transferred to an agent. So, we need to take the call away from the IVR cluster (possibly back to the OpenSER server) and redirect it to the telephone cluster so that an agent may handle the caller 4) CDR and billing purposes 5) Ideally, all our telephone users would register with OpenSER (instead of the Asterisk telephone cluster) and for "regular" calls, we could just handle everything within OpenSER, and only involve the telephone cluster if we need to redirect to voicemail or something like that We have read on the wiki about integrating SER with Asterisk for voicemail. We also have some snippets, mainly from the mailing list, about SER load balancing traffic to Asterisk. However, we really haven't found something that could address all items above. Could anyone provide some hits, snippets, or references to what we need in order to accomplish these goals? Thanks

1 0

[OpenSER-Users] Configuring openser to forward incoming to a conference agent
by Pascal Maugeri 09 Jan '08

09 Jan '08

Hi I'm developing a "conference agent" that will receive INVITEs and invites a list of invitees defined in the SIP INVITE message payload. Let's say that this agent is available behind the URI sip:conf@yourdomain.com and openser should forward all incoming "INVITE sip:conf@yourdomain.com SIP/2.0" to my agent. How should I configure openser to do that ? The easiest way would be to have my conference agent to REGISTER against the openser but I don't want to maintain this registration and I think there is a better way to do that. I've seen there is also a function called "ul_insert_urecord" in USRLOC module. Might it be a possible solution to add a new entry for my conference agent ? Any other solution or idea ? Regards, Pascal

2 1

[OpenSER-Users] presence caching and retrieval
by Sigrid Thijs 09 Jan '08

09 Jan '08

Hi, I was wondering, when a SIP user publishes his presence to the openser presence module, is this presence data cached in memory by the server or is this immediately stored in the database? Is this somehow configurable? If the presence data is cached in memory, is there an interface available for an external server to query the presence of one or more resources? In particular, we are looking for a performant way to query the presence of multiple users (for example 20). We would like to avoid the overhead of creating a SIP subscription for each resource: - to avoid traffic overhead - we only need to know the current presence of the resource(s) and are not interested in future presence changes. Do you have any suggestions on if/how this can be achieved with openser? kind regards, Sigrid Thijs

2 1

[OpenSER-Users] Working in TCP but NOT in UDP?
by David Villasmil Govea 09 Jan '08

09 Jan '08

Hello all, I've got a working conf on which RTPs work perfectly with TCP, but fails to work with UDP... can anyont tell me why?? Here's my conf: # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=10 #fifo="/tmp/openser_fifo" # --- module loading mpath="/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" #loadmodule "dispatcher.so" loadmodule "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "xlog.so" #loadmodule "/usr/lib/openser/modules/acc.so" #modparam("acc", "early_media", 0) #modparam("acc", "report_cancels", 1) #modparam("acc", "db_flag", 2) #modparam("acc", "db_url", "mysql://user:pwd@localhost/openser") #modparam("acc", "report_ack", 0) #modparam("acc", "db_missed_flag", 3) #modparam("acc", "failed_transaction_flag", 4) #modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") # --- setting module parameters modparam("usrloc|auth_db","db_url","mysql://user:pwd@localhost/openser") # --- setting module parameters #modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 3) # -- registrar params -- #modparam("registrar", "received_avp", "$avp(s:rcv)") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("registrar|nathelper", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@whatever.com") #modparam("nathelper", "received_avp", "$avp(s:rcv)") # -- xlog modparam("xlog", "buf_size", 8192) modparam("registrar", "max_contacts", 5) # --- main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("OPENSER_IP", "subscriber")) { www_challenge("OPENSER_IP", "0"); exit; }; if (isflagset(5)) { setbflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; # # - TERMINATIONS # if (method=="INVITE") { if (!proxy_authorize("OPENSER_IP","subscriber")) { if($si=="GW_IP"){ xlog("L_NOTICE", "\n************** ALLOWING INCOMING CALL AS IT COMES FROM $si ************\n"); }else{ proxy_challenge("OPENSER_IP","0"); exit; } } if (uri=~"^sip:00"){ xlog("L_NOTICE", "\n************** GOT OUTSIDE CALL ************\n"); strip(2); prefix("M3ll4m0d4v1d"); rewritehostport("GW_IP:5060"); sl_send_reply("100","Trying"); forward(); exit(); } } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; } MANY thanks to all. David

5 4

[OpenSER-Users] Carrirroute
by TCB 09 Jan '08

09 Jan '08

Would appreciate some help with CarrierRoute module I get the following error ERROR:carrierroute:carrier_rewrite_msg: error during rewrite_uri_recursor This is my Route route[22] { # route calls based on hash over callid # choose route domain 0 of the default carrier if(!cr_rewrite_uri("0", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In case of failure, re-route the request t_on_failure("3"); # Relay the request to the gateway t_relay(); } } -- TC

2 1

[OpenSER-Users] INVITE, exec_dset(), and BYE
by Juan Carlos Castro y Castro 09 Jan '08

09 Jan '08

I have a scenario in which an OpenSER machine distributes load among several Asterisk machines for outgoing PSTN. I use an external program which I call through exec_dset() to select which server a call goes to. I use record_route(). All calls are completed OK, but when I dial from some SIP user-agents and hang up from the caller side, OpenSER gets crazy and doesn't know where to route the BYE to. Here's the reason: Good scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:providercode#5551234@chosenserver SIP/2.0" - I get an OK back, all is well. (Sometimes the # in the uri above is sent as %23, but it works either way) Bad scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:5551234@mydomain SIP/2.0" - My openser.cfg doesn't know how to handle that and I get a "Loop Detected" back. The destination never gets a hangup signal. Is the second UA from the "bad" scenario disrespecting the RFC? Do I have any recourse to route a BYE that comes like that to the right Asterisk server? Thanks, Juan

3 4

[OpenSER-Users] OpenXCAP: no percent-encoded URI allowed? can't fetch element that beneath multiple-folded elements?
by KevinKinnan 09 Jan '08

09 Jan '08

The OpenXCAP 0.9.7(version 0.9.7) allows the HTTP URI that is not percent-encoded while the percent-encoded URI is rejected by the server. There is no problem of installation. Two types of http URIs are shown below: Accept: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour…"friends"] Reject: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… Another problem is about fetching element that within multiple-folded elements, for example: <list name="friend"> <entry uri="sip:alice@example.com"> <display-name>Alice</display-name> </entry> <entry uri="sip:bob@example.com"> <display-name>Bob</display-name> </entry> </list> <list name="closefriends"> <entry uri="sip:tom@example.com"> <display-name>Tom</display-name> </entry> <entry uri="sip:mike@example.com"> <display-name>Mike</display-name> </entry> </list> If fetching the element of "mike", the responsed to the following URI is "404 Not found". http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… However, an alternative way of uing element sequence number works. http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… Hope anyone can give me some clues, thanks a billion Regards, Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

2 1

[OpenSER-Users] Working in TCP but NOT in UDP?
by David Villasmil Govea 09 Jan '08

09 Jan '08

> > > Hello all, > > > > I've got a working conf on which RTPs work perfectly with TCP, but > > fails to work with UDP... can anyont tell me why?? > > Could you detail more what the issue is? Do you mean that the SIP works > fine > over TCP and fails over UDP? RTP has nothing to do here. > > > -- > I?aki Baz Castillo > Hello, That's exactly what I mean, signalling being TCP or UPD got nothing to do here.... but it DOES work on TCP and NOT over UDP! Weird... David

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Re: [OpenSER-Users] how to restrict invite when regiest unsuccessful
by liu york 09 Jan '08

09 Jan '08

yes, when i used same authenticating on both regiest and invite.the problem was resolved.many many thx. ======= 2008-01-08 15:16:35 您在来信中写道：======= > > >Alex Balashov schrieb: >> On Tue, 8 Jan 2008, liu york wrote: >> >>> I want to restrict invite when regiest unsuccessful even the invite >>> request have right authentication. actually,i just want to only one >>> person can call to pstn,others cant call even have sipid and >>> password.now i has set by modparam("registrar", "max_contacts", 10).but >>> cant restrict the invite by the sipphone no need regiestion. >> >> Are you authenticating INVITEs differently than you are authenticating >> registrations? In other words, are the authentication realm and required >> credentials different for your proxy_challenge() than they are for >> your www_challenge() for handling REGISTER requests? >> >> If they are, it seems that all you should have to do is do a lookup() >> and see if it fails. Presumably, you are doing a save() into your >> registrar/usrloc only on valid REGISTER challenges? >> >> >> i.e. >> >> if(is_method("INVITE")) { >> if(!lookup("location")) { > > if(!registered("location")) { >is sufficient > >btw: this does not give you any security > >regards >klaus > > >> sl_send_reply("403", "Forbidden -- REGISTER invalid."); >> exit; >> } >> } >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : +1-678-954-0670 >> Direct : +1-678-954-0671 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2008-01-09

1 0

Re: [OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

Sorry, not all of the text I originally sent has appeared on the list archives, so here is the full email another time! :P Hi, can anyone provide any advice on getting CDRTool to work? I have installed CDRTool 6.0 on a Debian Linux system 2.6 Kernel with Apache/2.2.6 (Unix) PHP/5.2.5 I am attempting to get it to work with OpenSER only (I have commented out the Asterisk and sip_trace sections in global.inc) and I am recieving the error: Error initializing CDRTool datasource >From in my web browser. It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

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Re: [OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

mmm, again! :S well here is the bottom half of my email.... It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

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[OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

Hi, can anyone provide any advice on getting CDRTool to work? I have installed CDRTool 6.0 on a Debian Linux system 2.6 Kernel with Apache/2.2.6 (Unix) PHP/5.2.5 I am attempting to get it to work with OpenSER only (I have commented out the Asterisk and sip_trace sections in global.inc) and I am recieving the error: Error initializing CDRTool datasource >From in my web browser. It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

1 0

[OpenSER-Users] Problem adding permanent user locations using openserctl
by Ashley Unitt 08 Jan '08

08 Jan '08

I've recently installed a vanilla OpenSER v1.3.0 (a new install) and I'm trying to set up permanent user locations using openserctl as I have some devices that won't register on their own. I am currently running without a database. When trying to add a user using the command line "openserctl ul add 100 sip:192.168.1.40" I get the error: ERROR: 400; check if you use aliases in OpenSER. Can someone please give me some pointers on what to try? Thanks Ashley I've modified the openserctlrc as follows: # $Id: openserctlrc,v 1.2 2006/07/05 19:37:20 miconda Exp $ # # openser control tool resource file # # here you can set variables used in the openserctl ## your SIP domain SIP_DOMAIN=openser.development.newvoicemedia.com ## database type: MYSQL or PGSQL, by defaulte none is loaded # DBENGINE=MYSQL ## database host # DBHOST=localhost ## database name # DBNAME=openser ## database read/write user # DBRWUSER=openser ## database read only user # DBROUSER=openserro ## password for database read only user # DBROPW=openserro ## database super user # DBROOTUSER="root" ## type of aliases used: DB - database aliases; UL - usrloc aliases ## - default: none ALIASES_TYPE=""UL ## control engine: FIFO or UNIXSOCK ## - default FIFO # CTLENGINE="FIFO" ## path to FIFO file # OSER_FIFO="FIFO" ## check ACL names; default on (1); off (0) # VERIFY_ACL=1 ## ACL names - if VERIFY_ACL is set, only the ACL names from below list ## are accepted # ACL_GROUPS="local ld int voicemail free-pstn" ## verbose - debug purposes - default '0' VERBOSE=1 Free Phone: 0800 280 2888 Tel: +44 (0)1256 892020 Fax: +44 (0)1256 892200 Registered Office: NewVoiceMedia Ltd, Laverstoke Grange, Laverstoke, Whitchurch, Hants, RG28 7PF NewVoiceMedia Registered in England No: 3602868

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[OpenSER-Users] Openser + Mediaproxy behind NAT
by Halomoan Chow 08 Jan '08

08 Jan '08

Hi Is it possible to put Openser + Mediaproxy behind NAT while serving internal phones and external phones ? Internal phone <-> Openser+MediaProxy <-> NAT <-> external phone (maybe behind nat also) Openser is act as a registrar in this case. If you have successful story with this setting, please share. Thank you so much Regards, Halomoan

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[OpenSER-Users] [OT] Error in Rome VON material about OpenSer and Asterisk Realtime
by Iñaki Baz Castillo 08 Jan '08

08 Jan '08

Hi, just as curiosity I comment that there is an error (an Asterisk error in fact) in the material of the Rome VON course of 2007: In file "config/08-asterisk.conf" it appears: CREATE VIEW sipusers AS SELECT username as name, [...] domain as defaultip, domain as ipaddr, [...] FROM openser.subscriber; There is a bug in case of SIP SRV domains because Asterisk doesn't perform a SRV DNS query for the location of registerd users (just for outbound calls if "srvlookup=yes") so if OpenSer has a SRV SIP domain "my_srv_domain.com" but DNS type A of that domain points to a different IP then Asterisk will fail (it will resolve to the A DNS entry). A simple solution is to modify the view as follows: CREATE VIEW sipusers AS SELECT username as name, [...] __OpenSer_IP__ as defaultip, __OpenSer_IP__ as ipaddr, [...] FROM openser.subscriber; Of course this is a bug in Asterisk since it should, according to RFC3261, perform a SRV query for the location of registered users in case they are not an IP. Just it ;) Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Question about append-hf and remove_hf
by Helmut Kuper 08 Jan '08

08 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I add some private headerfields (append-hf) during main route resp. sub-routes like "route(3)" and in certain cases I want to remove (remove_hf) some of them in main route befor sending request out. Unfortunately that doesn't work. Even when I add a header field and test for it directly afterwards or try to remove it, openser doesn't find anything. Test-Code is like this: append_hf("P-SI: sad\r\n"); if(is_present_hf("P-SI")) xlog("L_INFO", "######## Found\n"); if(remove_hf("P-SI")) xlog("L_INFO", "######## Removed\n"); textops doc has a hint at append_hf: " ... Note: Headers which are added in main route cannot be removed in further routes (e.g. failure routes). So, the idea is not to add there any headers that you might want to remove later. To add headers temporarely use the branch route because the changes you do there are per-branch. " Not sure if this is the reason for my problem. Is this a bug? Is there a way to add header fields temporarely? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHg1XQ4tZeNddg3dwRArKlAJ9vTWjg93/MeR8w+7R3XtP90p2eDwCgq/90 OfBsmaeapmBrGez4+Rt8NXQ= =3cte -----END PGP SIGNATURE-----

2 2

Re: [OpenSER-Users] how to restrict invite when regiest unsuccessful
by liu york 08 Jan '08

08 Jan '08

OK,i see,let me try. many thx. ======= 2008-01-08 15:16:35 您在来信中写道：======= > > >Alex Balashov schrieb: >> On Tue, 8 Jan 2008, liu york wrote: >> >>> I want to restrict invite when regiest unsuccessful even the invite >>> request have right authentication. actually,i just want to only one >>> person can call to pstn,others cant call even have sipid and >>> password.now i has set by modparam("registrar", "max_contacts", 10).but >>> cant restrict the invite by the sipphone no need regiestion. >> >> Are you authenticating INVITEs differently than you are authenticating >> registrations? In other words, are the authentication realm and required >> credentials different for your proxy_challenge() than they are for >> your www_challenge() for handling REGISTER requests? >> >> If they are, it seems that all you should have to do is do a lookup() >> and see if it fails. Presumably, you are doing a save() into your >> registrar/usrloc only on valid REGISTER challenges? >> >> >> i.e. >> >> if(is_method("INVITE")) { >> if(!lookup("location")) { > > if(!registered("location")) { >is sufficient > >btw: this does not give you any security > >regards >klaus > > >> sl_send_reply("403", "Forbidden -- REGISTER invalid."); >> exit; >> } >> } >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : +1-678-954-0670 >> Direct : +1-678-954-0671 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2008-01-08

1 0

[OpenSER-Users] how to restrict invite when regiest unsuccessful
by liu york 08 Jan '08

08 Jan '08

hi I want to restrict invite when regiest unsuccessful even the invite request have right authentication. actually,i just want to only one person can call to pstn,others cant call even have sipid and password.now i has set by modparam("registrar", "max_contacts", 10).but cant restrict the invite by the sipphone no need regiestion. so plz help,many thx.

3 3

[OpenSER-Users] presence(XCAP) permission rules?
by KevinKinnan 07 Jan '08

07 Jan '08

According to the instruction(README) of 'presence_xml', if using a integrated xcap server like OpenXCAP, the value of 'force_active' should set to 0. Then, the xcap server is queried and the subscription states is according to user defined permission rules. If no rules are defined for a certain watcher, the subscriptions remains in pending state and the Notify sent will have no body. Is the 'user defined permission rules' mentioned above specified in RFC 5025(Presence Authorization Rules) or 'XCAP usage for setting presence authorization'? They seems different. I have tried both of them and stored xml in xcap table for a user, say alice(she allows bob to wather her pre-info). But it was not working, subscription state was still 'pending'. Thank you! Regards, Kevin _________________________________________________________________ MSN圣诞礼物火热登场，免费发放中，快来领取吧！ http://im.live.cn/emoticons/?ID=18

1 0

[OpenSER-Users] XCAP permission rules?
by KevinKinnan 07 Jan '08

07 Jan '08

My OpenSER works with openxcap, force_active is set to 0. The Subscription query the "xcap" table for permission rules of specified watcher. I create a simple xml rule which is shown below, and store it in xcap table. But the subscription-state is still "pending". I found another presence authorization rules that uses different statments like <provide-services> or <transformations>. which one should I use? Thank you! <?xml version="1.0" encoding="UTF-8"?> <permission-statements> <statement id="as8f"> <applies-to> <domain>open-ims.test</domain> </applies-to> <permissions> <accept/> </permissions> </statement> </permission-statements>Regards, Lei _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

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[OpenSER-Users] new CDRTool release 6.1.0
by Adrian Georgescu 07 Jan '08

07 Jan '08

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog since 5.1.1: cdrtool (6.1.0) unstable; urgency=low * Fixed update of billing_rates entries from csv files * Overwrite defaults from sip_settings class per NGNPro connection * Added contrib directory * Added freeradius patch for freeradius cvs version from Norm Brandinger * Log progress of importing rating files * Show rating engine daemon uptime in syslog * Many fixes in provisioning client * Lookup customers and rates in MySQL, there is no memory limit anymore * Importing rating does not lock the rating engine anymore * Fixed billing of calls over the minimum duration cdrtool (6.0) unstable; urgency=low * Fixed links from Group-by output to discrete CDRs * After renormalization, re-cache the monthly usage only for the accounts that had CDRs changed during renormalization * Fixed application type for rating engine (default application is audio) * Added info to ENUM generator * Fixed node IP parsing in sip trace * Fixed default login account creation * Don't log ENUM tld in rate info if is set to none * Added RatingProfiles management in sip settings page * Added prepaid_cards management in Rating tables page * Fixed name of domain table when SIP Thor data-source is enabled * Added link in the trace window to jump to a different SIP Proxy * Import/export delimiter for rating tables can be adjusted, default is , * Must apply database structure changes from setup/mysql/ alter_tables.mysql * NGNPro client completed * Save settings before existing provisioning pages cdrtool (5.3.3) unstable; urgency=low * Fixed ENUM tld match and logging * Show visual ENUM usage * Provisioning engine update cdrtool (5.3.2) unstable; urgency=low * Fixed memory leak cdrtool (5.3.1) unstable; urgency=low * Removed reference to soapFilter which was unused * Skip ENUM tld 'none' in discounts * Update presence functions in provisioning client * Fixed initalization of gatewaygroups * Add function for showing customer text box * Allow specification of custom port names and classes per reseller * Fixed phplib typo that caused crash on php5 cdrtool (5.3.0) unstable; urgency=low * Added ENUM tld based discounts in the rating engine * Added ENUM tld management in the Rating page * Renamed field radius.Framed-Protocol to radius.ENUMtld to accomodate discounts based on ENUM top level domain. This functionality requires changes to the database structures and reconfigure several components: See doc/Upgrade.txt and setup/mysql/alter_tables.mysql for more information * Removed unused SOAP directory * Added NGNPro client provisioning library * Added sip settings page * Added new rights for provisioning tasks * Change main menu item names cdrtool (5.2.6) unstable; urgency=low * Log in syslog the duration of reloading of the rating tables and of the sip accounts into memcacche cdrtool (5.2.5) unstable; urgency=low * Replaced multiple SOAP definitions from global.inc with a pointer to an entry from the $soapEngines defined in soap_engines.inc * Fixed read of HTTP vars in sip trace, public traces are now visible when global vars are turned off cdrtool (5.2.4) unstable; urgency=low * Skip billing_rates.trafficRate absent from csv import * Update rating documentation * Added link to FreeRADIUS-CDRTool by Dan-Cristian Bogos * Fixed sip trace purging * Fix for virtualhost allowing .htaccess customization for cdrtool cdrtool (5.2.3) unstable; urgency=low * Allow usernames without domain part in quota check * Added publish of SIP domain statistics using SIP SIMPLE cdrtool (5.2.2) unstable; urgency=low * Added SIP trace for SIP Thor datasource cdrtool (5.2.1) unstable; urgency=low * Synced UPDATE queries for current and older month (in SQL stored procedures) (you must apply changes from setup/mysql/alter_tables.mysql) cdrtool (5.2.0) unstable; urgency=low * Eliminate the race condition that appears when normalization process is performed between the time radius database is updated when a BYE arrives and the MediaProxy updates the same radius record. cdrtool (5.1.9) unstable; urgency=low * Removed print of sql query cdrtool (5.1.8) unstable; urgency=low * Fixed saving the agreement with license page cdrtool (5.1.7) unstable; urgency=low * Added SOAP/XML management for LCR * Save when the user agrees with the usage license to avoid display the license agreement at each login cdrtool (5.1.6) unstable; urgency=low * Fixed calculus of historical rates when rate is found in alternative profile cdrtool (5.1.5) unstable; urgency=low * Hide price when grouping results for login accounts without showPrice right cdrtool (5.1.4) unstable; urgency=low * Fixed a typo in cdrlib.phtml cdrtool (5.1.3) unstable; urgency=low * Fixed pagination of exported records cdrtool (5.1.2) unstable; urgency=low * Show status of multiple mysql clusters Must migrate global.inc setting $CDRTool['replicated_databases'] to $CDRTool['mysql_clusters'] See setup/global.inc.in for example Regards, Adrian

1 0

[Serusers] SEMS- unix socket problem
by dim 06 Jan '08

06 Jan '08

hi everybody, You can see attached my conf files for ser, ser-sems and sems. I run two instances of ser : 1. the first on port 5060 2. the second which is the sip stack for sems, on 5070 My conf files show that when for example i call 101 or 102 , the call is directed to sems, but the debug of my 2nd ser shows this : 2(8658) write_to_unixsock: Error in connect: Connection refused 2(8658) ERROR:tm:t_write_unix: write_to_unixsock failed 2(8658) write_to_unixsock: Error in connect: Connection refused 2(8658) ERROR:tm:t_write_unix: write_to_unixsock failed Probably these is some kind of problem with my unix socket configuration in my conf files... can anyone help me ??

1 0

[OpenSER-Users] Run on multiple ports
by Arun Kumar 05 Jan '08

05 Jan '08

Hi, how do I run ser/openser on multiple ports. ? I've configured my ser with two listen stat. listen = ip:5060 listen = ip:5070 UA:5070 -------> Cisco (Blocked 5060) --------> SER -------------> UB:5060 UA is able to call UB, but UB is not able to call UA, I know 5060 is blocked but it should goto 5070 ? please guide on this ? Thanks, Arun

4 5

Re: [OpenSER-Users] newbie help - handeling 302s and 404s
by Jesus Rodriguez 05 Jan '08

05 Jan '08

Hi Lazer, >i need a simple config that achieves the following (w/ out a db) > >if a user is not registered then all calls should go to a default gw >192.168.10.1 > >the default gw will only respond w/ a 302 or a 404 > >if a 302 is received then openser should forward the call on to the >new gw listed in the reply > >if a 404 is received the call should be forwarded to a different gw 172.16.20.1 > >if the send gw is done then the call should die. Try with somehting like this (not tested): if (!lookup("location")) { route(5); exit; } route[5] { rewritehostport("192.168.10.1:5060"); t_on_failure("1"); route(1); exit; } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (t_check_status("404")) { rewritehostport("172.16.20.1:5060"); append_branch(); t_relay(); } else { get_redirects("*"); t_relay(); } } Saludos JesusR. >here is what i was thinking of but as a newbie i may be off my rocker >i seem to get an error > > ERROR: t_check_status: cannot check status for a reply which has no >T-state established > >but again my whole approach may be wrong. > > > if (!lookup("location")) { > >rewritehostport ("192.168.10.1:5060"); >if (t_check_status("(404)|(480)")) { > log("404 or 480 negative reply\n"); >rewritehostport ("172.16.20.1:5060"); >} >else{ >t_on_failure("1"); > t_relay(); >} > > > failure_route[1] { > > get_redirects("*"); > t_relay(); >} > } Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

3 4

[OpenSER-Users] Problem with "uac_replace_from()" and Linksys PAP: "vsf" parameter in Record-Route
by Iñaki Baz Castillo 05 Jan '08

05 Jan '08

Hi, I use "uac_replace_from" with: modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","vsf") So when a call arrives and must be sent to an Asterisk voicemail server I change the From: Linksys PAP -> OpenSer INVITE sip:*500@openser.ilimit.es SIP/2.0 Via: SIP/2.0/UDP 192.168.1.109:5061;branch=z9hG4bK-91c6d79a From: PAP809 <sip:809@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es> Call-ID: 148834d2-a640054a(a)192.168.1.109 OpenSer -> Asterisk INVITE sip:*500@80.94.0.111 SIP/2.0 Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw--> Via: SIP/2.0/UDP 80.94.0.110;branch=z9hG4bKb114.2e2e6ca5.0 Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a From: PAP809 <sip:809_openser.ilimit.es@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es> Call-ID: 148834d2-a640054a(a)192.168.1.109 Asterisk -> OpenSer SIP/2.0 200 OK Via: SIP/2.0/UDP 80.94.0.110;branch=z9hG4bKb114.2e2e6ca5.0;received=80.94.0.110 Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw--> From: PAP809 <sip:809_openser.ilimit.es@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es>;tag=as32a464eb Call-ID: 148834d2-a640054a(a)192.168.1.109 OpenSer -> Linksys PAP SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw--> From: PAP809 <sip:809@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es>;tag=as32a464eb Call-ID: 148834d2-a640054a(a)192.168.1.109 The problem is that this "200 OK" from OpenSer to Linksys PAP is not recognized by Linksys PAP who doesn't reply with an ACK (just ignores the "200 OK"), so the "200 OK" is resent again and again by Asterisk. This issue just occurs with the Linksys PAP: Product Name: PAP2-NA Software Version: 2.0.12(LS) Hardware Version: 0.03.4 There is no NAT problem and if I set: modparam("uac","from_restore_mode","none") then the problem dissapeares, so the problem is the existence of "vsf" parameter in Record-Route. If this parameter doesn't exist the problem doesn't occur. Again I repeat that this issue doesn't occur with others phones. Any idea of why this occurs with Linksys PAP? Thanks for any suggestion. PD: Changing the parameter name does solve nothing. PPD: What about if I don't restore the From in the replies? According to RFC3261 the only important is the matching of From_tag, To_tag and Call-id, so, maybe I just should set: modparam("uac","from_restore_mode","none") and forget this issue? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 3

[OpenSER-Users] Digest auth on openwrt without mysql ?
by Michael Bradley Jr 04 Jan '08

04 Jan '08

Hi, is it possible to run openser with digest auth on an openwrt router *without* mysql? If yes can someone please provide me a hint/link? Did try loading "/usr/lib/openser/module/auth.so" without great success, since my box lacks a mysql server version! Thanks in advance Michaek

4 3

[OpenSER-Users] # characters in URIs getting reformatted.
by Alex Balashov 04 Jan '08

04 Jan '08

I am using OpenSER to front some Cisco AS5300s. The way the trunk groups on those are set up, a prefix of #755 is required to dial out, so I transform the URI in OpenSER: $rU = "#755" + $rU; The problem is, certain messages never get acknowledged because OpenSER reformats the URI to use escaped sequences. I am not sure if this is RFC compliant. For instance, a 200 OK comes from the media gateway, and the ACK that OpenSER sends back has this for the URI scheme: ACK sip:%237557069999999@ip.address:5060 This is in response to an initial INVITE to: sip:#7557069999999@ip.address:5060 ... what should I make of this? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

4 5

Re: [OpenSER-Users] Digest auth on openwrt without mysql ?
by Henning Westerholt 04 Jan '08

04 Jan '08

On Friday 04 January 2008, Michael Bradley Jr wrote: > Hi Henning, > > I'm running openser on an openwrt (whiterussion 0.9) box and I'm quite new > to openser. > I did install openser and would like to quickly add a domain and some users > that could digest auth on my box. > I know RTFM but can you tell me the direction to go for this quick setup? Hi, look at http://www.openser.org/docs/modules/1.3.x/ for the modules docs. Basically you need to load a database driver of your choice and the db_auth module. Take a look at the docs for this module for the available functions for authentification, in the example config there are also some hints. I don't have experience with OpenSER on the openWRT, unfortunally, so i don't can give you more detailed hints about this special setup. You need to specify your database type in the /etc/openser/openserctlrc file, and then you could use the openserdbctl tool to create a new database. After that you could add a new user with the openserctl tool, and authentification should work. It it don't, increase the debug level to get some hints about the problems. Run this tools without any command to get some help, there is also some documentation about the installation on the wiki. Hope this helps - cheers, Henning

1 0

[OpenSER-Users] Fetching multiple rows from DB.
by Alex Balashov 04 Jan '08

04 Jan '08

So, is there a legitimate way to retrieve multiple rows using avp_db_query() and iterate through them with a loop structure? Does AVP support the notion of an "array" construction? And if not, how do I get OpenSER to apply a set of rules codified in custom database structures? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

3 2

[OpenSER-Users] UAC stops responding to opensers 401 unauthorized challenge after few hours
by Raju Abhyankar 03 Jan '08

03 Jan '08

I need some help and guidance here as I am fairly new to openser but have made some progress.... Can some one throw some light on this issue. I am using a standard openser script from sipwise.com. I have several ATA or UAC registered with openser-1.1.1-tls. THe registration process works just fine. I can make calls etc. no issues. I am using Mediaproxy for nat traversal and no issues there too. Since I am using the internet (not private LAN) to connect these devices, I have set my register expiration time to 1200 seconds. Now after a few hours of operation, I see that some of the ATA or UAC (HT486) deregister. I have used "ngrep" and I can see: 1. UAC sends REGISTER request 2. openser responds by saying "100" "Trying" 2. Openser generates a 401 unauthorized 3. UAC once again generates REGISTER request and this goes on. The only way to correct this issue is to reboot the ATA by powering it off and on. Any ideas why this may happen? When I run the same devices using Asterisk I dont see the devices deregister but I would like to use Asterisk only a PSTN gateway. I have included a trace of this below Thanks in advance Raju ==================================================== From: <sip:88@lmnopq.com>;tag=ed8cb29199123c47. To: <sip:88@lmnopq.com>. Contact: <sip:88@YY.YYY.YYY.YY:50826>. Supported: replaces, timer. Call-ID: a679dcef0eebdbec(a)192.168.1.5. CSeq: 100 REGISTER. Expires: 60. User-Agent: TigerNetCom 1.02.12.0.9.11. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE. Content-Length: 0. . U 2008/01/03 13:42:29.518524 XXX.XX.XXX.XXX:5060 -> YY.YYY.YYY.YY:50825 SIP/2.0 100 Trying. Via: SIP/2.0/UDP YY.YYY.YYY.YY:50826;branch=z9hG4bK0826bc66f89e21de;rport=50825. From: <sip:88@lmnopq.com>;tag=ed8cb29199123c47. To: <sip:88@lmnopq.com>. Call-ID: a679dcef0eebdbec(a)192.168.1.5. CSeq: 100 REGISTER. Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. . U 2008/01/03 13:42:29.518566 XXX.XX.XXX.XXX:5060 -> YY.YYY.YYY.YY:50825 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP YY.YYY.YYY.YY:50826;branch=z9hG4bK0826bc66f89e21de;rport=50825. From: <sip:88@lmnopq.com>;tag=ed8cb29199123c47. To:<sip:88@lmnopq.com>;tag=5cc123387d097550f225581b658b043b.4f69. Call-ID: a679dcef0eebdbec(a)192.168.1.5. CSeq: 100 REGISTER. WWW-Authenticate: Digest realm="lmnopq.com", nonce="477d57f148e6dc9dd21502b6c60d4bfe9d061657". Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. After this the register with same cseq shows up and the device does not register....

1 0

[Serusers] SER fails to match ACK to previous INVITE
by Srinivas Kotamarti 03 Jan '08

03 Jan '08

Hi, I have recently downloaded SER & SEMS and configured them for early_announce application as described in the early_announce module documentation. I am using SIPp to generate an incoming call to test the setup. Upon receiving ACK for some reason SER fails to match it to the previous INVITE and tries to forward it to the destination statelessly and picks it up again and forwards it. It continues to do this until the max forward limit is reached. I cannot figure out what is wrong. Any help is really appreciated. Here is the Route section from ser-sems.cfg file that I am using # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; if (uri==myself) { # filter unsupported requests if (!(method=="REGISTER" || method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" )) { sl_send_reply("501", "method not understood here"); break; } if (method=="REGISTER") { # make UAs which want to register unhappy sl_send_reply("501","registration couldn't be accepted"); break; }; if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method == "INVITE") { avp_write("Final-Reply-Code=405;Final-Reply-Reason=Prohiboted", "$sess_params"); if(!t_write_unix("/tmp/sems_sock","early_announce/early_headers")){ log("could not contact early_announce\n"); t_reply("404","not found"); }; break; } else if (method=="BYE" || method=="CANCEL") { if(!t_write_unix("/tmp/sems_sock","bye")) { t_reply("500","error contacting sems"); }; break; } else if (method=="ACK"){ # absorb ACKs t_relay(); break; }; break; } if (!t_relay()) { sl_reply_error(); }; } Debug logs from SER 0(27347) SIP Request: 0(27347) method: <ACK> 0(27347) uri: <sip:service@10.107.2.246> 0(27347) version: <SIP/2.0> 0(27347) parse_headers: flags=1 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=1 0(27347) parse_headers: this is the first via 0(27347) After parse_msg... 0(27347) preparing to run routing scripts... 0(27347) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(27347) parse_headers: flags=128 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) parse_headers: this is the second via 0(27347) Found param type 232, <branch> = <z9hG4bK-27351-1-3>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) DEBUG: add_param: tag=79EBE47B-477CE4230002578E-B7F6D6C0 0(27347) end of header reached, state=29 0(27347) DEBUG: get_hdr_field: <To> [76]; uri=[sip:service@10.107.2.246:5060]<mailto:[sip:service@10.107.2.246:5060]> 0(27347) DEBUG: to body [sut <sip:service@10.107.2.246:5060>] 0(27347) get_hdr_field: cseq <CSeq>: <1> <ACK> 0(27347) DEBUG:maxfwd:is_maxfwd_present: value = 15 0(27347) DEBUG: add_param: tag=27351SIPpTag001 0(27347) end of header reached, state=29 0(27347) parse_headers: flags=256 0(27347) DEBUG: get_hdr_body : content_length=0 0(27347) found end of header 0(27347) find_first_route: No Route headers found 0(27347) loose_route: There is no Route HF 0(27347) grep_sock_info - checking if host==us: 12==12 && [10.107.2.246] == [10.107.2.246] 0(27347) grep_sock_info - checking if port 5060 matches port 5060 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=3 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) SER: forwarding ACK statelessly 0(27347) DEBUG: mk_proxy: doing DNS lookup... 0(27347) check_via_address(10.107.2.246, 10.107.2.246, 0) 0(27347) Sending: ACK sip:service@10.107.2.246 SIP/2.0 Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246:5061;branch=z9hG4bK-27351-1-3 From: sipp <sip:sipp@10.107.2.246:5061>;tag=27351SIPpTag001 To: sut <sip:service@10.107.2.246:5060>;tag=79EBE47B-477CE4230002578E-B7F6D6C0 Call-ID: 1-27351(a)10.107.2.246<mailto:1-27351@10.107.2.246> CSeq: 1 ACK Contact: sip:sipp@10.107.2.246:5061 Max-Forwards: 14 Subject: Performance Test Content-Length: 0 --Srinivas Kotamarti

1 0

[Serusers] SER fails to match ACK to previous INVITE
by Srinivas Kotamarti 03 Jan '08

03 Jan '08

Hi, I have recently downloaded SER & SEMS and configured them for early_announce application as described in the early_announce module documentation. I am using SIPp to generate an incoming call to test the setup. Upon receiving ACK for some reason SER fails to match it to the previous INVITE and tries to forward it to the destination statelessly and picks it up again and forwards it. It continues to do this until the max forward limit is reached. I cannot figure out what is wrong. Any help is really appreciated. Here is the Route section from ser-sems.cfg file that I am using # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; if (uri==myself) { # filter unsupported requests if (!(method=="REGISTER" || method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" )) { sl_send_reply("501", "method not understood here"); break; } if (method=="REGISTER") { # make UAs which want to register unhappy sl_send_reply("501","registration couldn't be accepted"); break; }; if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method == "INVITE") { avp_write("Final-Reply-Code=405;Final-Reply-Reason=Prohiboted", "$sess_params"); if(!t_write_unix("/tmp/sems_sock","early_announce/early_headers")){ log("could not contact early_announce\n"); t_reply("404","not found"); }; break; } else if (method=="BYE" || method=="CANCEL") { if(!t_write_unix("/tmp/sems_sock","bye")) { t_reply("500","error contacting sems"); }; break; } else if (method=="ACK"){ # absorb ACKs t_relay(); break; }; break; } if (!t_relay()) { sl_reply_error(); }; } Debug logs from SER 0(27347) SIP Request: 0(27347) method: <ACK> 0(27347) uri: <sip:service@10.107.2.246> 0(27347) version: <SIP/2.0> 0(27347) parse_headers: flags=1 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=1 0(27347) parse_headers: this is the first via 0(27347) After parse_msg... 0(27347) preparing to run routing scripts... 0(27347) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(27347) parse_headers: flags=128 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) parse_headers: this is the second via 0(27347) Found param type 232, <branch> = <z9hG4bK-27351-1-3>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) DEBUG: add_param: tag=79EBE47B-477CE4230002578E-B7F6D6C0 0(27347) end of header reached, state=29 0(27347) DEBUG: get_hdr_field: <To> [76]; uri=[sip:service@10.107.2.246:5060]<mailto:[sip:service@10.107.2.246:5060]> 0(27347) DEBUG: to body [sut <sip:service@10.107.2.246:5060>] 0(27347) get_hdr_field: cseq <CSeq>: <1> <ACK> 0(27347) DEBUG:maxfwd:is_maxfwd_present: value = 15 0(27347) DEBUG: add_param: tag=27351SIPpTag001 0(27347) end of header reached, state=29 0(27347) parse_headers: flags=256 0(27347) DEBUG: get_hdr_body : content_length=0 0(27347) found end of header 0(27347) find_first_route: No Route headers found 0(27347) loose_route: There is no Route HF 0(27347) grep_sock_info - checking if host==us: 12==12 && [10.107.2.246] == [10.107.2.246] 0(27347) grep_sock_info - checking if port 5060 matches port 5060 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=3 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) SER: forwarding ACK statelessly 0(27347) DEBUG: mk_proxy: doing DNS lookup... 0(27347) check_via_address(10.107.2.246, 10.107.2.246, 0) 0(27347) Sending: ACK sip:service@10.107.2.246 SIP/2.0 Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246:5061;branch=z9hG4bK-27351-1-3 From: sipp <sip:sipp@10.107.2.246:5061>;tag=27351SIPpTag001 To: sut <sip:service@10.107.2.246:5060>;tag=79EBE47B-477CE4230002578E-B7F6D6C0 Call-ID: 1-27351(a)10.107.2.246<mailto:1-27351@10.107.2.246> CSeq: 1 ACK Contact: sip:sipp@10.107.2.246:5061 Max-Forwards: 14 Subject: Performance Test Content-Length: 0 --Srinivas Kotamarti

1 0

[OpenSER-Users] Daily Snapshots for 1.3.x release
by Anatoly Pidruchny 03 Jan '08

03 Jan '08

Hello, Is it planned to add daily snapshats for the 1.3.x release? They are missing here: http://www.openser.org/downloads/snapshots/ Thanks, /Anatoly.

1 0

[Serusers] Unix Socket Problem
by frq ser 03 Jan '08

03 Jan '08

Hi, Thanks for your reply Stefan. Here is the log from SEMS 0.10.0, which I just installed. This log is in two parts. The first one (with a few lines missing from the top) is displayed when sems is started. The second part is displayed when I send the INVITE message. Thanks again for your help. Frq. SEMS-0.10.0 log (Part 1) . . . /usr/local/lib/sems/plug-in': (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/stat s.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'stats' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/anno uncement.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'announcement' load ed. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/wav. so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 1 inserted (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 2 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMU' inserted with id 0 and order 100 (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMA' inserted with id 8 and order 108 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format Wav inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ilbc .so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 4 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'iLBC' inserted with id 97 and order 197 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format iLBC inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/webc onference.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'webconference' loa ded. (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'webconference' loaded . (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ann_ b2b.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'ann_b2b' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/sess ion_timer.so ... (24406) DEBUG: loadSehPlugIn (AmPlugIn.cpp:457): session component 'session_time r' loaded. (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'user_timer' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/uac_ auth.so ... (24406) DEBUG: loadSehPlugIn (AmPlugIn.cpp:457): session component 'uac_auth' lo aded. (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'uac_auth' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/echo .so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'echo' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/earl y_announce.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'early_announce' lo aded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/reg_ agent.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'reg_agent' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/regi strar_client.so ... (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'registrar_client' loa ded. (24406) DEBUG: loadSIPehPlugIn (AmPlugIn.cpp:502): sip event handler component ' registrar_client' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/voic email.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'voicemail' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/conf erence.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'conference' loaded . (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ivr. so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'ivr' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/gsm. so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 3 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'GSM' inserted with id 3 a nd order 103 (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/anno unce_transfer.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'announce_transfer' loaded. (24406) DEBUG: load (AmPlugIn.cpp:172): AmPlugIn: modules loaded. (24406) DEBUG: load (AmPlugIn.cpp:174): AmPlugIn: Initializing plugins... (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'first_parti cipant' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'join_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'drop_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'enter_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'wrong_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'entering_co nference' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '0' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '1' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '2' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '3' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '4' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '5' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '6' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '7' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '8' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '9' to promp t collection. (24406) DEBUG: onLoad (WebConference.cpp:104): Using adaptive playout buffer as playout technique. (24406) DEBUG: onLoad (WebConference.cpp:127): Webconference will strip 5 leadin g characters from direct room access usernames (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/ sems_rsp_sock (24406) DEBUG: onLoad (AnnounceB2B.cpp:66): AnnounceFile = default_en.wav (24406) DEBUG: onLoad (Conference.cpp:258): Using adaptive playout buffer as pla yout technique. (24406) DEBUG: onLoad (Echo.cpp:52): session_timer_f == 0x0000000009500490 (24406) DEBUG: add_env_path (AmUtils.cpp:891): setting PYTHONPATH to: '/usr/loca l/lib/sems/plug-in' (24406) DEBUG: Ivr-Python: Python-Ivr logging started (24406) DEBUG: onLoad (Ivr.cpp:469): ** IVR compile time configuration: (24406) DEBUG: onLoad (Ivr.cpp:470): ** built with PYTHON support. (24406) DEBUG: onLoad (Ivr.cpp:475): ** Text-To-Speech disabled (24406) DEBUG: onLoad (Ivr.cpp:478): ** IVR run time configuration: (24406) DEBUG: onLoad (Ivr.cpp:479): ** script path: '/usr/local/lib /sems/ivr/' (24406) DEBUG: onLoad (Ivr.cpp:495): directory '/usr/local/lib/sems/ivr/' opened (24406) DEBUG: _start (AmThread.cpp:68): Thread 3083512736 is starting. (24406) DEBUG: run (SIPRegistrarClient.cpp:190): SIPRegistrarClient starting... (24406) DEBUG: _start (AmThread.cpp:68): Thread 3084868512 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3083512736 is starting. (24406) DEBUG: run (SIPRegistrarClient.cpp:190): SIPRegistrarClient starting... (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: conf_auth. (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: mailbox. (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: mailbox_query . (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: pin_collect. (24406) ERROR: onLoad (RegistrationAgent.cpp:69): Account for registration not c orrectly configured. (24406) ERROR: onLoad (RegistrationAgent.cpp:70): RegistrationAgent will not reg ister any accounts. (24406) DEBUG: init (StatsUDPServer.cpp:144): udp_port = 5040 (24406) DEBUG: init (StatsUDPServer.cpp:186): socket bound at port 5040 (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: loadEmailTemplates (AnswerMachine.cpp:298): loading /usr/local/et c/sems//default.template ... (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'first_parti cipant' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'join_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'drop_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'enter_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'wrong_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'entering_co nference' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '0' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '1' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '2' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '3' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '4' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '5' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '6' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '7' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '8' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '9' to promp t collection. (24406) DEBUG: onLoad (WebConference.cpp:104): Using adaptive playout buffer as playout technique. (24406) DEBUG: onLoad (WebConference.cpp:127): Webconference will strip 5 leadin g characters from direct room access usernames (24406) DEBUG: load (AmPlugIn.cpp:221): AmPlugIn: Initialized plugins. (24406) DEBUG: main (sems.cpp:377): Starting session container (24406) DEBUG: _start (AmThread.cpp:68): Thread 3078294432 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3078294432 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3076889504 is starting. (24406) DEBUG: start (AmThread.cpp:135): Thread 3076889504 is just created. (24406) DEBUG: main (sems.cpp:380): Starting media processor (24406) DEBUG: init (AmMediaProcessor.cpp:61): Starting 1 MediaProcessorThreads. (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: main (sems.cpp:383): Starting mailer (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: main (sems.cpp:386): Starting RTP receiver (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/ sems_sock (24406) DEBUG: _start (AmThread.cpp:68): Thread 3075836832 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3074784160 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3073731488 is starting. Ser's unix socket: /tmp/ser_sock our unix socket: /tmp/sems_sock reply unix socket: /tmp/sems_rsp_sock plug-in path: /usr/local/lib/sems/plug-in daemon mode: 0 local IP: 192.168.3.208 default application: <not set> (24406) DEBUG: main (sems.cpp:371): Loading plug-ins (24406) DEBUG: AmPlugIn (AmPlugIn.cpp:102): adding built-in codecs... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 0 inserted (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 6 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'telephone-event' inserted with id 96 and order 196 (24406) DEBUG: load (AmPlugIn.cpp:136): AmPlugIn: loading modules in directory ' /usr/local/lib/sems/plug-in': (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/stat s.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'stats' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/anno uncement.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'announcement' load ed. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/wav. so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 1 inserted (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 2 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMU' inserted with id 0 and order 100 (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMA' inserted with id 8 and order 108 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format Wav inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ilbc .so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 4 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'iLBC' inserted with id 97 and order 197 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format iLBC inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/webc onference.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'webconference' loa --------------------------------------------------------------------------------------- And the second part of the log is this: (24406) DEBUG: run (AmServer.cpp:93): revents = 1 (24406) DEBUG: cacheMsg (AmCtrlInterface.cpp:186): recv-ed: <0.3 sip_request.announcement INVITE 200 192.168.3.208 192.168.3.208 5070 sip:200@192.168.3.208:5070 sip:rsf@192.168.3.77 sip:rsf@192.168.3.77 sip:200@192.168.3.208:5070 9787963-3319555348-76751(a)192.168.3.77 100 47668:470495812 <sip:192.168.3.208;lr=on> sip:192.168.3.208;lr=on P-MsgFlags: 0 . > (24406) DEBUG: execute (AmInterfaceHandler.cpp:127): req.method = <INVITE> (24406) DEBUG: execute (AmInterfaceHandler.cpp:128): req.user = <200> (24406) DEBUG: execute (AmInterfaceHandler.cpp:129): req.domain = <192.168.3.208> (24406) DEBUG: execute (AmInterfaceHandler.cpp:130): req.dstip = <192.168.3.208> (24406) DEBUG: execute (AmInterfaceHandler.cpp:131): req.port = <5070> (24406) DEBUG: execute (AmInterfaceHandler.cpp:132): req.r_uri = <sip:200@192.168.3.208:5070> (24406) DEBUG: execute (AmInterfaceHandler.cpp:133): req.from_uri = <sip:rsf@192.168.3.77> (24406) DEBUG: execute (AmInterfaceHandler.cpp:134): req.from = <sip:rsf@192.168.3.77> (24406) DEBUG: execute (AmInterfaceHandler.cpp:135): req.to = <sip:200@192.168.3.208:5070> (24406) DEBUG: execute (AmInterfaceHandler.cpp:136): req.callid = <9787963-3319555348-76751(a)192.168.3.77> (24406) DEBUG: execute (AmInterfaceHandler.cpp:137): req.from_tag = <> (24406) DEBUG: execute (AmInterfaceHandler.cpp:138): req.to_tag = <> (24406) DEBUG: execute (AmInterfaceHandler.cpp:141): cseq_str = <100> (24406) DEBUG: execute (AmInterfaceHandler.cpp:147): cseq = <100>(100) (24406) DEBUG: execute (AmInterfaceHandler.cpp:149): req.key = <47668:470495812> (24406) DEBUG: execute (AmInterfaceHandler.cpp:150): req.route = <<sip:192.168.3.208;lr=on>> (24406) DEBUG: execute (AmInterfaceHandler.cpp:151): req.next_hop = <sip:192.168.3.208;lr=on> (24406) DEBUG: execute (AmInterfaceHandler.cpp:156): hdrs = <P-MsgFlags: 0 > (24406) DEBUG: execute (AmInterfaceHandler.cpp:161): body = <> (24406) ERROR: run (AmServer.cpp:110): AmRequestHandler::execute: mandatory parameter is empty (from|to|callid|from_tag) (24406) DEBUG: run (AmServer.cpp:93): revents = 0 On 1/2/08, Stefan Sayer <stefan.sayer(a)iptego.de> wrote: > Hello, > > > frq ser wrote: > > Hi > > > > I am running the simple configuration in which SER 0.9.7-pre1 serves > > as Proxy (at port 5060) and SER-SEMS is the media server (at port > > 5070). At port 5070, the SER version is 0.9.6-pre1 and SEMS version is > > 0.10.0-rc2. I have placed log commands in the config files just to > > record the process flow. > > > > Using SIPSAK, I am sending an INVITE (for SER-SEMS at :5070) through > > the proxy. My sipsak message is this: > > C:>sipsak -f invite.txt -s sip:200@192.168.3.208:5060 -vvv > > > > The request enters the proxy ser.cfg file and is redirected to the > > SER-SEMS (see the log messages below). The user name (200@...) is used > > by if statement to invoke the announcement application. However, I > > don't see any response from the unix socket (no announcement played) > > nor do I see any error message, and the connection finally times out. > > > > Can anyone please tell me how I can check whether the unix sockets are > > doing their job or not? Why does the process not return any error and > > why does it just time out? > what does the sems log look like if you start sems with -D 3 -E ? does > it run and receive the message? > > thanks > Stefan > > > > > > > Thanks in advance for your help. > > Frq > > > > PS: Here are the (somewhat condensed) versions of SER and SEMS outputs > > (log messages mostly), SIPSAK responses, and the relevant config > > files. I apologise for sending a huge email. > p.s. I would recommend sems 0.10.0 - > ftp://ftp.iptel.org/pub/sems/sems-0.10.0.tar.gz > > -- > Stefan Sayer > VoIP Services > > stefan.sayer(a)iptego.de > www.iptego.de > > iptego GmbH > Am Borsigturm 40 > 13507 Berlin > Germany > > Amtsgericht Charlottenburg, HRB 101010 > Geschaeftsfuehrer: Alexander Hoffmann >

3 4

[OpenSER-Users] Meeting at CCC in Berlin, OSMB in Nuremberg
by Henning Westerholt 03 Jan '08

03 Jan '08

Hello all, i'll do two presentations about OpenSER at the Chaos Communication Congress [1], December 27th-30th, 2007 in Berlin and the Open Source Meets Business congress [2], January 22th–24th, 2008 in Nuremberg. If some people there are interested in a meeting for discussions around OpenSER and other related (or unrelated..) topics, please drop me a private email. Regards, Henning [1] http://events.ccc.de/congress/2007/Main_Page [2] http://www.heise.de/events/2008/open_source_meets_business/en/

3 4

[Serusers] Using SER with Microsoft OCS 2007
by Nolan Garrett 02 Jan '08

02 Jan '08

Hello! I am interested in using SER to convert TCP SIP traffic from OCS to UDP SIP traffic that I can use with an external PSTN gateway provider. Has anyone successfully managed this? Basically, the configuration would look like OCS -> Mediation Server -> SER -> External Provider Would the Mediation Server still need an external (non-NAT'd) IP? It seems like it would, unless I use mediaproxy or RTPproxy, which would add an additional layer of complexity. I guess I am looking for some validation that what I am trying to do will work. My next question, which I can't seem to find an answer to, is how I take the TCP SIP traffic from OCS, convert it to UDP SIP, and also authenticate with that external provider. Any ideas on how to use t_relay_to_udp with authentication? Thank you! Nolan

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Re: [OpenSER-Users] avp_pushto question
by Jeremy George 02 Jan '08

02 Jan '08

Perfect. Thank you. - Jeremy >-----Original Message----- >From: Iñaki Baz Castillo [mailto:ibc@aliax.net] >Sent: Wednesday, January 2, 2008 01:27 PM >To: users(a)lists.openser.org >Subject: Re: [OpenSER-Users] avp_pushto question > >El Miércoles, 2 de Enero de 2008, Jeremy George escribió: >> while($var(idx) < 5) { >> if ($(avp(i:1)[$var(idx)]) != NULL) { >> xlog("L_NOTICE", "avp(i:1)[$var(idx)]=$(avp(i:1)[$var(idx)]) >> avp(i:2)[$var(idx)]=$(avp(i:2)[$var(idx)])\n"); avp_pushto("$ru/username", >> "$(avp(i:2)[$var(idx)])"); >> rewritehostport("10.10.10.10:5060"); >> xlog("L_NOTICE", "$ru\n"); >> } else { >> xlog("L_NOTICE", "avp(i:1)[$var(idx)] == NULL\n"); >> } >> $var(idx) = $var(idx) + 1; >> } >> } else { >> xlog("L_NOTICE", "avp_db_query() returned no records\n"); >> }; > >Forget using avp_pushto, you can do: > > $rU = $(avp(i:2)[$var(idx)]); > >instead of: > > avp_pushto("$ru/username", "$(avp(i:2)[$var(idx)])"); > >-- >Iñaki Baz Castillo > >_______________________________________________ >Users mailing list >Users(a)lists.openser.org >http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] avp_pushto question
by Jeremy George 02 Jan '08

02 Jan '08

Hello all, I have a small problem with avp_db_query() and avp_pushto() that I could use some help with. The manual select statement returns the expected values: mysql> select username,attribute,value from usr_preferences where username='55555'; +----------+-----------+------------+ | username | attribute | value | +----------+-----------+------------+ | 55555 | 1 | 2035552222 | | 55555 | 2 | 2035551111 | +----------+-----------+------------+ 2 rows in set (0.00 sec) The problem I have is that, when I run the configuration segment below only the first instance of avp_pushto in the loop produces the correct value. avp_delete("*"); $var(idx) = 0; if (avp_db_query("SELECT attribute, value FROM usr_preferences WHERE username='$rU' ORDER BY attribute")) { while($var(idx) < 5) { if ($(avp(i:1)[$var(idx)]) != NULL) { xlog("L_NOTICE", "avp(i:1)[$var(idx)]=$(avp(i:1)[$var(idx)]) avp(i:2)[$var(idx)]=$(avp(i:2)[$var(idx)])\n"); avp_pushto("$ru/username", "$(avp(i:2)[$var(idx)])"); rewritehostport("10.10.10.10:5060"); xlog("L_NOTICE", "$ru\n"); } else { xlog("L_NOTICE", "avp(i:1)[$var(idx)] == NULL\n"); } $var(idx) = $var(idx) + 1; } } else { xlog("L_NOTICE", "avp_db_query() returned no records\n"); }; So, the avp_db_query() works exactly as expected. The problem is that the second avp_pushto() doesn't change the value in the ruri username. avp(i:1)[0]=1 avp(i:2)[0]=2035552222 sip:2035552222@10.10.10.10:5060;user=phone avp(i:1)[1]=2 avp(i:2)[1]=2035551111 <-- This is the mismatch sip:2035552222@10.10.10.10:5060;user=phone <-- This is the mismatch avp(i:1)[2] == NULL avp(i:1)[3] == NULL avp(i:1)[4] == NULL As always, thanks for any help. - Jeremy

2 1

[Serusers] Unix Socket Problem
by frq ser 02 Jan '08

02 Jan '08

Hi I am running the simple configuration in which SER 0.9.7-pre1 serves as Proxy (at port 5060) and SER-SEMS is the media server (at port 5070). At port 5070, the SER version is 0.9.6-pre1 and SEMS version is 0.10.0-rc2. I have placed log commands in the config files just to record the process flow. Using SIPSAK, I am sending an INVITE (for SER-SEMS at :5070) through the proxy. My sipsak message is this: C:>sipsak -f invite.txt -s sip:200@192.168.3.208:5060 -vvv The request enters the proxy ser.cfg file and is redirected to the SER-SEMS (see the log messages below). The user name (200@...) is used by if statement to invoke the announcement application. However, I don't see any response from the unix socket (no announcement played) nor do I see any error message, and the connection finally times out. Can anyone please tell me how I can check whether the unix sockets are doing their job or not? Why does the process not return any error and why does it just time out? Thanks in advance for your help. Frq PS: Here are the (somewhat condensed) versions of SER and SEMS outputs (log messages mostly), SIPSAK responses, and the relevant config files. I apologise for sending a huge email. ------------------------------------------------------------------------------------------- SER output [root@xxx ~]# ser Listening on udp: 192.168.3.208 [192.168.3.208]:5060 tcp: 192.168.3.208 [192.168.3.208]:5060 aliases... WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 5058 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(5056) Inside the main route.... 0(5056) just passed the sanity checks... 0(5056) inside method-register... 0(5056) inside uri-myself... 0(5056) Inside route1... 0(5056) Inside the main route.... 0(5056) just passed the sanity checks... 0(5056) inside method-register... 0(5056) inside uri-myself... 0(5056) Inside route1... ------------------------------------------------------------------------------------------- SER-SEMS output [root@xxx ~]# /opt/ser-sems/sbin/ser -f /opt/ser-sems/etc/ser/ser.cfg Listening on udp: 192.168.3.208 [192.168.3.208]:5070 tcp: 192.168.3.208 [192.168.3.208]:5070 aliases .... WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 5061 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(5059) Inside route block.... 0(5059) Inside if-invite... 0(5059) Inside if-uri-200... 0(5059) Inside route block.... ------------------------------------------------------------------------------------------- INVITE message used by SIPSAK (invite.txt ) is this: INVITE sip:200@192.168.3.208:5070 SIPi/2.0 From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 ------------------------------------------------------------------------------------------- SIPSAK responses: fqdnhostname: 192.168.3.77 our Via-Line: Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;a lias New message with Via-Line: INVITE sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 request: INVITE sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 send to: UDP:192.168.3.208:5060 message received received from: UDP:192.168.3.208:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport=2647;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Server: Sip EXpress router (0.9.7-pre1 (i386/linux)) Content-Length: 0 Warning: 392 192.168.3.208:5060 "Noisy feedback tells: pid=5056 req_src_ip=192. 168.3.77 req_src_port=2647 in_uri=sip:200@192.168.3.208:5070 out_uri=sip:200@192 .168.3.208:5070 via_cnt==1" ** reply received after 0.000 ms ** SIP/2.0 100 trying -- your call is important to us provisional received; still waiting for a final response message received received from: UDP:192.168.3.208:5060 SIP/2.0 101 Trying - just wait a minute! Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport=2647;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Server: Sip EXpress router (0.9.6-sems (i386/linux)) Content-Length: 0 Warning: 392 192.168.3.208:5070 "Noisy feedback tells: pid=5059 req_src_ip=192. 168.3.208 req_src_port=5060 in_uri=sip:200@192.168.3.208:5070 out_uri=sip:200@19 2.168.3.208:5070 via_cnt==2" ** reply received after 16.000 ms ** SIP/2.0 101 Trying - just wait a minute! provisional received; still waiting for a final response message received received from: UDP:192.168.3.208:5060 Content-Length set to 0 New message with changed Content-Length: ACK sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 ACK Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 reply with copyed To: ACK sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070;tag=cc619c805c36ff52db55353ff7c0aad3-be50 Contact: sip:rsf@192.168.3.77 CSeq: 100 ACK Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 request: ACK sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070;tag=cc619c805c36ff52db55353ff7c0aad3-be50 Contact: sip:rsf@192.168.3.77 CSeq: 100 ACK Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 send to: UDP:192.168.3.208:5060 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport=2647;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070;tag=cc619c805c36ff52db55353ff7c0aad3-be50 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Server: Sip EXpress router (0.9.6-sems (i386/linux)) Content-Length: 0 Warning: 392 192.168.3.208:5070 "Noisy feedback tells: pid=5060 req_src_ip=192. 168.3.208 req_src_port=5060 in_uri=sip:200@192.168.3.208:5070 out_uri=sip:200@19 2.168.3.208:5070 via_cnt==0" ** reply received 29235.000 ms after first send and -15.000 ms after last send ** SIP/2.0 408 Request Timeout final received ------------------------------------------------------------------------------------------- Ser.cfg (for proxy server at :5060) debug=3 fork=no log_stderror=yes listen=192.168.3.208 #2.13 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route { log(1, "Inside the main route....\n"); # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; log(3, "just passed the sanity checks...\n"); # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); log(3, "inside method-register...\n"); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { log(3, "inside loose route...\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { log(3, "inside uri-myself...\n"); route(1); break; }; . . . route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ log(3, "Inside route1...\n"); if (!t_relay()) { sl_reply_error(); }; } . . ----------------------------------------------------------------------------------------- SER (Stack for SEMS at :5070) config file is this: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) #/* Uncomment these lines to enter debugging mode fork=no log_stderror=yes #*/ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.3.208 port=5070 fifo="/tmp/ser_fifo" unix_sock="/tmp/ser_sock" # ------------------ module loading ---------------------------------- loadmodule "/opt/ser-sems/lib/ser/modules/sl.so" loadmodule "/opt/ser-sems/lib/ser/modules/tm.so" loadmodule "/opt/ser-sems/lib/ser/modules/rr.so" loadmodule "/opt/ser-sems/lib/ser/modules/maxfwd.so" loadmodule "/opt/ser-sems/lib/ser/modules/usrloc.so" loadmodule "/opt/ser-sems/lib/ser/modules/registrar.so" loadmodule "/opt/ser-sems/lib/ser/modules/textops.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("tm", "pass_provisional_replies", 1) # ------------------------- request routing logic ------------------- route{ log(1, "Inside route block....\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # ----------------------------------------------------------------------- # Redirection-related processing # ---------------------------------------------------------------------- if (method == "ACK" || method == "INVITE" || method == "BYE" || method == "CANCEL"){ # switch to stateful mode if (!t_newtran()){ sl_send_reply("500", "Could not create transaction"); break; }; # prevent timeout on the other side t_reply("101", "Trying - just wait a minute!"); if (method == "INVITE"){ log(2, "Inside if-invite...\n"); # Redirect the call to the 'conference' plugin # in case the URI begins with 100 if (uri =~"sip:100*@"){ # assumes that SMES configuration parameter # 'socket_name=' has been set to '/tmp/sems_sock' log(3, "Inside if-uri-100...\n"); if (!t_write_unix("/tmp/sems_sock","conference")){ t_reply("500","Error contacting SEMS 1"); }; break; }; # Redirect the call to the 'announcement' plugin in case # the URI begins with 200 if (uri =~"sip:200*@"){ log(3, "Inside if-uri-200...\n"); if (!t_write_unix("/tmp/sems_sock","announcement")){ t_reply("500","Error contacting SEMS 2"); log(3, "just past 200-unixsocket...\n"); }; break; }; # No service number, redirect to voicemail # Load email address into AVP so that voicemail gets the # callee's email address (see below for avpops modparams) #avp_db_load("$uri", "$email/$email_scheme"); #if(!t_write_unix("/tmp/sems_sock","voicemail")){ # t_reply("500","Error contacting SEMS"); #}; break; } . . } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ----------------------------------------------------------------------------------------- Finally, the relevant parts of sems.conf file are these: # $Id: sems.conf.sample 265 2007-03-07 21:31:54Z sayer $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # . . socket_name=/tmp/sems_sock reply_socket_name=/tmp/sems_rsp_sock ser_socket_name=/tmp/ser_sock listen=192.168.3.208 sip_port=5070 . . .

2 1

[OpenSER-Users] Some issues with Session-Timer module
by Franz Edler 31 Dec '07

31 Dec '07

Hi, I have just made some tests with sst-module (OpenSER v1.3) and I found the following two issues: I case the min-se value of sst-module is higher than the Session-Expires header field of a received INVITE OpenSER rejects the INVITE request with 422 response (that's as expected). But: 1. the Min-SE header field in 422 response (packet 18 in attached trace) does not end with CR/LF so that it is concatenated with the next header field (Server header) syslog prints in this case: ERROR:sst:sst_dialog_terminate_CB: DID 0xb5582d80 failed (canceled). Terminating session. 2. OpenSER in addition continues to forward the already rejected INVITE transaction with "100 Giving a try"(packet 19) and forwards the INVITE to the UAS (packet 20). The next issue I discovered is with the UAC (Xlite): The repeated INVITE (caused by the 422 Response) does not increment the CSeq header field so that it is regarded as same request leading to a "482 Merged request" rejection at the UAS (Eyebeam). I have attached the tcpdump-trace and the according SIP-scenario trace. Any comments appreciated. Regards Franz

1 1

[OpenSER-Users] (no subject)
by VoIP Forums www.Go4Calls.com 30 Dec '07

30 Dec '07

Hi Friends, How we can authorize Clients with their static IP Address in MySQL subscriber table? We don't want to provide them username or PWS and they also don't need to register. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

[OpenSER-Users] dialog module: dialog data remain in DB after "dlg_end_dlg"
by Franz Edler 29 Dec '07

29 Dec '07

Hi, I just found that after forced end of a dialog (MI command "dlg_end_dlg") the dialog data remain in the database. The dialog list in memory (MI command "dlg_list") is cleared after forced end of a dialog as expected. Shouldn't the dialog data in the DB also be deleted? Regards Franz

1 0

Re: [OpenSER-Users] ERROR:avpops:check_avp: value types don't match
by Jayesh Nambiar 29 Dec '07

29 Dec '07

Thanks a lot. Its working now. It had to be like: if(avp_check("$avp(i:103)", "eq/i:0/g")) { do certain things } else { do certain things } w/regards, Jayesh ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Friday, 28 December, 2007 2:54:17 PM Subject: Re: [OpenSER-Users] ERROR:avpops:check_avp: value types don't match Hi Jayesh, try to use the avp_print() debug function to see what avps you have loaded after the avp_db_query() and check the type. Note you need to use debug=6 . http://www.openser.org/docs/modules/1.3.x/avpops.html#AEN611 regards, bogdan Jayesh Nambiar wrote: > Hi All, > I am using the function avp_db_query to query a table and fetch few > values. The query is as follows: > "select subscriptionid, ip, media_status from domestic where > prefix=$rU", "$avp(i:101), $avp(i:102), $avp(i:103)". > > Now the media_status column is a column with datatype as integer which > will contain either 0 or 1. I need to check the value in this column > so I use the following: > > if(avp_check("$avp(i:103)", "eq/0/i")) { > do certain things; > } else { > do certain things; > }; > > Even when the value at media_status column is 0 it does not get into > that condition, it always executes the else condition and I see the > following in the logs: > ERROR:avpops:check_avp: value types don't match > > Can someone please help me to overcome this error. When I keep the > datatype of media_status column as varchar it works fine, but I dont > want to do that as that column is going to conatin only integer values. > > Thanks in advance. > > w/regards, > Jayesh > > > ------------------------------------------------------------------------ > Forgot the famous last words? Access your message archive online. > Click here. > <http://in.rd.yahoo.com/tagline_webmessenger_4/*http://in.messenger.yahoo.co…> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Get the freedom to save as many mails as you wish. To know how, go to http://help.yahoo.com/l/in/yahoo/mail/yahoomail/tools/tools-08.html

1 0

[Serusers] establishment call failure
by dimitris(yahoo) 29 Dec '07

29 Dec '07

hi, i am experiencing on little problem establishishing calls with my SER. First of all , i am running ser under linux opensuse, (ser version: 0.9.6 ) , i also installed rtpproxy and made the needed changes to my ser.cfg. When i "connetct" 2 different users with my SER locally , using softphone, everything looks fine. Gennerally the registration procedure using MySQL works fine(locally or not). When i am connected locally, a remote user can call me with no problems... The problem is the following, and i would appreciate if you could help me... When i try to call a remote user, the call cannot be established from my side. At the other side, the establishment of the call is ok. The message is the following: No ACK recieved... PS: I am only working now, till the level of the establishment of the calls... When i solve this problem , i will continue to the voice and video transmition, where i will need your help for sure... PS2: In order rtpproxy to run on the same computer with SER, i type rtpproxy with no parameters ?? can you help me with that too ??? Thank you in advance I send you also my ser.cfg

2 1

[OpenSER-Users] A Happy New Year
by Bogdan-Andrei Iancu 28 Dec '07

28 Dec '07

Hi everybody, I hope you found under the tree something that made you smile and happy, something that made you think about 2007 as a fulfilled year. I and the Voice System team wish you A Happy New Year, a good party and rest over the years and see you in 2008 ;) Best regards, Bogdan

1 0

[OpenSER-Users] xlog: color PV moved in xlog module ?
by Iñaki Baz Castillo 28 Dec '07

28 Dec '07

Hi, a change in 1.3 is: xlog: color PV moved in xlog module Could somebody explain what does it mean? Thanks a lot. -- Iñaki Baz Castillo

2 2

[OpenSER-Users] ERROR:avpops:check_avp: value types don't match
by Jayesh Nambiar 28 Dec '07

28 Dec '07

Hi All, I am using the function avp_db_query to query a table and fetch few values. The query is as follows: "select subscriptionid, ip, media_status from domestic where prefix=$rU", "$avp(i:101), $avp(i:102), $avp(i:103)". Now the media_status column is a column with datatype as integer which will contain either 0 or 1. I need to check the value in this column so I use the following: if(avp_check("$avp(i:103)", "eq/0/i")) { do certain things; } else { do certain things; }; Even when the value at media_status column is 0 it does not get into that condition, it always executes the else condition and I see the following in the logs: ERROR:avpops:check_avp: value types don't match Can someone please help me to overcome this error. When I keep the datatype of media_status column as varchar it works fine, but I dont want to do that as that column is going to conatin only integer values. Thanks in advance. w/regards, Jayesh Now you can chat without downloading messenger. Go to http://in.messenger.yahoo.com/webmessengerpromo.php

2 1

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