sr-users December 2007

sr-users@lists.kamailio.org

130 participants
234 discussions

by hammou moujane

Hello everybody, I am new in this mailing list and new user of OpenSER. I would like to know where can I find the doc which can help me to make OpenSER as a proxy server like P-CSCF of IMS core. what to be modified in openser.conf configuration file. Bests Regards. -- Hammou MOUJANE Tel: +212 62 46 33 89 INPT

16 years, 4 months

[OpenSER-Users] Parallel forking to loopback problem - for ringing multiple numbers at same time, simulring

by Dave Singer

by the way, I'm trying this using openser 1.3.0-pre1-notls from latest svn of trunk as of 12-4-07 I'm trying to do findme logic.. Call comes in, lookup settings for where it is supposed to go. >From the settings I find that the user wants it to ring a friends home phone, a cell phone, and the home ata associated with account. To keep voicemail handled by the home account and not by one of the places that they are forwarding to they want voicemail to be triggered after 15 seconds. The friends home phone is actually one of our accounts and has two ata's associated with it that register as the same sip user and they have it also ring a cell phone and the account sends the call to voicemail after 18 seconds. Psudo Call flow/Logic of what I have so far. Call comes in to openser for 555/123-4567 pull from usr_preferences settings for the call destination number. so I get a stack of avps $avp(s:brahch_uri) from usr_preferences like this: sip:5551239876@127.0.0.1 sip:localuser3@127.0.0.1 sip:5553216548@127.0.0.1 I do diversion header then branch for them with: avp_pushto("$br", "$avp(s:branch_uri)" I drop the local branch. The branched calls come in as expected and verify that it is a valid intentional loopback by testing: if ( ruri =~ "127.0.0.1" && dst_ip == 127.0.0.1 ) { < mark as valid loopback > } Like the beginning of the original call I lookup where they should go and if the user in the diversion header is allowed to send them there (PSTN) In this case one call would be sent out through the PSTN gateways with full failover logic, another would call the device associated with the account by doing a lookup("location") and the third, (the one to the friend that our customer) would add another diversion header and start another parallel fork with loopback for the cell phone and two ata's When I test with a simple forward to a single PSTN number, in the loopbacked session, I rewrite the ruri accordingly with real dest user and real ip address. the ruri shows what I want in the logs just before t_relay and exit and in branch_route[1] after last log entry from branch_route, I have in the log: ERROR:core:udp_send: sendto(sock,0xb619d954,1376,0,0xb619ba6c,16): Invalid argument(22) CRITICAL:core:udp_send: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net ERROR:tm:msg_send: udp_send failed ERROR:tm:t_forward_nonack: sending request failed Then I have log entrys from onreply_route[1]: Reply - S=100 M=INVITE RURI=<null> src_ip = 127.0.0.1 Reply - S=477 M=INVITE RURI=<null> src_ip = 127.0.0.1 not the real world IP address of the other server I set in the ruri and there is nothing in the logs on the intended server or the local server indicating routing logic for an invite failed I'm woundering if the TM module, while processing the invite in the loopback, is matching up the transaction with the session that originated the loopback and using the settings and status from it. Any Ideas, Thoughts, HELP!!? Or is OpenSER just not ready for this yet? Thanks Dave

16 years, 4 months

[OpenSER-Users] uac_restore_from problem

by Zoltán Kovács

Dear Colleagues, first of all, thank you for the great software which is a very nice and convenient way to build up a stable VoIP telephone system! We already have a working system based on OpenSER and Asterisk. After many experiments I finally raise this question to you because I cannot figure out the solution. The problem described on http://www.openser.org/pipermail/users/2006-January/002747.html seems to be very similar to mine. I am using openser-1.1.0-9etch1. I tried to replace the FROM variable as follows: loadmodule "uac.so" modparam("uac","from_restore_mode","auto") ... route { ... avp_write("$fU", "$avp(s:display)"); avp_write("$fU", "$avp(s:dispuri)"); avp_subst("$avp(s:display)", "/^36(.*)/06\1/"); avp_subst("$avp(s:dispuri)", "/^36(.*)/sip:06\1@datanet.hu/"); uac_replace_from("$avp(s:display)", "$avp(s:dispuri)"); ... } Unfortunately nothing seems to happen: no change is made on several VoIP telephones, including X-Lite as well. loose_route() and record_route() were also used in my configuration. Could you please help me what I am missing here? If needed, I can also send a full config file. Thank you in advance, Zoltan Kovacs sysadmin

16 years, 4 months

[Serusers] tm module problem

by Szasz Szabolcs

Hi I have a problem with tm module, I can't set the module parameter pass_provisional_replies . The module tm is loaded succesfully, I insert in to the configuration this line: modparam("tm", "pass_provisional_replies", 1) When I start ser I got this error message : 0(9618) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(9618) parse error (33,19-20): Can't set module parameter ser version used 0.9.6 , I couldn't found any solution on the google. I have to recompile the tm module with some modifications ? or I have other mistake in the configuration? Please help! Thank you Szasz Szabolcs

16 years, 4 months

[OpenSER-Users] Problem with setting the Acct-Authentic Radius Attribute

by Dan-Cristian Bogos

Folks, I am trying to make use of Acct-Authentic to identify authenticated sessions inside accounting packets. All ok, no problem with dictionaries, but when it comes to receiving the value in freeradius server, I get all sort of strange ones (integers). I am already using Acct-Authentic properly working with other software on the same radius server. Here is my config: modparam("acc", "radius_extra", " Called-Station-Id=$tu; Calling-Station-Id=$fu; Canonical-URI=$tu; User-Name=$au; Sip-User-Realm=$ar; Source-IP=$si; Source-Port=$sp; Acct-Authentic=$avp(s:acct_authentic); From-Header=$hdr(from); User-Agent=$hdr(user-agent); Contact=$hdr(contact); Event=$hdr(event)") I have tried different values for the $avp(s:acct_authentic): 1, "1", RADIUS, "RADIUS", etc. and each time I got a corresponding encoded. EG: If I will set the AVP through SIP-AVP or directly inside the routing script, for acct_authentic=1 I will receive in radius Acct-Authentic = 49. Are there any developers around who have implemented radius support in ser/openser and could help me troubleshooting this issue? My openser version: openser -V version: openser 1.2.2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:3003M @(#) $Id: main.c 2304 2007-05-25 16:36:07Z bogdan_iancu $ Ta, DanB

16 years, 4 months

Re: [OpenSER-Users] OpenSER Sending INVITE twice

by Ovidiu Sas

There is no per upstream server config. Best thing to do is upgrade to openser 1.2. Starting with this release the timer granularity has been increased. If you have a high RTT, you can increase the T1 timer: http://www.openser.org/docs/modules/1.2.x/tm#AEN168 Hope this helps, Ovidiu Sas On Dec 4, 2007 12:44 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Oh, and RTT as OpenSER determines it to where? Is it one RTT for all > outgoing traffic or is it a different RTT for every upstream server? > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: Users(a)lists.openser.org > Sent: Monday, November 26, 2007 5:47:23 PM > Subject: Re: [OpenSER-Users] OpenSER Sending INVITE twice > > This is the way that is supposed to work. If nothing is received from > upstream, the request is resent. > From the rfc3261: > > 17.1.1.1 Overview of INVITE Transaction > > The INVITE transaction consists of a three-way handshake. The client > transaction sends an INVITE, the server transaction sends responses, > and the client transaction sends an ACK. For unreliable transports > (such as UDP), the client transaction retransmits requests at an > interval that starts at T1 seconds and doubles after every > retransmission. T1 is an estimate of the round-trip time (RTT), and > it defaults to 500 ms. > > > Regards, > Ovidiu Sas > > On Nov 26, 2007 3:29 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > > > I have an OpenSER 1.1 install here, and for some reason, OpenSER is > sending > > the INVITE message twice to the upstream host. > > > > The time difference between them is about 1/5 of a second. Nothing is > > received between the first and second INVITE's. > > > > Why is OpenSER doing this? > > > > I have xlog() statements everywhere in openser.cfg, and OpenSER is only > > logging ONE outgoing INVITE message, eventhough it's sending two. It is > > logging the multiple TRYING messages that come back however. > > > > OpenSER is not calling failure_route because nothing is logged in there. > > What could be going wrong? Why is it doing this? > > > > Doug. > > > > > > > > ________________________________ > > Get easy, one-click access to your favorites. Make Yahoo! your homepage. > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now.

16 years, 4 months

[OpenSER-Users] invalid cseq for aor

by Gerson Matiolli

I used the SER for 2 years and not had this problem. I was using OPENSER for 20 days without problems. How can I fix this in Openser ? Thanks [OpenSER-Users] invalid cseq for aor Bogdan-Andrei Iancu bogdan at voice-system.ro Tue Dec 4 09:09:53 UTC 2007 * Previous message: [OpenSER-Users] invalid cseq for aor * Next message: [OpenSER-Users] perl * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi Gerson, The error is generated when a re-REGISTER (same contact and callid) is received but with invalid CSEQ number (equal or less than the previous). Regards, Bogdan Gerson Matiolli wrote: > Hi, All > > I am with the OPENSER running some time without problems. > > Yesterday, he went not to complete any call, and the system had the following message log: > > > ERROR:registry:update_contacts: invalid cseq for aor > ERROR:registry:update_contacts: invalid cseq for aor > ERROR:registry:update_contacts: invalid cseq for aor > > for all numbers > > Thanks >

16 years, 4 months

[OpenSER-Users] Remove header from Cancel

by Jarkko Nevala

Hello. Is it possible to remove Route: -header from Cancel requests. I have an user agent that inserts Route: -header in Cancel requests. I've already tried to get rid of it with remove_hf("Route") function, but it did not have any effect. From what I understand is, that this header is quite unnecessary in a stateful proxy, because the Cancel messages are handled hop-by-hop basis. It's just that it causes problems at the next hop proxy. I didn't even know about this problem with the UA before I started using the development version of OpenSER, the 1.2.2 version removed the Route: -header automatically before relaying Cancel forward. Thank you JN

16 years, 4 months

[OpenSER-Users] how does fix_nated_register() work?

by Daniel Grotti

Hi all, I have a user behind NAT with address : sip:userA@192.168.10.210:42276. When user send REGISTER, the proxy checks if user is behind NAT. If it is, proxy makes: fix_nated_register(); force_rport(); setbflag(6). When I check what kind of address proxy has saved on LOCATION database, I see at Contact column: sip:userA@IP_NAT:42276. So proxy mantains the same Port of the user and doesn't save the NAT's IP:PORT pair. Why? In this way, in fact, proxy is not able to keep connection alive. Regards, daniel

16 years, 4 months

[OpenSER-Users] Module Programming

by Daniel Corbe

Hello, Is there a better module programming guide than the one found on the docuwiki at http://www.openser.org/dokuwiki/doku.php/development:write-module ? It really doesn't explain much; for example the fact that functions are (or used to be) called with a paramater struct sip_msg *. The original SER 0.8.x and 0.9.x core had an extremely comprehensive guide to writing modules; however, I'm reluctant to use those texts as a guide for fear that the material is dated. Regards, Daniel

16 years, 4 months

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sr-users December 2007