September 2003 - sr-users

Re: [Serusers] rewrite & ACK forwarding problem
by lgfausak＠august.net 30 Apr '04

30 Apr '04

I have also had problems with getting the ACK back. I don't completely understand your configuration, you must allow for packets going both directions, right? Here is my config : route { # check to see if the message has been around too long # probably means that it is looping # if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; # # make sure the length of the message isn't too long! # if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; # # do the loose-routing thing, this is important! # if(loose_route()) { log(1,"doing top loose route"); t_relay(); break; }; # this is where I was dropping the ACKS. # I was simply dropping these, but they must be relayed # because they can be ACKs if(!(uri==myself)) { if(!t_relay()) { sl_reply_error(); break; }; break; }; This gets the ACKs through for me. By the way, I have this configured with Cisco ATAs, version 2.16. ---greg > >I have the same problem and posed it to the group yesterday ([Serusers] >Ignored 200 OK message.) So far the only workaround that I have found is to >use the rules in my gateway to rewrite the dialed digits before sending them >to the PSTN PRI, thus leaving the origianl URI intact for SIP >communications. > >One person told me that this is a bug in the Cisco ATA, but it happens on my >IPDialog phones also. It seems to me that the INVITE is being processed by >the SER dial rules and is rewritten, but the ACK is not. > >Sean >_______________________________________________ > >Sean Robertson > >NETXUSA >p. 800-289-6389 >f. 864-233-4344 "Ask me about Voice over IP." >http://www.netxusa.com/ > >----- Original Message ----- >From: "Alexander Mayrhofer" <axelm(a)nic.at> >To: <serusers(a)lists.iptel.org> >Sent: Friday, June 27, 2003 12:15 PM >Subject: [Serusers] rewrite & ACK forwarding problem > > >> >> Hi, >> >> we're running SER together with a PSTN Gateway. Before a call get's >> forwarded to the gateway, we are rewriting the request URI to make >> rewriting on the GW as simple as possible: >> >> route { >> ... >> strip(3); # +43xxx -> xxx >> prefix("0"); # xxx -> 0xxx >> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway >> route(1); >> break; >> ... >> >> SIP call flow looks like (record route enabled): >> >> (1) phone -> SER >> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0 >> >> (2) SER -> phone >> SIP/2.0 100 trying -- your call is important to us >> >> (3) SER -> GW >> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> (4) GW -> SER >> SIP/2.0 100 Trying >> >> (5) GW -> SER >> SIP/2.0 183 Session Progress >> >> (6) SER -> phone >> SIP/2.0 183 Session Progress >> >> (7) GW -> SER >> SIP/2.0 180 Ringing >> >> (8) SER -> phone >> SIP/2.0 180 Ringing >> >> (9) GW -> SER >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060> >> >> (10) SER -> phone >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060> >> >> [ call established, we can talk, but ... ] >> >> (11) phone -> SER >> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> --> Here starts the problem. That ACK (11) never gets forwarded to the >> Gateway, so after a few seconds, the GW starts over at (9). Those three >> packets (9-11) repeat a few times until GW runs into a timeout and drops >> the call. >> >> I have the impression that SER can't match the packet to the previous >> requests because of the rewritten URI. Is that correct? >> >> The only output at debug level 3 is: >> >> Warning: sl_send_reply: I won't send a reply for ACK!! >> >> Is that a routing goof somewhere in our scripts or is that a more >> generic problem? Is the problem that the warning indicates somehow >> related to the fact that the ACK is not being forwarded? >> >> Help appreciated. >> >> cheers >> >> axelm >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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RE: [Serusers] Latest STABLE build without using CVS
by Jiri Kuthan 20 Jan '04

20 Jan '04

At 07:39 PM 8/14/2003, Chad Brown wrote: >Perfect, > >Let me ask 2 quick follow-on questions... > >1. Can I go back to http://www.iptel.org/ser/tarball/ser_8_11_stable.tgz >to get the latest patched and STABLE builds? That's the latest, most stable source, you need to compile it myself. There will be a new complete distribution by end of this month -- we are now waiting to run SER through the upcoming SIPIT to release it. >2. What location / version of the modules should I use when running >builds for this location? (Serweb, mysql, jabber, etc) All SER modules are included there. uptodate SERWEB is now in the http://www.iptel.org/ser/tarball/ directory too. -jiri

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[Serusers] Radius accounting - radius_log_flag?
by Steven R. Bunin 16 Jan '04

16 Jan '04

Hi, I have radius authentication working thanks to Jan's help but I am still not receiving any accounting messages to my radius server. I found a mention of setting radius_log_flag but ser 0.9.11 tells me it isn't found in the "acc" module. How can I set invite and goodbye messages (or any others for that matter) to send information to my radius server for accounting purposes. I have read at leat 90% of the old messages but have yet to find an answer. Thanks in advance, Steve

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[Serusers] Xten and SER problem
by Jesus Rodriguez 17 Nov '03

17 Nov '03

Hello, I'm trying to register an Xten (1016 build) to SER and i always get a "401" response. Registering an ATA186 to the same SER with the same user and pass works with no problem (logs below Xten). We have tried SIPPS and registers ok too. Xten send the REGISTER, SER answers with a WWW-Authenticate challenge but Xten doesn't send anything, just sends a new REGISTER after few seconds: (c)2003 Xten Networks Inc. All rights reserved. Private build: 1016 SIP: 192.168.1.203:5060 RTP: 192.168.1.203:8000 NAT: 192.168.1.203 PXY#0: 192.168.1.2:5060 SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562200000000e06a238bd1858769f2e46b11d8d837f1",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" The same action works ok with ATA186: [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c579a00000000aa6fd2432e429f54e3 7e75981218834b",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13590 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" [0]Reg Resp 401; Unauthorized [0]REGISTER Retry 0 [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Authorization: Digest username="4070",realm="voztelecom.net",nonce="3f1c579a0000 0000aa6fd2432e429f54e37e75981218834b",uri="sip:192.168.1.2",response="fe07b8be83 0305cca7a28bf35c112270" Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;q=0.00;expires=6 0 Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" Thanks in advance. Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

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[Serusers] New release of seradmin v .03e
by Xten India 02 Nov '03

02 Nov '03

Hai, We Xten released a newer version (seradmin v .03e) of SERAdmin. Xten is building this for the ser community and we would very much like your feedback so that we can build you a better product The Source code is available at http://developer.berlios.de/projects/seradmin SERAdmin is a GUI interface between SIP Express Router (SER) and a SER administrator.SERAdmin has an intuitive look and feel. SERAdmin provides control over many SER tasks such as: Start, Stop, Pause, Re-start, Monitor SER, Add User, Change Password and EmailId Delete User, Add Alias , Edit Alias etc. With this we can also use FIFO commands,Access controls,User Location. The objective of the SERAdminv03e is to support the ser-0.8.11 version also. This latest SER Administration Application handles both ser-0.8.10 and ser-0.8.11. What is new?. The following commands available in this release ------------------------------------------------ Ping(uri) - Pinging a URI Cisco_restart(uri) - Restart a Cisco Phone arg - Arguments of SER pwd - Present Working Directory t_uac_dlg - Initiate a Transaction you can download the components from http://developer.berlios.de/projects/seradmin For any feedback & help pls. contact xten_india(a)yahoo.com Regards, Team, Xten_india --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software

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[Serusers] SER & rtpproxy
by Mathieu MALAISE 09 Oct '03

09 Oct '03

Hello All ! I'm new to SER, and I'd like to install rtpproxy. I've found the program to compile (https://demo.portaone.com/~sobomax/PortaSIP/rtpproxy/) but I don't know what to do with it once it is compiled, and I don't know what to change in SER to integrate the proxy. Thank you all for your help ! Mathieu

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Re: [Serusers] Re: Serusers Digest, Vol 5, Issue 63
by jaime.gil＠orange.co.uk 07 Oct '03

07 Oct '03

Steven, I remember having the same problem. In my case, the problem seemed to be in the loose_route processing. I don't know what it has to do with the accounting module, but commenting out this part made the trick in my case. Jaime "Steven R. Bunin" <steve(a)solaas.com> on 26/09/2003 17:11:41 To: serusers(a)lists.iptel.org cc: (bcc: Jaime GIL/EN/HTLUK) Subject: [Serusers] Re: Serusers Digest, Vol 5, Issue 63 Hi all, I have successfully gotten radius authentication working and I started getting Radius Start records for accounting but I am not sure what I am doing wrong in regards to getting radius Stop records. Below is the area I believe has the most affect on Radius Acccounting from my Log File. Any suggestions would be appreciated and if it would help to see the full Config file I will send it as well. record_route(); if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; if (method =="INVITE") { log(1,"INVITE\n"); setflag(1); }; if (method=="MESSAGE") { log(1,"MESSAGE\n"); setflag(1); }; if (method=="BYE"){ log (1, "BYE or CANCEL\n"); setflag(1); }; if (method=="CANCEL"){ log (1, "BYE or CANCEL\n"); setflag(1); }; Thanks in advance, Steve _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

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[Serusers] 400 fifo_uac: next hop address expected
by jaime.gil＠orange.co.uk 03 Oct '03

03 Oct '03

Hi, When trying to send a SIP message through the fifo server, it comes back with an error: "400 fifo_uac: next hop address expected" What could be causing this and how could we monitor what gets in and out of the fifo? Thanks! Jaime ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

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RE: [Serusers] Implementing Voicemail
by Steve Dolloff 03 Oct '03

03 Oct '03

>> I did place this portion inside the myself check >>and it still tries to transfer to vm after the time expires. >I'm puzzled -- did not you want to transfer to vm after the time expires? I will try to make this clearer. I am behind an ATA with a SIP proxy of 209.242.10.153. If I call someone else registered on my domain and they are not available, I want to go to voice mail. If I call 1-800-555-1212 from my phone, I do not want my sip proxy to reroute the call to voicemail after 10 seconds if no one answers(or ever for that matter). Right now if I dial 18005551212 from my handset, I see the destination as sip:18005551212@209.242.10.153 on the server which matches to myself and ser tries to send it to voicemail. Someone calling into the network is not a problem. They will never hit our server unless the destination is local. >>This is the part that I really need help with! When the call timer >>fails, the call goes to the route[1]. How do I get it into voice mail >>from that point? >See bellow, I think that should work. This is what I had originally, and I get the following syslog. Sep 10 16:36:36 voip2 ser: parse error (127,37-38): Command cannot be used in the block Sep 10 16:36:36 voip2 ser: ERROR: bad config file (1 errors) Sep 10 16:36:36 voip2 ser: ser startup failed Is says that vm is not valid in the block. According the admin guide, only certain commands can be used within a failure block. I assume that is the problem here. If not, please let me know as this is exactly what I want to do. >THE SAME STUFF LIKE ABOVE, YOU DON'T WANT TO t_relay ANYTHING >if(!vm("/tmp/am_fifo","voicemail")){ > t_reply("500", "SEMS >error"); > }; > break;

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[Serusers] Directory Services
by Ng, Soo Sim 03 Oct '03

03 Oct '03

I am looking for the following feature A list of Directory Services is displayed on the serweb, (for example, Hotel List, Emergency List, Airlines List ect). A registered user will subscribe to the intended list (Eg a hotel will subsribe to the hotel list). A normal user will be able to view the selected list. The seleted list show only online members at that time. Off-line members will only be shown when they are login in. Can this feature be done? Thanks

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