sr-users May 2007

sr-users@lists.kamailio.org

164 participants
296 discussions

[Users] changes in database setup scripts
by Henning Westerholt 01 Jun '07

01 Jun '07

Hello all, i've commited new database setup scripts for mysql and postgresql. Now are the SQL tables generated from a common XML definition, to avoid inconsistencies between the different databases. With the new system is not more necessary to update three different shell scripts for one database table change, hopefully this will lead to fewer bugs. The dbtext integration is still missing, i'll work on this and on further extension of the system in two weeks after the programming course in paris. I've tested the new system with mysql 5.0 and postgresql 8.2, but there are probably still some bugs present. So if you get an database setup script related error from trunk, don't hesitate to complain here or on the devel list. :-) You'll find further informations about this system in the wiki: http://www.openser.org/dokuwiki/doku.php/development:db-schema Best regards, Henning

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[Serusers] wrong password
by ram 01 Jun '07

01 Jun '07

Hi iam runing SER and my client is registering with Asterisk, running call center he keep getting the following error chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"M0601001458000013456" <sip:user@domain.com>;tag=as5bbc2eec' and voice going choppy, when he call from single user the voice go clear when he intiates the calls from number of people voice is very choppy and some time not able to hear all are public IP, not NAT here any suggestions ram

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[Users] Openser 1.2.1 with SERWeb 0.9.6-cannot open write fifo
by Lesley Liu 01 Jun '07

01 Jun '07

Hi Users, I'm setup SERWeb 0.9.6 with Openser 1.2.1. At first, when login with correct username/password, there is a "sorry -- cannot open write fifo" on the my_account.php. Referred from google, I changed the access right of openser_fifo to 666; that cause the my_account.php does not show. Please kindly advise, thanks in advance. Regards, Lesley

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[Serusers] Fwd: SIP 479 Regretfully
by flavio 31 May '07

31 May '07

---------- Forwarded message ---------- From: flavio <flavio.patria(a)gmail.com> Date: 31-mag-2007 18.04 Subject: Re: [Serusers] SIP 479 Regretfully To: olivier.taylor(a)hh174.be But If I try with my BT102 GrandStream (configured as Polycom on my SER) I'm able to start a call to PSTN Number. How is it possible? Have you any suggestions about? Thanks, ps I've also asterisk running on the same machine listening on port 5062. U 2007/05/31 18:00:52.457140 10.28.19.124:5060 -> 10.28.19.202:5060 INVITE sip:0672020949@10.28.19.202 SIP/2.0. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bK5654e41ba6af4166. From: <sip:0660522016@10.28.19.202>;tag=44dcaa39db672de9. To: <sip:0672020949@10.28.19.202>. Contact: <sip:0660522016@10.28.19.124>. Supported: replaces. Proxy-Authorization: Digest username="0660522016", realm="10.28.19.202", algorithm=MD5, uri="sip:0672020949@10.28.19.202", nonce="465f0e8002ccdf57a20d382a49036bd0e5c691d0", response="554b818d1ecfc8455f1bc2e774881508". Call-ID: 19d87a9eb5a4d432(a)10.28.19.124. CSeq: 17072 INVITE. User-Agent: Grandstream BT110 1.0.8.12. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 211. . v=0. o=0660522016 8000 8001 IN IP4 10.28.19.124. s=SIP Call. c=IN IP4 10.28.19.124. t=0 0. m=audio 5004 RTP/AVP 18 8 0. a=sendrecv. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=ptime:20. 2007/5/31, olivier.taylor <olivier.taylor(a)gmail.com>: > hi, > > Regretfully, we were not able to process the URI > > probably a malformed URI, don't you have to remove the leading 0 and add the international code? > > hope it helps, > > Olivier > > > > flavio a écrit : > > Hi to all. > > I've configured my polycom ip500 IPphone to use it with ser. > > If I try a call to users registred to ser all works fine. > > If I try to call PSTN Number through my gateway I've the follow sip message: > > > > > > > > INVITE sip:0672028405@10.28.19.202:5060;user=phone SIP/2.0. > > Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. > > From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. > > To: <sip:0672028405@10.28.19.202;user=phone>. > > CSeq: 2 INVITE. > > Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. > > Contact: <sip:0660522015@10.28.19.143>. > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, > > NOTIFY, PRACK, UPDATE, REFER. > > User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1. > > Supported: 100rel,replace. > > Allow-Events: talk,hold,conference. > > Proxy-Authorization: Digest username="0660522015", > > realm="10.28.19.202", > > nonce="465ef44bbb09c0155dc8555314519a20cac896f8", > > uri="sip:0672028405@10.28.19.202:5060;user=phone", > > response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5. > > Max-Forwards: 70. > > Content-Type: application/sdp. > > Content-Length: 237. > > . > > v=0. > > o=- 1180620549 1180620549 IN IP4 10.28.19.143. > > s=Polycom IP Phone. > > c=IN IP4 10.28.19.143. > > t=0 0. > > m=audio 2234 RTP/AVP 18 8 0 101. > > a=rtpmap:18 G729/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > > > # > > U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060 > > SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL). > > Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. > > From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. > > To: <sip:0672028405@10.28.19.202;user=phone>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7. > > CSeq: 2 INVITE. > > Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. > > Server: Sip EXpress router (0.9.6 (i386/linux)). > > Content-Length: 0. > > Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314 > > req_src_ip=10.28.19.143 req_src_port=5060 > > in_uri=sip:0672028405@10.28.19.202:5060;user=phone > > out_uri=sip:0672028405@10.28.52.105:5060:5060;user=phone via_cnt==1". > > > > > > Have you any suggestion about? > > I use my Polycom with Asterisk and BroadSoft without any problem. > > > > Thanks for your support and great patience :D > > > > Bye, > > > > F. > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ******************************** -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

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[Users] load testing presence server
by Toni Barata 31 May '07

31 May '07

Hi All, A test case with SUBSCRIBE winfo event message have reached values of hundreds of messages per second in a load test scenario. But a test case with SUBSCRIBE presence event message have only reached values of 20 messages per second. Is there any explainable reason for this big difference between the results of both test cases? Thanks, Toni

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Re: [Users] adding users
by Jeremy George 31 May '07

31 May '07

Okay, thanks. - Jeremy >-----Original Message----- >From: Ovidiu Sas [mailto:sip.nslu@gmail.com] >Sent: Thursday, May 31, 2007 10:42 AM >To: 'Jeremy George' >Cc: users(a)openser.org >Subject: Re: [Users] adding users > >Hi Jeremy, > >Have you installed the serweb tables? >If yes, you have to set HAS_SERWEB="yes" in your openserctlrc. Check >the mailing list archive, this issue was already discusses a few >times. > > >Regards, >Ovidiu Sas > >On 5/31/07, Jeremy George <jeremy(a)georgeco.us> wrote: >> >> I upgraded to 1.2.1 and completely re-created the db. Now I can add >> exactly one user to the db. Also, trying to delete that one user from >> openserctl hangs. Thoughts? >> >> Thanks much. >> >> - Jeremy >> >> Breathless-FC6# openserctl add jeremy xxxx murph(a)the.surf >> MySql password for user 'openser@localhost': >> new user 'jeremy' added >> Breathless-FC6# openserctl add helen xxxx murph(a)the.surf >> MySql password for user 'openser@localhost': >> ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 >> ERROR: introducing the new user 'helen' to the database failed >> Breathless-FC6# >> Breathless-FC6# openserctl rm jeremy >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >

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[Users] adding users
by Jeremy George 31 May '07

31 May '07

I upgraded to 1.2.1 and completely re-created the db. Now I can add exactly one user to the db. Also, trying to delete that one user from openserctl hangs. Thoughts? Thanks much. - Jeremy Breathless-FC6# openserctl add jeremy xxxx murph(a)the.surf MySql password for user 'openser@localhost': new user 'jeremy' added Breathless-FC6# openserctl add helen xxxx murph(a)the.surf MySql password for user 'openser@localhost': ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 ERROR: introducing the new user 'helen' to the database failed Breathless-FC6# Breathless-FC6# openserctl rm jeremy

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[Serusers] SIP 479 Regretfully
by flavio 31 May '07

31 May '07

Hi to all. I've configured my polycom ip500 IPphone to use it with ser. If I try a call to users registred to ser all works fine. If I try to call PSTN Number through my gateway I've the follow sip message: INVITE sip:0672028405@10.28.19.202:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. To: <sip:0672028405@10.28.19.202;user=phone>. CSeq: 2 INVITE. Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. Contact: <sip:0660522015@10.28.19.143>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1. Supported: 100rel,replace. Allow-Events: talk,hold,conference. Proxy-Authorization: Digest username="0660522015", realm="10.28.19.202", nonce="465ef44bbb09c0155dc8555314519a20cac896f8", uri="sip:0672028405@10.28.19.202:5060;user=phone", response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 237. . v=0. o=- 1180620549 1180620549 IN IP4 10.28.19.143. s=Polycom IP Phone. c=IN IP4 10.28.19.143. t=0 0. m=audio 2234 RTP/AVP 18 8 0 101. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. # U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060 SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL). Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. To: <sip:0672028405@10.28.19.202;user=phone>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7. CSeq: 2 INVITE. Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314 req_src_ip=10.28.19.143 req_src_port=5060 in_uri=sip:0672028405@10.28.19.202:5060;user=phone out_uri=sip:0672028405@10.28.52.105:5060:5060;user=phone via_cnt==1". Have you any suggestion about? I use my Polycom with Asterisk and BroadSoft without any problem. Thanks for your support and great patience :D Bye, F.

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[Users] openserctl ping broken?
by Papadopoulos Georgios 31 May '07

31 May '07

Hello all, I am having trouble getting openserct ping to work (version 1.2). The server is running but I get [1]+ Done cat <$path | filter_fl 400 I modified openserct at the options_ping() function to do echo $RET and I get [1]+ Done cat <$path | filter_fl 400 Too few or too many arguments I tried playing around with the arguments according to the documentation for tm's function t_uac_dlg, but no luck. Can someone help? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Serusers] SERVER REDIRECTION
by Cyprian Clement 31 May '07

31 May '07

Does any one know what modules(and corresponding functions if possible) that I need to use in order to make the SER do the Redirection by initiating a 302( after the reception of INVITE) that contains the contact info/uri that the caller will need to use to call the forwarded-to client? Can this be achieved without the need to use CPL language? Thanks in advance. Regards, Cyp. --------- ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

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sr-users May 2007