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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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[Users] openser: Call workflow
by Helmut Kuper 31 May '07

31 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is there a way in openser to get it work like this: 1. SIP-Request comes in 2. openser detects a special service for dialed number 3. openser sets that original sip session on hold and forwards the request to a remote service node 4. remote service node does it job and sends back a result message via network to openser 5. openser checks that result, get the session from hold and continues it's work on the session In the end I 'm looking for a way to let openser handles a sip session like a workflow, where I can inject some value added services by REMOTE service nodes (not by modules or plugins) without forwarding the complete session to a remote service node. I hope u understand what I mean. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGXrWO4tZeNddg3dwRAueLAJ48xkexHAz3GCkzKeNuRewV1hsb3ACeIr7p LST+x5cMjAuIvbzZl61PPyk= =Ita4 -----END PGP SIGNATURE-----

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[Serusers] SER Debug
by Cyprian Clement 31 May '07

31 May '07

Hi, Does any one know of a useful way of getting(turning on) SER debug report. I am trying to locate where my call is failing with the SER and currently use the "ngrep" but it does not seem to reveal much useful internal report. most of it looks external similar to ethereal. Will appreciate any share of info in this regard. Cyp. -------- ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

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[Serusers] register my asterisk on ser
by flavio 31 May '07

31 May '07

Hi, I've asterisk and ser runnig onver same machine.Is it wrong? I'm trying to register asterisk on ser as UAC, but I've the follow log error message: ERROR: forward_reply: no 2nd via found in reply sip log with ngrep are soon reported: U 2007/05/31 11:52:48.261775 10.28.19.202:5060 -> 10.28.19.202:5060 REGISTER sip:10.28.19.202 SIP/2.0. Via: SIP/2.0/UDP 10.28.19.202:5060;branch=z9hG4bK7bd3439f. From: <sip:0192091221@10.28.19.202>;tag=as7b047b0b. To: <sip:0192091221@10.28.19.202>. Call-ID: 41a53c8a015e385f4b6d951f6a0e07a5(a)127.0.0.1. CSeq: 102 REGISTER. User-Agent: Asterisk PBX. Max-Forwards: 70. Expires: 300. Contact: <sip:2000@10.28.19.202>. Event: registration. Content-Length: 0. . # U 2007/05/31 11:52:48.262493 10.28.19.202:5060 -> 10.28.19.202:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.28.19.202:5060;branch=z9hG4bK7bd3439f. From: <sip:0192091221@10.28.19.202>;tag=as7b047b0b. To: <sip:0192091221@10.28.19.202>. Call-ID: 41a53c8a015e385f4b6d951f6a0e07a5(a)127.0.0.1. CSeq: 102 REGISTER. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=7153 req_src_ip=10.28.19.202 req_src_port=5060 in_uri=sip:10.28.19.202 out_uri=sip:10.28.19.202 via_cnt==1". . # U 2007/05/31 11:52:48.263017 10.28.19.202:5060 -> 10.28.19.202:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.28.19.202:5060;branch=z9hG4bK7bd3439f. From: <sip:0192091221@10.28.19.202>;tag=as7b047b0b. To: <sip:0192091221@10.28.19.202>;tag=979d95a734c13f6db8b9e3a72b9f44a0.020d. Call-ID: 41a53c8a015e385f4b6d951f6a0e07a5(a)127.0.0.1. CSeq: 102 REGISTER. WWW-Authenticate: Digest realm="10.28.19.202", nonce="465eb83c86f495e14144d970e6bbabb7952b6579". Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=7153 req_src_ip=10.28.19.202 req_src_port=5060 in_uri=sip:10.28.19.202 out_uri=sip:10.28.19.202 via_cnt==1". Have you any suggestions? Thanks, F. -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

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