Low, Adam
2003-Jul-30 02:37 UTC
[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:4842@213.160.252.2;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: "611012210" <sip:611012210@213.160.252.50> To: <sip:4842@213.160.252.2;user=phone;phone-context=unknown> Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: 635D27D4-CB1D0233-0-8E9DB84@213.160.252.50 Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: <sip:611012210@213.160.252.50:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
yves.schaaf@restena.lu
2003-Jul-30 05:09 UTC
[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Hi, I am using the latest cvs release of asterisk, and the behaviour is in fact the same, outbound calls work fine, but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked" by asterisk, and never reach the phone. The setup is the same : 7960 <------> asterisk <------> C2651<-----> PSTN Yves |---------+-------------------------------------> | | "Low, Adam" | | | <ALow@Prioritytelecom.com>| | | Sent by: | | | asterisk-users-admin@lists| | | .digium.com | | | | | | | | | 30/07/2003 11:37 | | | Please respond to | | | asterisk-users | | | | |---------+-------------------------------------> >-----------------------------------------------------------------------------------------------------------------------| | | | To: "'asterisk-users@lists.digium.com'" <asterisk-users@lists.digium.com> | | cc: | | Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 | >-----------------------------------------------------------------------------------------------------------------------| All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:4842@213.160.252.2;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: "611012210" <sip:611012210@213.160.252.50> To: <sip:4842@213.160.252.2;user=phone;phone-context=unknown> Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: 635D27D4-CB1D0233-0-8E9DB84@213.160.252.50 Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: <sip:611012210@213.160.252.50:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Low, Adam
2003-Jul-30 05:31 UTC
[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Brenton, Yves, ... I've located the cause of the problem in chan_sip.c but am still trying to find the exact cause being completely new to the asterisk code. It seems that there was an added function in 1.135 called 'find_user' that is supposed to lookup the users incoming call limit but the routine is unable to find a matching user for my AS5300 which I suspect is because it does not REGISTER with the server prior to attempting to send calls. I'm going to continue debugging a little later and see if I can narrow it down more ... Adam -----Original Message----- From: yves.schaaf@restena.lu To: asterisk-users@lists.digium.com Sent: 30/07/03 14:09 Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 Hi, I am using the latest cvs release of asterisk, and the behaviour is in fact the same, outbound calls work fine, but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked" by asterisk, and never reach the phone. The setup is the same : 7960 <------> asterisk <------> C2651<-----> PSTN Yves |---------+-------------------------------------> | | "Low, Adam" | | | <ALow@Prioritytelecom.com>| | | Sent by: | | | asterisk-users-admin@lists| | | .digium.com | | | | | | | | | 30/07/2003 11:37 | | | Please respond to | | | asterisk-users | | | | |---------+------------------------------------->>-----------------------------------------------------------------------------------------------------------------------| | | | To: "'asterisk-users@lists.digium.com'" <asterisk-users@lists.digium.com> | | cc: | | Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 |>-----------------------------------------------------------------------------------------------------------------------| All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:4842@213.160.252.2;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: "611012210" <sip:611012210@213.160.252.50> To: <sip:4842@213.160.252.2;user=phone;phone-context=unknown> Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: 635D27D4-CB1D0233-0-8E9DB84@213.160.252.50 Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: <sip:611012210@213.160.252.50:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
Low, Adam
2003-Jul-30 06:45 UTC
[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Well found Patrick, that did the trick for me as well ! I had been trying to debug 1.135 where this portion of code wasn't added yet ... thats a lesson learnt ... -----Original Message----- From: Patrick To: 'asterisk-users@lists.digium.com ' Sent: 30/07/03 15:04 Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 It is in the find_user() routine. If it is not an extension on the PBX, it should return a zero if ( isfound ) { ast_log(LOG_DEBUG, "%s is not a local user\n", name); ast_pthread_mutex_unlock(&userl.lock); return 1; <--- this is the problem - change it to a 0. } It isn't an error, so it should just return. Change that and the function will work properly. I tested it using an AS5350 and successly made an inbound call. Patrick On Wed, 30 Jul 2003, Low, Adam wrote:> Brenton, Yves, ... > > I've located the cause of the problem in chan_sip.c but am stilltrying to find the exact cause being completely new to the asterisk code. It seems that there was an added function in 1.135 called 'find_user' that is supposed to lookup the users incoming call limit but the routine is unable to find a matching user for my AS5300 which I suspect is because it does not REGISTER with the server prior to attempting to send calls.> > I'm going to continue debugging a little later and see if I can narrowit down more ...> > Adam > > -----Original Message----- > From: yves.schaaf@restena.lu > To: asterisk-users@lists.digium.com > Sent: 30/07/03 14:09 > Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs1.134> > > Hi, > > I am using the latest cvs release of asterisk, and the behaviour is in > fact > the same, > > outbound calls work fine, > but for inbound calls (from C2651 over PSTN) , SIP messages get > "blocked" > by asterisk, and never reach the phone. > > The setup is the same : 7960 <------> asterisk <------> C2651<-----> > PSTN > > Yves > > > |---------+-------------------------------------> > | | "Low, Adam" | > | | <ALow@Prioritytelecom.com>| > | | Sent by: | > | | asterisk-users-admin@lists| > | | .digium.com | > | | | > | | | > | | 30/07/2003 11:37 | > | | Please respond to | > | | asterisk-users | > | | | > |---------+-------------------------------------> > > >----------------------------------------------------------------------- > ------------------------------------------------| > | > | > | To: "'asterisk-users@lists.digium.com'" > <asterisk-users@lists.digium.com> | > | cc: > | > | Subject: [Asterisk-Users] chan_sip.c problems problems from > cvs 1.134 | > > >----------------------------------------------------------------------- > ------------------------------------------------| > > > > > All, > > I've found problems in my setup with the latest couple of revisions > (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 > asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, > everything > is in the same VLAN and only running SIP. > > Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 > > But inbound calls fail, I see the initial INVITE from the AS5300 which > is > received by asterisk but not responded to and then the AS5300 sends > another > few INVITE's which are received but ignored assumable as they were > duplicates for the first. > > Unfortunately since I've been trying the different cvs revisions of > chan_sip.c I've got susbequent problems with the server crashing after > the > first INVITE from the AS5300 using anything greater than cvs 1.134 > > I suspect this is something to do with the per-user limits added incvs> 1.135 but I am curious to see if anyone has any problems with thelatest> cvs elease of asterisk with SIP ? > > Adam > > Sip read: > INVITE sip:4842@213.160.252.2;user=phone;phone-context=unknown SIP/2.0 > Via: SIP/2.0/UDP 213.160.252.50:53893 > From: "611012210" <sip:611012210@213.160.252.50> > To: <sip:4842@213.160.252.2;user=phone;phone-context=unknown> > Date: Wed, 30 Jul 2003 09:26:11 GMT > Call-ID: 635D27D4-CB1D0233-0-8E9DB84@213.160.252.50 > Cisco-Guid: 1667049428-3407675953-0-149543808 > User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled > CSeq: 101 INVITE > Max-Forwards: 6 > Timestamp: 1059557171 > Contact: <sip:611012210@213.160.252.50:5060;user=phone> > Expires: 180 > Content-Type: application/sdp > Content-Length: 149 > > v=0 > o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 > s=SIP Call > c=IN IP4 213.160.252.50 > t=0 0 > m=audio 20032 RTP/AVP 8 0 65535 18 > > 15 headers, 6 lines > Using latest request as basis request > Sending to 213.160.252.50 : 53893 (non-NAT) > Found audio format 8 > Found audio format 0 > Found audio format 65535 > Found audio format 18 > Capabilities: us - 524302, them - 268/0, combined - 12 > Non-codec capabilities: us - 1, them - 0, combined - 0 > AM00CM01*CLI> > Disconnected from Asterisk server > > > ********* DISCLAIMER ********* > > This message and any attachment are confidential and may be privileged > or > otherwise protected from disclosure and may include proprietary > information. If you are not the intended recipient, please telephoneor> email the sender and delete this message and any attachment from your > system. If you are not the intended recipient you must not copy this > message or attachment or disclose the contents to any other person > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ********* DISCLAIMER ********* > > This message and any attachment are confidential and may be privilegedor otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
James Sizemore
2003-Jul-30 08:03 UTC
[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
I also had the same problem with sip, I also moved back a couple of weeks in cvs. I also use a AS5300 Cisco in my call chain. I got a bunch of "Ignoring this request" in debug. I have not had time to trace the call path on this problem yet. Low, Adam wrote:>All, > >I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. > >Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 > >But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. > >Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 > >I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? > >Adam > >Sip read: >INVITE sip:4842@213.160.252.2;user=phone;phone-context=unknown SIP/2.0 >Via: SIP/2.0/UDP 213.160.252.50:53893 >From: "611012210" <sip:611012210@213.160.252.50> >To: <sip:4842@213.160.252.2;user=phone;phone-context=unknown> >Date: Wed, 30 Jul 2003 09:26:11 GMT >Call-ID: 635D27D4-CB1D0233-0-8E9DB84@213.160.252.50 >Cisco-Guid: 1667049428-3407675953-0-149543808 >User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled >CSeq: 101 INVITE >Max-Forwards: 6 >Timestamp: 1059557171 >Contact: <sip:611012210@213.160.252.50:5060;user=phone> >Expires: 180 >Content-Type: application/sdp >Content-Length: 149 > >v=0 >o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 >s=SIP Call >c=IN IP4 213.160.252.50 >t=0 0 >m=audio 20032 RTP/AVP 8 0 65535 18 > >15 headers, 6 lines >Using latest request as basis request >Sending to 213.160.252.50 : 53893 (non-NAT) >Found audio format 8 >Found audio format 0 >Found audio format 65535 >Found audio format 18 >Capabilities: us - 524302, them - 268/0, combined - 12 >Non-codec capabilities: us - 1, them - 0, combined - 0 >AM00CM01*CLI> >Disconnected from Asterisk server > > >********* DISCLAIMER ********* > >This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > >
yves.schaaf@restena.lu
2003-Jul-30 22:56 UTC
[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Inbound calls work pretty fine again, Thanks for you help Yves |---------+-------------------------------------> | | "Brenton D. Rothchild" | | | <brothchild@dstorage.com> | | | Sent by: | | | asterisk-users-admin@lists| | | .digium.com | | | | | | | | | 30/07/2003 16:15 | | | Please respond to | | | asterisk-users | | | | |---------+-------------------------------------> >-----------------------------------------------------------------------------------------------------------------------| | | | To: <asterisk-users@lists.digium.com> | | cc: | | Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 | >-----------------------------------------------------------------------------------------------------------------------| That also worked for me. My AudioCodes MP-104 FXO has no problem making inbound calls now. Thanks Patrick and Adam. -Brenton ----- Original Message ----- From: "Low, Adam" <ALow@Prioritytelecom.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 8:45 AM Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134> Well found Patrick, that did the trick for me as well ! > > I had been trying to debug 1.135 where this portion of code wasn't addedyet ... thats a lesson learnt ...> > -----Original Message----- > From: Patrick > To: 'asterisk-users@lists.digium.com ' > Sent: 30/07/03 15:04 > Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 > > > It is in the find_user() routine. If it is not an extension on the > PBX, > it should return a zero > > if ( isfound ) { > ast_log(LOG_DEBUG, "%s is not a local user\n", name); > ast_pthread_mutex_unlock(&userl.lock); > return 1; <--- this is the problem - change it to a 0. > } > > It isn't an error, so it should just return. Change that and the > function > will work properly. I tested it using an AS5350 and successly made an > inbound call. > > Patrick > > > On Wed, 30 Jul 2003, Low, Adam wrote: > > > Brenton, Yves, ... > > > > I've located the cause of the problem in chan_sip.c but am still > trying to find the exact cause being completely new to the asterisk > code. It seems that there was an added function in 1.135 called > 'find_user' that is supposed to lookup the users incoming call limit but > the routine is unable to find a matching user for my AS5300 which I > suspect is because it does not REGISTER with the server prior to > attempting to send calls. > > > > I'm going to continue debugging a little later and see if I can narrow > it down more ... > > > > Adam > > > > -----Original Message----- > > From: yves.schaaf@restena.lu > > To: asterisk-users@lists.digium.com > > Sent: 30/07/03 14:09 > > Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs > 1.134 > > > > > > Hi, > > > > I am using the latest cvs release of asterisk, and the behaviour is in > > fact > > the same, > > > > outbound calls work fine, > > but for inbound calls (from C2651 over PSTN) , SIP messages get > > "blocked" > > by asterisk, and never reach the phone. > > > > The setup is the same : 7960 <------> asterisk <------> C2651<-----> > > PSTN > > > > Yves > > > > > > |---------+-------------------------------------> > > | | "Low, Adam" | > > | | <ALow@Prioritytelecom.com>| > > | | Sent by: | > > | | asterisk-users-admin@lists| > > | | .digium.com | > > | | | > > | | | > > | | 30/07/2003 11:37 | > > | | Please respond to | > > | | asterisk-users | > > | | | > > |---------+-------------------------------------> > > > > > >----------------------------------------------------------------------- > > ------------------------------------------------| > > | > > | > > | To: "'asterisk-users@lists.digium.com'" > > <asterisk-users@lists.digium.com> | > > | cc: > > | > > | Subject: [Asterisk-Users] chan_sip.c problems problems from > > cvs 1.134 | > > > > > >----------------------------------------------------------------------- > > ------------------------------------------------| > > > > > > > > > > All, > > > > I've found problems in my setup with the latest couple of revisions > > (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 > > asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, > > everything > > is in the same VLAN and only running SIP. > > > > Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 > > > > But inbound calls fail, I see the initial INVITE from the AS5300 which > > is > > received by asterisk but not responded to and then the AS5300 sends > > another > > few INVITE's which are received but ignored assumable as they were > > duplicates for the first. > > > > Unfortunately since I've been trying the different cvs revisions of > > chan_sip.c I've got susbequent problems with the server crashing after > > the > > first INVITE from the AS5300 using anything greater than cvs 1.134 > > > > I suspect this is something to do with the per-user limits added in > cvs > > 1.135 but I am curious to see if anyone has any problems with the > latest > > cvs elease of asterisk with SIP ? > > > > Adam > > > > Sip read: > > INVITE sip:4842@213.160.252.2;user=phone;phone-context=unknown SIP/2.0 > > Via: SIP/2.0/UDP 213.160.252.50:53893 > > From: "611012210" <sip:611012210@213.160.252.50> > > To: <sip:4842@213.160.252.2;user=phone;phone-context=unknown> > > Date: Wed, 30 Jul 2003 09:26:11 GMT > > Call-ID: 635D27D4-CB1D0233-0-8E9DB84@213.160.252.50 > > Cisco-Guid: 1667049428-3407675953-0-149543808 > > User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled > > CSeq: 101 INVITE > > Max-Forwards: 6 > > Timestamp: 1059557171 > > Contact: <sip:611012210@213.160.252.50:5060;user=phone> > > Expires: 180 > > Content-Type: application/sdp > > Content-Length: 149 > > > > v=0 > > o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 > > s=SIP Call > > c=IN IP4 213.160.252.50 > > t=0 0 > > m=audio 20032 RTP/AVP 8 0 65535 18 > > > > 15 headers, 6 lines > > Using latest request as basis request > > Sending to 213.160.252.50 : 53893 (non-NAT) > > Found audio format 8 > > Found audio format 0 > > Found audio format 65535 > > Found audio format 18 > > Capabilities: us - 524302, them - 268/0, combined - 12 > > Non-codec capabilities: us - 1, them - 0, combined - 0 > > AM00CM01*CLI> > > Disconnected from Asterisk server > > > > > > ********* DISCLAIMER ********* > > > > This message and any attachment are confidential and may be privileged > > or > > otherwise protected from disclosure and may include proprietary > > information. If you are not the intended recipient, please telephone > or > > email the sender and delete this message and any attachment from your > > system. If you are not the intended recipient you must not copy this > > message or attachment or disclose the contents to any other person > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ********* DISCLAIMER ********* > > > > This message and any attachment are confidential and may be privileged > or otherwise protected from disclosure and may include proprietary > information. If you are not the intended recipient, please telephone or > email the sender and delete this message and any attachment from your > system. If you are not the intended recipient you must not copy this > message or attachment or disclose the contents to any other person > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ********* DISCLAIMER ********* > > This message and any attachment are confidential and may be privileged orotherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users