Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
Tried this... it got connected, but I can't hear any audio now whereas codec was allowed and both negotiated on alaw. Deepika -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif Sent: 15 June 2010 18:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Try setting insecure=port,invite in sip peer config. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Tuesday, June 15, 2010 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Deepika Nijhawan wrote: It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
Sounds like you have a firewall or NAT issue Deepika Nijhawan wrote: Tried this... it got connected, but I can't hear any audio now whereas codec was allowed and both negotiated on alaw. Deepika -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif Sent: 15 June 2010 18:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Try setting insecure=port,invite in sip peer config. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Tuesday, June 15, 2010 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Deepika Nijhawan wrote: It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
Deepika Nijhawan wrote: It just gives no matching peer error and doesn’t pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
Try setting insecure=port,invite in sip peer config. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Tuesday, June 15, 2010 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Deepika Nijhawan wrote: It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
