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
[asterisk-users] Asterisk reject SIP INTITE from different source ports
It's working now after giving nat=yes, thanks. Deepika -- _ -- 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
[asterisk-users] Asterisk reject SIP INTITE from different source ports
Hi, On some SIP interconnects with devices like Cisco, Dialogic we get SIP invite from different source port every time and asterisk rejects that INVITE. Does anyone knows solution for this? --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijha...@oxygen8.com Skype: deepika-nijhawan W: http://www.oxygen8.com/ www.oxygen8.com This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the intended recipient/s. If you are not the intended recipient/s please note that any distribution, copying or use of this communication or the information in it is strictly prohibited. If you have received this communication in error please notify us by email or by telephone (08082060808) and then delete the email and any copies of it. This communication is from Oxygen8 Communications UK Ltd - Company Number 03383285. Registered Address; 12th Floor, Lyndon House, 58-62 Hagley Road, Birmingham, B16 8PE. VAT Registration Number: 792 4494 89 -- _ -- 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
