Re: [asterisk-users] RTP not being switched between both SIP endpoints
Hi, Since opensips is not handling media (i presume) is the audio not already going direct from asterisk to the other endpoint? Thanks Kenny From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Gareth Blades [mailinglist+aster...@dns99.co.uk] Sent: 17 September 2013 11:17 To: asterisk-users@lists.digium.com Subject: [asterisk-users] RTP not being switched between both SIP endpoints We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call coming from opensips. Directmedia is not specifically defined so its using the asterisk default value. * Name : vmpubopensips3 Description : Secret : Not set MD5Secret: Not set Remote Secret: Not set Context : from-pubopensips Record On feature : automon Record Off feature : automon Subscr.Cont. : Not set Language : Tonezone : Not set AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : no Force rport : Auto (No) Symmetric RTP: No ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 88.x.x.x Addr-IP : 88.x.x.x:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (gsm|ulaw|alaw) Codec Order : (alaw:20,ulaw:20,gsm:20) Auto-Framing : No Status : Unmonitored Useragent: Reg. Contact : Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10428 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. When we send the outbound SIP information we advertise the following SDP :- v=0. o=root 431105643 431105643 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10144 RTP/AVP 8 3 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and the other end replies with :- v=0. o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x s=sip call. c=IN IP4 203.x.x.x t=0 0. m=audio 34146 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. In the Dial() command the only option we are using is M() which is used to run a macro when the call is answered. This is used to update cdr records and perform other features if they are enabled. In this case we are just updating the cdr records so I would expect the audio to be switched as soon as the macro finishes. Any ideas what could be wrong? We are running Asterisk PBX 11.2-cert2 Thanks Gareth -- _ -- 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
Re: [asterisk-users] RTP not being switched between both SIP endpoints
On 18/09/13 12:40, Kenny Watson wrote: Hi, Since opensips is not handling media (i presume) is the audio not already going direct from asterisk to the other endpoint? Thanks Kenny Opensips wasnt handling the media so the audio was between the original caller and asterisk (with the signalling being relayed by opensips). It was just when we dialled onto the final destination via SIP asterisk stayed in the loop and didnt issue a reinvite. Its all fixed now. Although we weren't using any features the AGI application was setting DYNAMIC_FEATURES to an empty string which was enough to keep asterisk in a loop. We stopped the AGI from setting the variable if there were no features and it started working. Thanks Gareth -- _ -- 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] RTP not being switched between both SIP endpoints
We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call coming from opensips. Directmedia is not specifically defined so its using the asterisk default value. * Name : vmpubopensips3 Description : Secret : Not set MD5Secret: Not set Remote Secret: Not set Context : from-pubopensips Record On feature : automon Record Off feature : automon Subscr.Cont. : Not set Language : Tonezone : Not set AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : no Force rport : Auto (No) Symmetric RTP: No ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 88.x.x.x Addr-IP : 88.x.x.x:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (gsm|ulaw|alaw) Codec Order : (alaw:20,ulaw:20,gsm:20) Auto-Framing : No Status : Unmonitored Useragent: Reg. Contact : Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10428 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. When we send the outbound SIP information we advertise the following SDP :- v=0. o=root 431105643 431105643 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10144 RTP/AVP 8 3 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and the other end replies with :- v=0. o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x s=sip call. c=IN IP4 203.x.x.x t=0 0. m=audio 34146 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. In the Dial() command the only option we are using is M() which is used to run a macro when the call is answered. This is used to update cdr records and perform other features if they are enabled. In this case we are just updating the cdr records so I would expect the audio to be switched as soon as the macro finishes. Any ideas what could be wrong? We are running Asterisk PBX 11.2-cert2 Thanks Gareth -- _ -- 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
