-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list,
2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having the same issues. In the trace below, I'm sending a fax from Hylafax server through iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw) connected to the PSTN via ISDN; the call is to my test fax machine, connected to the PSTN. chan_pjsip is used on Asterisk-13, and chan_sip is used on Asterisk-11. This how endpoint t0gw (Asterisk-11) is configured on tiare (Asterisk-13 ): tiare*CLI> pjsip show endpoint t0gw ... t38_udptl : true t38_udptl_ec : fec t38_udptl_ipv6 : false t38_udptl_maxdatagram : 400 t38_udptl_nat : false ... Could someone explain why I'm getting "Not acceptable" below? -- Accepting AUTHENTICATED call from 127.0.0.1:4570: -- > requested format = slin, -- > requested prefs = (), -- > actual format = slin, -- > host prefs = (slin), -- > priority = mine -- Executing [40ZZZZZZ@fax-sortant:1] NoOp("IAX2/iaxmodem0-7838", " calls 40ZZZZZZ (local)") in new stack -- Executing [40ZZZZZZ@fax-sortant:2] Set("IAX2/iaxmodem0-7838", "FAXOPT(gateway)=yes") in new stack -- Executing [40ZZZZZZ@fax-sortant:3] Dial("IAX2/iaxmodem0-7838", "PJSIP/40ZZZZZZ@t0gw") in new stack -- Called PJSIP/40ZZZZZZ@t0gw <--- Transmitting SIP request (936 bytes) to UDP:192.168.0.10:5060 ---> INVITE sip:40zzz...@gw.sysnux.pf SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e3 8e5f1 From: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 To: <sip:40zzz...@gw.sysnux.pf> Contact: <sip:63035284-ad7d-484f-8e54-f5ea54f39104@192.168.0.200:5060> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk GPL PBX Content-Type: application/sdp Content-Length: 238 v=0 o=- 1710591484 1710591484 IN IP4 192.168.0.200 s=Asterisk c=IN IP4 192.168.0.200 t=0 0 m=audio 8834 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (585 bytes) from UDP:192.168.0.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1; received=192.168.0.200;rport=5060 From: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 To: <sip:40zzz...@gw.sysnux.pf> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:40ZZZZZZ@192.168.0.10:5060> Content-Length: 0 <--- Received SIP response (895 bytes) from UDP:192.168.0.10:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1; received=192.168.0.200;rport=5060 From: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 To: <sip:40zzz...@gw.sysnux.pf>;tag=as7bba6b0d Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:40ZZZZZZ@192.168.0.10:5060> Content-Type: application/sdp Require: timer Content-Length: 236 v=0 o=root 2087714374 2087714374 IN IP4 192.168.0.10 s=Asterisk PBX 11.18.0 c=IN IP4 192.168.0.10 t=0 0 m=audio 16834 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- PJSIP/t0gw-0000001a is making progress passing it to IAX2/iaxmodem0-7838 <--- Received SIP response (601 bytes) from UDP:192.168.0.10:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1; received=192.168.0.200;rport=5060 From: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 To: <sip:40zzz...@gw.sysnux.pf>;tag=as7bba6b0d Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:40ZZZZZZ@192.168.0.10:5060> Content-Length: 0 -- PJSIP/t0gw-0000001a is ringing <--- Received SIP response (881 bytes) from UDP:192.168.0.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1; received=192.168.0.200;rport=5060 From: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 To: <sip:40zzz...@gw.sysnux.pf>;tag=as7bba6b0d Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:40ZZZZZZ@192.168.0.10:5060> Content-Type: application/sdp Require: timer Content-Length: 236 v=0 o=root 2087714374 2087714374 IN IP4 192.168.0.10 s=Asterisk PBX 11.18.0 c=IN IP4 192.168.0.10 t=0 0 m=audio 16834 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- Transmitting SIP request (412 bytes) to UDP:192.168.0.10:5060 ---> ACK sip:40ZZZZZZ@192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj8504e505-1222-4747-955f-4788fef f58d1 From: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 To: <sip:40zzz...@gw.sysnux.pf>;tag=as7bba6b0d Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 ACK Max-Forwards: 70 User-Agent: Asterisk GPL PBX Content-Length: 0 -- PJSIP/t0gw-0000001a answered IAX2/iaxmodem0-7838 -- Channel PJSIP/t0gw-0000001a joined 'simple_bridge' basic-bridge <56a7726f-44a3-4df3-aee0-d21020aa5be1> -- Channel IAX2/iaxmodem0-7838 joined 'simple_bridge' basic-bridge <56a7726f-44a3-4df3-aee0-d21020aa5be1> <--- Received SIP request (954 bytes) from UDP:192.168.0.10:5060 ---> UPDATE sip:63035284-ad7d-484f-8e54-f5ea54f39104@192.168.0.200:5060 SIP/2 .0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK4fd84f17;rport Max-Forwards: 70 From: <sip:40zzz...@gw.sysnux.pf>;tag=as7bba6b0d To: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 Contact: <sip:40ZZZZZZ@192.168.0.10:5060> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 102 UPDATE User-Agent: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 287 v=0 o=root 2087714374 2087714375 IN IP4 192.168.0.10 s=Asterisk PBX 11.18.0 c=IN IP4 192.168.0.10 t=0 0 m=image 5720 udptl t38 c=IN IP4 192.168.0.10 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC <--- Transmitting SIP response (376 bytes) to UDP:192.168.0.10:5060 ---> SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK4fd84f1 7 Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 From: <sip:40zzz...@gw.sysnux.pf>;tag=as7bba6b0d To: "SysNux" <sip:+68940XXXXXX@192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 CSeq: 102 UPDATE Server: Asterisk GPL PBX Content-Length: 0 Is anyone successfully using chan_pjsip and iaxmodem? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -----BEGIN PGP SIGNATURE----- iEYEARECAAYFAlW1ojoACgkQuu7Rv+oOo/iU1gCglmxl6Pe3igseOwpbWfWtZdqg qzAAoJ8/zxzP3Eg79DxT7cjyXJj2oP9h =FR59 -----END PGP SIGNATURE----- -- _____________________________________________________________________ -- 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