SDP issue receiving calls from SIP connection
Afternoon everyone, I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP provider (tesco's internet phone ... I know SIP isn't supported technically, but it sort of works...) Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but I'm not sure why ... here's a dump of my log / trace, anyone got any clues, I've tried adding extra codecs to my list and setting the late negotiation but that doesn't seem to fix it ... (registration all works fine by the way ...) ------------------------------------------------------------------------ INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone> Max-Forwards: 69 Content-Type: application/sdp Contact: <sip:ANOTHER_PHONE_NUMBER@77.75.1.10:5060;transport=udp> Allow: INVITE,CANCEL,ACK,BYE Accept: application/sdp Content-Length: 840 v=0 o=- 3250588766 0 IN IP4 77.75.1.10 s=- c=IN IP4 77.75.1.10 t=0 0 m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125 109 100 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G.729a/8000 a=rtpmap:101 G.726-16/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:4 G723/8000 a=rtpmap:105 G.723.1-L/8000 a=rtpmap:106 G.729b/8000 a=rtpmap:107 G.723.1a-H/8000 a=rtpmap:108 G.723.1a-L/8000 a=rtpmap:125 G.nX64/8000 a=rtpmap:109 AMR/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=maxptime:20 a=maxptime:30 a=ptime:20 a=ptime:30 a=X-cap: 1 audio RTP/AVP 100 a=X-cap: 2 image udptl t38 a=X-sqn:0 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 ------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8129468 (1416 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: canonizing sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060 with contact nta: INVITE (100) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80ef908, 0x80eccb8, 0x8120fe0) called soa_set_params(static::0x8111d08, ...) called nta_leg_tcreate(0x81200c0) soa_init_offer_answer(static::0x8111d08) called soa_set_remote_sdp(static::0x8111d08, (nil), 0x810ed20, 840) called nua(0x8120fe0): INVITE server: error parsing SDP nua: nua_invite_server_respond: entering tport_tsend(0x80fcbf0) tpn = UDP/77.75.1.10:5060 tport_resolve addrinfo = 77.75.1.10:5060 tport_by_addrinfo(0x80fcbf0): not found by name UDP/77.75.1.10:5060 tport_vsend returned 661 send 661 bytes to udp/[77.75.1.10]:5060 at 13:16:28.358779: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk Content-Length: 0 ------------------------------------------------------------------------ nta: sent 400 Bad Session Description for INVITE (100) nta_leg_destroy(0x81200c0) soa_destroy(static::0x8111d08) called tport_wakeup_pri(0x80fcbf0): events IN tport_recv_event(0x80fcbf0) tport_recv_iovec(0x80fcbf0) msg 0x8111d08 from (udp/192.168.1.10:5060) has 442 bytes, veclen = 1 recv 442 bytes from udp/[77.75.1.10]:5060 at 13:16:28.377628: ------------------------------------------------------------------------ ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 CSeq: 100 ACK Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8111d08 (442 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: ACK (100) is going to INVITE (100) nta: timer set next to 4820 ms nta: timer I fired, terminate 400 response incoming_reclaim_all((nil), (nil), 0xb174a1fc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set sofia loglevel 0 API CALL [sofia(loglevel 0)] output: Sofia-sip log level set to [0] Any clues would be welcomed ... Cheers Kirk Bateman
I assume it's "whining" about the SDP parse error encountered. nua(0x8120fe0): INVITE server: error parsing SDP I do not have the spec memorized but I'm sure if we show it to the sofia dev he can tell us the specific problem with the sdp. On Mon, Aug 11, 2008 at 8:30 AM, Kirk Bateman <kirk.bateman@gmail.com>wrote:
Afternoon everyone,
I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP provider (tesco's internet phone ... I know SIP isn't supported technically, but it sort of works...)
Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but I'm not sure why ... here's a dump of my log / trace, anyone got any clues, I've tried adding extra codecs to my list and setting the late negotiation but that doesn't seem to fix it ... (registration all works fine by the way ...)
------------------------------------------------------------------------ INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone> Max-Forwards: 69 Content-Type: application/sdp Contact: <sip:ANOTHER_PHONE_NUMBER@77.75.1.10:5060;transport=udp> Allow: INVITE,CANCEL,ACK,BYE Accept: application/sdp Content-Length: 840
v=0 o=- 3250588766 0 IN IP4 77.75.1.10 s=- c=IN IP4 77.75.1.10 t=0 0 m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125 109 100 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G.729a/8000 a=rtpmap:101 G.726-16/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:4 G723/8000 a=rtpmap:105 G.723.1-L/8000 a=rtpmap:106 G.729b/8000 a=rtpmap:107 G.723.1a-H/8000 a=rtpmap:108 G.723.1a-L/8000 a=rtpmap:125 G.nX64/8000 a=rtpmap:109 AMR/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=maxptime:20 a=maxptime:30 a=ptime:20 a=ptime:30 a=X-cap: 1 audio RTP/AVP 100 a=X-cap: 2 image udptl t38 a=X-sqn:0 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 ------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8129468 (1416 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: canonizing sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060 with contact nta: INVITE (100) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80ef908, 0x80eccb8, 0x8120fe0) called soa_set_params(static::0x8111d08, ...) called nta_leg_tcreate(0x81200c0) soa_init_offer_answer(static::0x8111d08) called soa_set_remote_sdp(static::0x8111d08, (nil), 0x810ed20, 840) called nua(0x8120fe0): INVITE server: error parsing SDP nua: nua_invite_server_respond: entering tport_tsend(0x80fcbf0) tpn = UDP/77.75.1.10:5060 tport_resolve addrinfo = 77.75.1.10:5060 tport_by_addrinfo(0x80fcbf0): not found by name UDP/77.75.1.10:5060 tport_vsend returned 661 send 661 bytes to udp/[77.75.1.10]:5060 at 13:16:28.358779: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk Content-Length: 0
------------------------------------------------------------------------ nta: sent 400 Bad Session Description for INVITE (100) nta_leg_destroy(0x81200c0) soa_destroy(static::0x8111d08) called tport_wakeup_pri(0x80fcbf0): events IN tport_recv_event(0x80fcbf0) tport_recv_iovec(0x80fcbf0) msg 0x8111d08 from (udp/192.168.1.10:5060) has 442 bytes, veclen = 1 recv 442 bytes from udp/[77.75.1.10]:5060 at 13:16:28.377628: ------------------------------------------------------------------------ ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 CSeq: 100 ACK Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Max-Forwards: 69 Content-Length: 0
------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8111d08 (442 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: ACK (100) is going to INVITE (100) nta: timer set next to 4820 ms nta: timer I fired, terminate 400 response incoming_reclaim_all((nil), (nil), 0xb174a1fc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set sofia loglevel 0 API CALL [sofia(loglevel 0)] output: Sofia-sip log level set to [0]
Any clues would be welcomed ...
Cheers
Kirk Bateman
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony, Yes I'd figured that out :) I couldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday). I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that. I'm hoping the sofia dev can point me in the right direction. I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that). Cheers Kirk 2008/8/11 Anthony Minessale <anthony.minessale@gmail.com>
I assume it's "whining" about the SDP parse error encountered. nua(0x8120fe0): INVITE server: error parsing SDP
I do not have the spec memorized but I'm sure if we show it to the sofia dev he can tell us the specific problem with the sdp.
On Mon, Aug 11, 2008 at 8:30 AM, Kirk Bateman <kirk.bateman@gmail.com>wrote:
Afternoon everyone,
I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP provider (tesco's internet phone ... I know SIP isn't supported technically, but it sort of works...)
Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but I'm not sure why ... here's a dump of my log / trace, anyone got any clues, I've tried adding extra codecs to my list and setting the late negotiation but that doesn't seem to fix it ... (registration all works fine by the way ...)
------------------------------------------------------------------------ INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone> Max-Forwards: 69 Content-Type: application/sdp Contact: <sip:ANOTHER_PHONE_NUMBER@77.75.1.10:5060;transport=udp> Allow: INVITE,CANCEL,ACK,BYE Accept: application/sdp Content-Length: 840
v=0 o=- 3250588766 0 IN IP4 77.75.1.10 s=- c=IN IP4 77.75.1.10 t=0 0 m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125 109 100 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G.729a/8000 a=rtpmap:101 G.726-16/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:4 G723/8000 a=rtpmap:105 G.723.1-L/8000 a=rtpmap:106 G.729b/8000 a=rtpmap:107 G.723.1a-H/8000 a=rtpmap:108 G.723.1a-L/8000 a=rtpmap:125 G.nX64/8000 a=rtpmap:109 AMR/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=maxptime:20 a=maxptime:30 a=ptime:20 a=ptime:30 a=X-cap: 1 audio RTP/AVP 100 a=X-cap: 2 image udptl t38 a=X-sqn:0 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202
------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8129468 (1416 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: canonizing sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060 with contact nta: INVITE (100) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80ef908, 0x80eccb8, 0x8120fe0) called soa_set_params(static::0x8111d08, ...) called nta_leg_tcreate(0x81200c0) soa_init_offer_answer(static::0x8111d08) called soa_set_remote_sdp(static::0x8111d08, (nil), 0x810ed20, 840) called nua(0x8120fe0): INVITE server: error parsing SDP nua: nua_invite_server_respond: entering tport_tsend(0x80fcbf0) tpn = UDP/77.75.1.10:5060 tport_resolve addrinfo = 77.75.1.10:5060 tport_by_addrinfo(0x80fcbf0): not found by name UDP/77.75.1.10:5060 tport_vsend returned 661 send 661 bytes to udp/[77.75.1.10]:5060 at 13:16:28.358779:
------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk Content-Length: 0
------------------------------------------------------------------------ nta: sent 400 Bad Session Description for INVITE (100) nta_leg_destroy(0x81200c0) soa_destroy(static::0x8111d08) called tport_wakeup_pri(0x80fcbf0): events IN tport_recv_event(0x80fcbf0) tport_recv_iovec(0x80fcbf0) msg 0x8111d08 from (udp/192.168.1.10:5060) has 442 bytes, veclen = 1 recv 442 bytes from udp/[77.75.1.10]:5060 at 13:16:28.377628:
------------------------------------------------------------------------ ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 CSeq: 100 ACK Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Max-Forwards: 69 Content-Length: 0
------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8111d08 (442 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: ACK (100) is going to INVITE (100) nta: timer set next to 4820 ms nta: timer I fired, terminate 400 response incoming_reclaim_all((nil), (nil), 0xb174a1fc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set sofia loglevel 0 API CALL [sofia(loglevel 0)] output: Sofia-sip log level set to [0]
Any clues would be welcomed ...
Cheers
Kirk Bateman
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =. Mike On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote:
ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).
Cheers
Kirk
Michael, Great :( so probably something I'd need to mod the sofia source for really then. Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :) Cheers Kirk 2008/8/11 Michael Jerris <mike@jerris.com>
The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =.
Mike
On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote:
ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).
Cheers
Kirk
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Try posting this to the sofia-sip mailing list, lets see if we can sort out a fix there. Mike On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote:
Michael,
Great :( so probably something I'd need to mod the sofia source for really then.
Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :)
Cheers
Kirk
2008/8/11 Michael Jerris <mike@jerris.com> The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =.
Mike
On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote:
ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).
Cheers
Kirk
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you mean sofia-sip-devel list ? Cheers Kirk 2008/8/11 Michael Jerris <mike@jerris.com>
Try posting this to the sofia-sip mailing list, lets see if we can sort out a fix there. Mike
On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote:
Michael,
Great :( so probably something I'd need to mod the sofia source for really then.
Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :)
Cheers
Kirk
2008/8/11 Michael Jerris <mike@jerris.com>
The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =.
Mike
On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote:
ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).
Cheers
Kirk
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Pekka Pessi from the sofia-sip project pushed a patch that is now synced to our tree. Please update and give it a try. Mike On Aug 11, 2008, at 11:41 AM, Michael Jerris wrote:
Try posting this to the sofia-sip mailing list, lets see if we can sort out a fix there.
Mike
On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote:
Michael,
Great :( so probably something I'd need to mod the sofia source for really then.
Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :)
Cheers
Kirk
2008/8/11 Michael Jerris <mike@jerris.com> The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =.
Mike
On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote:
ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).
Cheers
Kirk
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Mike, I'm just rebuilding FS with latest SVN... will let you know the result. Cheers Kirk 2008/8/12 Michael Jerris <mike@jerris.com>
Pekka Pessi from the sofia-sip project pushed a patch that is now synced to our tree. Please update and give it a try. Mike
On Aug 11, 2008, at 11:41 AM, Michael Jerris wrote:
Try posting this to the sofia-sip mailing list, lets see if we can sort out a fix there. Mike
On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote:
Michael,
Great :( so probably something I'd need to mod the sofia source for really then.
Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :)
Cheers
Kirk
2008/8/11 Michael Jerris <mike@jerris.com>
The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =.
Mike
On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote:
ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).
Cheers
Kirk
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Mike, Pekka ... All seems to work nicely thanks... Cheers Kirk 2008/8/13 Kirk Bateman <kirk.bateman@gmail.com>
Mike,
I'm just rebuilding FS with latest SVN... will let you know the result.
Cheers
Kirk
2008/8/12 Michael Jerris <mike@jerris.com>
Pekka Pessi from the sofia-sip project pushed a patch that is now synced to
our tree. Please update and give it a try. Mike
On Aug 11, 2008, at 11:41 AM, Michael Jerris wrote:
Try posting this to the sofia-sip mailing list, lets see if we can sort out a fix there. Mike
On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote:
Michael,
Great :( so probably something I'd need to mod the sofia source for really then.
Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :)
Cheers
Kirk
2008/8/11 Michael Jerris <mike@jerris.com>
The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =.
Mike
On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote:
ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).
Cheers
Kirk
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participants (3)
-
Anthony Minessale
-
Kirk Bateman
-
Michael Jerris