-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, has someone successfully used freeswitch for T.38 fax? If so, can you explain to me how to set freeswitch up? First call setup was successful, but then fax target sent a re-INVITE and freeswitch terminated the call because of a codec error (see attached log sequence). regards Helmut - --Console LOG-- nta: sent 200 OK for INVITE (118) nua(0x809fec8): event i_invite 200 OK nua(0x809fec8): ready call updated: completed received offer, sent answer soa_get_remote_sdp(static::0x80a4970, [0x413fedd8], [0x413fedd4], [(nil)]) called soa_get_local_sdp(static::0x80a4970, [0x413fedd0], [0x413fedcc], [(nil)]) called soa_get_params(static::0x80a4970, ...) called nua(0x809fec8): event i_state 200 OK nua: nua_event: entering nua: nua_event: entering 2007-07-17 11:12:08 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de entering state [completed] 2007-07-17 11:12:08 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Remote SDP: v=0 o=bell 1216809 1216810 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10844 UDPTL t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy 2007-07-17 11:12:08 [ERR] sofia.c:1218 sofia_handle_sip_i_state() Reinvite Codec Error! -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGnIxp4tZeNddg3dwRAt9EAJ0QpFyIrHjaFIxS46n2Px9SHHbKjwCfa2H9 ezoC6omq/XE1gw6alnGxn1Y= =EiNE -----END PGP SIGNATURE-----
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello again, Update: my environment looks like this: kapanga soft phone -> openser -> freeswitch -> SBC-> PSTN-Gateway T.38 call setup seems to work like this: 1. INVITE with SDP (voice) to PSTN Gateway (fax B) 2. Session in Progress with SDP (voice) back to fax A 3. OK with SDP (voice) back to fax A 4. Faxdetection in PSTN Gateway 5. re-INVITE with SDP (T.38) from PSTN-Gateway to fax A 6. fax exchange With freeswitch it looks like this: 1. INVITE with SDP (voice) to PSTN Gateway terminated by freeswitch 2. INVITE with SDP (voice) from freeswitch to PSTN-Gateway 3. Session in Progress with SDP (voice) back to freeswitch 4. Session in Progress with SDP (voice) from freeswitch back to fax A 5. OK with SDP (voice) back to freeswitch 6. OK with SDP (T.38) from freeswitch back to fax A 7. Voice exchange 8. Faxdetection in PSTN Gateway 9. re-INVITE with SDP (T.38) from PSTN-Gateway to freeswitch 10. Codec Error in freeswitch/Sofia It's seems to me that freeswitch should not touch the SDP in this case, but should send in transparently from pstn gateway to fax A. I looked into freeswutch's online docu and found this, which could help : "No Media Mode No media mode is an SDP Passthough feature that permits to endpoints that can see each other (no funky NAT's) to connect their media sessions directly while FreeSwitch maintains control of the SIP signalling. This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSwitch (video) or if you are using FreeSwitch in a high performance walled garden network and want to minimize the RTP handling FreeSwitch is doing to maximize call traffic. Before executing the bridge action you must set the "no-media" flag to true. " I changed my dialplan to this: - ----------------- <extension name="PrintVars" continue="true"> <condition field="destination_number" expression="(.*)"> <action application="set" data="no_media=true"/> </condition> </extension> <extension name="Fax dialout"> <condition field="destination_number" expression="^94(\d{4})$"> <action application="set" data="effective_caller_id_name=032223930016"/> <action application="set" data="effective_caller_id_number=032223930016"/> <action application="set" data="call_timeout=10"/> <action application="info"/> <action application="bridge" data="sofia/gateway/sip01.voice.ewetel.de/04418000$1@sip01.voice.ewetel.de"/> </condition> </extension> - -------------------------------- Unfortunately freeswitch still changes the SDP as a full B2BUA. I tried various values for no_media: true, TRUE, 1 but no change. I even put it directly into "Fax dialout" exttension. It seems freeswitch ignores this setting. At least it is printed out on console by info action. I attached the log of freswitch's answere to pstn gateway's re-INVITE. Any ideas what I do wrong ? regards helmut - ----------LOG----------------- recv 740 bytes from udp/[213.168.198.84]:5060 at 08:08:50.367892: ------------------------------------------------------------------------ INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Max-Forwards: 69 Contact: <sip:044180002799@213.168.198.84:5060;transport=udp> Content-Length: 274 Content-Type: application/sdp v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ tport_deliver(0x8091ed8): msg 0x809a2e0 (740 bytes) from udp/213.168.198.84:5090/sip next=(nil) nta: received INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 (CSeq 210) nta: canonizing sip:032223930016@212.6.123.118:5090 with contact nta: INVITE (210) going to existing leg nta: timer shortened to 200 ms nua: nua_stack_process_request: entering soa_init_offer_answer(static::0x80c59c8) called soa_set_remote_sdp(static::0x80c59c8, (nil), 0x809444a, 274) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x80c59c8) called soa_static_offer_answer_action(0x80c59c8, soa_generate_answer): called soa_static(0x80c59c8, soa_generate_answer): upgrade with remote description soa_static(0x80c59c8, soa_generate_answer): marking rejected media soa_sdp_mode_set_is_needed(0x413fec44, 0x80d1e50, ""): called soa_static(0x80c59c8, soa_generate_answer): storing local description soa_activate(static::0x80c59c8, (nil)) called soa_get_local_sdp(static::0x80c59c8, [(nil)], [0x413fed58], [0x413fed54]) called tport_tsend(0x8091ed8) tpn = UDP/213.168.198.84:5060 tport_resolve addrinfo = 213.168.198.84:5060 tport_by_addrinfo(0x8091ed8): not found by name UDP/213.168.198.84:5060 tport_vsend(0x8091ed8): 771 bytes of 771 to UDP/213.168.198.84:5060 tport_vsend returned 771 send 771 bytes to udp/[213.168.198.84]:5060 at 08:08:50.369576: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Contact: <sip:212.6.123.118:5090> User-Agent: FreeSWITCH(mod_sofia) Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Type: application/sdp Content-Disposition: session Content-Length: 156 v=0 o=FreeSWITCH 4751724862107259708 3984222328945838171 IN IP4 212.6.123.118 s=FreeSWITCH c=IN IP4 212.6.123.118 t=0 0 a=sendrecv m=image 0 UDPTL 9 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (210) nua(0x80c4e90): event i_invite 200 OK nua(0x80c4e90): ready call updated: completed received offer, sent answer soa_get_remote_sdp(static::0x80c59c8, [0x413fedd8], [0x413fedd4], [(nil)]) called soa_get_local_sdp(static::0x80c59c8, [0x413fedd0], [0x413fedcc], [(nil)]) called soa_get_params(static::0x80c59c8, ...) called nua(0x80c4e90): event i_state 200 OK nua: nua_event: entering nua: nua_event: entering 2007-07-18 10:08:50 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de entering state [completed] 2007-07-18 10:08:50 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Remote SDP: v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 UDPTL t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy 2007-07-18 10:08:50 [ERR] sofia.c:1218 sofia_handle_sip_i_state() Reinvite Codec Error! -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGndrk4tZeNddg3dwRAqnEAKCUC9OmSZJn86x/EwzAc4LNCJu8rQCgwyxh z/YOykDnV/VcqZPUsAUpYQE= =8hIO -----END PGP SIGNATURE-----
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, correction: 6. OK with SDP (T.38) from freeswitch back to fax A is wrong. It should be 6. OK with SDP (voice) from freeswitch back to fax A Helmut Kuper schrieb:
Hello again,
Update:
my environment looks like this:
kapanga soft phone -> openser -> freeswitch -> SBC-> PSTN-Gateway
T.38 call setup seems to work like this:
1. INVITE with SDP (voice) to PSTN Gateway (fax B) 2. Session in Progress with SDP (voice) back to fax A 3. OK with SDP (voice) back to fax A 4. Faxdetection in PSTN Gateway 5. re-INVITE with SDP (T.38) from PSTN-Gateway to fax A 6. fax exchange
With freeswitch it looks like this:
1. INVITE with SDP (voice) to PSTN Gateway terminated by freeswitch 2. INVITE with SDP (voice) from freeswitch to PSTN-Gateway 3. Session in Progress with SDP (voice) back to freeswitch 4. Session in Progress with SDP (voice) from freeswitch back to fax A 5. OK with SDP (voice) back to freeswitch 6. OK with SDP (T.38) from freeswitch back to fax A 7. Voice exchange 8. Faxdetection in PSTN Gateway 9. re-INVITE with SDP (T.38) from PSTN-Gateway to freeswitch 10. Codec Error in freeswitch/Sofia
It's seems to me that freeswitch should not touch the SDP in this case, but should send in transparently from pstn gateway to fax A.
I looked into freeswutch's online docu and found this, which could help :
"No Media Mode
No media mode is an SDP Passthough feature that permits to endpoints that can see each other (no funky NAT's) to connect their media sessions directly while FreeSwitch maintains control of the SIP signalling. This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSwitch (video) or if you are using FreeSwitch in a high performance walled garden network and want to minimize the RTP handling FreeSwitch is doing to maximize call traffic.
Before executing the bridge action you must set the "no-media" flag to true. "
I changed my dialplan to this:
----------------- <extension name="PrintVars" continue="true"> <condition field="destination_number" expression="(.*)"> <action application="set" data="no_media=true"/> </condition> </extension>
<extension name="Fax dialout"> <condition field="destination_number" expression="^94(\d{4})$"> <action application="set" data="effective_caller_id_name=032223930016"/> <action application="set" data="effective_caller_id_number=032223930016"/> <action application="set" data="call_timeout=10"/> <action application="info"/> <action application="bridge" data="sofia/gateway/sip01.voice.ewetel.de/04418000$1@sip01.voice.ewetel.de"/> </condition> </extension> --------------------------------
Unfortunately freeswitch still changes the SDP as a full B2BUA.
I tried various values for no_media: true, TRUE, 1 but no change. I even put it directly into "Fax dialout" exttension. It seems freeswitch ignores this setting. At least it is printed out on console by info action.
I attached the log of freswitch's answere to pstn gateway's re-INVITE.
Any ideas what I do wrong ?
regards helmut
----------LOG-----------------
recv 740 bytes from udp/[213.168.198.84]:5060 at 08:08:50.367892: ------------------------------------------------------------------------ INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Max-Forwards: 69 Contact: <sip:044180002799@213.168.198.84:5060;transport=udp> Content-Length: 274 Content-Type: application/sdp
v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ tport_deliver(0x8091ed8): msg 0x809a2e0 (740 bytes) from udp/213.168.198.84:5090/sip next=(nil) nta: received INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 (CSeq 210) nta: canonizing sip:032223930016@212.6.123.118:5090 with contact nta: INVITE (210) going to existing leg nta: timer shortened to 200 ms nua: nua_stack_process_request: entering soa_init_offer_answer(static::0x80c59c8) called soa_set_remote_sdp(static::0x80c59c8, (nil), 0x809444a, 274) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x80c59c8) called soa_static_offer_answer_action(0x80c59c8, soa_generate_answer): called soa_static(0x80c59c8, soa_generate_answer): upgrade with remote description soa_static(0x80c59c8, soa_generate_answer): marking rejected media soa_sdp_mode_set_is_needed(0x413fec44, 0x80d1e50, ""): called soa_static(0x80c59c8, soa_generate_answer): storing local description soa_activate(static::0x80c59c8, (nil)) called soa_get_local_sdp(static::0x80c59c8, [(nil)], [0x413fed58], [0x413fed54]) called tport_tsend(0x8091ed8) tpn = UDP/213.168.198.84:5060 tport_resolve addrinfo = 213.168.198.84:5060 tport_by_addrinfo(0x8091ed8): not found by name UDP/213.168.198.84:5060 tport_vsend(0x8091ed8): 771 bytes of 771 to UDP/213.168.198.84:5060 tport_vsend returned 771 send 771 bytes to udp/[213.168.198.84]:5060 at 08:08:50.369576: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Contact: <sip:212.6.123.118:5090> User-Agent: FreeSWITCH(mod_sofia) Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Type: application/sdp Content-Disposition: session Content-Length: 156
v=0 o=FreeSWITCH 4751724862107259708 3984222328945838171 IN IP4 212.6.123.118 s=FreeSWITCH c=IN IP4 212.6.123.118 t=0 0 a=sendrecv m=image 0 UDPTL 9 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (210) nua(0x80c4e90): event i_invite 200 OK nua(0x80c4e90): ready call updated: completed received offer, sent answer soa_get_remote_sdp(static::0x80c59c8, [0x413fedd8], [0x413fedd4], [(nil)]) called soa_get_local_sdp(static::0x80c59c8, [0x413fedd0], [0x413fedcc], [(nil)]) called soa_get_params(static::0x80c59c8, ...) called nua(0x80c4e90): event i_state 200 OK nua: nua_event: entering nua: nua_event: entering 2007-07-18 10:08:50 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de entering state [completed] 2007-07-18 10:08:50 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Remote SDP: v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 UDPTL t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy
2007-07-18 10:08:50 [ERR] sofia.c:1218 sofia_handle_sip_i_state() Reinvite Codec Error!
_______________________________________________ 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 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGneq74tZeNddg3dwRAup5AJoCUxXjLjpBayv8lRO2RdKjWg6zOwCfT5qn JqkcjK2LQ7Ol+xHkk7vcCwE= =VGEd -----END PGP SIGNATURE-----
Try bypass_media instead of no_media /b On Jul 18, 2007, at 4:18 AM, Helmut Kuper wrote:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hello again,
Update:
my environment looks like this:
kapanga soft phone -> openser -> freeswitch -> SBC-> PSTN-Gateway
T.38 call setup seems to work like this:
1. INVITE with SDP (voice) to PSTN Gateway (fax B) 2. Session in Progress with SDP (voice) back to fax A 3. OK with SDP (voice) back to fax A 4. Faxdetection in PSTN Gateway 5. re-INVITE with SDP (T.38) from PSTN-Gateway to fax A 6. fax exchange
With freeswitch it looks like this:
1. INVITE with SDP (voice) to PSTN Gateway terminated by freeswitch 2. INVITE with SDP (voice) from freeswitch to PSTN-Gateway 3. Session in Progress with SDP (voice) back to freeswitch 4. Session in Progress with SDP (voice) from freeswitch back to fax A 5. OK with SDP (voice) back to freeswitch 6. OK with SDP (T.38) from freeswitch back to fax A 7. Voice exchange 8. Faxdetection in PSTN Gateway 9. re-INVITE with SDP (T.38) from PSTN-Gateway to freeswitch 10. Codec Error in freeswitch/Sofia
It's seems to me that freeswitch should not touch the SDP in this case, but should send in transparently from pstn gateway to fax A.
I looked into freeswutch's online docu and found this, which could help :
"No Media Mode
No media mode is an SDP Passthough feature that permits to endpoints that can see each other (no funky NAT's) to connect their media sessions directly while FreeSwitch maintains control of the SIP signalling. This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSwitch (video) or if you are using FreeSwitch in a high performance walled garden network and want to minimize the RTP handling FreeSwitch is doing to maximize call traffic.
Before executing the bridge action you must set the "no-media" flag to true. "
I changed my dialplan to this:
- ----------------- <extension name="PrintVars" continue="true"> <condition field="destination_number" expression="(.*)"> <action application="set" data="no_media=true"/> </condition> </extension>
<extension name="Fax dialout"> <condition field="destination_number" expression="^94(\d{4})$"> <action application="set" data="effective_caller_id_name=032223930016"/> <action application="set" data="effective_caller_id_number=032223930016"/> <action application="set" data="call_timeout=10"/> <action application="info"/> <action application="bridge" data="sofia/gateway/sip01.voice.ewetel.de/04418000 $1@sip01.voice.ewetel.de"/> </condition> </extension> - --------------------------------
Unfortunately freeswitch still changes the SDP as a full B2BUA.
I tried various values for no_media: true, TRUE, 1 but no change. I even put it directly into "Fax dialout" exttension. It seems freeswitch ignores this setting. At least it is printed out on console by info action.
I attached the log of freswitch's answere to pstn gateway's re-INVITE.
Any ideas what I do wrong ?
regards helmut
- ----------LOG-----------------
recv 740 bytes from udp/[213.168.198.84]:5060 at 08:08:50.367892:
---------------------------------------------------------------------- -- INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Max-Forwards: 69 Contact: <sip:044180002799@213.168.198.84:5060;transport=udp> Content-Length: 274 Content-Type: application/sdp
v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy
---------------------------------------------------------------------- -- tport_deliver(0x8091ed8): msg 0x809a2e0 (740 bytes) from udp/213.168.198.84:5090/sip next=(nil) nta: received INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 (CSeq 210) nta: canonizing sip:032223930016@212.6.123.118:5090 with contact nta: INVITE (210) going to existing leg nta: timer shortened to 200 ms nua: nua_stack_process_request: entering soa_init_offer_answer(static::0x80c59c8) called soa_set_remote_sdp(static::0x80c59c8, (nil), 0x809444a, 274) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x80c59c8) called soa_static_offer_answer_action(0x80c59c8, soa_generate_answer): called soa_static(0x80c59c8, soa_generate_answer): upgrade with remote description soa_static(0x80c59c8, soa_generate_answer): marking rejected media soa_sdp_mode_set_is_needed(0x413fec44, 0x80d1e50, ""): called soa_static(0x80c59c8, soa_generate_answer): storing local description soa_activate(static::0x80c59c8, (nil)) called soa_get_local_sdp(static::0x80c59c8, [(nil)], [0x413fed58], [0x413fed54]) called tport_tsend(0x8091ed8) tpn = UDP/213.168.198.84:5060 tport_resolve addrinfo = 213.168.198.84:5060 tport_by_addrinfo(0x8091ed8): not found by name UDP/ 213.168.198.84:5060 tport_vsend(0x8091ed8): 771 bytes of 771 to UDP/213.168.198.84:5060 tport_vsend returned 771 send 771 bytes to udp/[213.168.198.84]:5060 at 08:08:50.369576:
---------------------------------------------------------------------- -- SIP/2.0 200 OK Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Contact: <sip:212.6.123.118:5090> User-Agent: FreeSWITCH(mod_sofia) Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Type: application/sdp Content-Disposition: session Content-Length: 156
v=0 o=FreeSWITCH 4751724862107259708 3984222328945838171 IN IP4 212.6.123.118 s=FreeSWITCH c=IN IP4 212.6.123.118 t=0 0 a=sendrecv m=image 0 UDPTL 9
---------------------------------------------------------------------- -- nta: sent 200 OK for INVITE (210) nua(0x80c4e90): event i_invite 200 OK nua(0x80c4e90): ready call updated: completed received offer, sent answer soa_get_remote_sdp(static::0x80c59c8, [0x413fedd8], [0x413fedd4], [(nil)]) called soa_get_local_sdp(static::0x80c59c8, [0x413fedd0], [0x413fedcc], [(nil)]) called soa_get_params(static::0x80c59c8, ...) called nua(0x80c4e90): event i_state 200 OK nua: nua_event: entering nua: nua_event: entering 2007-07-18 10:08:50 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de entering state [completed] 2007-07-18 10:08:50 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Remote SDP: v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 UDPTL t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy
2007-07-18 10:08:50 [ERR] sofia.c:1218 sofia_handle_sip_i_state() Reinvite Codec Error! -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFGndrk4tZeNddg3dwRAqnEAKCUC9OmSZJn86x/EwzAc4LNCJu8rQCgwyxh z/YOykDnV/VcqZPUsAUpYQE= =8hIO -----END PGP SIGNATURE-----
_______________________________________________ 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
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi brian, thanks for the hint. I added "bypass_media" to my dialplan: <extension name="PrintVars" continue="true"> <condition field="destination_number" expression="(.*)"> <action application="set" data="effective_caller_id_name=032223930016"/> <action application="set" data="effective_caller_id_number=032223930016"/> <action application="set" data="call_timeout=10"/> </condition> </extension> <extension name="FAX Dialout"> <condition field="destination_number" expression="^94(\d{4})$"> <action application="set" data="bypass_media=true"/> <action application="info"/> <action application="bridge" data="sofia/gateway/sip01.voice.ewetel.de/04418000$1@sip01.voice.ewetel.de"/> </condition> </extension> With this dialplan I get a "480 Temporarily Unavailable" from freeswitch without forwarding the INVITE to PSTN Gateway. I use build 5536. LOG: recv 1208 bytes from udp/[212.6.123.118]:5060 at 07:47:46.687933: ------------------------------------------------------------------------ INVITE sip:942799@sip0.en.ewetel.de SIP/2.0 Record-Route: <sip:212.6.123.118;lr=on;ftag=1184830476> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK1028.95646442.0 Via: SIP/2.0/UDP 212.6.95.198:5070;branch=z9hG4bKB1A69A926C26EEF0C9A11FD3C7379087;rport=5070 From: "Your Long Name" <sip:v101@sip0.en.ewetel.de>;tag=1184830476 To: <sip:942799@sip0.en.ewetel.de> Contact: <sip:v101@212.6.95.198:5070;transport=udp> Call-ID: 990023F94830E6CDBB54ACDF59BF7929@sip0.en.ewetel.de User-Agent: Kapanga Softphone Desktop 1.00/2161c+1184674739_000CF1B3D117 Supported: timer, replaces CSeq: 1 INVITE Max-Forwards: 69 Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Length: 492 P-SI: Alias v=0 o=v101 1184830473 1184831265 IN IP4 212.6.95.198 s=Kapanga [1184830473] c=IN IP4 212.6.95.198 t=0 0 m=audio 5128 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=sendrecv a=silenceSupp:off - - - a=rtcp:5129 a=maxptime:20 a=ptime:20 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,36 m=video 5130 RTP/AVP 34 98 a=rtpmap:34 h263/90000 a=fmtp:34 QCIF=2 a=sendrecv a=rtcp:5131 a=rtpmap:98 theora/90000 a=fmtp:98 sampling=YCbCr-4:2:0; width=176, height=144; ------------------------------------------------------------------------ tport_deliver(0x8091560): msg 0x8093348 (1208 bytes) from udp/212.6.123.118:5090/sip next=(nil) nta: received INVITE sip:942799@sip0.en.ewetel.de SIP/2.0 (CSeq 1) nta: INVITE (1) 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::0x808ebf8, 0x808e920, 0x80ae9d0) called soa_set_params(static::0x80a49d0, ...) called soa_set_params(static::0x80a49d0, ...) called nta_leg_create(0x80a1198) soa_init_offer_answer(static::0x80a49d0) called soa_set_remote_sdp(static::0x80a49d0, (nil), 0x80a0b8c, 492) called nua(0x80ae9d0): adding session usage tport_tsend(0x8091560) tpn = UDP/212.6.123.118:5060 tport_resolve addrinfo = 212.6.123.118:5060 tport_by_addrinfo(0x8091560): not found by name UDP/212.6.123.118:5060 tport_vsend(0x8091560): 470 bytes of 470 to UDP/212.6.123.118:5060 tport_vsend returned 470 send 470 bytes to udp/[212.6.123.118]:5060 at 07:47:46.690076: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK1028.95646442.0 Via: SIP/2.0/UDP 212.6.95.198:5070;branch=z9hG4bKB1A69A926C26EEF0C9A11FD3C7379087;rport=5070 Record-Route: <sip:212.6.123.118;lr=on;ftag=1184830476> From: "Your Long Name" <sip:v101@sip0.en.ewetel.de>;tag=1184830476 To: <sip:942799@sip0.en.ewetel.de> Call-ID: 990023F94830E6CDBB54ACDF59BF7929@sip0.en.ewetel.de CSeq: 1 INVITE User-Agent: FreeSWITCH(mod_sofia) Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (1) nua(0x80ae9d0): event i_invite 100 Trying nua(0x80ae9d0): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x80a49d0, [0x413fedd8], [0x413fedd4], [(nil)]) called nua(0x80ae9d0): event i_state 100 Trying nua: nua_event: entering nua: nua_handle_bind: entering nua: nua_event: entering 2007-07-19 09:47:46 [NOTICE] switch_channel.c:385 switch_channel_set_name() New Chan sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 [57342754-35cc-11dc-821f-299d3d3cef0c] 2007-07-19 09:47:46 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 entering state [received] 2007-07-19 09:47:46 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Remote SDP: v=0 o=v101 1184830473 1184831265 IN IP4 212.6.95.198 s=Kapanga [1184830473] c=IN IP4 212.6.95.198 t=0 0 m=audio 5128 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,36 a=silenceSupp:off - - - a=rtcp:5129 a=maxptime:20 a=ptime:20 m=video 5130 RTP/AVP 34 98 a=rtpmap:34 h263/90000 a=fmtp:34 QCIF=2 a=rtpmap:98 theora/90000 a=fmtp:98 sampling=YCbCr-4:2:0; width=176, height=144; a=rtcp:5131 2007-07-19 09:47:46 [DEBUG] sofia_glue.c:1202 sofia_glue_negotiate_sdp() Audio Codec Compare [pcmu:0:8000]/[PCMU:0:8000] 2007-07-19 09:47:46 [DEBUG] sofia_glue.c:812 sofia_glue_tech_set_codec() Set Codec sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 pcmu/8000 20 ms 160 samples 2007-07-19 09:47:46 [DEBUG] sofia_glue.c:1166 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2007-07-19 09:47:46 [DEBUG] sofia.c:1132 sofia_handle_sip_i_state() sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 State Change CS_NEW - -> CS_INIT 2007-07-19 09:47:46 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 [BREAK] 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060) State INIT 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:92 sofia_on_init() SOFIA INIT 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:105 sofia_on_init() sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 State Change CS_INIT - -> CS_RING 2007-07-19 09:47:46 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 [BREAK] 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060) State RING 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:120 sofia_on_ring() SOFIA RING 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_ring() Standard RING sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 2007-07-19 09:47:46 [INFO] mod_dialplan_xml.c:316 dialplan_hunt() Processing Your Long Name->942799! 2007-07-19 09:47:46 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(942799) =~ /(.*)/ 2007-07-19 09:47:46 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(942799) =~ /^94(\d{4})$/ 2007-07-19 09:47:46 [DEBUG] mod_dialplan_xml.c:372 dialplan_hunt() sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 State Change CS_RING - -> CS_EXECUTE 2007-07-19 09:47:46 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 [BREAK] 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060) State EXECUTE 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:137 sofia_on_execute() SOFIA EXECUTE 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:121 switch_core_standard_on_execute() Standard EXECUTE 2007-07-19 09:47:46 [NOTICE] switch_core_state_machine.c:131 switch_core_standard_on_execute() Execute set(effective_caller_id_name=032223930016) 2007-07-19 09:47:46 [DEBUG] mod_dptools.c:344 set_function() SET [effective_caller_id_name]=[032223930016] 2007-07-19 09:47:46 [NOTICE] switch_core_state_machine.c:131 switch_core_standard_on_execute() Execute set(effective_caller_id_number=032223930016) 2007-07-19 09:47:46 [DEBUG] mod_dptools.c:344 set_function() SET [effective_caller_id_number]=[032223930016] 2007-07-19 09:47:46 [NOTICE] switch_core_state_machine.c:131 switch_core_standard_on_execute() Execute set(call_timeout=10) 2007-07-19 09:47:46 [DEBUG] mod_dptools.c:344 set_function() SET [call_timeout]=[10] 2007-07-19 09:47:46 [NOTICE] switch_core_state_machine.c:131 switch_core_standard_on_execute() Execute set(bypass_media=true) 2007-07-19 09:47:46 [DEBUG] mod_dptools.c:344 set_function() SET [bypass_media]=[true] 2007-07-19 09:47:46 [NOTICE] switch_core_state_machine.c:131 switch_core_standard_on_execute() Execute info() 2007-07-19 09:47:46 [CRIT] mod_dptools.c:448 info_function() CHANNEL_DATA: Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/sip0.en.ewetel.de/v101%40sip0.en.ewetel.de%3A5060 Unique-ID: 57342754-35cc-11dc-821f-299d3d3cef0c Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: v101 Caller-Dialplan: XML Caller-Caller-ID-Name: Your%20Long%20Name Caller-Caller-ID-Number: v101 Caller-Network-Addr: 212.6.123.118 Caller-Destination-Number: 942799 Caller-Unique-ID: 57342754-35cc-11dc-821f-299d3d3cef0c Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/sip0.en.ewetel.de/v101%40sip0.en.ewetel.de%3A5060 Caller-Channel-Created-Time: 1184831266691155 Caller-Channel-Answered-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: yes Caller-Privacy-Hide-Name: no Caller-Privacy-Hide-Number: no variable_sip_req_uri: 942799%40sip0.en.ewetel.de%3A5060 variable_call_timeout: 10 variable_sip_from_port: 5060 variable_remote_media_ip: 212.6.95.198 variable_sip_contact_uri: v101%40212.6.95.198%3A5070 variable_sip_from_user_stripped: v101 variable_sip_via_host: 212.6.123.118 variable_switch_r_sdp: v%3D0%20%20o%3Dv101%201184830473%201184831265%20IN%20IP4%20212.6.95.198%20%20s%3DKapanga%20%5B1184830473%5D%20%20c%3DIN%20IP4%20212.6.95.198%20%20t%3D0%200%20%20m%3Daudio%205128%20RTP/AVP%200%208%20101%20%20a%3Drtpmap%3A0%20pcmu/8000%20%20a%3Drtpmap%3A8%20pcma/8000%20%20a%3Drtpmap%3A101%20telephone-event/8000%20%20a%3Dfmtp%3A101%200-15,36%20%20a%3DsilenceSupp%3Aoff%20-%20-%20-%20%20a%3Drtcp%3A5129%20%20a%3Dmaxptime%3A20%20%20a%3Dptime%3A20%20%20m%3Dvideo%205130%20RTP/AVP%2034%2098%20%20a%3Drtpmap%3A34%20h263/90000%20%20a%3Dfmtp%3A34%20QCIF%3D2%20%20a%3Drtpmap%3A98%20theora/90000%20%20a%3Dfmtp%3A98%20sampling%3DYCbCr-4%3A2%3A0%3B%20width%3D176,%20height%3D144%3B%20%20a%3Drtcp%3A5131%20%20 variable_sip_to_host: sip0.en.ewetel.de variable_sip_h_P-SI: Alias variable_sip_to_user: 942799 variable_max_forwards: 69 variable_sip_from_uri: v101%40sip0.en.ewetel.de%3A5060 variable_remote_media_port: 5128 variable_effective_caller_id_name: 032223930016 variable_sip_req_host: sip0.en.ewetel.de variable_sip_req_user: 942799 variable_bypass_media: true variable_sip_to_port: 5060 variable_sip_call_id: 990023F94830E6CDBB54ACDF59BF7929%40sip0.en.ewetel.de variable_sip_contact_host: 212.6.95.198 variable_sip_contact_user: v101 variable_endpoint_disposition: RECEIVED variable_sip_req_port: 5060 variable_effective_caller_id_number: 032223930016 variable_sip_contact_port: 5070 variable_sip_from_host: sip0.en.ewetel.de variable_sip_to_uri: 942799%40sip0.en.ewetel.de%3A5060 variable_sip_from_user: v101 2007-07-19 09:47:46 [NOTICE] switch_core_state_machine.c:131 switch_core_standard_on_execute() Execute bridge(sofia/gateway/sip01.voice.ewetel.de/044180002799@sip01.voice.ewetel.de) 2007-07-19 09:47:46 [NOTICE] switch_channel.c:385 switch_channel_set_name() New Chan sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de [573582ac-35cc-11dc-821f-299d3d3cef0c] 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:1431 sofia_outgoing_channel() sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de State Change CS_NEW -> CS_INIT 2007-07-19 09:47:46 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de [BREAK] 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de) State INIT 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:92 sofia_on_init() SOFIA INIT nua: nh_create_handle: entering nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x8099fa0): signal r_invite nua(0x8099fa0): recv r_invite nua(0x8099fa0): signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x808ebf8, 0x808e920, 0x8099fa0) called soa_set_params(static::0x8099d98, ...) called soa_set_user_sdp(static::0x8099d98, (nil), 0x8099663, -1) called nua(0x8099fa0): event i_error 900 Error Setting SOA Parameters nta_leg_create(0x809a030) nua(0x8099fa0): adding session usage soa_init_offer_answer(static::0x8099d98) called soa_generate_offer(static::0x8099d98, 0) called nua(0x8099fa0): event r_invite 900 Internal NUA Error nua(0x8099fa0): call state changed: init -> init nua(0x8099fa0): event i_state 900 Internal NUA Error nua(0x8099fa0): event i_terminated 900 Internal NUA Error nua(0x8099fa0): removing session usage soa_destroy(static::0x8099d98) called nta_leg_destroy(0x809a030) nua: nua_event: entering nua: nua_event: entering nua: nua_event: entering nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua(0x8099fa0): signal r_destroy nua: nua_event: entering nua(0x8099fa0): event i_terminated dropped nua(0x8099fa0): recv r_destroy nua(0x8099fa0): signal r_destroy nta_leg_destroy((nil)) 2007-07-19 09:47:46 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_i_error] status [900][Error Setting SOA Parameters] session: sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de 2007-07-19 09:47:46 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_r_invite] status [900][Internal NUA Error] session: sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de 2007-07-19 09:47:46 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_i_state] status [900][Internal NUA Error] session: sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de 2007-07-19 09:47:46 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de entering state [terminated] 2007-07-19 09:47:46 [NOTICE] sofia.c:1317 sofia_handle_sip_i_state() Hangup sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de [CS_INIT] [NORMAL_UNSPECIFIED] 2007-07-19 09:47:46 [DEBUG] switch_channel.c:1076 switch_channel_perform_hangup() Kill sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de [KILL] 2007-07-19 09:47:46 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de [BREAK] 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:339 switch_core_session_run() (sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de) State HANGUP 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:217 sofia_on_hangup() Channel sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de hanging up, cause: NORMAL_UNSPECIFIED 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de, cause: NORMAL_UNSPECIFIED 2007-07-19 09:47:46 [DEBUG] switch_core_session.c:706 switch_core_session_thread() Session 2 (sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de) Locked, Waiting on external entities 2007-07-19 09:47:46 [DEBUG] switch_ivr_originate.c:882 switch_ivr_originate() Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] 2007-07-19 09:47:46 [INFO] mod_dptools.c:1056 audio_bridge_function() Originate Failed. Cause: NORMAL_UNSPECIFIED 2007-07-19 09:47:46 [NOTICE] mod_dptools.c:1060 audio_bridge_function() Hangup sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2007-07-19 09:47:46 [DEBUG] switch_channel.c:1076 switch_channel_perform_hangup() Kill sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 [KILL] 2007-07-19 09:47:46 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 [BREAK] 2007-07-19 09:47:46 [DEBUG] switch_core_state_machine.c:339 switch_core_session_run() (sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060) State HANGUP 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:217 sofia_on_hangup() Channel sofia/sip0.en.ewetel.de/v101@sip0.en.ewetel.de:5060 hanging up, cause: NORMAL_UNSPECIFIED 2007-07-19 09:47:46 [DEBUG] mod_sofia.c:239 sofia_on_hangup() Responding to INVITE with: 480 nua: nua_respond: entering nua(0x80ae9d0): signal r_respond nua(0x80ae9d0): recv r_respond nua(0x80ae9d0): signal r_respond 480 nua: nua_stack_set_params: entering soa_set_params(static::0x80a49d0, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x80a49d0) called tport_tsend(0x8091560) tpn = UDP/212.6.123.118:5060 tport_resolve addrinfo = 212.6.123.118:5060 tport_by_addrinfo(0x8091560): not found by name UDP/212.6.123.118:5060 tport_vsend(0x8091560): 614 bytes of 614 to UDP/212.6.123.118:5060 tport_vsend returned 614 send 614 bytes to udp/[212.6.123.118]:5060 at 07:47:46.721087: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK1028.95646442.0 Via: SIP/2.0/UDP 212.6.95.198:5070;branch=z9hG4bKB1A69A926C26EEF0C9A11FD3C7379087;rport=5070 From: "Your Long Name" <sip:v101@sip0.en.ewetel.de>;tag=1184830476 To: <sip:942799@sip0.en.ewetel.de>;tag=8eNSv0vN2pU5c Call-ID: 990023F94830E6CDBB54ACDF59BF7929@sip0.en.ewetel.de CSeq: 1 INVITE User-Agent: FreeSWITCH(mod_sofia) Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Length: 0 ------------------------------------------------------------------------ nta: sent 480 Temporarily Unavailable for INVITE (1) regards Helmut Brian West schrieb:
Try bypass_media instead of no_media
/b
On Jul 18, 2007, at 4:18 AM, Helmut Kuper wrote:
-----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGnx5B4tZeNddg3dwRAhq2AJ4mFxM0fUd5C6u6GBaPYwYCE8JRWwCbBS7y S6ZPHFgK+GeRmIbua3Rs800= =soMI -----END PGP SIGNATURE-----
participants (2)
-
Brian West
-
Helmut Kuper