November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Knock knock knock is this thing on?
by Ken Rice 24 Nov '08

24 Nov '08

Test... Is it just me or has this thing been slow today? K

2 1

SIP read only and writable variables
by Iñaki Baz Castillo 24 Nov '08

24 Nov '08

Hi, how could I know which variables related to SIP are writable variables and which ones just read variables? For example, I'm trying to set the From domain and User-Agent in leg B by doing: <action application="bridge" data="{sip_invite_domain=${sip_from_host},sip_user_agent=CUSTOM}sofia/gateway/gw1/xxx(a)xxxxxx.xxx"/> Unfortunatelly just the From domain is set while the User-Agent remains the value in 'user-agent-string' of the Sofia profile. -- Iñaki Baz Castillo

3 6

SIP/2.0 500 Cannot Get IP Address for Media.
by Peter P GMX 24 Nov '08

24 Nov '08

I have set up a FS behind a NAT. Calling internal numbers, VM etc. works from a different network (Caller's phone is also natted). However if I call an external Gateway the call is terminated as soon as the remote party lifts the handset. And Freeswitch is delivering "SIP/2.0 500 Cannot Get IP Address for Media" to the caller and then sends a BYE. I can see that RTP packets are coming in from the external gateway from port 26726 and they are sent to the callers phone before hangup, but no RTP packets are coming from the callers phone and are send to the external gateway. But as I mentionned, calling internal number from this phone works. Also the gateway works from another FS installation. But what does this message mean in detail? Does he complain about the IP adress of the caller or about the IP of the getaway, or does he complain that he may not reach a certain port for the media? Best regards Peter Here's the part of the FS log ==================== 2008-11-21 20:49:27 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/06XX16XXXX3(a)sip.qsc.de entering state [proceeding] 2008-11-21 20:49:27 [DEBUG] switch_ivr_bridge.c:292 audio_bridge_thread() sofia/internal/1000(a)my.domain.com receive message [SWITCH_MESSAGE_INDICATE_RINGING] 2008-11-21 20:49:27 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal sofia/internal/1000(a)my.domain.com [BREAK] 2008-11-21 20:49:30 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/06XX16XXXX3(a)sip.qsc.de entering state [ready] 2008-11-21 20:49:30 [DEBUG] switch_channel.c:1657 switch_channel_perform_mark_answered() Send signal sofia/internal/1000(a)my.domain.com [BREAK] 2008-11-21 20:49:30 [NOTICE] sofia.c:2741 sofia_handle_sip_i_state() Channel [sofia/external/06XX16XXXX3(a)sip.qsc.de] has been answered 2008-11-21 20:49:30 [DEBUG] switch_channel.c:1712 switch_channel_perform_answer() sofia/internal/1000(a)my.domain.com receive message [SWITCH_MESSAGE_INDICATE_ANSWER] 2008-11-21 20:49:30 [DEBUG] mod_sofia.c:502 sofia_answer_channel() Local SDP sofia/internal/1000(a)my.domain.com: v=0 o=FreeSWITCH 1227284490 1227284492 IN IP4 my.domain.com s=FreeSWITCH c=IN IP4 my.domain.com t=0 0 m=audio 12476 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 2008-11-21 20:49:30 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal sofia/internal/1000(a)my.domain.com [BREAK] 2008-11-21 20:49:30 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)my.domain.com entering state [terminated] 2008-11-21 20:49:30 [NOTICE] sofia.c:2857 sofia_handle_sip_i_state() Hangup sofia/internal/1000(a)my.domain.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2008-11-21 20:49:30 [NOTICE] switch_ivr_bridge.c:303 audio_bridge_thread() Channel [sofia/internal/1000(a)my.domain.com] has been answered 2008-11-21 20:49:30 [DEBUG] switch_channel.c:1441 switch_channel_perform_hangup() Send signal sofia/internal/1000(a)my.domain.com [KILL] 2008-11-21 20:49:30 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal sofia/internal/1000(a)my.domain.com [BREAK] 2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:361 audio_bridge_thread() sofia/internal/1000(a)my.domain.com ending bridge by request from read function 2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/1000(a)my.domain.com] 2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/external/06XX16XXXX3(a)sip.qsc.de [BREAK] 2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:913 switch_ivr_multi_threaded_bridge() (sofia/external/06XX16XXXX3(a)sip.qsc.de) State Change CS_EXCHANGE_MEDIA -> CS_RESET 2008-11-21 20:49:30 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal sofia/external/06XX16XXXX3(a)sip.qsc.de [BREAK] 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1000(a)my.domain.com) State EXECUTE going to sleep 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (sofia/internal/1000(a)my.domain.com) Running State Change CS_HANGUP 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403 switch_core_session_run() (sofia/internal/1000(a)my.domain.com) State HANGUP 2008-11-21 20:49:30 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/1000(a)my.domain.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1000(a)my.domain.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403 switch_core_session_run() (sofia/internal/1000(a)my.domain.com) State HANGUP going to sleep 2008-11-21 20:49:30 [DEBUG] switch_core_session.c:911 switch_core_session_thread() Session 15 (sofia/internal/1000(a)my.domain.com) Locked, Waiting on external entities 2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/external/06XX16XXXX3(a)sip.qsc.de receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2008-11-21 20:49:30 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal sofia/external/06XX16XXXX3(a)sip.qsc.de [BREAK] 2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/external/06XX16XXXX3(a)sip.qsc.de] 2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/internal/1000(a)my.domain.com [BREAK] 2008-11-21 20:49:30 [NOTICE] switch_ivr_bridge.c:472 audio_bridge_on_exchange_media() Hangup sofia/external/06XX16XXXX3(a)sip.qsc.de [CS_RESET] [NORMAL_CLEARING] 2008-11-21 20:49:30 [DEBUG] switch_channel.c:1441 switch_channel_perform_hangup() Send signal sofia/external/06XX16XXXX3(a)sip.qsc.de [KILL] 2008-11-21 20:49:30 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal sofia/external/06XX16XXXX3(a)sip.qsc.de [BREAK] 2008-11-21 20:49:30 [NOTICE] switch_core_session.c:929 switch_core_session_thread() Session 15 (sofia/internal/1000(a)my.domain.com) Ended 2008-11-21 20:49:30 [NOTICE] switch_core_session.c:931 switch_core_session_thread() Close Channel sofia/internal/1000(a)my.domain.com [CS_HANGUP] 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:436 switch_core_session_run() (sofia/external/06XX16XXXX3(a)sip.qsc.de) State EXCHANGE_MEDIA going to sleep 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (sofia/external/06XX16XXXX3(a)sip.qsc.de) Running State Change CS_HANGUP 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403 switch_core_session_run() (sofia/external/06XX16XXXX3(a)sip.qsc.de) State HANGUP 2008-11-21 20:49:30 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/external/06XX16XXXX3(a)sip.qsc.de Overriding SIP cause 480 with 500 from the other leg 2008-11-21 20:49:30 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/06XX16XXXX3(a)sip.qsc.de hanging up, cause: NORMAL_CLEARING 2008-11-21 20:49:30 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/external/06XX16XXXX3(a)sip.qsc.de 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/06XX16XXXX3(a)sip.qsc.de Standard HANGUP, cause: NORMAL_CLEARING 2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403 switch_core_session_run() (sofia/external/06XX16XXXX3(a)sip.qsc.de) State HANGUP going to sleep 2008-11-21 20:49:30 [DEBUG] switch_core_session.c:911 switch_core_session_thread() Session 16 (sofia/external/06XX16XXXX3(a)sip.qsc.de) Locked, Waiting on external entities 2008-11-21 20:49:30 [NOTICE] switch_core_session.c:929 switch_core_session_thread() Session 16 (sofia/external/06XX16XXXX3(a)sip.qsc.de) Ended 2008-11-21 20:49:30 [NOTICE] switch_core_session.c:931 switch_core_session_thread() Close Channel sofia/external/06XX16XXXX3(a)sip.qsc.de [CS_HANGUP] Network-Scan on the FS machine: =========================================== U 213.148.136.2:5060 -> 10.101.0.201:5080 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 195.xxx.xxx.2xx:5080;branch=z9hG4bKZ0pFrtcctSS4m;rport=5080. Call-ID: 1820daf3-325d-122c-e4a8-001d096ac1c9. From: "Extension 1000"<sip:07141xxxxxx@sip.qsc.de;transport=udp>;tag=pevpy0DUNSQpS. To: <sip:06XX16XXXX3@sip.qsc.de>;tag=207dc7d8. CSeq: 107513988 INVITE. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Contact: <sip:213.148.136.2:5060;user=phone>. Content-Length: 0. . # U 213.148.136.2:5060 -> 10.101.0.201:5080 SIP/2.0 200 OK. Via: SIP/2.0/UDP 195.xxx.xxx.2xx:5080;branch=z9hG4bKZ0pFrtcctSS4m;rport=5080. Call-ID: 1820daf3-325d-122c-e4a8-001d096ac1c9. From: "Extension 1000"<sip:07141xxxxxx@sip.qsc.de;transport=udp>;tag=pevpy0DUNSQpS. To: <sip:06XX16XXXX3@sip.qsc.de>;tag=207dc7d8. CSeq: 107513988 INVITE. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Supported: 100rel,replaces,timer,precondition,histinfo. Contact: <sip:213.148.136.2:5060;user=phone>. Content-Length: 221. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 10118984 10118985 IN IP4 213.148.136.2. s=Sip Call. c=IN IP4 213.148.136.2. t=0 0. m=audio 26726 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. # U 10.101.0.201:5080 -> 213.148.136.2:5060 ACK sip:213.148.136.2:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 195.xxx.xxx.2xx:5080;rport;branch=z9hG4bK09F8SNXFQ2FQg. Max-Forwards: 70. From: "Extension 1000" <sip:07141xxxxxx@sip.qsc.de;transport=udp>;tag=pevpy0DUNSQpS. To: <sip:06XX16XXXX3@sip.qsc.de>;tag=207dc7d8. Call-ID: 1820daf3-325d-122c-e4a8-001d096ac1c9. CSeq: 107513988 ACK. Contact: <sip:07141xxxxxx@195.xxx.xxx.2xx:5080;transport=udp>. Proxy-Authorization: Digest username="07141xxxxxx", realm="qsc.de", nonce="492693bf3670f4927f1ad7f6310fb22b65e10c17", cnonce="GCLZVTJdEiyo5AAdCWrByQ", algorithm=MD5, uri="sip:06XX16XXXX3@sip.qsc.de", response="79c35e38c30eb7c25a92f9b90526abd5", qop=auth, nc=00000001. Content-Length: 0. . # U 10.101.0.201:5060 -> 217.XX.XX.189:2048 *SIP/2.0 500 Cannot Get IP Address for Media.* Via: SIP/2.0/UDP 217.XX.XX.189:2048;branch=z9hG4bK-kgkuy3yyoq8a;rport=2048. From: "1000" <sip:1000@my.domain.com>;tag=44eejmbuny. To: <sip:06XX16XXXX3@my.domain.com;user=phone>;tag=56v0XXNSBSHNc. Call-ID: 3c2718517d92-pifrb7rpiz4f. CSeq: 2 INVITE. Contact: <sip:mod_sofia@my.domain.com:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10438. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk. Content-Length: 0.

4 5

Some questions about FS with external users and domains
by Iñaki Baz Castillo 24 Nov '08

24 Nov '08

Hi, I'm trying to configure FS as a simple and transparent B2BUA without local users, but I have some issues: - FS always adds "Remote-Party-Id" header. This header is a deprecated draft (the standar way is "P-Asserted-Identity"). Anyway I want FS not adding this header, how could I get it? - When FS generates the INVITE in leg_b it replaces the original From domain with its own domain/IP. How could I avoid it? (It's strange since I've checked that sending a MESSAGE instead of INVITE the From domain remains the original). -- Iñaki Baz Castillo

3 5

Freeswitch hangs up after 30 s when using Record-Route
by Woody Dickson 24 Nov '08

24 Nov '08

Hi I am using Openser as the sip proxy in front of freeswitch. When using Record-Route, Freeswitch hangs of every call after 30 s. 277.32.22.33:5060 is the public ip of openser and 192.168.1.101:5800 is freeswitch's external profile port. Both openser and freeswitch are within the same box. In the console, I am getting: 2008-11-25 01:31:24 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/1000(a)61.141.158.178 entering state [terminating] freeswitch(a)localhost.localdomain> freeswitch(a)localhost.localdomain> 2008-11-25 01:31:24 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/1000(a)61.141.158.178entering state [terminated] Here is the sip trace: U 192.168.1.101:5800 -> 277.32.22.33:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.101 ;branch=z9hG4bK022f.a4e32a57.0;received=277.32.22.33. Via: SIP/2.0/UDP 192.168.1.102:12334;received=121.15.98.134 ;branch=z9hG4bK-d87543-a5439229f1204a4e-1--d87543-;rport=14392. Record-Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>. From: "1000" <sip:1000@277.32.22.33>;tag=c947d86b. To: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 2 INVITE. Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk. Session-Expires: 120;refresher=uas. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 268. . v=0. o=FreeSWITCH 6527595211019529703 806853432324137362 IN IP4 277.32.22.33. s=FreeSWITCH. c=IN IP4 277.32.22.33. t=0 0. m=audio 11046 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 277.32.22.33:5800 -> 192.168.1.101:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.101 ;branch=z9hG4bK022f.a4e32a57.0;received=277.32.22.33. Via: SIP/2.0/UDP 192.168.1.102:12334;received=121.15.98.134 ;branch=z9hG4bK-d87543-a5439229f1204a4e-1--d87543-;rport=14392. Record-Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>. From: "1000" <sip:1000@277.32.22.33>;tag=c947d86b. To: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 2 INVITE. Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk. Session-Expires: 120;refresher=uas. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 268. . v=0. o=FreeSWITCH 6527595211019529703 806853432324137362 IN IP4 277.32.22.33. s=FreeSWITCH. c=IN IP4 277.32.22.33. t=0 0. m=audio 11046 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 192.168.1.101:5800 -> 277.32.22.33:5060 BYE sip:1000@121.15.98.134:14392 SIP/2.0. Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg. Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>. Max-Forwards: 70. From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. . U 277.32.22.33:5800 -> 192.168.1.101:5060 BYE sip:1000@121.15.98.134:14392 SIP/2.0. Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg. Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>. Max-Forwards: 70. From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. . U 192.168.1.101:5800 -> 277.32.22.33:5060 BYE sip:1000@121.15.98.134:14392 SIP/2.0. Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg. Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>. Max-Forwards: 70. From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. . U 277.32.22.33:5800 -> 192.168.1.101:5060 BYE sip:1000@121.15.98.134:14392 SIP/2.0. Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg. Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>. Max-Forwards: 70. From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. . U 192.168.1.101:5060 -> 277.32.22.33:5800 SIP/2.0 200 OK. Via: SIP/2.0/UDP 277.32.22.33:5800;received=277.32.22.33;rport=5800;branch=z9hG4bK10ttgjpr9KeQg. Record-Route: <sip:192.168.1.101;lr;ftag=4Uve20t8p31Ba>. Contact: <sip:1000@121.15.98.134:14392>. To: "1000"<sip:1000@277.32.22.33>;tag=c947d86b. From: "0"<sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 0. P-hint: (3)passed thru onreply_route[1]. . U 277.32.22.33:5060 -> 192.168.1.101:5800 SIP/2.0 200 OK. Via: SIP/2.0/UDP 277.32.22.33:5800;received=277.32.22.33;rport=5800;branch=z9hG4bK10ttgjpr9KeQg. Record-Route: <sip:192.168.1.101;lr;ftag=4Uve20t8p31Ba>. Contact: <sip:1000@121.15.98.134:14392>. To: "1000"<sip:1000@277.32.22.33>;tag=c947d86b. From: "0"<sip:0@277.32.22.33>;tag=4Uve20t8p31Ba. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 0. P-hint: (3)passed thru onreply_route[1]. . Thanks in advance for any help or suggestion on resolving the problem. Woody

2 1

Using an outbound proxy
by Iñaki Baz Castillo 24 Nov '08

24 Nov '08

Hi, I want FS receiving an INVITE and generating a leg_b like this: INVITE sip:alice@domain.org SIP/2.0 but I need this INVITE being sent to a specific IP X.X.X.X (instead of resolving "domain.org" and so). This is: I need the Request URI being "sip:alice@domain.org" but FS sending the request to IP X.X.X.X, is it possible? Thanks a lot. -- Iñaki Baz Castillo

3 7

mod_perl multiple bindings
by regs＠kinetix.gr 24 Nov '08

24 Nov '08

Hi, Is there a way to declare more than one script with its binding in perl.conf.xml? Because from what I understood by reading the documentation, is that there are no different sections to define different perl scripts with bindings like for example in the xml_curl.conf.xml : <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="binding1"> ... </binding> <binding name="binding2"> ... </binding> ... </bindings> </configuration>

3 7

c compiler cant create executables
by Faisal Maqsoodi 24 Nov '08

24 Nov '08

HELP.... While trying to install freeswitch on my system. The error msg displayed is configure: 3087: error: C compiler can not create executables What should i try. faisal

4 21

Transfer/Bridge
by henkoegema 23 Nov '08

23 Nov '08

What is the difference between <action application="transfer" data="1000"/> and <action application="bridge" data=/sofia/internal/1000%${{domain}"/> ? -- View this message in context: http://www.nabble.com/Transfer-Bridge-tp20647162p20647162.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 5

openzap reliability question
by Chad Leigh - Pengar Enterprises Inc 23 Nov '08

23 Nov '08

Hi I am new to FS. I have limited Asterisk experience from a few years ago when I had a rudimentary setup working but I dropped that project and the computer system sat off and unused. I fired it up a week ago, installed FreeBSD 7.0-RELEASE and started looking for a VoIP setup that supports SIP and my analog cards through my Digium TD400 cards (one is 4 fxs and one is 1 fxs and 4 fxo though I currently have no PSTN hookup to use with the FXO). So I got the latest zaptel driver code from the FreeBSD ports (1.4. ) and installed it and then compiled FS with OpenZAP. (Much easier and more friendly than trying to compile CallWeaver). After struggling for several hours with the configuration and terms I have a simple dial plan that lets me dial out (through gafachi) as well as receive calls. I have a Panasonic 5.8ghz wireless system plugged into the first port on my digium cards. However, in the 4 or 5 tests of incoming I have done, about half of them start to ring and then get a [ERR] zap_analog.c:526 zap_analog_channel_run() READ ERROR [] error -- full call log here: 2008-11-23 01:40:21 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing unknown->18663710235@public 2008-11-23 01:40:21 [NOTICE] mod_sofia.c:1058 sofia_receive_message() Ring-Ready sofia/external/8016354650(a)sip.gafachi.com! 2008-11-23 01:40:21 [NOTICE] mod_dptools.c:373 ring_ready_function() Ring Ready sofia/external/8016354650(a)sip.gafachi.com! 2008-11-23 01:40:21 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/1/1 [97881b5d-3ab9- dd11-a567-00e08161e9af] 2008-11-23 01:40:23 [ERR] zap_analog.c:526 zap_analog_channel_run() READ ERROR [] 2008-11-23 01:40:37 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/external/8016354650(a)sip.gafachi.com [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-11-23 01:40:37 [NOTICE] switch_ivr_originate.c:1277 switch_ivr_originate() Hangup OpenZAP/1:1/1/1 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] ----- The Panasonic stops ringing and I cannot answer it but FS does not hang up until the caller hangs up. Half the time this does not happen. I was wondering what the reliability of openZAP is and where the problem might lie with these read errors. (I plan on setting up some SIP phones for main use but would like to use the analog phones for the incoming home number versus the office SIP phones for business use). Thanks Chad

1 0