May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unable to successfully bridge calls to an "external" user
by David Robinson 18 May '09

18 May '09

Ok, my FS box is on a LAN IP (10.0.0.12), but it is setup as the DMZ machine on my modem so it can receive incoming connections without any NAT related problems. I'm trying to get a user outside on the internet to connect to my FS box and register as an internal user. He is using X-Lite on his laptop behind his own NAT. His external IP is 203.206.171.118. His registration looks like this: Call-ID: NGU1NjE4ZmIxODMwMWM1OTZlMDYzMzk0NjMyMjM1YWY. User: 1001(a)10.0.0.12 Contact: "124.254.81.250" <sip:1001@203.206.171.118:40168;rinstance=c5779e159bbe8bc7 > Agent: X-Lite release 1014k stamp 47051 Status: Registered(UDP)(unknown) EXP(2009-05-18 19:32:03) Host: kira IP: 203.206.171.118 Port: 40168 Auth-User: 1001 Auth-Realm: 124.254.81.250 I note that it's registered as plain UDP, not UDP-NAT like my own internal extensions are. The dialplan is set to route this DID (0746029001) to user 1001@$$ {domain} as follows: <extension name="Jake"> <condition field="destination_number" expression="^(0746029001)$"> <action application="bridge" data="USER/1001@$${domain}"/> </condition> </extension> When I try and make a call from my mobile (0451282630) to the DID, it says it's bridging to USER/1001(a)10.0.0.12, but when the person answers, we get no audio in either direction. It rings and answers fine, it just doesn't send any audio in either direction so I'm suspecting a bridging problem. The log file of the connection is on the web at http://pastebin.freeswitch.org/8990 The bridge line is: EXECUTE sofia/external/0451282630(a)203.161.130.132 bridge(USER/1001(a)10.0.0.12 ) But the sofia address for the connection is shown as sofia/internal/sip:1001@203.206.171.118:40168;rinstance=c5779e159bbe8bc7 Is this correct ? Am I missing something fundamental ? His user address is @10.0.0.12, but his sofia address is sip: 1001(a)203.206.171.118. Is this OK or should his user ID be at his actual ip address ? This seems normal to me as I believe the 10.0.0.12 address is the "domain" of the FS box. Is it OK that he's in the same domain as my own users on my LAN or am I supposed to configure a different domain for him because he's "outside". I thought maybe it was a double-NAT problem, but the log doesn't show any fs_nat=yes entries so I assume it's not trying to NAT him (as it shouldn't). The situation is an external mobile rings my DID, so the call comes in from my provider's address, hits my FS box, which successfully sends at least the ringing information out to his softphone at his external IP, but then when it bridges, it seems not to send the audio to the right place. I'm terrible with FS log files so I have no idea whether any of the entries are wrong. What's likely to be my issue here ? Is it NAT- related, or routing related ? Any suggestions appreciated. David

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Freeswitch Drops Call after 30 Seconds
by adamF 18 May '09

18 May '09

I am having an issue where freeswitch drops an incoming call after being connected for 30 seconds. Here is my console output starting from the termination sequence. Any help would be greatly appreciated. 2009-05-17 10:59:41 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/2062842721(a)64.24.35.74 entering state [terminating] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:355 audio_bridge_thread() sofia/external/2062842721(a)64.24.35.74 ending bridge by request from write function 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/external/2062842721(a)64.24.35.74 [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:471 audio_bridge_on_exchange_media() Hangup sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:467 switch_core_session_run() (sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) State EXCHANGE_MEDIA going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) Running State Change CS_HANGUP 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) State HANGUP 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes hanging up, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes Standard HANGUP, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) State HANGUP going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 4 (sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) Locked, Waiting on external entities 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/external/2062842721(a)64.24.35.74 receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/external/2062842721(a)64.24.35.74 [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/external/2062842721(a)64.24.35.74] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:960 switch_ivr_multi_threaded_bridge() Hangup sofia/external/2062842721(a)64.24.35.74 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/external/2062842721(a)64.24.35.74 [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/external/2062842721(a)64.24.35.74 [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:464 switch_core_session_run() (sofia/external/2062842721(a)64.24.35.74) State EXECUTE going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/external/2062842721(a)64.24.35.74) Running State Change CS_HANGUP 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/2062842721(a)64.24.35.74) State HANGUP 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/2062842721(a)64.24.35.74 hanging up, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/2062842721(a)64.24.35.74 Standard HANGUP, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/2062842721(a)64.24.35.74) State HANGUP going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 3 (sofia/external/2062842721(a)64.24.35.74) Locked, Waiting on external entities 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 3 (sofia/external/2062842721(a)64.24.35.74) Ended 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/external/2062842721(a)64.24.35.74 [CS_HANGUP] 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 4 (sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) Ended 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/sip:1000@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [CS_HANGUP] -- View this message in context: http://www.nabble.com/Freeswitch-Drops-Call-after-30-Seconds-tp23586016p235… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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SIP dump to DB
by kokoska rokoska 18 May '09

18 May '09

Hi all, I'm facing the problem I need all SIP messages "going thru" FreeSWITCH (I know FS i B2BUA - so, better to say just "all SIP messages") logged somewhere and this log have to be "searchable" (by call-id etc) and I should be able to simply delete "old" messages... And more over - it should be done on not trivial SIP messages amount - say hundreds messages per second. My questions are: 1. Do you have any suggestion how to do it with FreeSWITCH? 2. Or it is not possible now, and "bounty" is necessary? :-) 3. How hard it will be to implement? ---------- FYI: I think something like SER/Kamailio/OpenSIPS siptrace is what I'am (probably) looking for: http://www.kamailio.org/docs/modules/1.4.x/siptrace.html It could go (based on my tests) up to 7-10.000 messages per second till MySQL "dies"... ---------- Thanks for your time :-) Best regards, kokoska.rokoska

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Running FreeSwitch in the background
by Adam Falcone 17 May '09

17 May '09

after starting freeswitch I am able to receive incoming calls but if I let freeswitch run in the background it stops picking up incoming calls. The only way I am able to get freeswitch to recognize incoming calls again is by making an outbound call and it then afterwards receives inbound calls. Not sure what if any log information would be beneficial to post here. I am new to freeswitch.

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Call For Participants: Lightning Talks at ClueCon 2009
by Michael Collins 17 May '09

17 May '09

*ClueCon 2009 is coming soon!* We are interested in your thoughts on subjects for lighting talks. We would love to have a number of 5-10 minute presentations by members of the community. If you would like to give a talk, or just have an idea for a talk, please let us know. How do lightning talks work? Quite simply, the presenter has just a few minutes to speak on a particular subject, usually no more than 10 minutes. He or she will deliver the information rapidly, which means keeping the presentation focused tightly on the subject being discussed. Lightning talks usually do not have enough time for audience Q&A. However, ClueCon has a long lunch period that is designed to allow attendees plenty of time to interact. Those are perfect times to discuss lightning talks or any other presentations. Those who give presentations enjoy interacting with other attendees in a relaxed atmosphere during lunch or in the evening at dinner. If you haven't already registered for ClueCon 2009 then please call us at 877.742.CLUE right away and we will complete your registration. Also, don't forget that expedia.com has some nice hotel deals for the Wyndham Chicago. Book your room today! We look forward to hearing from you and seeing you all at ClueCon in Chicago. -Michael http://www.cluecon.com 877.742.CLUE

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freeswitch and invites without sdp in initial invite
by Uwe Kastens 17 May '09

17 May '09

Hi there, Freeswitch looks really interesting. I am trying to connect a softswitch wich does some strange things. If a call arrives from pots the softswitch won't send sdp information in the initial invite. Is this something I can change in the sip-profile with freeswitch? BR Uwe

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Dial-string to a registerd gatway
by Nameer Kazzaz 17 May '09

17 May '09

Hi there, I'm using a quintum gateway for 16 pots phones, the gateway registers with freeswitch, calls into freeswitch work but, how do i dial into the gateway from freeswitch dialplan. My dialplan <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> Can I get some help please ??

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pocketsphinx and event socket
by Peter P GMX 16 May '09

16 May '09

Hello, I have tried the pizza demo and did it get to work so far. However I would like to use pocketsphinx through event socket. I saw in the wiki that there is a chapter for Speech Synthesis Commands: http://wiki.freeswitch.org/wiki/Mod_commands#Speech_Synthesis_Commands However this is empty. Also http://wiki.freeswitch.org/wiki/ASR didn't give me a hint. As I am not a Java programmer, it's hard for me to determine how the pizza demo actually works. Anybody has a sample how he did it e.g. in Php/Perl or so? (I am working with Ruby) Or back to the basics: Is it possible to use pocketsphinx through event socket? Best regards Peter

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Re: [Freeswitch-users] FS loses registration (Double Re-Register problem)
by Diego Toro 16 May '09

16 May '09

Greetings, I see Wireshark trace a double re-REGISTER in minus of the 1 second between one and other REGISTER, then siemens pbx return "Status: 100 Trying (0 bindings)" and then FS loses registration. Wireshark trace: http://pastebin.freeswitch.org/8978 170 11:40:17.98 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 171 11:40:18.00 192.168.0.120 192.168.0.125 SIP Status: 100 Trying (0 bindings) 172 11:40:18.02 192.168.0.120 192.168.0.125 SIP Status: 200 OK (1 bindings) 365 11:42:15.69 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 366 11:42:15.69 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 367 11:42:15.72 192.168.0.120 192.168.0.125 SIP Status: 100 Trying (0 bindings) 368 11:42:15.72 192.168.0.120 192.168.0.125 SIP Status: 200 OK (1 bindings) 369 11:42:19.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 370 11:42:19.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying (0 bindings) 399 11:42:23.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 400 11:42:23.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying (0 bindings) 406 11:42:27.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 407 11:42:27.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying (0 bindings) 408 11:42:31.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 409 11:42:31.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying (0 bindings) 415 11:42:35.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 416 11:42:35.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying (0 bindings) 417 11:42:35.35 192.168.0.120 192.168.0.125 SIP Status: 408 Request Timeout (0 bindings) FreeSWITCH Version 1.0.4pre6 (13176) FS IP: 192.168.0.125 siemes 3500 IP: 192.168.0.120 I would appreciate any suggestions. Thank you Diego /09, Diego Toro <dftoro(a)yahoo.com> wrote: From: Diego Toro <dftoro(a)yahoo.com> Subject: Re: [Freeswitch-users] FS loses registration To: freeswitch-users(a)lists.freeswitch.org Date: Friday, May 15, 2009, 7:58 AM I have been changing gateway parameters but i don't know what is the problem. any suggestion ? Diego --- On Thu, 5/14/09, Brian West <brian(a)freeswitch.org> wrote: From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] FS loses registration To: freeswitch-users(a)lists.freeswitch.org Date: Thursday, May 14, 2009, 6:49 PM Seem to have left out a few things... /b On May 14, 2009, at 6:39 PM, Diego Toro wrote: I have a trouble, when FS start it register whit Siemens pbx, but after FS loses registration. I have a x-lite registered whit Siemens pbx and x-lite keeps register. I add whireshark trace thank you Diego Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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JavaScript session conference
by Baskar 16 May '09

16 May '09

Hi, In JavaScript session i have one question: Step1: I have written one small JavaScript program first dial the one mobile number and one extension example: 9841798874====>1001 Step2: In that same JavaScript itself i want to transfer both the mobile number and extension into conference room 3001 Call Flow JavaScript session == (Mobile Number (9841799874) ==calls ==Extenson(1001)==calls ==Conference Room(3001)) *Javascript:* session = new Session(); session.originate(session, "sofia/default/sip:9841799874@192.168.1.135:5066 "); session.execute("bridge", "sofia/internal/1001(a)192.168.1.77"); session.execute("transfer", 9841799874); How can i use this ??? Session.execute("conference", conf_nr + "@default+" + memberflags); I want to transfer both the Extension and mobile Number in conference room in JavaScript session. How can i add it can any one assist me to solve this problem. Thanks in advance. -- Warm Regards, N.Baskar

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