March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102
by zhaoxxqq 23 Mar '09

23 Mar '09

HI, friend, I added <param name="dtmf-type" value="rfc2833" /> to my sip profile in external , like below. <gateway name="vos3000"> <param name="username" value="anything"/>--> <param name="realm" value="221.192.133.51"/> <param name="password" value="anything"/> <param name="dtmf-type" value="rfc2833" /> <param name="extension" value="87263150"/> <param name="register" value="false"/> </gateway> but. the problem is still exist. Can you help me. Zhao Xiaoqiang ------------------------------------------------------------------------------------------------------------ Hi - It looks like you're getting digits both in the RTP stream and as SIP INFO. Try adding <param name="dtmf-type" value="rfc2833" /> to the SIP profile you're using for inbound calls. --Dave > I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS > to PSTN with DID numbers. For inband I connect it to FS's demo_ivr. > When I press the key, the FS accept always DOUBLE of key. The debug > information like below. > > 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 > 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file > 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 > 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms > 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 > > > Can any friend can help me? > > Zhao Xiaoqiang > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 >

4 5

Two portaudio UA's, one FS. Possible?
by Gerry Hull 23 Mar '09

23 Mar '09

I have an app where I would like to run two portaudio user agents on the same computer (two sound cards). I want one UA to be feeding a conference, and the other as a softphone. I don't see a way to run two portaudio ua's on the same instance of FS. Is this possible. If not, OK to run two FS instances? TIA

2 1

Cepstral and RSS feeds
by HarryK 23 Mar '09

23 Mar '09

I have Cepstral working. Can someone please tell me how to go about having it read RSS feeds? I can have the dialplan direct it np. But I really dont have a clue how to point it at an RSS. Any help would be great, ddint find anything in the wiki. -- View this message in context: http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22594923.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

8 11

Problem about always Double accept DTMFs
by zhaoxxqq 22 Mar '09

22 Mar '09

I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS to PSTN with DID numbers. For inband I connect it to FS�s demo_ivr. When I press the key, the FS accept always DOUBLE of key. The debug information like below. 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 Can any friend can help me? Zhao Xiaoqiang

2 1

ptime problem with provider (Cisco hardware)
by Łukasz Czerpak 21 Mar '09

21 Mar '09

Hi, I have some troubles with provider configuration. The are warnings in logs: 2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 40 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. My provider uses SER and Cisco hardware (Cisco-SIPGateway/IOS-12.x). Moreover, above warning appears only when someone calls to FreeSWITCH (voice quality is poor) - connection from FreeSWITCH works without any warnings and has perfect quality. Other my providers works perfect. Is there any solution of this problem? regards, Lukasz

5 14

proper way to clean source tree after build?
by Jason White 21 Mar '09

21 Mar '09

I'm building under Debian. To clean the source tree after building, I've tried make clean debuild clean but what often happens is that after updating to a later svn revision and then trying to build, I get various errors during the compilation process that disappear if I make a clean checkout from the repository and run the build again. I have also found that using svn export to reproduce the tree in another directory, then building from that directory, doesn't always solve the problem. Is there a better way to clean up the source tree after updating it, to avoid these issues? I assume there are extraneous modifications or files that simply aren't being eliminated by the "clean" makefile targets.

2 1

Fifo feature request -- no caller disconnect after agent hangup
by Matt Hunter 20 Mar '09

20 Mar '09

I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the "fifo in" execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt

3 12

not hanging up
by Steven Ward 20 Mar '09

20 Mar '09

I have phones registered to a FS box, and an * box. There is a sip trunk between the two boxes. A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone while it's still ringing, this is what I get on the sip trace on FS: ... 2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/12345(a)11.2.22.45! recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: ------------------------------------------------------------------------ CANCEL sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345(a)b-pbx-sip-3.abc.xyz.net>SIP/2.0 Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport From: "Steve" <sip:54321@11.2.22.45 <sip%3A54321(a)11.2.22.45> >;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345(a)b-pbx-sip-3.abc.xyz.net> > Call-ID: 0c0614d866a62841546cbf3340224682(a)11.2.22.45 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 From: "Steve" <sip:54321@11.2.22.45 <sip%3A54321(a)11.2.22.45> >;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345(a)b-pbx-sip-3.abc.xyz.net> >;tag=c5Z8Q1e93p7KD Call-ID: 0c0614d866a62841546cbf3340224682(a)11.2.22.45 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------------------------- The effect is that the FS keeps on ringing - it doesn't detect the hangup. When I call from a FS phone (1000) to another FS phone (12345), and I hang up the calling phone while it's still ringing, this is what I get on the sip trace: ... send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: ------------------------------------------------------------------------ CANCEL sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0 Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a Max-Forwards: 69 From: "Extension 1000" <sip:1000@11.2.22.46 <sip%3A1000(a)11.2.22.46> >;tag=meK8yUgpgU2Zc To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3> Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a Contact: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3> To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a From: "Extension 1000"<sip:1000@11.2.22.46 <sip%3A1000(a)11.2.22.46> >;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a From: "Extension 1000"<sip:1000@11.2.22.46 <sip%3A1000(a)11.2.22.46> >;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ... It works just fine. Any ideas? I'm not sure where to go with this. Thanks.

2 1

Re: [Freeswitch-users] (OT) SPA-922 unlock
by Mark 20 Mar '09

20 Mar '09

I was never able to unlock these phones using some sort of sniffing/back door stuff. I was, however, lucky enough to have the previous owner discover the password. If it helps anybody, the now-defunct Webnet Global Communications (where the phones came from at a bankruptcy auction) used the username Admin and password 'webnet'. Thanks to all who provided information on this endeavor. Library Mark Quoting Gabriel Kuri <gkuri(a)ieee.org>: > On the slight chance they're not doing remote provisioning and the phone > is just simply locked with a username/password, you'll need to feed the > phone a TFTP server via DHCP Option 66 and setup a config file on that > tftp server with the name spa922.cfg. > > Contact me off list about generating a config file for the phone. > > Gabe > > Mark wrote: >> I did unplug the ethernet cable. I have never been able to make the >> IVR work on any of the Linksys phones that I have. I must be doing >> something wrong. >> >> I will try to sniff the traffic on the phone when I start it up. I >> will report back when I do. >> >> Thanks so much - >> >> Library Mark >> >> Quoting "Gabriel Kuri" <gkuri(a)ieee.org>: >> >>> I believe you need to make sure the Ethernet cable is unplugged from the >>> phone when trying to dial that string. >>> >>> Now I've never tried this, but it should theoretically be possible ... >>> >>> Sniff the traffic of the phone and see where it's attempting to pickup >>> the config file. Then setup a local network with your own DNS server, >>> and re-direct the phone (via DNS) to your own web server (assuming it's >>> picking up the config via http) and have a config file on the web server >>> with a username and password you specify to reset the config and get >>> into the phone. Let's hope they didn't setup the phone to provision via >>> https, otherwise you're really SOL >>> >>> If you need help generating a config for the phone, with Linksys' >>> special config tool, contact me offlist. >>> >>> Gabe >>> >>> Mark wrote: >>>> Sadly, ****73738# does not work. >>>> >>>> Is there a jumper on the board or some other hardware fix for this? >>>> >>>> Quoting "Gabriel Kuri" <gkuri(a)ieee.org>: >>>> >>>>> Have you tried resetting the phone via the built-in IVR menu? >>>>> >>>>> Pick up the handset and dial ****73738# >>>>> >>>>> This should reset the phone to factory defaults, assuming that company >>>>> didn't lock this feature out. >>>>> >>>>> Gabe >>>>> >>>>> >>>>> >>>>> Mark wrote: >>>>>> Hello, folks - I hope that I can reach someone who knows the answer to >>>>>> this one: >>>>>> >>>>>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>>>>> locked by Webnet global Communications. From what I can tell, this >>>>>> company went bankrupt, and the ebay seller bought the phones from a >>>>>> bankruptcy auction. He does not know the admin username or password. >>>>>> Nowhere on the linksys site is there a solution to how to unlock these >>>>>> phones. >>>>>> >>>>>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>>>>> than the password thing, they function fine. >>>>>> >>>>>> Thanks - >>>>>> >>>>>> Library Mark >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> 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 >>>>> _______________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> 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 >>> >>> _______________________________________________ >>> 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 >>> >> >> >> >> >> _______________________________________________ >> 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 > > > _______________________________________________ > 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 >

1 0

SIP registration fails when using hostname in sip_profile ?
by ludovic 20 Mar '09

20 Mar '09

2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:141 switch_core_standard_on_execute() sofia/internal/ludovic@192.168.1.1:5061 Execute db(insert/spymap/${caller_id_number}/) 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:1059 switch_core_session_execute_application() sofia/internal/ludovic@192.168.1.1:5061 Expanded String db(insert/spymap/ludovic/0) 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:141 switch_core_standard_on_execute() sofia/internal/ludovic@192.168.1.1:5061 Execute db(insert/last_dial/${caller_id_numbe) 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:1059 switch_core_session_execute_application() sofia/internal/ludovic@192.168.1.1:5061 Expanded String db(insert/last_dial/ludovi) 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:141 switch_core_standard_on_execute() sofia/internal/ludovic@192.168.1.1:5061 Execute db(insert/last_dial/global/${uuid}) 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:1059 switch_core_session_execute_application() sofia/internal/ludovic@192.168.1.1:5061 Expanded String db(insert/last_dial/global) 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:141 switch_core_standard_on_execute() sofia/internal/ludovic@192.168.1.1:5061 Execute bridge(sofia/gateway/freephonie.net/0) 2009-03-16 18:29:42 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/external/0123456789 [08ee8f46-1250-11de-9e6d-07496a8df75d] 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:2029 sofia_outgoing_channel() sofia/external/0123456789 State Change CS_NEW -> CS_INIT 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/0123456789 [BREAK] 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:366 switch_core_session_run() sofia/external/0123456789 Running State Change CS_INIT 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:416 switch_core_session_run() (sofia/external/0123456789) State INIT 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:85 sofia_on_init() sofia/external/0123456789 SOFIA INIT 2009-03-16 18:29:42 [DEBUG] sofia_glue.c:488 sofia_glue_ext_address_lookup() Stun Success [89.226.241.9]:[18992] 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:112 sofia_on_init() sofia/external/0123456789 State Change CS_INIT -> CS_ROUTING 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/0123456789 [BREAK] 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:416 switch_core_session_run() (sofia/external/0123456789) State INIT going to sleep 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:366 switch_core_session_run() sofia/external/0123456789 Running State Change CS_ROUTING 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:421 switch_core_session_run() (sofia/external/0123456789) State ROUTING 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:124 sofia_on_routing() sofia/external/0123456789 SOFIA ROUTING 2009-03-16 18:29:42 [DEBUG] switch_ivr_originate.c:57 originate_on_routing() sofia/external/0123456789 State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/0123456789 [BREAK] 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:421 switch_core_session_run() (sofia/external/0123456789) State ROUTING going to sleep 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:366 switch_core_session_run() sofia/external/0123456789 Running State Change CS_CONSUME_MEDIA 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:443 switch_core_session_run() (sofia/external/0123456789) State CONSUME_MEDIA 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/external/0123456789 2009-03-16 18:29:42 [DEBUG] sofia.c:2177 sofia_handle_sip_i_state() Channel sofia/external/0123456789 entering state [calling] 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event [nua_r_invite] status [503][DNS Error] session: sofia/external/0123456789 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event [nua_i_state] status [503][DNS Error] session: sofia/external/0123456789 2009-03-16 18:29:42 [DEBUG] sofia.c:2177 sofia_handle_sip_i_state() Channel sofia/external/0123456789 entering state [terminated] 2009-03-16 18:29:42 [NOTICE] sofia.c:2598 sofia_handle_sip_i_state() Hangup sofia/external/0123456789 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-03-16 18:29:42 [DEBUG] switch_channel.c:1368 switch_channel_perform_hangup() Kill sofia/external/0123456789 [KILL] 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/0123456789 [BREAK] 2009-03-16 18:29:42 [DEBUG] switch_ivr_originate.c:1448 switch_ivr_originate() Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-03-16 18:29:42 [INFO] mod_dptools.c:1814 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2009-03-16 18:29:42 [NOTICE] mod_dptools.c:1841 audio_bridge_function() Hangup sofia/internal/ludovic@192.168.1.1:5061 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:443 switch_core_session_run() (sofia/external/0123456789) State CONSUME_MEDIA going to sleep 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:366 switch_core_session_run() sofia/external/0123456789 Running State Change CS_HANGUP 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:394 switch_core_session_run() (sofia/external/0123456789) State HANGUP 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:269 sofia_on_hangup() Channel sofia/external/0123456789 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:47 switch_core_standard_on_hangup() Standard HANGUP sofia/external/0123456789, cause: NORMAL_TEMPORARY_FAILURE 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:394 switch_core_session_run() (sofia/external/0123456789) State HANGUP going to sleep 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:794 switch_core_session_thread() Session 22 (sofia/external/0123456789) Locked, Waiting on external entities 2009-03-16 18:29:42 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 22 (sofia/external/0123456789) Ended 2009-03-16 18:29:42 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/external/0123456789 [CS_HANGUP] 2009-03-16 18:29:42 [DEBUG] switch_channel.c:1368 switch_channel_perform_hangup() Kill sofia/internal/ludovic@192.168.1.1:5061 [KILL] 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/internal/ludovic@192.168.1.1:5061 [BREAK] 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:431 switch_core_session_run() (sofia/internal/ludovic@192.168.1.1:5061) State EXECUTE going to sleep 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:366 switch_core_session_run() sofia/internal/ludovic@192.168.1.1:5061 Running State Change CS_HANGUP 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:394 switch_core_session_run() (sofia/internal/ludovic@192.168.1.1:5061) State HANGUP 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:235 sofia_on_hangup() sofia/internal/ludovic@192.168.1.1:5061 Overriding SIP cause 503 with 503 from the other leg 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:269 sofia_on_hangup() Channel sofia/internal/ludovic@192.168.1.1:5061 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-03-16 18:29:42 [DEBUG] mod_sofia.c:343 sofia_on_hangup() Responding to INVITE with: 503 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:47 switch_core_standard_on_hangup() Standard HANGUP sofia/internal/ludovic@192.168.1.1:5061, cause: NORMAL_TEMPORARY_FAILURE 2009-03-16 18:29:42 [DEBUG] switch_core_state_machine.c:394 switch_core_session_run() (sofia/internal/ludovic@192.168.1.1:5061) State HANGUP going to sleep 2009-03-16 18:29:42 [DEBUG] switch_core_session.c:794 switch_core_session_thread() Session 21 (sofia/internal/ludovic@192.168.1.1:5061) Locked, Waiting on external entities 2009-03-16 18:29:42 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 21 (sofia/internal/ludovic@192.168.1.1:5061) Ended 2009-03-16 18:29:42 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/ludovic@192.168.1.1:5061 [CS_HANGUP] 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event [nua_i_state] status [503][Service Unavailable] session: n/a

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