I think this is fixed in latest git. I had a similar problem, and it was fixed yesterday. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Alex Crow Skickat: den 1 december 2011 14:30 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... Hi, I think Tony might have something here. I built a new FS from git of 29/11/2011 and calling between registered extensions is broken. The called phone will ring and I get ringing on the caller, but as soon and the callee goes offhook the caller gets sent to voicemail. Here are is a log snippet of such a call: 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3175 (sofia/internal/sip:1000@192.168.44.150:5060) Callstate Change RINGING -> ACTIVE 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/1001@192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [NOTICE] sofia.c:6077 Channel [sofia/internal/sip:1000@192.168.44.150:5060] has been answered 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3140 AUDIO RTP [sofia/internal/1001@192.168.44.171] 192.168.44.171 port 19398 -> 192.168.44.145 port 5034 codec: 0 ms: 20 2011-12-01 12:58:48.206623 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3404 Set 2833 dtmf send payload to 101 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3410 Set 2833 dtmf receive payload to 101 2011-12-01 12:58:48.206623 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1001@192.168.44.171: v=0 o=FreeSWITCH 1322724930 1322724931 IN IP4 192.168.44.171 s=FreeSWITCH c=IN IP4 192.168.44.171 t=0 0 m=audio 19398 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 a=sendrecv 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1001@192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3175 (sofia/internal/1001@192.168.44.171) Callstate Change RINGING -> ACTIVE 2011-12-01 12:58:48.206623 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/1001@192.168.44.171] has been answered 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1001@192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] sofia.c:5368 Channel sofia/internal/1001@192.168.44.171 entering state [completed][200] 2011-12-01 12:58:48.206623 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [sofia/internal/sip:1000@192.168.44.150:5060] 2011-12-01 12:58:48.206623 [DEBUG] switch_ivr_originate.c:3514 (sofia/internal/sip:1000@192.168.44.150:5060) State Change CS_CONSUME_MEDIA -> CS_RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1000@192.168.44.150:5060 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1000@192.168.44.150:5060) Running State Change CS_RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1000@192.168.44.150:5060) State RESET 2011-12-01 12:58:48.206623 [DEBUG] mod_sofia.c:166 sofia/internal/sip:1000@192.168.44.150:5060 SOFIA RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:93 sofia/internal/sip:1000@192.168.44.150:5060 Standard RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1000@192.168.44.150:5060) State RESET going to sleep 2011-12-01 12:58:48.226638 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2011-12-01 12:58:48.226638 [INFO] mod_dptools.c:2897 Originate Failed. Cause: ORIGINATOR_CANCEL This also happens on a call in from freetdm to an extension, as soon as the destination picks up the caller gets voicemail. Very odd. Cheers Alex On 01/12/11 09:57, obbyone wrote:
Hi,
I've just installed Freeswitch from GIT on Linux UBUNTU 8.04 LTS. Registering is ok, so every softphone or ATA are registered on the server BUT, as I try to place a call from a softphone to another, it doesn't work.
Has someone a solution to my problem,
Thanks,
Tony
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