[Freeswitch-users] Freeswitch installed, ATA registered but no call are possible...

Thu Dec 1 16:37:04 MSK 2011

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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För Alex Crow
Skickat: den 1 december 2011 14:30
Till: freeswitch-users at 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 at 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 at 192.168.44.171 [BREAK]
2011-12-01 12:58:48.206623 [NOTICE] sofia.c:6077 Channel [sofia/internal/sip:1000 at 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 at 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 at 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 at 192.168.44.171 [BREAK]
2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3175
(sofia/internal/1001 at 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 at 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 at 192.168.44.171 [BREAK]
2011-12-01 12:58:48.206623 [DEBUG] sofia.c:5368 Channel
sofia/internal/1001 at 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 at 192.168.44.150:5060]
2011-12-01 12:58:48.206623 [DEBUG] switch_ivr_originate.c:3514
(sofia/internal/sip:1000 at 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 at 192.168.44.150:5060 [BREAK]
2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/sip:1000 at 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 at 192.168.44.150:5060) State RESET
2011-12-01 12:58:48.206623 [DEBUG] mod_sofia.c:166
sofia/internal/sip:1000 at 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 at 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 at 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
>
> --
> View this message in context: 
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> -registered-but-no-call-are-possible-tp7049935p7049935.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
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