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

[Alex Crow]

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
>
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>
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