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 To :  Marian Podgoreanu <marian@v...>
 From :  Milan Rukavina <rukavinamilan@g...>
 Subject :  Re: [yate] Sangoma USBfxo analog card
 Date :  Sat, 29 May 2010 19:58:20 +0200
Hi,

I'm still not able to make outbound call. Soon after I try to call - it
says "connected" and I hear dialtone from analog line but no actual
dialing is made. Any help or hint would be helpful. Here is console
output. I'm calling 995 - rewritten by regex to 95 - but just hear
dialtone. You can notice line where analog returned status 'answered' 
where the call is not really answered. Disregard 'asterisk' word it's
just realm property:

 Received 650 bytes SIP message from 192.168.1.2:5061
------
INVITE sip:995@asterisk SIP/2.0
Via: SIP/2.0/UDP
192.168.1.2:5061;rport;branch=z9hG4bK57F15274F410AB8CBFBAAB810E6EE6BA
From: 1001 ;tag=162244127
To: 
Contact: 
Call-ID: 0ADFB37B-86DF-983E-B4E9-EEE0D43A044D@1...
CSeq: 38311 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 231

v=0
o=1001 3844635156 3844635208 IN IP4 192.168.1.2
s=X-Lite
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
------
 Sending code 100 0x6d0c20 to 192.168.1.2:5061
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.2:5061;rport=5061;branch=z9hG4bK57F15274F410AB8CBFBAAB810E6EE6BA;received=192.168.1.2
From: 1001 ;tag=162244127
To: 
Call-ID: 0ADFB37B-86DF-983E-B4E9-EEE0D43A044D@1...
CSeq: 38311 INVITE
Server: YATE/2.2.0
Content-Length: 0

------
 Could not classify call from '1001', wasted 15563 usec
 Routing call to '995' in context 'default' via 'analog/house' in
2379 usec
 Executing call. caller=1001 called=95 line=house/1
 Outgoing call on line house/1 caller=1001 called=95
[0x6e6200]
 status=answered [0x6e6200]
 Guessed local IP '192.168.1.2' for remote '192.168.1.2'
 Session 'yrtp/2118801173' 0x6eba60 bound to
192.168.1.2:27750 +RTCP [0x6eaf10]
 DataTranslator::attachChain [0x697080] '(null)' -> [0x6a8c80]
'mulaw' not possible
 DataTranslator::attachChain [0x6d14f0] 'mulaw' -> [0x6971b0]
'(null)' not possible
 Sending code 200 0x6a3c10 to 192.168.1.2:5061
------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5061;rport=5061;branch=z9hG4bK57F15274F410AB8CBFBAAB810E6EE6BA;received=192.168.1.2
From: 1001 ;tag=162244127
To: ;tag=610486506
Call-ID: 0ADFB37B-86DF-983E-B4E9-EEE0D43A044D@1...
CSeq: 38311 INVITE
Server: YATE/2.2.0
Contact: 
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 205

v=0
o=yate 1275154956 1275154956 IN IP4 192.168.1.2
s=SIP Call
c=IN IP4 192.168.1.2
t=0 0
m=audio 27750 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
------
 Received 368 bytes SIP message from 192.168.1.2:5061
------
ACK sip:995@1...:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.2:5061;rport;branch=z9hG4bK101D604D5CAD0E8D0C67398D76A77330
From: 1001 ;tag=162244127
To: ;tag=610486506
Contact: 
Call-ID: 0ADFB37B-86DF-983E-B4E9-EEE0D43A044D@1...
CSeq: 38311 ACK
Max-Forwards: 70
Content-Length: 0

------
 RTP starting format 'mulaw' payload 0 [0x6eaf10]
 Choosing started 'audio' format 'mulaw' [0x6e8490]
 ZapCircuit(1). Buffer overrun old=640 channel=1 (11:
Resource temporarily unavailable) [0x6ace20]
 ZapCircuit(1). Buffer overrun old=640 channel=1 (11:
Resource temporarily unavailable) [0x6ace20]
 ZapCircuit(1). Buffer overrun old=640 channel=1 (11:
Resource temporarily unavailable) [0x6ace20]
 Received 402 bytes SIP message from 192.168.1.2:5061
------
BYE sip:995@1...:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.2:5061;rport;branch=z9hG4bK12D4A5734F21B3C36B3D464960A4DE45
From: 1001 ;tag=162244127
To: ;tag=610486506
Contact: 
Call-ID: 0ADFB37B-86DF-983E-B4E9-EEE0D43A044D@1...
CSeq: 38312 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1105d
Content-Length: 0

------
 Sending code 100 0x6d0c20 to 192.168.1.2:5061
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.2:5061;rport=5061;branch=z9hG4bK12D4A5734F21B3C36B3D464960A4DE45;received=192.168.1.2
From: 1001 ;tag=162244127
To: ;tag=610486506
Call-ID: 0ADFB37B-86DF-983E-B4E9-EEE0D43A044D@1...
CSeq: 38312 BYE
Server: YATE/2.2.0
Content-Length: 0

------
 YRTPWrapper::terminate() [0x6eaf10]
 status=disconnected reason=normal [0x6e6200]
 status=hangup reason=normal [0x6e6200]
 Sending code 200 0x6a3c10 to 192.168.1.2:5061
------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5061;rport=5061;branch=z9hG4bK12D4A5734F21B3C36B3D464960A4DE45;received=192.168.1.2
From: 1001 ;tag=162244127
To: ;tag=610486506
Call-ID: 0ADFB37B-86DF-983E-B4E9-EEE0D43A044D@1...
CSeq: 38312 BYE
P-RTP-Stat: PS=1116,OS=178560,PR=1116,OR=178560,PL=0
Server: YATE/2.2.0
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
 ZapCircuit(1). Consumer errors: 3. Lost: 1920/178560 [0x6ace20]
 status=destroyed reason=normal [0x6e6200]

> Hi
>
> This is an FXO card so in zapcard.conf type must be FXO not FXS.
>
> Marian
>
> Milan Rukavina wrote:
>> Hi All,
>>
>> I bought recently Sangoma USBfxo device
>> http://wiki.sangoma.com/sangoma-wanpipe-usbfxo.I succeeded to install
>> drivers /zaptel and run USBfxo with Asterisk, but my target platform is
>> Yate with zapcard/analog modules. The only thing I made to work for now
>> is to make my SIP phone ring on incoming calls, but when I pick up a
>> call nothing happens. Also, when caller hangs up - sip still rings.
>> Caller ID is not present, but it works with Asterisk.
>>
>> Also, I'm not able to make outgoing call - it's routed well to analog
>> module but it freezes in status trying.
>>
>> Is there anything particular I should look at and change in
>> configuration files:
>>
>> *zapcard.conf:*
>> [span1]
>> type=FXS
>> offset=0
>> voicechans=1
>>
>> [span2]
>> type=FXS
>> offset=1
>> voicechans=1
>>
>> *analog.conf:*
>> [house-fxs]
>> type=FXS
>> spans=span1,span2
>>
>> *regexroute.conf:*
>> ;outbound calls
>> ^9\(.*\)$=analog/house-fxs;called=
>> ;inbound calls
>> ${address}^house-fxs/\(.*\)$=;called=1001
>>
>> Thanks,
>> Milan
>