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 To :  Milan Rukavina <rukavinamilan@g...>
 From :  Marian Podgoreanu <marian@v...>
 Subject :  Re: [yate] Sangoma USBfxo analog card
 Date :  Wed, 02 Jun 2010 11:40:13 +0300
Hi,

Please enable the message sniffer (yate.conf, msgsniff=on) to see all the 
messages sent by the analog channel.
Let the call ringing, don't hangup, for at least twice the delay dial value 
(defaults to 2s).

For tone detector: it seems the device does not support it.
You can disable tone detection in zapcard.conf (yate will attach its own 
detector on incoming calls).
[house]
dtmfdetect=disable

Marian

Milan Rukavina wrote:
> Hi Marian,
> 
> When I set on answer-on-polarity I can make outgoing call - other party
> rings, but when it picks up a call, yate doesn't make it answered, so
> nothing happens. So as you explained seems like line polarity changes
> are not supported.
> 
> I tried to tweak delaydial but without success. If it's grater than 0
> calls becomes answered and I can hear dialtone, if it's 0 call is
> answered as well but no dialtone. In any case seems like no actual
> dialing is done. I tried to change this parameter in analog and in route
> in regexroute but no dialing is done.
> 
> One more thing I noticed - on startup there's note:
> 
>  ZapCircuit driver=zaptel type=FXO channel=1 cic=1
> dtmfdetect=true echotaps=0 echotrain=400 buflen=255 readonly=false
> idlevalue=255 priority=normal [0x6ad150]
>  ZapSpan('') driver=zaptel section=span1 type=FXO channels=1
> circuits=1 [0x6ac670]
>  ZapCircuit driver=zaptel type=FXO channel=2 cic=2
> dtmfdetect=true echotaps=0 echotrain=400 buflen=255 readonly=false
> idlevalue=255 priority=normal [0x6ad6a0]
>  ZapSpan('') driver=zaptel section=span2 type=FXO channels=2
> circuits=2 [0x6ad540]
> * ZapCircuit(1). IOCTL(SetToneDetect) failed on channel 1
> (param=3). 38: Function not implemented [0x6ac100]
>  ZapCircuit(1). IOCTL(SetToneDetect) failed on channel 1
> (param=3). 38: Function not implemented [0x6ac100]
>  ZapCircuit(2). IOCTL(SetToneDetect) failed on channel 2
> (param=3). 38: Function not implemented [0x6ac100]
>  ZapCircuit(2). IOCTL(SetToneDetect) failed on channel 2
> (param=3). 38: Function not implemented [0x6ac100]*
> 
> Does it have anything to do with this dialing problem?
> 
> Thanks,
> Milan
>> Hi,
>>
>> Try to set
>> delaydial=0
>> in the line's group in analog.conf.
>> This will send the called number just after the off hook notification.
>>
>> You should also test if your provider and device support line polarity
>> changes.
>> Set
>> answer-on-polarity=yes
>> hangup-on-polarity=yes
>> in the line's group in analog.conf.
>> If line polarity changes are not supported by your provider or device
>> there is no way to detect when an outgoing call is answered so it will
>> be declared answered as soon as the number is sent.
>>
>> Marian
>>
>> Milan Rukavina wrote:
>>> 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
> 
>