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 To :  Milan Rukavina <rukavinamilan@g...>
 From :  Marian Podgoreanu <marian@v...>
 Subject :  Re: [yate] Sangoma USBfxo analog card
 Date :  Mon, 31 May 2010 10:58:27 +0300
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
>