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 To :  Marian Podgoreanu <marian@v...>
 From :  Milan Rukavina <rukavinamilan@g...>
 Subject :  Re: [yate] Sangoma USBfxo analog card
 Date :  Fri, 04 Jun 2010 20:41:11 +0200
Hi Marian Podgoreanu,

Seems like dtmfinband=enable is exactly what I needed. Still, I have one
more problem :(

It turns out when I set format=alaw in zapcard span configuration -
inbound calls don't work, no message is received. When I comment out
this then inbound calls work, but not outbound - I hear dialing but I
suppose dial tone is mulaw encoded so it doesn't work, which makes
sense, but why inbound calls doesn't work.

This is my zaptel conf:
loadzone=de
defaultzone=de
#Sangoma USB U100  [bus:5-1 span:1] 
fxsks=1
fxsks=2

and sangoma wanpipe1.conf
[devices]
wanpipe1 = WAN_USB_ANALOG, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE       = USB
AUTO_DETECT     = NO
USB_BUSID       = 5-1
FE_MEDIA        = FXO/FXS
TDMV_LAW        = ALAW
TDMV_OPERMODE   = TBR21
RM_BATTTHRESH   = 3
RM_BATTDEBOUNCE = 16
TDMV_SPAN       = 1

[w1g1]
ACTIVE_CH       = ALL
TDMV_HWEC       = YES


Thanks,
Milan
> Hi,
>
> When call succeeded I noticed a 14 seconds interval between
> chan.startup and call.progress messages (this one is sent when a dial
> complete notification is received from driver).
> It took about 12 seconds to the driver to dial the number!
>
> So, wait for about 20 seconds to be sure the number wasn't sent.
>
> If still nothing or you are not ok with this interval try the followig:
>
> Let delaydial to its default value.
> Set in line's section
> [house]
> dtmfinband=enable
> This will tell the analog channel to use yate tone generator when
> dialing the number.
>
> Marian
>
> Milan Rukavina wrote:
>> Hi Marian Podgoreanu,
>>
>> Here are attached logs with answer-on-polarity yes/no and delay
>> default/0.
>>
>> Is there anything more I could configure?
>>
>> Thanks,
>> Milan
>>> 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
>>>>
>