no sound with Dial() to Betamax providers

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What steps will reproduce the problem?
1. install latest asterisk 1.8, latest chan_datacard
2. setup datacard.conf like this:
[datacard0]
audio=/dev/ttyUSB1         
data=/dev/ttyUSB2  
;imei=XXX - also possible, the same result

add extension (incoming calls to gsm):
exten => +1234567890,1,Dial(SIP/BETAMAX_PROVIDER/987654321)
exten => +1234567890,n,Hangup
where +1234567890 - is GSM sim card number, BETAMAX_PROVIDER - Nonoh, Rynga, 
Voipgain etc, 987654321 - destination number

3. we cannot hear any voice! 
the voice appears in two cases:
I. exten => +1234567890,Noop()
exten => +1234567890,Answer
exten => +1234567890,Background(beep)
exten => +1234567890,n,Dial(SIP/BETAMAX_PROVIDER/987654321)
II. use non-betamax provider, for example, PCTEL works great
III. Makhutov's driver has voice! 

What is the expected output? What do you see instead?
I compared all call logs in those 3 options - they are the same!! no 
differences in logs. only IDs and IPs of course.

What version of the product are you using? On what operating system?
asterisk 1.8.2.2
Datacard from googlecode - r183
Makhutov - r184
OS Ubuntu 10.04 amd64, 2.6.35-22-generic
Please provide any additional information below.
tested on 2 totally different Hardware configurations.
the problem in russian is also posted here:
http://asteriskforum.ru/viewtopic.php?p=49170#49170

Original issue reported on code.google.com by ruslan.l...@gmail.com on 1 Feb 2011 at 10:28

[GoogleCodeExporter]

did you try asterisk 1.6.2?

in any case, did you check result of rtp debug?  one way audio or two way audio 
problems?

Original comment by pag...@gmail.com on 2 Feb 2011 at 4:38

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no, I didnt try asterisk 1.6.2 and I didnt try rtp debug. 
two way problem.
I'll try it and write down the result here.

Original comment by ruslan.l...@gmail.com on 2 Feb 2011 at 8:38

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so, the same issue with latests Asterisk 1.6.2.16.1 and latests datacard.
rtp debug shows no UDP.. just a couple of 5060 in both ways..

Original comment by ruslan.l...@gmail.com on 4 Feb 2011 at 1:09

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5060 is SIP port, RTP use other port as defined in rtp.conf

Original comment by bg_...@mail.ru on 4 Feb 2011 at 2:13

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thanks for your reply, but I know rtp port range and know the file rtp.conf.
as I have noticed, there is NO RTP (UDP) trafic in tcpdump only 5060 trafic 
(SIP), that is why I cannot hear any voice.
can you advise anything?

Original comment by ruslan.l...@gmail.com on 4 Feb 2011 at 2:33

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the channel is already answered??  can you see in CLI something like:

SIP/XXXX answered XXXXX  ??

Original comment by pag...@gmail.com on 4 Feb 2011 at 2:46

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SIP/2.0 100 Trying
...
SIP/2.0 183 Session progress

v=0
o=nonohlogin 1296082916 1296082916 IN IP4 77.72.168.13
s=SIP Call
c=IN IP4 77.72.168.13
t=0 0
m=audio 57484 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 
(nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 77.72.168.13:57484
but no sound, like there are no connections/requests to/from udp 57484..
no "Answer".
moreover, as I have already mentioned, when I do so:
exten => +1234567890,Answer
and then Dial() - I can hear voice.

Original comment by ruslan.l...@gmail.com on 4 Feb 2011 at 3:45

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your problem is that channel is not answered, so there is no voice. If you put,

exten => _X.,1,Answer()

then channel is answered and can hear voice..

Original comment by pag...@gmail.com on 4 Feb 2011 at 4:07

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yes I know this )) but I think it is a big problem.
I want to Dial() without Answer().
it works:
1. Makhutov drivers
2. with non-Betamax provider.
it work wothout any Answer()..

Original comment by ruslan.l...@gmail.com on 4 Feb 2011 at 4:10

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can you trace sip dialog using Betamax clone and PCTEL, to see differences?

Original comment by pag...@gmail.com on 5 Feb 2011 at 6:20

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have already done it. almost no differences. i'll post it to pastebin

Original comment by ruslan.l...@gmail.com on 5 Feb 2011 at 7:13

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http://pastebin.com/kirpNrp3 - PCTEL - works great.
http://pastebin.com/uZ0XHkKN - VOIPGAIN (or Nonoh, doesnt matter) - no sound.

Original comment by ruslan.l...@gmail.com on 5 Feb 2011 at 7:26

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no feedback

Original comment by pag...@gmail.com on 5 May 2011 at 10:17

• Changed state: WontFix