[Asterisk-video] [Posible SPAM (Header Check)] - Re: [Posible SPAM (Header Check)] - Re: Problems with a loopback scenario, initial negotiation fails. - Email found in subject - Email found in subject
Klaus Darilion
klaus.mailinglists at pernau.at
Fri Jan 9 05:47:25 CST 2009
[Jan 8 11:56:34] DEBUG[19649] chan_dahdi.c: Enabled echo cancellation
on channel 32
This sounds very bad!!! Make sure that the echo cancellers are disabled.
Further, if it still does not work, I suggest to turn other version: I
use asterisk 1.4.21.1, zaptel-1.4.11 and libpri-1.4.5
regards
klaus
Hernan Rajchert schrieb:
> Doh!, I've replied the message yesterday but didn't noticed it awaited
> moderators queue because of its size, so I reply again but pasting my
> log in a website that hopefully never expires.
>
> The full log can be found in http://paste-it.net/public/j02ee96/
>
> --- My orig msg ---
>
>
> Well changing the code from ALAW to ULAW in the app_h324m.c gave a
> positive show channel info. Now no transcoding occurs, but I still have
> the same problem. Ive enabled core set debug 1 for the fist time (I was
> only aware of verbose) and found the following that might be
> interesting. Also notice that ive increased the debug in libh324m
> through app_h324m.c and added a few of my own.
>
> I include the output in an attachment, I hope it works :P
>
> The I've notice are
> 1)
> [Jan 8 11:56:34] DEBUG[19639] channel.c: Avoiding initial deadlock for
> channel '0x919cdf0'
>
> Maybe a problem of having the call and gw in the same asterisk?
>
> 2)
> [Jan 8 11:56:34] DEBUG[19671] app_h324m.c: H324M changed state 1 ....
> [Jan 8 11:56:34] DEBUG[19669] app_h324m.c: H324M changed state 1
>
> Both gw and call stays in the state 1, which is SETUP
>
> 3)
> [Jan 8 11:56:34] DEBUG[19648] chan_dahdi.c: Enabled echo cancellation
> on channel 1
>
> Do I need echo cancellation? Does it interfear with digital comm?
>
> 4)
>
> [Jan 8 11:56:34] DEBUG[19669] channel.c: Released clone lock on
> 'Local/201670 at from-internal-custom-c31e,1<ZOMBIE>'
> [Jan 8 11:56:34] DEBUG[19670] channel.c: Bridge stops bridging channels
> Local/201670 at from-internal-custom-c31e,2 and
> Local/201670 at from-internal-custom-c31e,1<ZOMBIE>
> [Jan 8 11:56:34] DEBUG[19670] channel.c: Hanging up zombie
> 'Local/201670 at from-internal-custom-c31e,1<ZOMBIE>'
> [Jan 8 11:56:34] DEBUG[19670] rtp.c: Channel
> 'Local/201670 at from-internal-custom-c31e,2' has no RTP, not doing
> anything [Jan 8 11:56:34] DEBUG[19670] app_dial.c: Exiting with
> DIALSTATUS=ANSWER. [Jan 8 11:56:34] DEBUG[19670] pbx.c: Spawn extension
> (from-internal-custom,201670,1) exited non-zero on
> 'Local/201670 at from-internal-custom-c31e,2'
> [Jan 8 11:56:34] DEBUG[19670] channel.c: Soft-Hanging up channel
> 'Local/201670 at from-internal-custom-c31e,2'
> [Jan 8 11:56:34] DEBUG[19670] channel.c: Hanging up channel
> 'Local/201670 at from-internal-custom-c31e,2'
> [Jan 8 11:56:34] DEBUG[19669] channel.c: Done Masquerading DAHDI/1-1
> (6) [Jan 8 11:56:53] DEBUG[19643] chan_sip.c: Auto destroying SIP
> dialog '2F47F4E203262D656005D3C9B69E158C at ats-ar.com.ar'
>
> That's not good.
>
> -----Original Message-----
> From: asterisk-video-bounces at lists.digium.com
> [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Klaus
> Darilion
> Sent: Jueves, 08 de Enero de 2009 02:50 p.m.
> To: Development discussion of video media support in Asterisk
> Subject: [Posible SPAM (Header Check)] - Re: [Asterisk-video] [Posible
> SPAM (Header Check)] - Re: Problems with a loopback scenario, initial
> negotiation fails. - Email found in subject - Email found in subject
>
>
> comment inline ...
>
> Hernan Rajchert schrieb:
>> Also here is the show channel output of one of the test with h324m.
>> You can notice that TRANSFERCAPABILITY=SPEECH and in DAHDI/1-1
>> WriteFormat and ReadFormat is alaw and I don't know why.
>>
>>
>> ======================================================================
>> ==
>> =======================================
>> =====================================From 402 calling to
>> 301670============================================
>>
> ========================================================================
>> =======================================
>> ---
>> --- core show channels
>> ---
>> Channel Location State Application(Data)
>> Local/migw at from-pstn migw at from-pstn-cpe:1 Up h324m_gw_answer()
>> Local/migw at from-pstn migw at from-pstn-cpe:1 Up (None)
>> DAHDI/32-1 670 at from-pstn-cpe:3 Up
>> h324m_gw(migw at from-pstn-cpe)
>> DAHDI/1-1 201670 at from-internal Up (None)
>> SIP/402-08ed4508 301670 at from-internal Ring
>> h324m_call(201670 at from-interna
>>
>> ---
>> --- core show channel DAHDI/32-1
>> ---
>>
>> -- General --
>> Name: DAHDI/32-1
>> Type: DAHDI
>> UniqueID: 1231431919.83
>> Caller ID: 402
>> Caller ID Name: zultys 402
>> DNID Digits: 670
>> State: Up (6)
>> Rings: 1
>> NativeFormats: 0x44 (ulaw|slin)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: No
>> 1st File Descriptor: 41
>> Frames in: 2126
>> Frames out: 2125
>> Time to Hangup: 0
>> Elapsed Time: 0h0m42s
>> Direct Bridge: <none>
>> Indirect Bridge: <none>
>> -- PBX --
>> Context: from-pstn-cpe
>> Extension: 670
>> Priority: 3
>> Call Group: 0
>> Pickup Group: 0
>> Application: h324m_gw
>> Data: migw at from-pstn-cpe
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> CALLEDTON=33
>> ANI2=0
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: clid="zultys 402" <402>
>> level 1: src=402
>> level 1: dst=670
>> level 1: dcontext=from-pstn-cpe
>> level 1: channel=DAHDI/32-1
>> level 1: lastapp=h324m_gw
>> level 1: lastdata=migw at from-pstn-cpe
>> level 1: start=2009-01-08 10:25:19
>> level 1: answer=2009-01-08 10:25:19
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231431919.83
>>
>>
>> ---
>> --- core show channel DAHDI/1-1
>> ---
>>
>> -- General --
>> Name: DAHDI/1-1
>> Type: DAHDI
>> UniqueID: 1231431919.80
>> Caller ID: 201670
>> Caller ID Name: (N/A)
>> DNID Digits: (N/A)
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x44 (ulaw|slin)
>> WriteFormat: 0x8 (alaw)
>> ReadFormat: 0x8 (alaw)
>> WriteTranscode: Yes
>> ReadTranscode: Yes
>
>
> That does not look good. As you see the write/read format is different
> to the native supported formats thus Asterisk does transcoding which of
> course breaks the digital data stream.
>
> IIRC zaptel used as default codec alaw for E1 and ulaw for T1. Thus I
> wonder why the DAHDI channel uses ulaw.
>
> Can you try an older Asterisk version which uses zaptel?
>
> You can also try to change the local channel generated during h324m_call
>
> to request ulaw instead of alaw (just grep for LAW in app_h324m.c and
> you will find the corresponding lines of code).
>
> This is all a mess - especially as the zaptel/dahdi modul include a
> transcoder too and you never know what happens in the kernel modul. It
> is just a shame that Asterisk does not support digital calls.
>
> regards
> klaus
>
>> 1st File Descriptor: 11
>> Frames in: 4265
>> Frames out: 4262
>> Time to Hangup: 0
>> Elapsed Time: 0h1m25s
>> Direct Bridge: <none>
>> Indirect Bridge: <none>
>> -- PBX --
>> Context: from-internal-custom
>> Extension: 201670
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: (N/A)
>> Data: (None)
>> Blocking in: ast_waitfor_nandfds
>> Variables: BRIDGEPEER=Local/201670 at from-internal-custom-08bc,2
>> DIALEDPEERNUMBER=1/670
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: dst=s
>> level 1: dcontext=from-pstn-net
>> level 1: channel=DAHDI/1-1
>> level 1: start=2009-01-08 10:25:19
>> level 1: answer=2009-01-08 10:25:19
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>>
>> ---
>> --- core show channel SIP/402-08ed4508
>> ---
>> -- General --
>> Name: SIP/402-08ed4508
>> Type: SIP
>> UniqueID: 1231431919.79
>> Caller ID: 402
>> Caller ID Name: zultys 402
>> DNID Digits: 301670
>> State: Ring (4)
>> Rings: 0
>> NativeFormats: 0x4 (ulaw)
>> WriteFormat: 0x2000 (amr)
>> ReadFormat: 0x2000 (amr)
>> WriteTranscode: Yes
>> ReadTranscode: Yes
>> 1st File Descriptor: 87
>> Frames in: 0
>> Frames out: 0
>> Time to Hangup: 0
>> Elapsed Time: 0h2m35s
>> Direct Bridge: <none>
>> Indirect Bridge: <none>
>> -- PBX --
>> Context: from-internal-custom
>> Extension: 301670
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: h324m_call
>> Data: 201670 at from-internal-custom
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> SIPCALLID=1666386936-36
>> SIPUSERAGENT=Zultys ZIP4x4 1.4.2
>> SIPDOMAIN=172.16.101.36
>> SIPURI=sip:402 at 172.16.101.2:5060
>>
>> CDR Variables:
>> level 1: clid="zultys 402" <402>
>> level 1: src=402
>> level 1: dst=301670
>> level 1: dcontext=from-internal-custom
>> level 1: channel=SIP/402-08ed4508
>> level 1: lastapp=h324m_call
>> level 1: lastdata=201670 at from-internal-custom
>> level 1: start=2009-01-08 10:25:19
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=NO ANSWER
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231431919.79
>>
>> -----Original Message-----
>> From: asterisk-video-bounces at lists.digium.com
>> [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Hernan
>> Rajchert
>> Sent: Jueves, 08 de Enero de 2009 02:07 p.m.
>> To: Development discussion of video media support in Asterisk
>> Subject: Re: [Asterisk-video] [Posible SPAM (Header Check)] - Re:
>> Problems with a loopback scenario, initial negotiation fails. - Email
>> found in subject
>>
>>
>> Sry for the fast answer but I have to go for a while:
>>
>>>>> Note: you should never have an audio problem because
>>>>> of different codecs as Asterisk should do the transcoding.
>> For some reason it did, I found one of your post that said that
>> outgoing pri calls was hardcoded to alaw.
>>
>> This is a log from the time it went wrong...
>>
>> The problem was in 403 calling 201402
>>
>>
>>
>>
>> ======================================================================
>> ==
>> =======================================
>> =====================================From 402 calling to
>> 201403============================================
>>
> ========================================================================
>> =======================================
>> ---
>> --- core show channels
>> ---
>>
>> Channel Location State Application(Data)
>> SIP/403-09c633d0 (None) Up AppDial((Outgoing
>> Line))
>> DAHDI/32-1 403 at from-pstn-cpe:1 Up Dial(SIP/403)
>> DAHDI/1-1 (None) Up AppDial((Outgoing
>> Line))
>> SIP/402-09c75308 201403 at from-internal Up Dial(DAHDI/1/403)
>> 4 active channels
>> 2 active calls
>>
>> ---
>> --- core show channel SIP/403-09c633d0
>> ---
>>
>> -- General --
>> Name: SIP/403-09c633d0
>> Type: SIP
>> UniqueID: 1231354355.54
>> Caller ID: 403
>> Caller ID Name: (N/A)
>> DNID Digits: (N/A)
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x80004 (ulaw|h263)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x8 (alaw)
>> WriteTranscode: No
>> ReadTranscode: Yes
>> 1st File Descriptor: 92
>> Frames in: 6903
>> Frames out: 7017
>> Time to Hangup: 0
>> Elapsed Time: 0h2m21s
>> Direct Bridge: DAHDI/32-1
>> Indirect Bridge: DAHDI/32-1
>> -- PBX --
>> Context: from-internal-custom
>> Extension:
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: AppDial
>> Data: (Outgoing Line)
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=DAHDI/32-1
>> DIALEDPEERNUMBER=403
>> SIPCALLID=40f92e08748e5a35142064c27d3dfd1b at 172.16.101.36
>>
>> CDR Variables:
>> level 1: dst=s
>> level 1: dcontext=from-internal-custom
>> level 1: channel=SIP/403-09c633d0
>> level 1: start=2009-01-07 12:52:35
>> level 1: answer=2009-01-07 12:52:39
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354355.54
>>
>> ---
>> --- core show channel DAHDI/32-1
>> ---
>> -- General --
>> Name: DAHDI/32-1
>> Type: DAHDI
>> UniqueID: 1231354355.53
>> Caller ID: 402
>> Caller ID Name: zultys 402
>> DNID Digits: 403
>> State: Up (6)
>> Rings: 1
>> NativeFormats: 0x48 (alaw|slin)
>> WriteFormat: 0x8 (alaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: Yes
>> 1st File Descriptor: 41
>> Frames in: 14753
>> Frames out: 14590
>> Time to Hangup: 0
>> Elapsed Time: 0h4m55s
>> Direct Bridge: SIP/403-09c633d0
>> Indirect Bridge: SIP/403-09c633d0
>> -- PBX --
>> Context: from-pstn-cpe
>> Extension: 403
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: Dial
>> Data: SIP/403
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=SIP/403-09c633d0
>> DIALEDPEERNUMBER=403
>> DIALEDPEERNAME=SIP/403-09c633d0
>> CALLEDTON=33
>> ANI2=0
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: clid="zultys 402" <402>
>> level 1: src=402
>> level 1: dst=403
>> level 1: dcontext=from-pstn-cpe
>> level 1: channel=DAHDI/32-1
>> level 1: dstchannel=SIP/403-09c633d0
>> level 1: lastapp=Dial
>> level 1: lastdata=SIP/403
>> level 1: start=2009-01-07 12:52:35
>> level 1: answer=2009-01-07 12:52:39
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354355.53
>>
>> ---
>> --- core show channel DAHDI/1-1
>> ---
>> -- General --
>> Name: DAHDI/1-1
>> Type: DAHDI
>> UniqueID: 1231354355.52
>> Caller ID: 201403
>> Caller ID Name: (N/A)
>> DNID Digits: (N/A)
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x48 (alaw|slin)
>> WriteFormat: 0x8 (alaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: Yes
>> 1st File Descriptor: 11
>> Frames in: 15652
>> Frames out: 15490
>> Time to Hangup: 0
>> Elapsed Time: 0h5m13s
>> Direct Bridge: SIP/402-09c75308
>> Indirect Bridge: SIP/402-09c75308
>> -- PBX --
>> Context: from-pstn-net
>> Extension:
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: AppDial
>> Data: (Outgoing Line)
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=SIP/402-09c75308
>> DIALEDPEERNUMBER=1/403
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: dst=s
>> level 1: dcontext=from-pstn-net
>> level 1: channel=DAHDI/1-1
>> level 1: start=2009-01-07 12:52:35
>> level 1: answer=2009-01-07 12:52:39
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354355.52
>>
>> ---
>> --- core show channel SIP/402-09c75308
>> ---
>> -- General --
>> Name: SIP/402-09c75308
>> Type: SIP
>> UniqueID: 1231354355.51
>> Caller ID: 402
>> Caller ID Name: zultys 402
>> DNID Digits: 201403
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x4 (ulaw)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x8 (alaw)
>> WriteTranscode: No
>> ReadTranscode: Yes
>> 1st File Descriptor: 87
>> Frames in: 16928
>> Frames out: 17063
>> Time to Hangup: 0
>> Elapsed Time: 0h5m42s
>> Direct Bridge: DAHDI/1-1
>> Indirect Bridge: DAHDI/1-1
>> -- PBX --
>> Context: from-internal-custom
>> Extension: 201403
>> Priority: 2
>> Call Group: 0
>> Pickup Group: 0
>> Application: Dial
>> Data: DAHDI/1/403
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=DAHDI/1-1
>> DIALEDPEERNUMBER=1/403
>> DIALEDPEERNAME=DAHDI/1-1
>> SIPCALLID=465185821-36
>> SIPUSERAGENT=Zultys ZIP4x4 1.4.2
>> SIPDOMAIN=172.16.101.36
>> SIPURI=sip:402 at 172.16.101.2:5060
>>
>> CDR Variables:
>> level 1: clid="zultys 402" <402>
>> level 1: src=402
>> level 1: dst=201403
>> level 1: dcontext=from-internal-custom
>> level 1: channel=SIP/402-09c75308
>> level 1: dstchannel=DAHDI/1-1
>> level 1: lastapp=Dial
>> level 1: lastdata=DAHDI/1/403
>> level 1: start=2009-01-07 12:52:35
>> level 1: answer=2009-01-07 12:52:39
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354355.51
>>
>>
>>
>>
>> ======================================================================
>> ==
>> =======================================
>> =====================================From 403 calling to
>> 201402============================================
>>
> ========================================================================
>> =======================================
>> ---
>> --- core show channels
>> ---
>> Channel Location State Application(Data)
>> SIP/402-09c633d0 (None) Up AppDial((Outgoing
>> Line))
>> DAHDI/32-1 402 at from-pstn-cpe:1 Up Dial(SIP/402)
>> DAHDI/1-1 (None) Up AppDial((Outgoing
>> Line))
>> SIP/403-09c75308 201402 at from-internal Up Dial(DAHDI/1/402)
>>
>> ---
>> --- core show channel SIP/402-09c633d0
>> ---
>> -- General --
>> Name: SIP/402-09c633d0
>> Type: SIP
>> UniqueID: 1231354728.58
>> Caller ID: 402
>> Caller ID Name: (N/A)
>> DNID Digits: (N/A)
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x8 (alaw)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x8 (alaw)
>> WriteTranscode: Yes
>> ReadTranscode: No
>> 1st File Descriptor: 92
>> Frames in: 2
>> Frames out: 4708
>> Time to Hangup: 0
>> Elapsed Time: 0h1m34s
>> Direct Bridge: DAHDI/32-1
>> Indirect Bridge: DAHDI/32-1
>> -- PBX --
>> Context: from-internal-custom
>> Extension:
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: AppDial
>> Data: (Outgoing Line)
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=DAHDI/32-1
>> DIALEDPEERNUMBER=402
>> SIPCALLID=4a7375a32439b466389fb47915b893f8 at 172.16.101.36
>>
>> CDR Variables:
>> level 1: dst=s
>> level 1: dcontext=from-internal-custom
>> level 1: channel=SIP/402-09c633d0
>> level 1: start=2009-01-07 12:58:48
>> level 1: answer=2009-01-07 12:58:50
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354728.58
>>
>> ---
>> --- core show channel DAHDI/32-1
>> ---
>> -- General --
>> Name: DAHDI/32-1
>> Type: DAHDI
>> UniqueID: 1231354728.57
>> Caller ID: 403
>> Caller ID Name: SOFT PHONE 403
>> DNID Digits: 402
>> State: Up (6)
>> Rings: 1
>> NativeFormats: 0x48 (alaw|slin)
>> WriteFormat: 0x8 (alaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: Yes
>> 1st File Descriptor: 41
>> Frames in: 5431
>> Frames out: 0
>> Time to Hangup: 0
>> Elapsed Time: 0h1m49s
>> Direct Bridge: SIP/402-09c633d0
>> Indirect Bridge: SIP/402-09c633d0
>> -- PBX --
>> Context: from-pstn-cpe
>> Extension: 402
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: Dial
>> Data: SIP/402
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=SIP/402-09c633d0
>> DIALEDPEERNUMBER=402
>> DIALEDPEERNAME=SIP/402-09c633d0
>> CALLEDTON=33
>> ANI2=0
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: clid="SOFT PHONE 403" <403>
>> level 1: src=403
>> level 1: dst=402
>> level 1: dcontext=from-pstn-cpe
>> level 1: channel=DAHDI/32-1
>> level 1: dstchannel=SIP/402-09c633d0
>> level 1: lastapp=Dial
>> level 1: lastdata=SIP/402
>> level 1: start=2009-01-07 12:58:48
>> level 1: answer=2009-01-07 12:58:50
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354728.57
>>
>> ---
>> --- core show channel DAHDI/1-1
>> ---
>> -- General --
>> Name: DAHDI/1-1
>> Type: DAHDI
>> UniqueID: 1231354728.56
>> Caller ID: 201402
>> Caller ID Name: (N/A)
>> DNID Digits: (N/A)
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x48 (alaw|slin)
>> WriteFormat: 0x8 (alaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: Yes
>> 1st File Descriptor: 11
>> Frames in: 6406
>> Frames out: 6401
>> Time to Hangup: 0
>> Elapsed Time: 0h2m8s
>> Direct Bridge: SIP/403-09c75308
>> Indirect Bridge: SIP/403-09c75308
>> -- PBX --
>> Context: from-pstn-net
>> Extension:
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: AppDial
>> Data: (Outgoing Line)
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=SIP/403-09c75308
>> DIALEDPEERNUMBER=1/402
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: dst=s
>> level 1: dcontext=from-pstn-net
>> level 1: channel=DAHDI/1-1
>> level 1: start=2009-01-07 12:58:48
>> level 1: answer=2009-01-07 12:58:50
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354728.56
>>
>> ---
>> --- core show channel SIP/403-09c75308
>> ---
>> -- General --
>> Name: SIP/403-09c75308
>> Type: SIP
>> UniqueID: 1231354728.55
>> Caller ID: 403
>> Caller ID Name: SOFT PHONE 403
>> DNID Digits: 201402
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x4 (ulaw)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x8 (alaw)
>> WriteTranscode: No
>> ReadTranscode: Yes
>> 1st File Descriptor: 87
>> Frames in: 7277
>> Frames out: 7237
>> Time to Hangup: 0
>> Elapsed Time: 0h2m25s
>> Direct Bridge: DAHDI/1-1
>> Indirect Bridge: DAHDI/1-1
>> -- PBX --
>> Context: from-internal-custom
>> Extension: 201402
>> Priority: 2
>> Call Group: 0
>> Pickup Group: 0
>> Application: Dial
>> Data: DAHDI/1/402
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> BRIDGEPEER=DAHDI/1-1
>> DIALEDPEERNUMBER=1/402
>> DIALEDPEERNAME=DAHDI/1-1
>> SIPCALLID=YmE1MjkyZmQ0MTU5NTk0MWVjNDFkN2NjM2Q2YzZmMDM.
>> SIPUSERAGENT=X-Lite 4.0 release 4.0 RC4 stamp 51016
>> SIPDOMAIN=172.16.101.36 SIPURI=sip:403 at 172.16.97.199:25484
>>
>> CDR Variables:
>> level 1: clid="SOFT PHONE 403" <403>
>> level 1: src=403
>> level 1: dst=201402
>> level 1: dcontext=from-internal-custom
>> level 1: channel=SIP/403-09c75308
>> level 1: dstchannel=DAHDI/1-1
>> level 1: lastapp=Dial
>> level 1: lastdata=DAHDI/1/402
>> level 1: start=2009-01-07 12:58:48
>> level 1: answer=2009-01-07 12:58:50
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1231354728.55
>> -----Original Message-----
>> From: asterisk-video-bounces at lists.digium.com
>> [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Klaus
>> Darilion
>> Sent: Jueves, 08 de Enero de 2009 01:53 p.m.
>> To: Development discussion of video media support in Asterisk
>> Subject: [Posible SPAM (Header Check)] - Re: [Asterisk-video] Problems
>
>> with a loopback scenario, initial negotiation fails. - Email found in
>> subject
>>
>>
>> So, if I understand it right, the PRI (E1) connection is fine as audio
>> calls are working. Note: you should never have an audio problem
> because
>> of different codecs as Asterisk should do the transcoding.
>>
>> use "pri debug span 1"
>> and span 2
>> to enable q931 debuging.
>>
>> You should have at least the Progress and Alerting messages.
>>
>> In your loopback scenario the libpri patch is not needed as you can
>> differ video from audio calls based on the extension. But if you want
> to
>> make real outgoing videocalls, you have to patch libpri to signal
>> H324M
>> to the switch of your telephony provider.
>>
>> regards
>> klaus
>>
>> Hernan Rajchert schrieb:
>>> Hi, I'm having problems establishing an h324m call. We don't have
>>> access yet to an external PRI connection so we bought a Digium TE205P
>
>>> with 2 PRI and connected them with a crossed cable to make a loopback
>
>>> (one
>> acts
>>> like a net and the other as a cpe).
>>>
>>>
>>>
>>> My idea is to make a SIP call to asterisk, forward it with h324m_call
>>> through the span 1, receive it with h324m_gw from the span 2 and then
>
>>> make a new SIP call to another phone (or use the h324m_loopback). The
>
>>> problem is that the negotiation stays in the state SETUP and thus the
>
>>> initial SIP phone stays in "calling". I've think the problem might be
>
>>> either in the configuration of the PRI loopback, in my selection of
>>> install steps or in the weird dialplan, so i include some of the
>> things
>>> I've tried to see if I'm missing something.
>>>
>>>
>>>
>>> Selection of install steps:
>>>
>>> *I've decided to use Dahdi instead of Zap (haven't seen any post on
>>> this but I don't think it should be a problem). I use the version
>>> 2.1.0.3
>>>
>>>
>>>
>>> *At first I've tried with Asterisk 1.6.0.1, but there were 3
>>> problems:
>>>
>>> 1. I had to manually relink app_h324m.so because of Makefile
>>> differences with version 1.4.X
>>>
>>> 2. Had to comment line 1560 of app_h324m.c
>>> "ast_cli_register(&cli_debug);" because it causes a segfault at
>> loading.
>>> 3. Had to modify AMR code in several places to compile.
>>>
>>>
>>>
>>> * Then I've changed to Asterisk 1.4.22 and those problems went away.
>>>
>>>
>>>
>>> * I'm using latest version from svn, revision 241. Don't know how
>>> stable
>>> this is.
>>>
>>>
>>>
>>> * I'm using libpri-1.4.8 without patching with
>>> http://bugs.digium.com/view.php?id=10217 (I think this could be a
>>> problem, but I haven't seen a tutorial that says the patch its
>> required)
>>>
>>>
>>>
>>>
>>> Configuration of the PRI loopback.
>>>
>>> I include at the bottom of the mail the configuration im currently
>>> using. In general all examples follow the same logic. With the prefix
>
>>> 201 i can make a voice call through the span 1, with the prefix 301 I
>
>>> create an h324 pseudo channel and then go through the span 1. All
>>> incoming calls from the span 2 are answered with different dialplans
>> in
>>> the context [from-pstn-cpe], some of them expect normal voice, and
>> other
>>> expects h324m. I have two SIP phones connected to asterisk, 402 is a
>>> physical and 403 is a soft phone.
>>>
>>>
>>>
>>> * If from 402 I call 2011234, the loopback works and I can hear 1, 2,
>>> 3, 4.
>>>
>>>
>>>
>>> * If from 403 I call 201402 or if from 402 I call 201403 it works
>>> two, but at some point I had audio problems between in the first
>>> scenario because of alaw and ulaw.
>>>
>>>
>>>
>>> * From any SIP phone I call 3016XX it does not work, the SIP phone
>>> stays
>>> on the state "calling" and from what I've seen debugging the app,
> both
>>> h324m_call and h324m_gw stays in state 1 (SETUP).
>>>
>>>
>>>
>>> * Strange things I've noticed in the loopback looking at core show
>>> channel/s are:
>>>
>>> * Different law handling, I couldn't hardcore chan_dahdi.c to
>>> make it always ulaw.
>>>
>>> * TRANSFERCAPABILITY=SPEECH, I don't know exactly what this is,
>>> but
>>> couldn't make both DIGITAL, just the receiving end.
>>>
>>> * Using dahdi_monitor I could dump the pri channel, but couldn't
>>> find any sequence i would recognize from the mailing list.
>>>
>>>
>>>
>>>
>>>
>>> ---------------------- Configuration files---------------------
>>>
>>> /etc/asterisk/extensions.conf
>>>
>>>
>>>
>>> [from-sip]
>>> exten => 402,1,Dial(SIP/402)
>>> exten => 403,1,Dial(SIP/403)
>>>
>>>
>>>
>>> ;;;;exten => _201.,1,Set(CHANNEL(transfercapability)=DIGITAL)
>>> ;;;;exten => _201.,n,Dial(DAHDI/1/${EXTEN:3})
>>>
>>>
>>>
>>> exten => _201.,1,Dial(DAHDI/1/${EXTEN:3})
>>>
>>>
>>>
>>> exten => _301.,1,h324m_call(201${EXTEN:3}@from-internal-custom)
>>>
>>>
>>>
>>> [from-pstn-cpe]
>>> exten => 1234,1,Answer()
>>> exten => 1234,n,SayDigits(${EXTEN})
>>>
>>>
>>>
>>> exten => 402,1,Dial(SIP/402)
>>> exten => 403,1,Dial(SIP/403)
>>>
>>> exten => 666,1,Answer()
>>> exten => 666,n,h324m_loopback(v)
>>>
>>> exten => 667,1,h324m_gw_answer()
>>> exten => 667,n,h324m_loopback()
>>>
>>> exten => 668,1,h324m_gw_answer()
>>> exten => 668,n,Playback(tt-monkeys)
>>>
>>> exten => 669,1,Set(CHANNEL(transfercapability)=DIGITAL)
>>> exten => 669,n,h324m_gw(migw at from-pstn-cpe
>>> <mailto:migw at from-pstn-cpe>)
>>>
>>>
>>>
>>> exten => 670,1,Answer()
>>> exten => 670,n,h324m_gw(migw at from-pstn-cpe
>>> <mailto:migw at from-pstn-cpe>)
>>>
>>>
>>>
>>>
>>> exten => 671,1,Answer()
>>> exten => 671,n,h324m_gw(migw2 at from-pstn-cpe
>>> <mailto:migw2 at from-pstn-cpe>)
>>>
>>>
>>>
>>> exten => migw,1,h324m_gw_answer()
>>> exten => migw,2,Echo()
>>>
>>>
>>>
>>> exten => migw2,1,h324m_gw_answer()
>>> exten => migw2,2,Dial(SIP/402)
>>>
>>>
>>>
>>> ----------------------------------------------------------
>>>
>>> /etc/dahdi/system.conf
>>>
>>>
>>>
>>> span=1,0,0,ccs,hdb3,crc4
>>> # termtype: te
>>> bchan=1-15,17-31
>>> dchan=16
>>> mulaw=1-15
>>> mulaw=17-31
>>> echocanceller=mg2,1-15,17-31
>>>
>>>
>>>
>>> # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
>>> span=2,1,0,ccs,hdb3,crc4 # termtype: te bchan=32-46,48-62
>>> dchan=47
>>> mulaw=32-46
>>> mulaw=48-62
>>> echocanceller=mg2,32-46,48-62
>>>
>>>
>>>
>>> # Global data
>>>
>>>
>>>
>>> loadzone = us
>>> defaultzone = us
>>>
>>> ----------------------------------------------------------------
>>>
>>> /etc/asterisk/chan_dahdi.conf
>>>
>>>
>>>
>>> [channels]
>>>
>>> ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=11
>>> context=from-pstn-net switchtype = euroisdn
>>> signalling = pri_net
>>> transfer=yes
>>> ;threewaycalling=yes
>>> ;cancallforward=yes
>>> facilityenable = yes
>>> channel => 1-15,17-31
>>>
>>>
>>>
>>> ; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
>>> group=12
>>> context=from-pstn-cpe
>>> switchtype = euroisdn
>>> signalling = pri_cpe
>>> transfer=yes
>>> ;threewaycalling=yes
>>> ;cancallforward=yes
>>> facilityenable = yes
>>> channel => 32-46,48-62
>>>
>>> ----------------------------------------------------------------
>>>
>>> /etc/asterisk/sip.conf
>>>
>>>
>>>
>>> [general]
>>>
>>> context=default
>>>
>>> allowoverlap=no
>>>
>>> bindport=5060
>>>
>>> bindaddr=0.0.0.0
>>>
>>> srvlookup=yes
>>>
>>>
>>>
>>> videosupport=yes
>>>
>>>
>>>
>>> disable=all
>>> allow=ulaw
>>> allow=h263
>>> allow=h263p
>>>
>>> [402]
>>> type=friend
>>> qualify=no
>>> port=5060
>>> nat=never
>>> host=dynamic
>>> dtmfmode=rfc2833
>>> context=from-internal-custom
>>> canreinvite=yes
>>> callerid="zultys 402" <402>
>>>
>>>
>>>
>>> [403]
>>> type=friend
>>> secret=403
>>> qualify=no
>>> port=5060
>>> nat=never
>>> host=dynamic
>>> dtmfmode=rfc2833
>>> context=from-internal-custom
>>> canreinvite=yes
>>> callerid="SOFT PHONE 403" <403>
>>>
>>>
>>>
>>>
>>> ---------------------------------------------------------------------
>>> -
>>> --
>>>
>>> _______________________________________________
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>>>
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>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-video
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>
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