[Asterisk-video] [Posible SPAM (Header Check)] - Re: Problems with a loopback scenario, initial negotiation fails. - Email found in subject
Klaus Darilion
klaus.mailinglists at pernau.at
Thu Jan 8 10:50:19 CST 2009
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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