[Asterisk-video] [Posible SPAM (Header Check)] - Re: Problems with a loopback scenario, initial negotiation fails. - Email found in subject
Hernan Rajchert
hrajchert at ats-connection.com
Thu Jan 8 10:07:17 CST 2009
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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