[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 10:40:38 CST 2009
Hernan Rajchert schrieb:
> 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.
That is if the SETUP message does not include alaw or ulaw (as in the
case of H324M), a default value somewhere in dahdi will be used.
but for normal voice calls it should work fine
klaus
>
> 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>
>>
>>
>>
>>
>> ----------------------------------------------------------------------
>> --
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
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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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