[prev in list] [next in list] [prev in thread] [next in thread]
List: asterisk-video
Subject: Re: [Asterisk-video] 3G <-->SIP audio problems
From: "aster vdo" <astervdo () gmail ! com>
Date: 2008-08-25 10:48:30
Message-ID: 302c6de30808250336l75deec30j9caa2267131a7882 () mail ! gmail ! com
[Download RAW message or body]
[Attachment #2 (multipart/alternative)]
Hi Klaus,
I had done the patch from sip.fontventa.com.
and also added the line
[amr]
octet-aligned=1
in my codecs.conf file.
but that did not help me.
regards,
aster
On 8/25/08, Klaus Darilion <klaus.mailinglists@pernau.at> wrote:
>
> Have you patched Asterisk with the AMR patch from sip.fontventa.com?
>
> klaus
>
> aster vdo schrieb:
>
>> Hi,
>>
>> I am doing a video call from a 3G to SIP.
>>
>> The video works fine, but there is no audio for both the parties.
>>
>> I using Asterisk 1.4.20.1 <http://1.4.20.1>
>>
>> and i get the following warning message on asterisk
>>
>> WARNING[15840] chan_sip.c: Asked to transmit frame type 8192, while native
>> formats is 0x4 (ulaw)(4) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
>>
>> and the show channel commands results in the following output.
>>
>>
>> *CLI> core show channel Zap/1-1
>> -- General --*CLI>
>> Name: Zap/1-1
>> Type: Zap
>> UniqueID: 1219243635.23
>> Caller ID: XXXXXXXX
>> Caller ID Name: (N/A)
>> DNID Digits: XXXXXXX
>> State: Up (6)
>> Rings: 1
>> NativeFormats: 0x44 (ulaw|slin)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: No
>> 1st File Descriptor: 19
>> Frames in: 3275
>> Frames out: 2532
>> Time to Hangup: 0
>> Elapsed Time: 0h1m5s
>> Direct Bridge: <none>
>> Indirect Bridge: <none>
>> -- PBX --
>> Context: default
>> Extension: s
>> Priority: 2
>> Call Group: 0
>> Pickup Group: 0
>> Application: h324m_gw
>> Data: dial@cell_to_sip
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> ul1=65535
>> CALLEDTON=33
>> ANI2=0
>> TRANSFERCAPABILITY=DIGITAL
>>
>> CDR Variables:LI>
>> level 1: clid=XXXXXXXX
>> level 1: src=XXXXXXXX
>> level 1: dst=s
>> level 1: dcontext=default
>> level 1: channel=Zap/1-1
>> level 1: lastapp=h324m_gw
>> level 1: lastdata=dial@cell_to_sip
>> level 1: start=2008-08-20 15:47:15
>> level 1: answer=2008-08-20 15:47:30
>> level 1: end=2008-08-20 15:47:30
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1219243635.23
>>
>>
>>
>> Is there any thing i am doing wrong..
>>
>>
>> regards
>>
>> aster
>>
>
[Attachment #5 (text/html)]
Hi Klaus,<br><br> I had done the patch from <a \
href="http://sip.fontventa.com">sip.fontventa.com</a>.<br><br> and also added \
the line <br>[amr]<br>octet-aligned=1<br><br>in my codecs.conf file.<br><br>but that \
did not help me.<br> <br><br>regards,<br>aster<br><br><br><div><span \
class="gmail_quote">On 8/25/08, <b class="gmail_sendername">Klaus Darilion</b> <<a \
href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>> \
wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; \
margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; \
border-left-width: 1px; border-left-style: solid; padding-left: 1ex"> Have you \
patched Asterisk with the AMR patch from <a href="http://sip.fontventa.com" \
target="_blank" onclick="return \
top.js.OpenExtLink(window,event,this)">sip.fontventa.com</a>?<br><br> klaus<br><br> \
aster vdo schrieb:<br> <blockquote class="gmail_quote" style="margin-top: 0; \
margin-right: 0; margin-bottom: 0; margin-left: 0.80ex; border-left-color: #cccccc; \
border-left-width: 1px; border-left-style: solid; padding-left: 1ex"><span class="q"> \
Hi,<br> <br> I am doing a video call from a 3G to SIP.<br><br> The video works fine, \
but there is no audio for both the parties.<br><br></span> I using Asterisk <a \
href="http://1.4.20.1" target="_blank" onclick="return \
top.js.OpenExtLink(window,event,this)">1.4.20.1</a> <<a href="http://1.4.20.1" \
target="_blank" onclick="return \
top.js.OpenExtLink(window,event,this)">http://1.4.20.1</a>><div> <span class="e" \
id="q_11bf909e04bcb2c4_3"><br><br> and i get the following warning message on \
asterisk<br><br> WARNING[15840] chan_sip.c: Asked to transmit frame type 8192, while \
native formats is 0x4 (ulaw)(4) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)<br> \
<br> and the show channel commands results in the following output.<br><br><br> \
*CLI> core show channel Zap/1-1<br> -- General --*CLI><br> Name: Zap/1-1<br> \
Type: Zap<br> UniqueID: 1219243635.23<br> Caller ID: XXXXXXXX<br> Caller ID Name: \
(N/A)<br> DNID Digits: XXXXXXX<br> State: Up (6)<br> Rings: 1<br> NativeFormats: 0x44 \
(ulaw|slin)<br> WriteFormat: 0x4 (ulaw)<br> ReadFormat: 0x4 (ulaw)<br> \
WriteTranscode: No<br> ReadTranscode: No<br> 1st File Descriptor: 19<br> Frames in: \
3275<br> Frames out: 2532<br> Time to Hangup: 0<br> Elapsed Time: 0h1m5s<br> Direct \
Bridge: <none><br> Indirect Bridge: <none><br> -- PBX --<br> Context: \
default<br> Extension: s<br> Priority: 2<br> Call Group: 0<br> Pickup Group: 0<br> \
Application: h324m_gw<br> Data: dial@cell_to_sip<br> Blocking in: \
ast_waitfor_nandfds<br> Variables:<br> ul1=65535<br> CALLEDTON=33<br> ANI2=0<br> \
TRANSFERCAPABILITY=DIGITAL<br><br> CDR Variables:LI><br> level 1: \
clid=XXXXXXXX<br> level 1: src=XXXXXXXX<br> level 1: dst=s<br> level 1: \
dcontext=default<br> level 1: channel=Zap/1-1<br> level 1: lastapp=h324m_gw<br> level \
1: lastdata=dial@cell_to_sip<br> level 1: start=2008-08-20 15:47:15<br> level 1: \
answer=2008-08-20 15:47:30<br> level 1: end=2008-08-20 15:47:30<br> level 1: \
duration=0<br> level 1: billsec=0<br> level 1: disposition=ANSWERED<br> level 1: \
amaflags=DOCUMENTATION<br> level 1: uniqueid=1219243635.23<br> <br><br><br> Is there \
any thing i am doing wrong..<br><br><br> regards<br><br> \
aster<br></span></div></blockquote></blockquote></div><br>
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
[prev in list] [next in list] [prev in thread] [next in thread]
Configure |
About |
News |
Add a list |
Sponsored by KoreLogic