[Asterisk-video] asterisk-video Digest, Vol 15, Issue 33
Walter Rodrigues Filho
walter at telebit.net.br
Mon Jul 23 13:02:48 CDT 2007
Hi,
I was wondering if somebody could tell me whether my E1 (ISDN PRI signaling
Euro ISDN) with libpri,zaptel...would be able to bridge h324 video calls to
SIP.
Thanks for the help.
At.
Walter Rodrigues Filho
Brasil
Em 23.07.07 14:00, "asterisk-video-request at lists.digium.com"
<asterisk-video-request at lists.digium.com> escreveu:
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> Today's Topics:
>
> 1. how to setup SIP-324M gateway (Arnold P. Siboro)
> 2. Re: how to setup SIP-324M gateway (Arnold P. Siboro)
> 3. Re: how to setup SIP-324M gateway (Klaus Darilion)
> 4. h324m_loopback no audio problem (QQ D)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 23 Jul 2007 14:54:08 +0900
> From: "Arnold P. Siboro" <asiboro at maltech.jp>
> Subject: [Asterisk-video] how to setup SIP-324M gateway
> To: asterisk-video at lists.digium.com
> Message-ID: <20070723144204.7DB4.ASIBORO at maltech.jp>
> Content-Type: text/plain; charset="US-ASCII"
>
>
> I got my Asterisk box running and tested with ISDN line. Furthermore,
> h324m_loopback()
> also worked perfectly. I want to setup a SIP-324M gateway, following the
> instruction on libh324m gateway, I set it as follows:
>
> [from-zaptel]
> exten => _.,1,Answer
> ;exten => s,10,h324m_gw(SIP/1002)
> exten => _X.,1,h324m_gw(s at threegvideo)
>
> [threegvideo]
> exten => s,1,Dial(SIP/1002)
>
> However, it does not work (caller keeps ringing but callee does not get
> does not response), giving the following message:
>
>
> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on
> asterisk1 (pid = 22790)
> ionerisk1*CLI>
> Verbosity is at least 3
> == Parsing '/etc/asterisk/manager.conf': Found
> == Parsing '/etc/asterisk/manager_custom.conf': Found
> == Manager 'admin' logged on from 127.0.0.1
> -- Going to extension s|1 because of immediate=yes
> -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
> -- Executing [s at from-zaptel:1] Answer("Zap/1-1", "") in new stack
> == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
> -- Executing [h at from-zaptel:1] Answer("Zap/1-1", "") in new stack
> -- Hungup 'Zap/1-1'
>
> Is it codec problem? I was kind of expecting some message if it's about
> codec. BTW, on the SIP end I am using X-Lite, which I think does not
> have the right audio codec to talk with h324m endpoint.
>
>
> Arnold P. Siboro (asiboro at maltech.jp)
>
> "Imagination is more important than knowledge."
> --Albert Einstein.
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 23 Jul 2007 15:15:29 +0900
> From: "Arnold P. Siboro" <asiboro at maltech.jp>
> Subject: Re: [Asterisk-video] how to setup SIP-324M gateway
> To: asterisk-video at lists.digium.com
> Message-ID: <20070723151323.7DC5.ASIBORO at maltech.jp>
> Content-Type: text/plain; charset="US-ASCII"
>
>
> Actually the instruction does not mention the Answer line, so I fixed
> the configuration to as follows:
>
> [from-zaptel]
> exten => s,1,h324m_gw(s at threegvideo)
>
> [threegvideo]
> exten => s,1,Dial(SIP/1002)
>
> But the caller keeps ringing while callee (1002) receives nothing, as
> follows:
>
> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on
> asterisk1 (pid = 22790)
> ionerisk1*CLI>
> Verbosity is at least 3
> -- Going to extension s|1 because of immediate=yes
> -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
> -- Executing [s at from-zaptel:1] h324m_gw("Zap/1-1", "s at threegvideo") in new
> stack
> -- Executing [s at threegvideo:1] Dial("Local/s at threegvideo-3a33,2",
> "SIP/1002") in new stack
> -- Couldn't call 1002
> == Everyone is busy/congested at this time (0:0/0/0)
> == Auto fallthrough, channel 'Local/s at threegvideo-3a33,2' status is
> 'CHANUNAVAIL'
> == Spawn extension (from-zaptel, s, 1) exited non-zero on 'Zap/1-1'
> -- Hungup 'Zap/1-1'
>
>
> Pada Mon, 23 Jul 2007 14:54:08 +0900
> si "Arnold P. Siboro" <asiboro at maltech.jp> bilang:
>
>>
>> I got my Asterisk box running and tested with ISDN line. Furthermore,
>> h324m_loopback()
>> also worked perfectly. I want to setup a SIP-324M gateway, following the
>> instruction on libh324m gateway, I set it as follows:
>>
>> [from-zaptel]
>> exten => _.,1,Answer
>> ;exten => s,10,h324m_gw(SIP/1002)
>> exten => _X.,1,h324m_gw(s at threegvideo)
>>
>> [threegvideo]
>> exten => s,1,Dial(SIP/1002)
>>
>> However, it does not work (caller keeps ringing but callee does not get
>> does not response), giving the following message:
>>
>>
>> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on
>> asterisk1 (pid = 22790)
>> ionerisk1*CLI>
>> Verbosity is at least 3
>> == Parsing '/etc/asterisk/manager.conf': Found
>> == Parsing '/etc/asterisk/manager_custom.conf': Found
>> == Manager 'admin' logged on from 127.0.0.1
>> -- Going to extension s|1 because of immediate=yes
>> -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
>> -- Executing [s at from-zaptel:1] Answer("Zap/1-1", "") in new stack
>> == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
>> -- Executing [h at from-zaptel:1] Answer("Zap/1-1", "") in new stack
>> -- Hungup 'Zap/1-1'
>>
>> Is it codec problem? I was kind of expecting some message if it's about
>> codec. BTW, on the SIP end I am using X-Lite, which I think does not
>> have the right audio codec to talk with h324m endpoint.
>>
>>
>> Arnold P. Siboro (asiboro at maltech.jp)
>>
>> "Imagination is more important than knowledge."
>> --Albert Einstein.
>>
>>
>> _______________________________________________
>> --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
>>
>
>
> Arnold P. Siboro (asiboro at maltech.jp)
>
> The opinions expressed herein are not necessarily those of my employer,
> not necessarily mine, and probably not necessary.
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 23 Jul 2007 13:54:54 +0200
> From: Klaus Darilion <klaus.mailinglists at pernau.at>
> Subject: Re: [Asterisk-video] how to setup SIP-324M gateway
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Message-ID: <46A4970E.8010202 at pernau.at>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Have you installed the AMR codec?
>
> regards
> klaus
>
> Arnold P. Siboro wrote:
>> Actually the instruction does not mention the Answer line, so I fixed
>> the configuration to as follows:
>>
>> [from-zaptel]
>> exten => s,1,h324m_gw(s at threegvideo)
>>
>> [threegvideo]
>> exten => s,1,Dial(SIP/1002)
>>
>> But the caller keeps ringing while callee (1002) receives nothing, as
>> follows:
>>
>> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on
>> asterisk1 (pid = 22790)
>> ionerisk1*CLI>
>> Verbosity is at least 3
>> -- Going to extension s|1 because of immediate=yes
>> -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
>> -- Executing [s at from-zaptel:1] h324m_gw("Zap/1-1", "s at threegvideo") in
>> new stack
>> -- Executing [s at threegvideo:1] Dial("Local/s at threegvideo-3a33,2",
>> "SIP/1002") in new stack
>> -- Couldn't call 1002
>> == Everyone is busy/congested at this time (0:0/0/0)
>> == Auto fallthrough, channel 'Local/s at threegvideo-3a33,2' status is
>> 'CHANUNAVAIL'
>> == Spawn extension (from-zaptel, s, 1) exited non-zero on 'Zap/1-1'
>> -- Hungup 'Zap/1-1'
>>
>>
>> Pada Mon, 23 Jul 2007 14:54:08 +0900
>> si "Arnold P. Siboro" <asiboro at maltech.jp> bilang:
>>
>>> I got my Asterisk box running and tested with ISDN line. Furthermore,
>>> h324m_loopback()
>>> also worked perfectly. I want to setup a SIP-324M gateway, following the
>>> instruction on libh324m gateway, I set it as follows:
>>>
>>> [from-zaptel]
>>> exten => _.,1,Answer
>>> ;exten => s,10,h324m_gw(SIP/1002)
>>> exten => _X.,1,h324m_gw(s at threegvideo)
>>>
>>> [threegvideo]
>>> exten => s,1,Dial(SIP/1002)
>>>
>>> However, it does not work (caller keeps ringing but callee does not get
>>> does not response), giving the following message:
>>>
>>>
>>> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on
>>> asterisk1 (pid = 22790)
>>> ionerisk1*CLI>
>>> Verbosity is at least 3
>>> == Parsing '/etc/asterisk/manager.conf': Found
>>> == Parsing '/etc/asterisk/manager_custom.conf': Found
>>> == Manager 'admin' logged on from 127.0.0.1
>>> -- Going to extension s|1 because of immediate=yes
>>> -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
>>> -- Executing [s at from-zaptel:1] Answer("Zap/1-1", "") in new stack
>>> == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
>>> -- Executing [h at from-zaptel:1] Answer("Zap/1-1", "") in new stack
>>> -- Hungup 'Zap/1-1'
>>>
>>> Is it codec problem? I was kind of expecting some message if it's about
>>> codec. BTW, on the SIP end I am using X-Lite, which I think does not
>>> have the right audio codec to talk with h324m endpoint.
>>>
>>>
>>> Arnold P. Siboro (asiboro at maltech.jp)
>>>
>>> "Imagination is more important than knowledge."
>>> --Albert Einstein.
>>>
>>>
>>> _______________________________________________
>>> --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
>>>
>>
>>
>> Arnold P. Siboro (asiboro at maltech.jp)
>>
>> The opinions expressed herein are not necessarily those of my employer,
>> not necessarily mine, and probably not necessary.
>>
>>
>> _______________________________________________
>> --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
>
>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 23 Jul 2007 14:34:47 +0100
> From: "QQ D" <qiaoqiaode at gmail.com>
> Subject: [Asterisk-video] h324m_loopback no audio problem
> To: asterisk-video at lists.digium.com
> Message-ID:
> <f35030be0707230634l5976df69mb81e953d43b13846 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi, all
> I had managed to make a loopback video call on a Motorola K3 phone but
> there is no audio coming with the video loopback. Is this h324m_loopback
> designed to loop video only or a bug?
>
> Anybody elso has this loopback function working with audio? Another question
> is that what is the difference between h324m_loopback and the video_loopback
> function?
>
> As I know, mobiles are using AMR as default audio codec. Is this no audio in
> h324m_loopback mode problem to do with AMR codec not officially supported in
> this project? (I guess maybe not because h324m_loopback is looping back
> content without decoding them, I'm I right here?)
>
> Thanks a lot for your time!
> Qiao
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> ------------------------------
>
> _______________________________________________
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>
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> End of asterisk-video Digest, Vol 15, Issue 33
> **********************************************
WALTER RODRIGUES FILHO
( SIP:WALTER at SIPTEL.NET.BR
* WALTER at SIPTEL.NET.BR
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