Re: [Asterisk-video] hot to know a mobile device supports video call when trying to do outbound 3g video call
I have thought of that, but normally people when answering a
videocall they dont say hello, they expect to see a video and then
start talkng. Hence maybe AMD can help us understand if the call has
been answered by voicemail (not really sure if it is possible, did
not try it) but it will not help start playing the video to a
capable device as it will expect some word, and then a pause. What
do we do about that?
thank you for your reply
Regards
Dinos
On 13/9/2011 4:22 μμ, Robson Felix wrote:
when transferring to an operator service (voice mail) you can perform
an AMD to detect that.
Enviado do meu iPhone
On 13/09/2011, at 10:13, Konstantinos Liadakis wrote:
Again this thread is dead although i feel a proper reply is not given yet.
In most cases, a call will hangup if it is sent to a non video capable device.
There are 2 cases where mobile device is not video capable, and has got voicemail enabled, and is also in an area with no 3g coverage, or device is not video capable, has got voicemail enabled and is switched off.
>From my experience and tests, i have found that the call will be answered in the same way as a video capable device would do, and the dialplan proceeds exactly as if the video is played on the recipient. In this case the sender would think that the video was correctly displayed on recipients device, but in fact it isn't.
Sergio's earlier reply to this thread was:"Quick answer, the only way to know if someone supports 3G video call is making a call." This is true for most cases except from the ones describe above.
Also Borja from i6net said: "If you tag the video correctly (signaling marks), the non capable phone shoudn't ring and not be able get the call (the call is rejected or transferred to an operator service)." How do we handle this if the call is diverted to an operator service,in my case cellphone voicemail.
If there was a way to know that call has been diverted to voicemail, then it could hangup, and not leave the disturbing buzzing voicemail.
No matter what i have tried, i cannot distinguish a successful video transmission over 3g from an unsuccessful divert to voicemail when a device is not video capable.
I hope that someone can help on this, as i cannot think of anything else to try
Best regards
Dinos
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Re: [Asterisk-video] hot to know a mobile device supports video call when trying to do outbound 3g video call
Again this thread is dead although i feel a proper reply is not given yet. In most cases, a call will hangup if it is sent to a non video capable device. There are 2 cases where mobile device is not video capable, and has got voicemail enabled, and is also in an area with no 3g coverage, or device is not video capable, has got voicemail enabled and is switched off. From my experience and tests, i have found that the call will be answered in the same way as a video capable device would do, and the dialplan proceeds exactly as if the video is played on the recipient. In this case the sender would think that the video was correctly displayed on recipients device, but in fact it isn't. Sergio's earlier reply to this thread was:"Quick answer, the only way to know if someone supports 3G video call is making a call." This is true for most cases except from the ones describe above. Also Borja from i6net said: "If you tag the video correctly (signaling marks), the non capable phone shoudn't ring and not be able get the call (the call is rejected or transferred to an operator service)." How do we handle this if the call is diverted to an operator service,in my case cellphone voicemail. If there was a way to know that call has been diverted to voicemail, then it could hangup, and not leave the disturbing buzzing voicemail. No matter what i have tried, i cannot distinguish a successful video transmission over 3g from an unsuccessful divert to voicemail when a device is not video capable. I hope that someone can help on this, as i cannot think of anything else to try Best regards Dinos -- _ -- 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
Re: [Asterisk-video] Generate outbound 3g call will .call file and local channel
Hello again, After some more reading regarding voicemail problem, i realized that this problem is also faced in voice calls when someone wants to ring his desk phone and mobile phone at the same time when receiving a call. It appears that "followme" requires the recipient to press 1 in order to accept the call. Voicemails cannot press 1 hence it is known when a call goes to voicemail. It seems that there is no solution to this problem unless someone else has another idea Thank you very much for your valuable help Regards Dinos -- _ -- 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
Re: [Asterisk-video] Generate outbound 3g call will .call file and local channel
Hello,
I finally managed to make it. Jack was right, we needed to use h324m_call
This is how i did it
callfile
Channel: Local/XX@from-dialer1/n
CallerId: 2111811499
Context: video-campaign1
Extension: play
Priority: 1
Set: CHANNEL(transfercapability)=VIDEO
Set: CHANNEL(userinformationlayer1)=38
Set: phonenumber=XX
[from-dialer1]
exten => _X.,1,Set(__phonenumber=${phonenumber})
exten => _X.,n,h324m_call(play@from-dialer1)
exten => play,1,Set(CHANNEL(transfercapability)=VIDEO)
exten => play,n,Set(CHANNEL(userinformationlayer1)=38)
exten => play,n,Dial(Zap/g0/${phonenumber},30)
exten => play,n,Hangup()
[video-campaign1]
exten => play,1,Wait(1)
exten =>
play,n,mp4play(/var/lib/asterisk/sounds/autodialer/user_recordings/video/26/NEW_Mercedes_SLK_2012.3gp)
exten => play,n,Hangup()
Regarding Mituls suggestions, below is the .3gp file mp4info
3gvideo:~# mp4info
/var/lib/asterisk/sounds/autodialer/user_recordings/video/26/gauloises1.3gp
mp4info version 1.6.1
/var/lib/asterisk/sounds/autodialer/user_recordings/video/26/gauloises1.3gp:
Track TypeInfo
1 video H.263, 19.750 secs, 55 kbps, 176x144 @ 8.00 fps
2 audio AMR, 19.620 secs, 13 kbps, 8000 Hz
3 hintPayload H263-2000 for track 1
4 hintPayload AMR for track 2
Tool: mp4creator 1.6.1
What is the difference between h324m=bc and h324m=llc?. I had llc and
changed it to bc. Now i think that video is smoother. Is it just my
idea, or could it be true?
Now that dial command is in the dialplan, ${HANGUPCAUSE} gets populated,
and one can have better idea why a call failed. Now if a call fails one
can distinguish if device is switched off or device is not capable to
receive video call (as long as voicemail is disabled).
The problem that remains now is when phonenumber has voicemail enabled
and device is not video capable. I am still not able to understand that
the call has failed. Hangup cause and dialstatus indicate that call
proceeds ok, but call goes into voicemail.
Nobody has ever come to this problem?
Thank you very much Mitul and Jack for helping me implement this way of
autodialeing for 3g video. Wish there was a way to handle better failed
calls.
Your experiences are welcome
Best regards
Dinos
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Re: [Asterisk-video] Generate outbound 3g call will .call file and local channel
Hello Jack, I have already tried to use h324m_call instead of h324m_gw, but again when h324m_gw_answer() is reached, call again hangs up Thank you very much for your input Regards Dinos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
Re: [Asterisk-video] Generate outbound 3g call will .call file and local channel
Hello again Mitul and thank you for your reply,
I followed your instructions but still again when h324m_gw_answer() is
reached, call hungs up. Any more suggestions?
Thank you again, your help is very much appreciated!
Best regards
Dinos
My call file
Channel: Local/X@from-dialer1/n
CallerId: 1506
Context: video-campaign1
Extension: play
Priority: 1
Set: CHANNEL(transfercapability)=VIDEO
Set: CHANNEL(userinformationlayer1)=38
My dialplan
[from-dialer1]
exten => _X.,1,Set(CHANNEL(transfercapability)=VIDEO)
exten => _X.,n,Set(CHANNEL(userinformationlayer1)=38)
exten => _X.,n,Dial(Zap/g0/${EXTEN},30)
exten => _X.,n,Hangup()
[video-campaign1]
exten => play,1,h324m_gw(play2@video-campaign1)
exten => play2,1,h324m_gw_answer()
exten => play2,n,mp4play(file.3gp)
exten => play2,n,Hangup()
-- Attempting call on Local/XX@from-dialer1/n for
play@video-campaign1:1 (Retry 1)
-- Executing [XX@from-dialer1:1]
Set("Local/XX@from-dialer1-fa03,2",
"CHANNEL(transfercapability)=VIDEO") in new stack
-- Executing [XX@from-dialer1:2]
Set("Local/XX@from-dialer1-fa03,2",
"CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [XX@from-dialer1:3]
Dial("Local/XX@from-dialer1-fa03,2", "Zap/g0/XX|30") in
new stack
-- digital call, setting user information layer 1 to 38 (0x26)
-- zap call: h324musellc=0, ast->userinformationlayer1=38
-- Requested transfer capability: 0x18 - VIDEO
-- Called g0/XX
-- Zap/1-1 is proceeding passing it to
Local/XX@from-dialer1-fa03,2
-- Zap/1-1 is ringing
-- Zap/1-1 answered Local/XX@from-dialer1-fa03,2
> Channel Local/XX@from-dialer1-fa03,1 was answered.
-- Executing [play@video-campaign1:1]
h324m_gw("Local/XX@from-dialer1-fa03,1",
"play2@video-campaign1") in new stack
-- Executing [play2@video-campaign1:1]
h324m_gw_answer("Local/play2@video-campaign1-5a2e,2", "") in new stack
-- Channel 0/1, span 1 got hangup request, cause 16
-- Hungup 'Zap/1-1'
== Spawn extension (from-dialer1, XX, 3) exited non-zero on
'Local/XX@from-dialer1-fa03,2'
-- Executing [h@from-dialer1:1]
NoOp("Local/XX@from-dialer1-fa03,2", "Hangupcause1:16") in new stack
-- Executing [h@from-dialer1:2]
NoOp("Local/XX@from-dialer1-fa03,2", "Dialstatus1:ANSWER") in
new stack
-- Executing [h@from-dialer1:3]
NoOp("Local/XX@from-dialer1-fa03,2", "Reason1:") in new stack
-- Executing [h@from-dialer1:4]
Hangup("Local/XX@from-dialer1-fa03,2", "") in new stack
== Spawn extension (from-dialer1, h, 4) exited non-zero on
'Local/XX@from-dialer1-fa03,2'
== Spawn extension (video-campaign1, play2, 1) exited non-zero on
'Local/play2@video-campaign1-5a2e,2'
-- Executing [h@video-campaign1:1]
NoOp("Local/play2@video-campaign1-5a2e,2", "Hangupcause2:0") in new stack
-- Executing [h@video-campaign1:2]
NoOp("Local/play2@video-campaign1-5a2e,2", "Dialstatus2:") in new stack
-- Executing [h@video-campaign1:3]
NoOp("Local/play2@video-campaign1-5a2e,2", "Reason2:") in new stack
-- Executing [h@video-campaign1:4]
Hangup("Local/play2@video-campaign1-5a2e,2", "") in new stack
== Spawn extension (video-campaign1, h, 4) exited non-zero on
'Local/play2@video-campaign1-5a2e,2'
== Spawn extension (video-campaign1, play, 1) exited non-zero on
'Local/XX@from-dialer1-fa03,1'
-- Executing [h@video-campaign1:1]
NoOp("Local/XX@from-dialer1-fa03,1", "Hangupcause2:0") in new stack
-- Executing [h@video-campaign1:2]
NoOp("Local/XX@from-dialer1-fa03,1", "Dialstatus2:") in new stack
-- Executing [h@video-campaign1:3]
NoOp("Local/XX@from-dialer1-fa03,1", "Reason2:") in new stack
-- Executing [h@video-campaign1:4]
Hangup("Local/XX@from-dialer1-fa03,1", "") in new stack
== Spawn extension (video-campaign1, h, 4) exited non-zero on
'Local/XX@from-dialer1-fa03,1'
[2011-09-08 16:59:53] NOTICE[3895]: pbx_spool.c:351 attempt_thread: Call
completed to Local/XX@from-dialer1/n
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Re: [Asterisk-video] Generate outbound 3g call will .call file and local channel
My best guess regarding outbound video dialling with .call files using a
localchannel is the following
Call file
Channel: Local/XX@from-dialer1/n
CallerId: 1506
Context: from-dialer1
Extension: XX
Priority: 1
Set: CHANNEL(transfercapability)=VIDEO
Set: CHANNEL(userinformationlayer1)=38
[from-dialer1]
exten => _X.,1,Set(CHANNEL(transfercapability)=VIDEO)
exten => _X.,n,Set(CHANNEL(userinformationlayer1)=38)
exten => _X.,n,Dial(Zap/g0/${EXTEN},30)
exten => _X.,n,h324m_gw(play@video-campaign1)
exten => _X.,n,Hangup()
[video-campaign1]
exten => play,1,h324m_gw_answer()
exten =>
play,n,mp4play(/var/lib/asterisk/sounds/autodialer/user_recordings/video/26/gauloises1.3gp)
exten => play,n,Hangup()
But yet, it does not work. When execution reaches h324m_gw_answer call
hangs up.
Below is the cli
-- Attempting call on Local/XX@from-dialer1/n for
XX@from-dialer1:1 (Retry 1)
-- Executing [XX@from-dialer1:1]
Set("Local/XX@from-dialer1-90dd,2",
"CHANNEL(transfercapability)=VIDEO") in new stack
-- Executing [XX@from-dialer1:2]
Set("Local/XX@from-dialer1-90dd,2",
"CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [XX@from-dialer1:3]
Dial("Local/XX@from-dialer1-90dd,2", "Zap/g0/XX|30") in
new stack
-- digital call, setting user information layer 1 to 38 (0x26)
-- zap call: h324musellc=0, ast->userinformationlayer1=38
-- Requested transfer capability: 0x18 - VIDEO
-- Called g0/XX
-- Zap/1-1 is proceeding passing it to
Local/XX@from-dialer1-90dd,2
-- Zap/1-1 is ringing
-- Zap/1-1 answered Local/XX@from-dialer1-90dd,2
> Channel Local/XX@from-dialer1-90dd,1 was answered.
-- Executing [XX@from-dialer1:1]
Set("Local/XX@from-dialer1-90dd,1",
"CHANNEL(transfercapability)=VIDEO") in new stack
-- Executing [XX@from-dialer1:2]
Set("Local/XX@from-dialer1-90dd,1",
"CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [XX@from-dialer1:3]
Dial("Local/XX@from-dialer1-90dd,1", "Zap/g0/XX|30") in
new stack
-- digital call, setting user information layer 1 to 38 (0x26)
-- zap call: h324musellc=0, ast->userinformationlayer1=38
-- Requested transfer capability: 0x18 - VIDEO
-- Called g0/XX
-- Local/XX@from-dialer1-90dd,1 requested special control
20, passing it to Zap/2-1
-- Zap/2-1 is proceeding passing it to
Local/XX@from-dialer1-90dd,1
-- Local/XX@from-dialer1-90dd,1 requested special control
20, passing it to Zap/2-1
-- Local/XX@from-dialer1-90dd,1 requested special control
20, passing it to Zap/2-1
-- Channel 0/2, span 1 got hangup request, cause 17
-- Zap/2-1 is busy
-- Hungup 'Zap/2-1'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [XX@from-dialer1:4]
h324m_gw("Local/XX@from-dialer1-90dd,1", "play@video-campaign1")
in new stack
-- Executing [play@video-campaign1:1]
h324m_gw_answer("Local/play@video-campaign1-8ead,2", "") in new stack
-- Channel 0/1, span 1 got hangup request, cause 16
-- Hungup 'Zap/1-1'
== Spawn extension (from-dialer1, XX, 3) exited non-zero on
'Local/XX@from-dialer1-90dd,2'
== Spawn extension (from-dialer1, XX, 4) exited non-zero on
'Local/XX@from-dialer1-90dd,1'
== Spawn extension (video-campaign1, play, 1) exited non-zero on
'Local/play@video-campaign1-8ead,2'
[2011-09-08 15:42:43] NOTICE[2713]: pbx_spool.c:351 attempt_thread: Call
completed to Local/XX@from-dialer1/n
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Re: [Asterisk-video] Generate outbound 3g call will .call file and local channel
Hello Mitul and thank you for your reply,
I tried your example but did not work for me as i am looking to play an
mp4 file using mp4play on the receiving device. I added mp4 play command
at the end but call hangs up. Could you modify dialplan to play mp4 file?
${Dialstatus} does not give much information as to what happened to the
call.
During my tests i realized that the only way to know in more detail what
happened to the call and especially if 3g call was sent to a 3g video
capable device is ${HANGUPCAUSE}.
From my test i have found the following:
If device is switched off (*this number has not got voicemail
activated*), but device*is video capable* then you get *Hangupcause=20
subscriber absent 480 Temporarily unavailable*
If device is switched on (*this number has not got voicemail
activated*), but *is not video capable* then you get *Hangupcause=88
incompatible destination 503 Service unavailable*
If device is switched on (*this number has not got voicemail
activated*), but *is busy whether device is video capable or not*, then
you get *Hangupcause=17 user busy 486 Busy here*
A problem arises when mobile number has got *voicemail enabled*.
If device is*not video capable*, and *voicemail is activated* then:
a. if device is on 3g network, then call jusr fails with *Hangupcause=88
incompatible destination 503 Service unavailable*
b. if device is not on 3g network, then call is redirected to voicemail,
without any hint about it. Major problem here, because from the dialplan
point of view it seems that video was actually played at the device
although it didnt. Receiver ends up with an annoying voicemail with a
buzzing sound. Also call does not hangup until voicemail times out
Has anybody got a better idea of how to handle this?
Thank you in advance
Regards
Dinos
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Re: [Asterisk-video] Generate outbound 3g call will .call file and local channel
Any help would be appreciated
Still haven't managed to do it.
Anybody?
On 6/9/2011 4:01 ??, Konstantinos Liadakis wrote:
Hello everybody,
I have been trying lately to generate an outbound call from a .call
file *but *using a local channel.
I need the local channel in order populate ${HANGUPCAUSE} asterisk
variable.
Hangupcause gives a better explanation than ${REASON} as to why call
has failed, and hence one can find out if receiving device has not 3g
video capability, or is out of 3g coverage e.t.c
No matter what i try i cannot make it work.
I have managed to do outbound calls generated by .call file, but not
with a local channel
I am using the following:
Channel: Zap/g0/number
CallerId: $strCallerId
Context: from-dialer
Extension: outbound
Priority: 1
Set: CHANNEL(transfercapability)=VIDEO
Set: CHANNEL(userinformationlayer1)=38
[from-dialer]
exten => outbound,1,h324m_gw(play@video-campaign)
[video-campaign]
exten => play,1,h324m_gw_answer()
exten => play,n,mp4play(${filename})
exten => play,n,Hangup()
Could someone modify the above in order to work with local channel, so
the dial command can be in the dialplan
Thank you very much in advance for your help
Regards
Dinos
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[Asterisk-video] Generate outbound 3g call will .call file and local channel
Hello everybody,
I have been trying lately to generate an outbound call from a .call file
*but *using a local channel.
I need the local channel in order populate ${HANGUPCAUSE} asterisk
variable.
Hangupcause gives a better explanation than ${REASON} as to why call has
failed, and hence one can find out if receiving device has not 3g video
capability, or is out of 3g coverage e.t.c
No matter what i try i cannot make it work.
I have managed to do outbound calls generated by .call file, but not
with a local channel
I am using the following:
Channel: Zap/g0/number
CallerId: $strCallerId
Context: from-dialer
Extension: outbound
Priority: 1
Set: CHANNEL(transfercapability)=VIDEO
Set: CHANNEL(userinformationlayer1)=38
[from-dialer]
exten => outbound,1,h324m_gw(play@video-campaign)
[video-campaign]
exten => play,1,h324m_gw_answer()
exten => play,n,mp4play(${filename})
exten => play,n,Hangup()
Could someone modify the above in order to work with local channel, so
the dial command can be in the dialplan
Thank you very much in advance for your help
Regards
Dinos
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Re: [Asterisk-video] hot to know a mobile device supports video call when trying to do outbound 3g video call
One more note, I realized that when non capable phone has 3g coverage then when trying to call it using h324 then the call fails straight away. If the non capable phone has no 3g coverage then call will go through to voicemail. I tested this using h324 and asterisk, but i double chcked that using my 3g video capable phone trying to call a non capable phone using video call. I dont have problem calling the number to see if it supports video call, but i dont want to leave voicemails of buzzing sounds... Has anybody ever seen this before? Is there a workaround Thanks again Regards Dinos On 24/8/2011 2:04 ??, Sergio Garcia Murillo wrote: Hi Dinos, Quick answer, the only way to know if someone supports 3G video call is making a call. Best regards Sergio El 24/08/2011 12:56, Konstantinos Liadakis escribió: Although this post is 4 weeks old, i never got some answers from the list. There is no way to know if device has 3g video capability or if it is in an area with 3g coverage? How can we avoid sending 3g video calls to non capable devices? Could someone post a dialplan that could do that? Hope this time i get some answer. Thank you very much in advance Regards Dinos On 25/7/2011 11:53 ??, Konstantinos Liadakis wrote: Hello list, Is there any way to know if device supports video call, or if device supports video calls but is not within 3g coverage when trying to make outbound video call and call fails. During my tests i have seen that sometimes calls go through to devices without video capability. In this case i found a message in mobiles voicemails with just a buzzing sound. How can we avoid sending 3g video calls to non capable devices? Could someone post a dialplan that could do that? thank you in advance dinos -- _ -- 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 - No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1391 / Virus Database: 1520/3820 - Release Date: 08/07/11 -- -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video -- _ -- 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 No virus found in this message. Checked by AVG - www.avg.com <http://www.avg.com> Version: 10.0.1392 / Virus Database: 1520/3853 - Release Date: 08/23/11 -- _ -- 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
Re: [Asterisk-video] hot to know a mobile device supports video call when trying to do outbound 3g video call
Although this post is 4 weeks old, i never got some answers from the list. There is no way to know if device has 3g video capability or if it is in an area with 3g coverage? How can we avoid sending 3g video calls to non capable devices? Could someone post a dialplan that could do that? Hope this time i get some answer. Thank you very much in advance Regards Dinos On 25/7/2011 11:53 πμ, Konstantinos Liadakis wrote: Hello list, Is there any way to know if device supports video call, or if device supports video calls but is not within 3g coverage when trying to make outbound video call and call fails. During my tests i have seen that sometimes calls go through to devices without video capability. In this case i found a message in mobiles voicemails with just a buzzing sound. How can we avoid sending 3g video calls to non capable devices? Could someone post a dialplan that could do that? thank you in advance dinos -- _ -- 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 - No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1391 / Virus Database: 1520/3820 - Release Date: 08/07/11 -- Konstantinos Liadakis Managing Director TEL: (+30) 211 11 44 111 DIRECT: (+30) 211 11 44 101 MOBILE: (+30) 6936 52 51 13 FAX: (+30) 210 68 12 386 E-MAIL: kliada...@yuboto.com WEBSITE: www.yuboto.com Click 2 Call me Free -- _ -- 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
[Asterisk-video] hot to know a mobile device supports video call when trying to do outbound 3g video call
Hello list, Is there any way to know if device supports video call, or if device supports video calls but is not within 3g coverage when trying to make outbound video call and call fails. During my tests i have seen that sometimes calls go through to devices without video capability. In this case i found a message in mobiles voicemails with just a buzzing sound. How can we avoid sending 3g video calls to non capable devices? Could someone post a dialplan that could do that? thank you in advance dinos -- _ -- 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
Re: [Asterisk-video] can i install h324M in asterisk 1.6
Hello, I used the following link in order to find the information needed http://asterisk-party.net/index.php/Asterisk_Video_3G_FR Good luck On 4/3/2011 1:54 μμ, keshu dhanda wrote: hello dear, can u tell me the process of installation h324m on 1.4 On Fri, Mar 4, 2011 at 5:14 PM, Konstantinos Liadakis <kliada...@yuboto.com> wrote: Hello list, Has anybody managed to install h324M tools on asterisk 1.6? I have succesfully installed on 1.4, is it wortht rying 1.6 or it will never work Thank you in advance Regards Dinos -- _ -- 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 -- _ -- 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 No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1435/3480 - Release Date: 03/03/11 -- Konstantinos Liadakis Managing Director TEL: (+30) 211 11 44 111 DIRECT: (+30) 211 11 44 101 MOBILE: (+30) 6936 52 51 13 FAX: (+30) 210 68 12 386 E-MAIL: kliada...@yuboto.com WEBSITE: www.yuboto.com Click 2 Call me Free -- _ -- 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
[Asterisk-video] can i install h324M in asterisk 1.6
Hello list, Has anybody managed to install h324M tools on asterisk 1.6? I have succesfully installed on 1.4, is it wortht rying 1.6 or it will never work Thank you in advance Regards Dinos -- _ -- 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
[Asterisk-video] Video gateway with sip or iax trunk for outbound calls - is it possible?
Hello List,
I have successfully tested the following scenario
3g video calls --- Pri -- Asterisk with h324M
What i would like to accomplish now is
3g video calls --- Pri --Asterisk 1.4 BoxB --- SIP or IAX2 trunk
over gigabit LAN --- Asterisk 1.4 with h324M BoxA
I made some tests, using iax2 and sip trunk.
BoxA generates a call and sends it to boxB over the SIP or IAX2 trunk.
When call comes out mobile phone receives it as voice and soon after it
hangs up.
BoxA when connected directly to pri can make and receive 3g video calls.
I have read in the lists that people have used sip and iax2 trunks
successfully but did not give much details in how they succeeded it.
I have enabled in sip.conf videosupport=yes and have set the following
codecs
videosupport = yes
disallow=all
allow=alaw
allow=h263p
allow=h261
allow=h263
allow=h264
My pri uses alaw, and this is what i used when making the trunks.
Below is the dialplan i use. I am using eyebeam as softphone
[outgoing_calls]
exten => _X.,1,h324m_call(${ext...@to3g)
[to3g]
exten => _X.,1,Set(CHANNEL(transfercapability)=VIDEO)
exten => _X.,n,NoOp(transfer=${CHANNEL(transfercapability)})
exten => _X.,n,Set(CHANNEL(userinformationlayer1)=38)
exten => _X.,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
;exten => _X.,n,Dial,SIP/prodpbx/${EXTEN}
exten => _X.,n,Dial(SIP/astvideo/${EXTEN})
exten => _X.,n,Playtones(congestion)
exten => _X.,n,Hangup
Could someone guide me to right direction?
Thank you in advance
Merry Christmas to all
Dinos
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Re: [Asterisk-video] 3G Video Call PRI Debug Output
During my test using eyebeam i realised that in order for your video to be seen on the other side, you have to only enable 1 video codec in eyebeam and i think its th h264+ only. If you add all 3 codecs, then it did not work for me Hope this might help Dinos On 10/12/2010 9:38 πμ, Mitul Limbani wrote: Klaus, Quoting Klaus Darilion : Am 03.12.2010 17:40, schrieb Mitul Limbani: However, Interestingly, I am able to make Video Calls out from Eyebeam, but My video is not seen on remote end, but the remote video is seen on my EyeBeam. Does that mean that 3G-negotiation works on outbound calls, but not on inbound calls? Yes, you are right. My video doesnt go and same is the case when I try to play any Video File. So I suspect some issue with my side, remote side video is seen on EyeBeam. Regards, Mitul Limbani -- Konstantinos Liadakis Managing Director TEL: (+30) 211 11 44 111 DIRECT: (+30) 211 11 44 101 MOBILE: (+30) 6936 52 51 13 FAX: (+30) 210 68 12 386 E-MAIL: kliada...@yuboto.com WEBSITE: www.yuboto.com Click 2 Call me Free -- _ -- 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
Re: [Asterisk-video] 3G Video Call PRI Debug Output
you can also install mpeg4ip from multimedia sources. Which is simpler and works perfectly #vi /etc/apt/sources.list # Vi / etc / apt / sources.list deb deb http://mirror.home-dn.net/debian-multimedia http://mirror.home-dn.net/debian-multimedia lenny main lenny main deb-src deb-src http://mirror.home-dn.net/debian-multimedia http://mirror.home-dn.net/debian-multimedia lenny main lenny main #aptitude update && aptitude dist-upgrade # Aptitude update & & aptitude dist-upgrade #aptitude install mpeg4ip mpeg4ip-server mpeg4ip-utils # Aptitude install mpeg4ip mpeg4ip mpeg4ip-server-utils Regards Konstantinos Liadakis On 6/12/2010 6:29 μμ, Klaus Darilion wrote: Am 02.12.2010 16:21, schrieb Mitul Limbani: Hi Klaus, Quoting Klaus Darilion : Actually this looks fine. Where are you located - what is your default 'law'? Are you using E1/T1? India, E1, default is ulaw Have you tried if h324m_loopback() works? e.g. [from-pstn] exten => _7642,1,h324m_loopback() Yes, even this doesnt work. BTW, I have slight update here, I am using mp4v2-1.9.0 from : http://code.google.com/p/mp4v2/ coz MPEG4IP 1.5.0 refused to compile on my Ubuntu system. Also that I am using DAHDHI 2.4.0, coz Zaptel refused to compile on my system as well. As long as h324m_loopback() does not work it is useless to test video streaming. Maybe there is a problem with the phone? Have you tried another phone? regards Klaus -- Konstantinos Liadakis Managing Director TEL: (+30) 211 11 44 111 DIRECT: (+30) 211 11 44 101 MOBILE: (+30) 6936 52 51 13 FAX: (+30) 210 68 12 386 E-MAIL: kliada...@yuboto.com WEBSITE: www.yuboto.com Click 2 Call me Free -- _ -- 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
Re: [Asterisk-video] h324m_gw_answer() disconnects me from asterisk cli and call get dropped
hello Klaus Finally i managed to do some tests For h324m to work i had to do the following I removed asterisk 1.4.15 and used asterisk 1.4.21.1 with the patch as for your instructions. I used libpri 1.4.5 and patched it. I used zaptel 1.4.12.1 because on my debian machine i could not compile 1.4.11 (there is a bug about kernel, don't know much) I compiled with my asterisk amr codec, app_mp4, app_transcode, app_rtsp Now almost everything seem to work. These are the scenarios i tested 1. 3g --> pri --> asterisk --> mp4play works like a charm video and audio 2. .call file --> asterisk --> pri --> 3g works like a charm - video and audio 3. sip softphone bria --> asterisk --> pri --> 3g video works fine, no audio, i noticed the following errors on different tries a. Got SIP response 415 "Unsupported Media Type" back from 10.0.0.21 b. rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.21' c. chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! i suspect that bria might not have required codecs 4. 3g --> pri --> asterisk --> sip again caller can see my video but i cannot see the caller, no audio Tomorrow i want to try ivvr, and let you know. I havent managed to try app_rtsp as i dont have anything to stream from, would vlc work? i havnet managed to use transcoding as again i dont know how to use it, any example or any help? Thank you in advance for your replies, hope you can help me solve the audio problem Best regards, to all Dinos On 2/12/2010 11:43 πμ, Klaus Darilion wrote: Ok, some more questions I am using asterisk 1.4.15 with 1.4.11.4 libpri and 1.4.12.1 zaptel The production system I have is still running asterisk-1.4.21.1 with libpri-1.4.5 and zaptel-1.4.11. I do not know if newer/older versions work fine too. Do you think i should change to a more recent asterisk and libpri and dahdi instead of zaptel? I suspect that zaptel might be causing the problem, as i am having trouble to start it again after an asterisk disconnect So, what happens exactly - does Asterisk crash? asterisk-video*CLI> module load chan_zaptel.so [2010-12-02 11:25:27] WARNING[9033]: loader.c:362 load_dynamic_module: Error loading module 'chan_zaptel.so': /usr/lib/asterisk/modules/chan_zaptel.so: cannot open shared object file: No such file or directory [2010-12-02 11:25:27] WARNING[9033]: loader.c:646 load_resource: Module 'chan_zaptel.so' could not be loaded. If i reboot then everything will be fine and i will be able to make normal calls what about reloading the zaptel kernel modules? (rmmod, insmod/modprobe) Also i read about a patch in libpri in order to make outbound 3g calls, which in fact is what i want to do. I havent applied any patch to libpri. Should i and if yes where do i get it The patch is onyl necessary for proper signaling of outbound calls. Thus, in your case where you test with inbound calls there is no need to patch yet. Finally it is a pitty that there is no little wiki about this wonderfull project, i am having difficulty to find dialplans to test, like how to use app_transcode and how to set up test environment e.g. how to make sip to 3g and vice versa True. Klaus -- Konstantinos Liadakis Managing Director TEL: (+30) 211 11 44 111 DIRECT: (+30) 211 11 44 101 MOBILE: (+30) 6936 52 51 13 FAX: (+30) 210 68 12 386 E-MAIL: kliada...@yuboto.com WEBSITE: www.yuboto.com Click 2 Call me Free -- _ -- 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
Re: [Asterisk-video] h324m_gw_answer() disconnects me from asterisk cli and call get dropped
On 2/12/2010 11:43 πμ, Klaus Darilion wrote: >> Ok, some more questions >> >> I am using asterisk 1.4.15 with 1.4.11.4 libpri and 1.4.12.1 zaptel > > The production system I have is still running asterisk-1.4.21.1 with > libpri-1.4.5 and zaptel-1.4.11. > > I do not know if newer/older versions work fine too. > >> Do you think i should change to a more recent asterisk and libpri and >> dahdi instead of zaptel? I suspect that zaptel might be causing the >> problem, as i am having trouble to start it again after an asterisk >> disconnect > > So, what happens exactly - does Asterisk crash? > >> >> asterisk-video*CLI> module load chan_zaptel.so >> [2010-12-02 11:25:27] WARNING[9033]: loader.c:362 load_dynamic_module: >> Error loading module 'chan_zaptel.so': >> /usr/lib/asterisk/modules/chan_zaptel.so: cannot open shared object >> file: No such file or directory >> [2010-12-02 11:25:27] WARNING[9033]: loader.c:646 load_resource: Module >> 'chan_zaptel.so' could not be loaded. >> >> If i reboot then everything will be fine and i will be able to make >> normal calls > > what about reloading the zaptel kernel modules? (rmmod, insmod/modprobe) > >> Also i read about a patch in libpri in order to make outbound 3g calls, >> which in fact is what i want to do. I havent applied any patch to >> libpri. Should i and if yes where do i get it > > The patch is onyl necessary for proper signaling of outbound calls. > Thus, in your case where you test with inbound calls there is no need > to patch yet. > >> Finally it is a pitty that there is no little wiki about this wonderfull >> project, i am having difficulty to find dialplans to test, like how to >> use app_transcode and how to set up test environment e.g. how to make >> sip to 3g and vice versa > > True. > > Klaus > > > > - > No virus found in this message. > Checked by AVG - www.avg.com > Version: 10.0.1170 / Virus Database: 426/3291 - Release Date: 12/01/10 > > hello Klaus Finally i managed to do some tests For h324m to work i had to do the following I removed asterisk 1.4.15 and used asterisk 1.4.21.1 with the patch as for your instructions. I used libpri 1.4.5 and patched it. I used zaptel 1.4.12.1 because on my debian machine i could not compile 1.4.11 (there is a bug about kernel, don't know much) I compiled with my asterisk amr codec, app_mp4, app_transcode, app_rtsp Now almost everything seem to work. These are the scenarios i tested 1. 3g --> pri --> asterisk --> mp4play works like a charm video and audio 2. .call file --> asterisk --> pri --> 3g works like a charm - video and audio 3. sip softphone bria --> asterisk --> pri --> 3g video works fine, no audio, i noticed the following errors on different tries a. Got SIP response 415 "Unsupported Media Type" back from 10.0.0.21 b. rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.21' c. chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! i suspect that bria might not have required codecs 4. 3g --> pri --> asterisk --> sip again caller can see my video but i cannot see the caller, no audio Tomorrow i want to try ivvr, and let you know. I havent managed to try app_rtsp as i dont have anything to stream from, would vlc work? i havnet managed to use transcoding as again i dont know how to use it, any example or any help? Thank you in advance for your replies, hope you can help me solve the audio problem Best regards, to all Dinos -- _ -- 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
Re: [Asterisk-video] h324m_gw_answer() disconnects me from asterisk cli and call get dropped
On 1/12/2010 8:42 μμ, Klaus Darilion wrote:
>
>
> Am 01.12.2010 15:57, schrieb Konstantinos Liadakis:
>> Hello,
>>
>> This is my first post to this list.
>>
>> I have a debian box with 1.4.15 asterisk and all video stuff installed
>> using directions from
>> http://asterisk-party.org/index.php/Asterisk_Video_3G_FR
>>
>> I am using zap fro my PRI
>>
>> I can see all modules in asterisk module directory.
>>
>> I have *succeded *in making a loopback test 3g phone --> PRI -->
>> Asterisk using dialplan below
>>
>> ;exten => _1XXX,1,Answer
>> ;exten => _1XXX,n,h324m_loopback()
>> ;exten => _1XXX,n,Hangup
>>
>> Now, i am trying to play a video using mp4play to a 3g phone calling my
>> asterisk.
>>
>> I record mp4 file using my bria softphone and the following dialplan:
>>
>> exten => 700,1,Wait,0 ; Wait a second, just for fun
>> exten => 700,2,Answer ; Answer the line
>> exten => 700,3,mp4play(/usr/src/videomenu/menu.mp4)
>>
>> file is saved alright, and can be replayed to my bria softphone with
>> mp4play with no problem.
>
>
> As next step, before streaming video, you should make an echo test.
> Similar to the loopback test, but this time the H324m stream is
> encoded/decoded, e.g:
>
> [incoming_calls]
> exten => _1XXX,1,h324m_gw(m...@3gin)
>
> [3gin]
>
> exten => menu,1,h324m_gw_answer()
> exten => menu,2,Wait(1)
> exten => menu,3,Echo()
>
>> But as soon as call comes in i get disconnected from asterisk cli and
>> call is dropped
>
> Dropping from CLI happens when Asterisk crashes - maybe Asterisk
> crashes? Are you automatically starting ASterisk when it crashes? Take
> a look at the uptime after reconnecting.
>
> regards
> Klaus
>
>
>>
>> asterisk-video*CLI> -- Accepting call from '6936525113' to '1400' on
>> channel 0/1, span 1
>> asterisk-video*CLI> -- Executing [1...@incoming_calls:1]
>> h324m_gw("Zap/1-1", "m...@3gin") in new stack
>> asterisk-video*CLI> asterisk-video*CLI>
>> asterisk-video*CLI> Disconnected from Asterisk server
>> No such command '-- Accepting' (type 'help' for help)
>> No such command '-- Executing' (type 'help' for help)
>> No such command 'asterisk-video*CLI>' (type 'help' for help)
>> No such command 'Disconnected from' (type 'help' for help)
>> asterisk-video*CLI>
>> asterisk-video*CLI>
>> asterisk-video*CLI>
>> asterisk-video*CLI> pri debug span 1
>> Enabled debugging on span 1
>> asterisk-video*CLI> h324m debug level 4
>> app_h324m Debugging enabled level: 4
>> asterisk-video*CLI>
>> < Protocol Discriminator: Q.931 (8) len=42
>> < TEI=0 Call Ref: len= 2 (reference 26/0x1A) (Sent from originator)
>> < Message Type: SETUP (5)
>> < [a1]
>> < Sending Complete (len= 1)
>> < [04 03 88 90 a6]
>> < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
>> capability: Unrestricted digital information (8)
>> < Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
>> < User information layer 1: H.223/H.245 Multimedia (38)
>> < [18 03 a1 83 81]
>> < Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0
>> Preferred Dchan: 0
>> < ChanSel: As indicated in following octets
>> < Ext: 1 Coding: 0 Number Specified Channel Type: 3
>> < Ext: 1 Channel: 1 Type: CPE]
>> < [6c 0c 21 83 36 39 33 36 35 32 35 31 31 33]
>> < Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
>> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>> < Presentation: Presentation allowed of network provided number (3)
>> '6936525113' ]
>> < [70 05 a1 31 34 30 30]
>> < Called Number (len= 7) [ Ext: 1 TON: National Number (2) NPI:
>> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1400' ]
>> < [7c 03 88 90 a6]
>> < IE: Low-layer Compatibility (len = 5)
>> -- Making new call for cref 26
>> Received message for call 0x9807510 on 0x97c61b0 TEI/SAPI 0/0, call->pri
>> is 0x97c61b0 TEI/SAPI 0/0
>> -- Processing Q.931 Call Setup
>> -- Processing IE 161 (cs0, Sending Complete)
>> -- Processing IE 4 (cs0, Bearer Capability)
>> -- Processing IE 24 (cs0, Channel Identification)
>> -- Processing IE 108 (cs0, Calling Party Number)
>> -- Processing IE 112 (cs0, Called Party Number)
>> -- Processing IE 124 (cs0, Low-layer Compatibility)
>>
[Asterisk-video] h324m_gw_answer() disconnects me from asterisk cli and call get dropped
Hello,
This is my first post to this list.
I have a debian box with 1.4.15 asterisk and all video stuff installed
using directions from
http://asterisk-party.org/index.php/Asterisk_Video_3G_FR
I am using zap fro my PRI
I can see all modules in asterisk module directory.
I have *succeded *in making a loopback test 3g phone --> PRI -->
Asterisk using dialplan below
;exten => _1XXX,1,Answer
;exten => _1XXX,n,h324m_loopback()
;exten => _1XXX,n,Hangup
Now, i am trying to play a video using mp4play to a 3g phone calling my
asterisk.
I record mp4 file using my bria softphone and the following dialplan:
exten => 700,1,Wait,0 ; Wait a second, just for fun
exten => 700,2,Answer ; Answer the line
exten => 700,3,mp4play(/usr/src/videomenu/menu.mp4)
file is saved alright, and can be replayed to my bria softphone with
mp4play with no problem.
i use the following:
[incoming_calls]
exten => _1XXX,1,h324m_gw(m...@3gin)
[3gin]
exten => menu,1,h324m_gw_answer()
exten => menu,2,Wait(1)
exten => menu,3,mp4play(/usr/src/videomenu/phonevideo.mp4)
exten => menu,4,Hangup
But as soon as call comes in i get disconnected from asterisk cli and
call is dropped
asterisk-video*CLI> -- Accepting call from '6936525113' to '1400' on
channel 0/1, span 1
asterisk-video*CLI> -- Executing [1...@incoming_calls:1]
h324m_gw("Zap/1-1", "m...@3gin") in new stack
asterisk-video*CLI> asterisk-video*CLI>
asterisk-video*CLI> Disconnected from Asterisk server
No such command '-- Accepting' (type 'help' for help)
No such command '-- Executing' (type 'help' for help)
No such command 'asterisk-video*CLI>' (type 'help' for help)
No such command 'Disconnected from' (type 'help' for help)
asterisk-video*CLI>
asterisk-video*CLI>
asterisk-video*CLI>
asterisk-video*CLI> pri debug span 1
Enabled debugging on span 1
asterisk-video*CLI> h324m debug level 4
app_h324m Debugging enabled level: 4
asterisk-video*CLI>
< Protocol Discriminator: Q.931 (8) len=42
< TEI=0 Call Ref: len= 2 (reference 26/0x1A) (Sent from originator)
< Message Type: SETUP (5)
< [a1]
< Sending Complete (len= 1)
< [04 03 88 90 a6]
< Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (8)
< Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
pri
is 0x97c61b0 TEI/SAPI 0/0
-- Processing Q.931 Call Setup
-- Processing IE 161 (cs0, Sending Complete)
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
-- Processing IE 124 (cs0, Low-layer Compatibility)
q931.c:6966 post_handle_q931_message: Call 26 enters state 6 (Call
Present). Hold state: Idle
q931.c:4594 q931_call_proceeding: Call 26 enters state 9 (Incoming Call
Proceeding). Hold state: Idle
> DL-DATA request
> Protocol Discriminator: Q.931 (8) len=10
> TEI=0 Call Ref: len= 2 (reference 26/0x1A) (Sent to originator)
> Message Type: CALL PROCEEDING (2)
TEI=0 Transmitting N(S)=30, window is open V(A)=30 K=7
> Protocol Discriminator: Q.931 (8) len=10
> TEI=0 Call Ref: len= 2 (reference 26/0x1A) (Sent to originator)
> Message Type: CALL PROCEEDING (2)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0
Exclusive Dchan: 0
> ChanSel: As indicated in following octets
> Ext: 1 Coding: 0 Number Specified Channel
Type: 3
> Ext: 1 Channel: 1 Type: CPE]
-- Accepting call from 'xxx' to '1400' on channel 0/1, span 1
-- Executing [1...@incoming_calls:1] h324m_gw("Zap/1-1",
"m...@3gin") in new stack
asterisk-video*CLI>
Disconnected from Asterisk server
asterisk-video:~#
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