[asterisk-users] Playback with noanswer in AGI
Zohair Raza
engineerzuhairraza at gmail.com
Tue Feb 7 03:55:46 CST 2012
Yes,
Thanks
Regards,
Zohair Raza
On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind <govoiper at gmail.com> wrote:
> Exactly that's what I expected.
> Great - now have fun
>
>
> On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza <engineerzuhairraza at gmail.com>wrote:
>
>> Sammy,
>>
>> Problem is at phones, with a linksys phone it works but with eyebeam and
>> fanvill it doesn't
>>
>> Maybe they don't support early media.
>>
>> I think i will have to stick with ResetCDR and that will be okay now as
>> I've modified the code for that
>>
>> Thank you
>>
>> Regards,
>> Zohair Raza
>>
>>
>> On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza <
>> engineerzuhairraza at gmail.com> wrote:
>>
>>> Hi Sammy,
>>>
>>> Thanks for input.
>>>
>>> I have an eyebeam softphone registered with Asterisk 1.8.6 locally and
>>> from agi, I pass this
>>>
>>> $filetoplay = 'congestion';
>>> $agi->exec("Progress");
>>> $agi->exec("Playback $filetoplay,noanswer");
>>>
>>> Have tried putting file in .gsm and .wav formats, I hear ringing tone
>>> instead of playback
>>>
>>> Please have a look at sip-trace
>>>
>>> <--- SIP read from UDP:176.249.0.50:8721 --->
>>> INVITE sip:100 at 176.249.0.77 SIP/2.0
>>> To: <sip:100 at 176.249.0.77>
>>> From: Zohair<sip:1000 at 176.249.0.77>;tag=7f222672
>>> Via: SIP/2.0/UDP 176.249.0.50:8721
>>> ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport
>>> Call-ID: 2932f90ef302332b
>>> CSeq: 2 INVITE
>>> Contact: <sip:1000 at 176.249.0.50:8721>
>>> Max-Forwards: 70
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE, INFO
>>> Content-Type: application/sdp
>>> User-Agent: eyeBeam release 3006o stamp 17551
>>> Authorization: Digest
>>> username="1000",realm="asterisk",nonce="2abce759",uri="
>>> sip:100 at 176.249.0.77
>>> ",response="c1a2dbcf1b51d839521b1ee848bea055",algorithm=MD5
>>> Content-Length: 269
>>>
>>> v=0
>>> o=- 4333518 4333604 IN IP4 176.249.0.50
>>> s=eyeBeam
>>> c=IN IP4 176.249.0.50
>>> t=0 0
>>> m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101
>>> a=alt:1 1 : 119610F1 000000B3 176.249.0.50 6506
>>> a=fmtp:101 0-15
>>> a=rtpmap:100 speex/16000
>>> a=rtpmap:101 telephone-event/8000
>>> a=sendrecv
>>> <------------->
>>> --- (13 headers 11 lines) ---
>>> Sending to 176.249.0.50:8721 (no NAT)
>>> sing INVITE request as basis request - 2932f90ef302332b
>>> Found peer '1000' for '1000' from 176.249.0.50:8721
>>> == Using SIP RTP CoS mark 5
>>> Found RTP audio format 100
>>> Found RTP audio format 6
>>> Found RTP audio format 0
>>> Found RTP audio format 8
>>> Found RTP audio format 3
>>> Found RTP audio format 18
>>> Found RTP audio format 5
>>> Found RTP audio format 101
>>> Found audio description format speex for ID 100
>>> Found audio description format telephone-event for ID 101
>>> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20000012e
>>> (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing),
>>> combined - 0xc (ulaw|alaw)
>>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
>>> (telephone-event|), combined - 0x1 (telephone-event|)
>>> Peer audio RTP is at port 176.249.0.50:6506
>>> Looking for 100 in default (domain 176.249.0.77)
>>> list_route: hop: <sip:1000 at 176.249.0.50:8721>
>>>
>>> <--- Transmitting (no NAT) to 176.249.0.50:8721 --->
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 176.249.0.50:8721
>>> ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
>>> From: Zohair<sip:1000 at 176.249.0.77>;tag=7f222672
>>> To: <sip:100 at 176.249.0.77>
>>> Call-ID: 2932f90ef302332b
>>> CSeq: 2 INVITE
>>> Server: Asterisk PBX 1.8.0
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>> INFO, PUBLISH
>>> Supported: replaces, timer
>>> Contact: <sip:100 at 176.249.0.77:5060>
>>> Content-Length: 0
>>>
>>>
>>> <------------>
>>> -- Executing [100 at default:1] AGI("SIP/1000-00000019", "agi.php,DID")
>>> -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
>>> -- AGI Script Executing Application: (Progress) Options: ()
>>> Audio is at 5060
>>> Adding codec 0x4 (ulaw) to SDP
>>> Adding codec 0x8 (alaw) to SDP
>>> Adding non-codec 0x1 (telephone-event) to SDP
>>>
>>> <--- Transmitting (no NAT) to 176.249.0.50:8721 --->
>>> SIP/2.0 183 Session Progress
>>> Via: SIP/2.0/UDP 176.249.0.50:8721
>>> ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
>>> From: Zohair<sip:1000 at 176.249.0.77>;tag=7f222672
>>> To: <sip:100 at 176.249.0.77>;tag=as01491743
>>> Call-ID: 2932f90ef302332b
>>> CSeq: 2 INVITE
>>> Server: Asterisk PBX 1.8.0
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>> INFO, PUBLISH
>>> Supported: replaces, timer
>>> Contact: <sip:100 at 176.249.0.77:5060>
>>> Content-Type: application/sdp
>>> Content-Length: 258
>>>
>>> v=0
>>> o=root 1225456982 1225456982 IN IP4 176.249.0.77
>>> s=Asterisk PBX 1.8.0
>>> c=IN IP4 176.249.0.77
>>> t=0 0
>>> m=audio 15918 RTP/AVP 0 8 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> <------------>
>>> -- AGI Script Executing Application: (Playback) Options:
>>> (congestion,noanswer)
>>> -- <SIP/1000-00000019> Playing 'congestion.slin' (language 'en')
>>> -- <SIP/1000-00000019>AGI Script agi.php completed, returning 0
>>>
>>>
>>> Regards,
>>> Zohair Raza
>>>
>>>
>>> On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind <govoiper at gmail.com>wrote:
>>>
>>>> Hey Danny,
>>>>
>>>> I've this thing exactly running and working as Zohair mentioned! i.e I
>>>> do not answer() the call rather put a progress() and soon after that
>>>> playing back the sound file from playback with noanswer and then I get the
>>>> file streaming as 183-Session progress file.
>>>>
>>>> I do understand that playing any sound file before establishing any
>>>> audio session between two end point will result in no-adio from playback()
>>>> BUT the combination of progress() and playback(,noanswer) works fine for me.
>>>>
>>>> What I think the issue could be for Zohair is that its
>>>> requesting/incoming session(carrier) isn't allowing the 183-Session
>>>> progress.
>>>>
>>>> Zohair can you do a SIP trace for this particular call along with the
>>>> dialplan executing for it!?
>>>>
>>>> Regards,
>>>> Sammy.
>>>>
>>>>
>>>> On Tue, Feb 7, 2012 at 11:55 AM, Zohair Raza <
>>>> engineerzuhairraza at gmail.com> wrote:
>>>>
>>>>> Thanks for this explanation Dany!
>>>>>
>>>>> Regards,
>>>>> Zohair Raza
>>>>>
>>>>>
>>>>> On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas <danny at debsinc.com>wrote:
>>>>>
>>>>>> You are mis-understanding the concept – the noanswer option is
>>>>>> playing the file as you requested, but since you aren’t answering the call,
>>>>>> no channel is established to actually present the sound to you.****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>>>>>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Zohair Raza
>>>>>> *Sent:* Monday, February 06, 2012 12:06 PM
>>>>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>>>>> *Subject:* [asterisk-users] Playback with noanswer in AGI****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> Hi All, ****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> I want to play a file in agi but dont want to answer the call****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> I am dialing through sip phone and running asterisk 1.8.6,****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> I tried following with no luck****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> $agi->exec("Progress");****
>>>>>>
>>>>>> $agi->exec("Playback $filetoplay,noanswer");****
>>>>>>
>>>>>> $agi->hangup();****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> When I dial I can't hear the audio but if I answer the call or remove
>>>>>> noanswer argument I can hear the audio.****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> phpAGI's stream_file didn't help either. ****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> I ended up with ResetCDR() before hangup to reset billsec, duration
>>>>>> and disposition but don't want to do it this way.****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> What could be the problem?****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> From Voip-info.org :****
>>>>>>
>>>>>> *noanswer*: Play the sound file, but don't answer the channel first
>>>>>> (if hasn't been answered already). Not all channels support playing
>>>>>> messages while still on hook.****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> Is it because the channel is not supported?****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> Regards,****
>>>>>>
>>>>>> Zohair Raza****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>> http://www.asterisk.org/hello
>>>>>
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>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>> http://www.asterisk.org/hello
>>>>
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>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
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