[asterisk-users] Playback with noanswer in AGI

Sammy Govind govoiper at gmail.com
Tue Feb 7 03:37:53 CST 2012 

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:
>>>>>               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:
>>>>               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:
>>>               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:
>               http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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