[asterisk-users] Playback with noanswer in AGI

Tue Feb 7 02:09:33 CST 2012

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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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