<div dir="ltr">Yes, <div><br></div><div>Thanks </div><div><br></div><div><br clear="all"><div dir="ltr">Regards,<br>Zohair Raza<div><br></div></div><div class="gmail_quote">On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Exactly that's what I expected.<div>Great - now have fun<div><div class="h5"><br><br><div class="gmail_quote">On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza <span dir="ltr"><<a href="mailto:engineerzuhairraza@gmail.com" target="_blank">engineerzuhairraza@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Sammy, <div><br></div><div>Problem is at phones, with a linksys phone it works but with eyebeam and fanvill it doesn't</div>
<div><br></div><div>Maybe they don't support early media. </div><div><br></div>
<div>I think i will have to stick with ResetCDR and that will be okay now as I've modified the code for that</div><div><br></div><div>Thank you</div><div><br></div><div>Regards,<span><font color="#888888"><div dir="ltr">
Zohair Raza<div><br></div>
<div><br></div></div></font></span><div><div><div class="gmail_quote">On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza <span dir="ltr"><<a href="mailto:engineerzuhairraza@gmail.com" target="_blank">engineerzuhairraza@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hi Sammy, <div><br></div><div>Thanks for input. </div><div><br></div><div>I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this</div><div><br></div><div>$filetoplay = 'congestion';</div>
<div>
<div><div>$agi->exec("Progress");<span style="white-space:pre-wrap"> </span></div><div>$agi->exec("Playback $filetoplay,noanswer");</div></div><div><br></div></div><div>Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback</div>
<div><br></div><div>Please have a look at sip-trace</div><div><br></div><div><div><--- SIP read from UDP:<a href="http://176.249.0.50:8721" target="_blank">176.249.0.50:8721</a> ---></div><div>INVITE <a href="mailto:sip%3A100@176.249.0.77" target="_blank">sip:100@176.249.0.77</a> SIP/2.0</div>
<div>To: <<a href="mailto:sip%3A100@176.249.0.77" target="_blank">sip:100@176.249.0.77</a>></div><div>From: Zohair<<a href="mailto:sip%3A1000@176.249.0.77" target="_blank">sip:1000@176.249.0.77</a>>;tag=7f222672</div>
<div>Via: SIP/2.0/UDP 176.249.0.50:8721;branch=z9hG4bK-d87543-521938753-1--d87543-;rport</div>
<div>Call-ID: 2932f90ef302332b</div><div>CSeq: 2 INVITE</div><div>Contact: <<a href="http://sip:1000@176.249.0.50:8721" target="_blank">sip:1000@176.249.0.50:8721</a>></div><div>Max-Forwards: 70</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO</div>
<div>Content-Type: application/sdp</div><div>User-Agent: eyeBeam release 3006o stamp 17551</div><div>Authorization: Digest username="1000",realm="asterisk",nonce="2abce759",uri="<a href="mailto:sip%3A100@176.249.0.77" target="_blank">sip:100@176.249.0.77</a>",response="c1a2dbcf1b51d839521b1ee848bea055",algorithm=MD5</div>
<div>Content-Length: 269</div><div><br></div><div>v=0</div><div>o=- 4333518 4333604 IN IP4 176.249.0.50</div><div>s=eyeBeam</div><div>c=IN IP4 176.249.0.50</div><div>t=0 0</div><div>m=audio 6506 RTP/AVP 100 <a href="tel:6%200%208%203%2018%205%20101" value="+16083185101" target="_blank">6 0 8 3 18 5 101</a></div>
<div>a=alt:1 1 : 119610F1 000000B3 176.249.0.50 6506</div><div>a=fmtp:101 0-15</div><div>a=rtpmap:100 speex/16000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=sendrecv</div><div><-------------></div><div>
--- (13 headers 11 lines) ---</div><div>Sending to <a href="http://176.249.0.50:8721" target="_blank">176.249.0.50:8721</a> (no NAT)</div><div>sing INVITE request as basis request - 2932f90ef302332b</div><div>Found peer '1000' for '1000' from <a href="http://176.249.0.50:8721" target="_blank">176.249.0.50:8721</a></div>
<div> == Using SIP RTP CoS mark 5</div><div>Found RTP audio format 100</div><div>Found RTP audio format 6</div><div>Found RTP audio format 0</div><div>Found RTP audio format 8</div><div>Found RTP audio format 3</div><div>
Found RTP audio format 18</div><div>Found RTP audio format 5</div><div>Found RTP audio format 101</div><div>Found audio description format speex for ID 100</div><div>Found audio description format telephone-event for ID 101</div>
<div>Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20000012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)</div><div>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)</div>
<div>Peer audio RTP is at port <a href="http://176.249.0.50:6506" target="_blank">176.249.0.50:6506</a></div><div>Looking for 100 in default (domain 176.249.0.77)</div><div>list_route: hop: <<a href="http://sip:1000@176.249.0.50:8721" target="_blank">sip:1000@176.249.0.50:8721</a>></div>
<div><br></div><div><--- Transmitting (no NAT) to <a href="http://176.249.0.50:8721" target="_blank">176.249.0.50:8721</a> ---></div><div>SIP/2.0 100 Trying</div><div>Via: SIP/2.0/UDP 176.249.0.50:8721;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721</div>
<div>From: Zohair<<a href="mailto:sip%3A1000@176.249.0.77" target="_blank">sip:1000@176.249.0.77</a>>;tag=7f222672</div><div>To: <<a href="mailto:sip%3A100@176.249.0.77" target="_blank">sip:100@176.249.0.77</a>></div>
<div>Call-ID: 2932f90ef302332b</div>
<div>CSeq: 2 INVITE</div><div>Server: Asterisk PBX 1.8.0</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>Supported: replaces, timer</div><div>Contact: <<a href="http://sip:100@176.249.0.77:5060" target="_blank">sip:100@176.249.0.77:5060</a>></div>
<div>Content-Length: 0</div><div><br></div><div><br></div><div><------------></div><div> -- Executing [100@default:1] AGI("SIP/1000-00000019", "agi.php,DID")</div><div> -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php</div>
<div> -- AGI Script Executing Application: (Progress) Options: ()</div><div>Audio is at 5060</div><div>Adding codec 0x4 (ulaw) to SDP</div><div>Adding codec 0x8 (alaw) to SDP</div><div>Adding non-codec 0x1 (telephone-event) to SDP</div>
<div><br></div><div><--- Transmitting (no NAT) to <a href="http://176.249.0.50:8721" target="_blank">176.249.0.50:8721</a> ---></div><div>SIP/2.0 183 Session Progress</div><div>Via: SIP/2.0/UDP 176.249.0.50:8721;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721</div>
<div>From: Zohair<<a href="mailto:sip%3A1000@176.249.0.77" target="_blank">sip:1000@176.249.0.77</a>>;tag=7f222672</div><div>To: <<a href="mailto:sip%3A100@176.249.0.77" target="_blank">sip:100@176.249.0.77</a>>;tag=as01491743</div>
<div>Call-ID: 2932f90ef302332b</div>
<div>CSeq: 2 INVITE</div><div>Server: Asterisk PBX 1.8.0</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>Supported: replaces, timer</div><div>Contact: <<a href="http://sip:100@176.249.0.77:5060" target="_blank">sip:100@176.249.0.77:5060</a>></div>
<div>Content-Type: application/sdp</div><div>Content-Length: 258</div><div><br></div><div>v=0</div><div>o=root 1225456982 1225456982 IN IP4 176.249.0.77</div><div>s=Asterisk PBX 1.8.0</div><div>c=IN IP4 176.249.0.77</div>
<div>t=0 0</div><div>m=audio 15918 RTP/AVP 0 8 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=sendrecv</div>
<div><br></div><div><------------></div><div> -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer)</div><div> -- <SIP/1000-00000019> Playing 'congestion.slin' (language 'en')</div>
<div> -- <SIP/1000-00000019>AGI Script agi.php completed, returning 0</div></div><div><br></div><div><br clear="all"><div dir="ltr">Regards,<span><font color="#888888"><br>Zohair Raza<div><br></div>
<div><br></div></font></span></div><div><div><div class="gmail_quote">
On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hey Danny,<div><br></div><div>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. </div>
<div><br></div><div>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.</div>
<div><br></div><div>What I think the issue could be for Zohair is that its requesting/incoming session(carrier) isn't allowing the 183-Session progress.</div><div><br></div><div>Zohair can you do a SIP trace for this particular call along with the dialplan executing for it!?</div>
<div><br></div><div>Regards,</div><div>Sammy.<div><div><br><br><div class="gmail_quote">On Tue, Feb 7, 2012 at 11:55 AM, Zohair Raza <span dir="ltr"><<a href="mailto:engineerzuhairraza@gmail.com" target="_blank">engineerzuhairraza@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Thanks for this explanation Dany!<br clear="all"><div dir="ltr"><br></div><div dir="ltr">Regards,<br>Zohair Raza<div>
<br></div><div><br></div></div><div class="gmail_quote"><div><div>On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com" target="_blank">danny@debsinc.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">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.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Zohair Raza<br>
<b>Sent:</b> Monday, February 06, 2012 12:06 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] Playback with noanswer in AGI<u></u><u></u></span></p><div><div>
<p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">Hi All, <u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I want to play a file in agi but dont want to answer the call<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I am dialing through sip phone and running asterisk 1.8.6,<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div>
<p class="MsoNormal">I tried following with no luck<u></u><u></u></p></div><div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">$agi->exec("Progress");<u></u><u></u></p></div><div>
<p class="MsoNormal">$agi->exec("Playback $filetoplay,noanswer");<u></u><u></u></p></div></div><div><p class="MsoNormal">$agi->hangup();<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p>
</div><div><p class="MsoNormal">When I dial I can't hear the audio but if I answer the call or remove noanswer argument I can hear the audio.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div>
<div>
<p class="MsoNormal">phpAGI's stream_file didn't help either. <u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I ended up with ResetCDR() before hangup to reset billsec, duration and disposition but don't want to do it this way.<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">What could be the problem?<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">From Voip-info.org :<u></u><u></u></p>
</div><div><p class="MsoNormal"><strong><span style="font-size:9.0pt;font-family:"Verdana","sans-serif";background:white">noanswer</span></strong><span style="font-size:9.0pt;font-family:"Verdana","sans-serif";background:white">: 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.</span><u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Is it because the channel is not supported?<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">
<u></u> <u></u></p></div><div><p class="MsoNormal">Regards,<u></u><u></u></p></div><div><div><p class="MsoNormal">Zohair Raza<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">
<u></u> <u></u></p>
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