<div dir="ltr">Confirmed as well, played back with wireshark and audio was there but phone was ringing.<div><br><div>Thanks again.</div><div><br></div><div><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">
Hi,<div><br><div>Given invites seems fine, can you take a wireshark trace of the call on your eyebeam machine! from that wireshark trace use telephony calls options and hear if you are actually receiving RTPs on your system. If you could hear the played back sound file on your eyembeam machine . this would mean that your eyebeam client is not good enough to play media while its in 183 session progress.</div>
<div><br></div><div>Also can you send me the short sample php-agi script you are executing so i actually test this on my virtual machines as well.</div><div><br></div><div>Regards,</div><div>Sammy</div><div><div class="h5">
<div><br><div class="gmail_quote">
On Tue, Feb 7, 2012 at 1:09 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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