Yes, Thanks
Regards, Zohair Raza On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind <[email protected]> wrote: > Exactly that's what I expected. > Great - now have fun > > > On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza > <[email protected]>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 < >> [email protected]> 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:[email protected] SIP/2.0 >>> To: <sip:[email protected]> >>> From: Zohair<sip:[email protected]>;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:[email protected]: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:[email protected] >>> ",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:[email protected]: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:[email protected]>;tag=7f222672 >>> To: <sip:[email protected]> >>> 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:[email protected]:5060> >>> Content-Length: 0 >>> >>> >>> <------------> >>> -- Executing [100@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:[email protected]>;tag=7f222672 >>> To: <sip:[email protected]>;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:[email protected]: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 <[email protected]>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 < >>>> [email protected]> wrote: >>>> >>>>> Thanks for this explanation Dany! >>>>> >>>>> Regards, >>>>> Zohair Raza >>>>> >>>>> >>>>> On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas <[email protected]>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:* [email protected] [mailto: >>>>>> [email protected]] *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 >>>>>> >>>>>> 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 >>>>> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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
