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
