I had problems when I was playing with the ExtenSpy command as well. The issue for me was that the context for the extension that I was using was not the same as the one that Asterisk showed in the console output when I called the phone. This is because I have various contexts included in other contexts so it was a bit confusing as to which context the extension was in at some given point in time.
After changing things to match contexts stuff worked as expected. -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ > From: Marco Signorini <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Date: Thu, 06 Nov 2008 10:38:11 +0100 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] ExtenSpy? am I doing it correctly? > > Hi Steve. > I'm still trying the same because I'm interested in the subject. > For what I can understand the ExtenSpy application is working properly > if the selected extension receives a call. Seems not working, instead, > if the selected extension originates the call. > My actual setup is like that: > > Ext12(Soggiorno) <==> Ext13(Camera) > ^ > | > Ext911-> ExtSpy(12) > > Here is the log when the 13 calls the 12 and 911 is called by an other > phone (StudioAV): > -- Executing [EMAIL PROTECTED]:1] Dial("SIP/Camera-08231e60", > "SIP/Soggiorno") in new stack > -- Called Soggiorno > -- SIP/Soggiorno-082560f8 is ringing > -- SIP/Soggiorno-082560f8 answered SIP/Camera-08231e60 > -- Packet2Packet bridging SIP/Camera-08231e60 and SIP/Soggiorno-082560f8 > -- Executing [EMAIL PROTECTED]:1] ExtenSpy("SIP/StudioAV-0822f350", > "12") in new stack > -- <SIP/StudioAV-0822f350> Playing 'beep' (language 'it') > -- <SIP/StudioAV-0822f350> Playing 'spy-sip' (language 'it') > == Spying on channel SIP/Camera-08231e60 > > Unfortunately, in the opposite direction: > > -- Executing [EMAIL PROTECTED]:1] Dial("SIP/Soggiorno-0822f350", > "SIP/Camera") in new stack > -- Called Camera > -- SIP/Camera-08231e60 is ringing > -- SIP/Camera-08231e60 answered SIP/Soggiorno-0822f350 > -- Packet2Packet bridging SIP/Soggiorno-0822f350 and SIP/Camera-08231e60 > -- Executing [EMAIL PROTECTED]:1] ExtenSpy("SIP/StudioAV-082560f8", > "12") in new stack > -- <SIP/StudioAV-082560f8> Playing 'beep' (language 'it') > == Spawn extension (from-sip, 911, 1) exited non-zero on > 'SIP/StudioAV-082560f8' > == Spawn extension (from-sip, 13, 1) exited non-zero on > 'SIP/Soggiorno-0822f350' > > The application ExtSpy seems to hang just before playing the 'spy-sip' > and I can't hear anything coming from the selected extension. > > I'm using Asterisk version "Asterisk 1.4.20.1 built by root @ Gateway on > a i686". > Is this the correct behavior or a bug? > > Thank you and best regards. > Marco Signorini. > > Steve Gladden wrote: >> Scratching my head and trying this. >> Asterisk Version: Asterisk 1.4.21.2 >> >> Tried: >> exten => 4771,1,ExtenSpy([EMAIL PROTECTED]) >> exten => 4771,2,Hangup >> >> Also tried: >> exten => 4771,1,Answer >> exten => 4771,2,ExtenSpy([EMAIL PROTECTED]) >> exten => 4771,3,Hangup >> >> Also tried many variations including option ,b >> I think most calls I make are 'bridged' >> extensions 4771 and 4724 are both in mbb context. >> Tried 'cycling' though the channels and volule "*" "#" no change. >> >> Test: >> 4724 places outbound or extension call. >> I dial "4771" from 4772 >> I expect to hear audio from 4724's in progress call but hear nothing. >> I hear a recording "beep" when I dial 4771. >> I expect to hear audio from call being made from ext. 4724 >> I've obviously got this wrong or the feature is not working :-) >> >> Ao far I've been unable to find much information on the net of anyone >> documenting >> a problem or a working configuration. >> Is there something I'm completely missing here? >> >> Thanks! >> >> Steve >> >> >> >> > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
