I got an idea. If you only have 1 sip trunk, just do chanspy(SIP/) -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Thursday, September 27, 2007 10:17 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy issue
Good point, but the deal is that I have a remote call center with their own Nortel PBX. I get these calls from my DID provided via Zap and I send them VoIP to the gateway connected to the Nortel PBX. This is what I refer to my SIP trunk. When I specify Sip/SIPTRUNK (SIPTRUNK) is the name of the trunk. Asterisk only monitors one call at a time in the whole trunk, and you can Cycle through the calls by pressing "*". -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Wednesday, September 26, 2007 8:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy issue I am not an expert on chanspy, but it seems to me spying on the trunk would not work very well, would not you hear multiple conversations mixed if more than one extension were calling? Seems best to me to spy on an extension. YOu also can do a show channels to see who is talking to whom. on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote > The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by "sip trunk". It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses "SIP/extension+xxx" as the channel name of the call. > > > -----Original Message----- > From: [EMAIL PROTECTED] on behalf of Ed Nuñez > Sent: Wed 9/26/2007 4:48 PM > > To: [EMAIL PROTECTED] > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] ChanSpy issue > > > > Hello list > > > > > I am having an issue with Chanspy/SIP that I'm hoping someone has come > > across and resolved in the past. > > > > I am sending calls that come in TDM through T1 ZAP channels and go out to a > > SIP trunk. > > > > If I spy on the SIP channel, I can hear the person on the SIP side of the > > call just fine, but the person on the ZAP channel fades in and out. > > If I spy on the ZAP channel, and can hear both sides just fine, but I don't > > know who I am spying on since I have other calls coming in on the same T1. > > > > If I spy on a SIP extension instead of a SIP trunk, I hear both sides just > > fine. > > > > I am using a recent version of Asterisk 1.2 and I am using g729 licenses. > > > > This is the command I am using to spy. > > > > exten => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) > > > > > > > > > > > <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN"> > <HTML> > <HEAD> > > <META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1"> > > <META NAME="Generator" CONTENT="MS Exchange Server version 6.5.7638.1"> > > <TITLE>RE: [asterisk-users] ChanSpy issue</TITLE> > </HEAD> > <BODY> > > <!-- Converted from text/plain format --> > > <P><FONT SIZE=2>The > parameter to Chanspy should be the whole or part of the channel name. I do > not understand what you mean by "sip trunk". It make perfect sense > that you can hear both streams of voice when you use the phone's extension > as Asterisk usually uses "SIP/extension+xxx" as the channel name > of the call.<BR> > <BR> > <BR> > -----Original Message-----<BR> > From: > [EMAIL PROTECTED] on behalf of Ed Nuñez<BR> > Sent: Wed 9/26/2007 4:48 > PM<BR> > To: [EMAIL PROTECTED]<BR> > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'<BR> > > Subject: Re: [asterisk-users] ChanSpy issue<BR> > <BR> > <BR> > <BR> > > Hello list<BR> > <BR> > <BR> > <BR> > I am having an issue with > Chanspy/SIP that I'm hoping someone has come<BR> > across and resolved in > the past.<BR> > <BR> > <BR> > <BR> > I am sending calls that come in TDM > through T1 ZAP channels and go out to a<BR> > SIP trunk.<BR> > <BR> > > <BR> > <BR> > If I spy on the SIP channel, I can hear the person on the > SIP side of the<BR> > call just fine, but the person on the ZAP channel > fades in and out.<BR> > <BR> > If I spy on the ZAP channel, and can hear > both sides just fine, but I don't<BR> > know who I am spying on since I > have other calls coming in on the same T1.<BR> > <BR> > <BR> > <BR> > If > I spy on a SIP extension instead of a SIP trunk, I hear both sides just<BR> > > fine.<BR> > <BR> > <BR> > <BR> > I am using a recent version of > Asterisk 1.2 and I am using g729 licenses.<BR> > <BR> > <BR> > <BR> > > This is the command I am using to spy.<BR> > <BR> > <BR> > <BR> > exten > => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))<BR> > <BR> > <BR> > <BR> > <BR> > <BR> > <BR> > <BR> > <BR> > <BR> > <BR> > </FONT> > </P> > > </BODY> > </HTML>_______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
