I got an idea. If you only have 1 sip trunk, just do chanspy(SIP/)
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Ed Nu?ez
Sent: Thursday, September 27, 2007 10:17 AM
To: covici at ccs.covici.com; '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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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(wkwu at calltrol.com) 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: asterisk-users-bounces at lists.digium.com on behalf of Ed Nu?ez
> Sent: Wed 9/26/2007 4:48 PM > To: asterisk-users-bounces at
lists.digium.com
> 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: asterisk-users-bounces at lists.digium.com on behalf of Ed
Nu?ez<BR> > Sent: Wed 9/26/2007 4:48 PM<BR> > To:
asterisk-users-bounces at lists.digium.com<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>_______________________________________________
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--
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you spend it?
John Covici
covici at ccs.covici.com
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