Re: [asterisk-users] Newb Question
On 11/30/07, Vivek Shrivastava <[EMAIL PROTECTED]> wrote: > > you can try Cain & Abel ( to route calls) and Wireshark to record all the > calls. > > On 11/29/07, Adam Moffett <[EMAIL PROTECTED]> wrote: > > > > I'm pretty sure asterisk won't do that without modification. You'll > > need to do packet sniffing and decode the datathere may be products > > that do this, but asterisk is not it. > > > > And we're assuming the calls are unencrypted? > > > I inherited an office with phones that are hosted off-site. Everything > > > is skinny and G729. I see that the FreeBSD asterisk port comes with a > > > G729 codec. > > > > > > I want to record everything. If I use port mirroring on my switch, is > > > it possible to configure asterisk to record and assemble packets that > > > it doesn't otherwise route? Is it insane to user asterisk for this > > > purpose? Advice or a link to a howto would be greatly appreciated. > > > > > > > > > > > > > Thanks everyone. You can indeed use cain and abel to convert g729 to .wav (wireshark doesn't have that codec just yet), and it's easy enough to capture packets with tcpdump or wireshark. I've done this a few times as an experiment. >I suspect either you want to insert an Asterisk system in-between as a >"tap" (requiring re-configuring your phones and your outside provider) or >using a "voip sniffer" plugged into the management port of your Ethernet >switch. That's more or less it. I know how to duplicate the packets. What I want now is to automatically reassemble, decode and archive each rtp stream (every call) on this network of 20 users or so. There are open source applications for this but they don't have G729 support. Asterisk has G729 support and it can record calls (at least from the command line it appears to have that feature). All the pieces are there but I'm not even sure where to begin configuring it to do only that. Surely some adventurous soul has done this already. ___ --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
Re: [asterisk-users] Newb Question
you can try Cain & Abel ( to route calls) and Wireshark to record all the calls. On 11/29/07, Adam Moffett <[EMAIL PROTECTED]> wrote: > > I'm pretty sure asterisk won't do that without modification. You'll > need to do packet sniffing and decode the datathere may be products > that do this, but asterisk is not it. > > And we're assuming the calls are unencrypted? > > I inherited an office with phones that are hosted off-site. Everything > > is skinny and G729. I see that the FreeBSD asterisk port comes with a > > G729 codec. > > > > I want to record everything. If I use port mirroring on my switch, is > > it possible to configure asterisk to record and assemble packets that > > it doesn't otherwise route? Is it insane to user asterisk for this > > purpose? Advice or a link to a howto would be greatly appreciated. > > > > > > > > > ___ > --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
Re: [asterisk-users] Newb Question
I'm pretty sure asterisk won't do that without modification. You'll need to do packet sniffing and decode the datathere may be products that do this, but asterisk is not it. And we're assuming the calls are unencrypted? > I inherited an office with phones that are hosted off-site. Everything > is skinny and G729. I see that the FreeBSD asterisk port comes with a > G729 codec. > > I want to record everything. If I use port mirroring on my switch, is > it possible to configure asterisk to record and assemble packets that > it doesn't otherwise route? Is it insane to user asterisk for this > purpose? Advice or a link to a howto would be greatly appreciated. > > > ___ --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
Re: [asterisk-users] Newb Question
On Fri, 30 Nov 2007, ram wrote: > chan spy does the job i belive > > ram > > On Nov 30, 2007 7:37 AM, Jeff Adams <[EMAIL PROTECTED]> wrote: > >> I inherited an office with phones that are hosted off-site. Everything is >> skinny and G729. I see that the FreeBSD asterisk port comes with a G729 >> codec. >> I want to record everything. If I use port mirroring on my switch, is it >> possible to configure asterisk to record and assemble packets that it >> doesn't otherwise route? Is it insane to user asterisk for this purpose? >> Advice or a link to a howto would be greatly appreciated. Chanspy lets you listen to a channel. While it will record to a file, it would be a manual operation for every call. I suspect either you want to insert an Asterisk system in-between as a "tap" (requiring re-configuring your phones and your outside provider) or using a "voip sniffer" plugged into the management port of your Ethernet switch. Of course, I've done neither, so my advice is worth every penny you paid for it :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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
Re: [asterisk-users] Newb Question
chan spy does the job i belive ram On Nov 30, 2007 7:37 AM, Jeff Adams <[EMAIL PROTECTED]> wrote: > I inherited an office with phones that are hosted off-site. Everything is > skinny and G729. I see that the FreeBSD asterisk port comes with a G729 > codec. > I want to record everything. If I use port mirroring on my switch, is it > possible to configure asterisk to record and assemble packets that it > doesn't otherwise route? Is it insane to user asterisk for this purpose? > Advice or a link to a howto would be greatly appreciated. > > ___ > --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
[asterisk-users] Newb Question
I inherited an office with phones that are hosted off-site. Everything is skinny and G729. I see that the FreeBSD asterisk port comes with a G729 codec. I want to record everything. If I use port mirroring on my switch, is it possible to configure asterisk to record and assemble packets that it doesn't otherwise route? Is it insane to user asterisk for this purpose? Advice or a link to a howto would be greatly appreciated. ___ --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
Re: [Asterisk-Users] newb question regarding DTMF
On Tue, 24 Aug 2004, Erik Anderson wrote:
> On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
> <[EMAIL PROTECTED]> wrote:
> > x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to
> > transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
> > for your x-lite user.
>
> That's what I've read, and I have added dtmfmode=rfc2833 in my
> sip.conf...see this snippet:
>
> [xlite1]
> ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
> ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
> type=friend
> username=xlite
> callerid="Jane Smith" <5678>
> host=dynamic
> nat=yes ; X-Lite is behind a NAT router
> canreinvite=no; Typically set to NO if behind NAT
> disallow=all
> allow=gsm ; GSM consumes far less bandwidth than ulaw
> allow=ulaw
> allow=alaw
> dtmfmode=rfc2833
>
> I've applied that change and restarted asterisk, but no dice...
Dial the extension, then on the * CLI use 'sip show channels' to get the
name of the active channel. Next use 'sip show channel ___' to get info on
that particular channel (you can type the first few characters and use tab
completion; no need to type the whole string!). Scan through the output to
see whether asterisk is really using rfc2833 for that channel. If it is,
then the problem is likely in the x-lite config. If not, try moving
dtmfmode to the general section of sip.conf
Greg
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Re: [Asterisk-Users] newb question regarding DTMF
On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
<[EMAIL PROTECTED]> wrote:
> x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to
> transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
> for your x-lite user.
That's what I've read, and I have added dtmfmode=rfc2833 in my
sip.conf...see this snippet:
[xlite1]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=xlite
callerid="Jane Smith" <5678>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
dtmfmode=rfc2833
I've applied that change and restarted asterisk, but no dice...
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Re: [Asterisk-Users] newb question regarding DTMF
On Mon, 23 Aug 2004, Erik Anderson wrote: > Hello all - I'm just starting to play around w/ asterisk, and I've run > into a seemingly simple problem that has really manged to frustrate > me... > > I'm running the latest cvs version of *, and am trying to dial in to > the default extention 1000 demo using x-lite. I can dial and hear the > greeting no problem, but when I try and send any DTMF tones, I don't > get any response. Is there something specific I need to set in my > sip.conf to allow DTMF? x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to transmit inband). You need dtmfmode=rfc2833 in [general] or in the section for your x-lite user. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newb question regarding DTMF
Check the wiki for dtmfmode. It is explained here: http://voip-info.org/tiki-index.php?page=Asterisk%20sip%20dtmfmode -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Monday, August 23, 2004 7:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] newb question regarding DTMF Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there something specific I need to set in my sip.conf to allow DTMF? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newb question regarding DTMF
Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there something specific I need to set in my sip.conf to allow DTMF? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
