Hi!
> >>Oh right.. I remember seeing that.. yeah that looked a whole lot more
> >>elegant than *8. Why isn't it in HEAD?
> >
> >I'm not sure. Once it started getting some testing BKW closed it. If
> >someone is interested in testing the patch I'm sure the bug could be
> >reopened.
> >
> I'll te
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote:
> Peter,
> Do you know what the current status of app_intercept is?
No, not really. See below for the errors listed.
> I got it working on 1.0.2, but can't get it to complie on
> CVS-HEAD-01/26/05-02:14:44
> I get:
> app_intercept.c: In function `int
Peter Svensson wrote:
Perhaps the app_intercept patch would work better? It is a lot less of a
kludge and more flexible than the *8 that is in Asterisk.
Peter,
Do you know what the current status of app_intercept is?
I got it working on 1.0.2, but can't get it to complie on
CVS-HEAD-01/26/05-0
Peter Svensson wrote:
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote:
Oh right.. I remember seeing that.. yeah that looked a whole lot more
elegant than *8. Why isn't it in HEAD?
I'm not sure. Once it started getting some testing BKW closed it. If
someone is interested in testing the patch I
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote:
> Oh right.. I remember seeing that.. yeah that looked a whole lot more
> elegant than *8. Why isn't it in HEAD?
I'm not sure. Once it started getting some testing BKW closed it. If
someone is interested in testing the patch I'm sure the bug could be
Peter Svensson wrote:
On Wed, 9 Feb 2005, Matthew Boehm wrote:
I'm still not sure how to provide services that interact one phone with
another phone's RTP stream. Like call pickup. How can I pickup a call on
another asterisk server? Hmm Hm
Aw crap. I completly forgot about call pi
On Wed, 9 Feb 2005, Matthew Boehm wrote:
> > I'm still not sure how to provide services that interact one phone with
> > another phone's RTP stream. Like call pickup. How can I pickup a call on
> > another asterisk server? Hmm Hm
>
> Aw crap. I completly forgot about call pickup. Good poi
[EMAIL PROTECTED] wrote:
I think most people probably do something like:
AddQueueMember(techsupport|SIP/${CALLERIDNUM}), but I bet you can put
any valid channel name in there.
And you would win that bet :-)
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Matthew Boehm wrote:
I'm still not sure how to provide services that interact one phone with
another phone's RTP stream. Like call pickup. How can I pickup a call on
another asterisk server? Hmm Hm
-Brett
Aw crap. I completly forgot about call pickup. Good point. If you have a
call co
Matthew Boehm wrote:
Why not let asterisk be your PSTN GW? It is in our case, just throwing out
my $0.02.
Most of the cases I can think of I can get around. The one I can't seem to
figure out is 'Agents'.
Agents will need to login/logout using 1 number. I can forward that number
from SER to asteris
Matthew Boehm wrote:
If I can get re-invites working great, then I should have no worries about
inter-office communication. SER should be able to connect 2 office-mates to
eachother even if they are both behind the same NAT, or behind different
NATs.
You can accomplish that with a low-end box runni
> I'm still not sure how to provide services that interact one phone with
> another phone's RTP stream. Like call pickup. How can I pickup a call on
> another asterisk server? Hmm Hm
>
> -Brett
Aw crap. I completly forgot about call pickup. Good point. If you have a
call come into one of y
> [EMAIL PROTECTED] wrote:
>
> >
> > I think you might be missing the point here. SER is a raw SIP processor.
> > So for a second throw everything you know about Asterisk + SIP out the
> > window and go back to vanilla SIP. Getting used to a B2BUA in the call
> > path kinda beats some of the raw po
states and that seems difficult to accomplish.
-Matthew
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 08, 2005 4:33 PM
Subject: Re: [Asterisk-Users] SER Interaction: Agents and Ext
Michael Welter wrote:
SER newbie here. Why do you need Asterisk for Sip->SIP setup? And if
there is a reinvite, is that for the RTP stream only or for the SIP
transactions as well? Will you lose the BYE transaction if there is a
reinvite?
Also, how many SIP registrations do you expect to ma
[EMAIL PROTECTED] wrote:
I think you might be missing the point here. SER is a raw SIP processor.
So for a second throw everything you know about Asterisk + SIP out the
window and go back to vanilla SIP. Getting used to a B2BUA in the call
path kinda beats some of the raw power of SIP up. Think
Matthew Boehm wrote:
With all of these caveats, it seems to me that a SER->Asterisk solution
isn't that great. If anyone else out there can show me otherwise...
Thanks,
Matthew
I think you might be missing the point here. SER is a raw SIP processor.
So for a second throw everything you know abo
Hey gang,
I'm trying to work out all possible scenarios using SER & Asterisk in our
upcomming deployment. The example scenario is 50 different customers, all
with different numbers of SIP UAs. All UAs would register with SER; This
will help keep any inter-office conversations off our bandwidth sin
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