Displaying 20 results from an estimated 10000 matches similar to: "app_conference compile?"
2005 Jul 21
1
account code missing in csv cdr
My cdrs are missing accountcodes for incoming calls from other asterisk
servers..
I've seen a few people mentioning this on the list and the solution
seems to be setting up a dialplan for incoming calls from a particular
sip peer.. in my opinion this does not scale well at all and I am
looking for a solution to correct this problem.
example sip peer:
[asterisk_gw]
type=friend
2005 Mar 16
2
t.38 support news?
Maybe I've missed it but I'm wondering if there has been any movement
towards getting t.38 support into asterisk.. has there been any news?
Where is t.38 support at? will it even happen?
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2005 Aug 01
4
test message - ignore me
Haven't seen email since the 29th.. just testing.
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2004 Dec 01
2
voicemail cuts off / hangs up
I'm having a problem with voicemail where the system will allow me to
login to the vm box no problem but when it starts tell tell me the
number of messages I have it hangs up.. I get "you have" and it dies
right there.. I'm running cvs tag v1-0.. what might be causing this?
I looked through my mail list archive and didn't notice anything like this..
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2005 Jul 13
0
tiny audio drops (blips)
We are receiving multiple audio drop outs on calls .. I've done quite a
bit of troubleshooting and it only involves calls that require the
Dial(SIP/xxx,,t) for transfers.. as long as the media path goes through
the server the audio blips happen.. using ulaw codec, btw. I have been
able to align the blips in audio to a specific point involving
asterisk.. it seems to happen right at about
2005 Oct 18
1
sip rfc bye violated?
I have this in sip show history for a particular channel marked as dead
(should be removed) in sip show channels:
1. TxReqRel INVITE / 102 INVITE
2. Rx SIP/2.0 / 102 INVITE
3. CancelDestroy
4. Rx SIP/2.0 / 102 INVITE
5. CancelDestroy
6. Unhold SIP/2.0
7. Rx SIP/2.0 / 102 INVITE
8. CancelDestroy
9. Unhold SIP/2.0
10. Rx
2005 Aug 02
0
codec question
I'm looking for opinions on g726-32 vs. g711u..
They both have decent audio quality.. and looking at the wiki I get the
impression that g726 is like the little brother to g711. Yet, I've run
into quite a few sip termination vendors who don't support it. Does
anyone on the list actively use g726 for anything and what have those
experiences been?
The g726 codec for me at least
2006 Jun 04
1
Help with compilation of app_conference in x86_64
Any C gurus out there that can tell me if this code compiled ok to be
used in x86_64 (Pentium Dual Core). It's for the app_conference
application.
Im using Centos 4.3 x86_64
kernel: 2.6.9-34.ELsmp
libgcc-3.4.5-2
gcc-3.4.5-2
after the compilation part is the makefile
************begin compilation*******************
[root@centos app_conference]# make clean
rm -f *.so *.o app_conference.o
2005 Jun 29
1
App_conference in dial plan?
Hi all,
I've been trying to get meetme working for a while now (complie problems
- will probably try again later on another machine) but have given up
and started looking at alternatives.
I've managed to get app_conference compiled and installed - show modules
shows its there in asterisk, but I don't know how too actually use it in
the dial plan...
The info on voip-info
2005 Jul 05
1
app_conference, CVS HEAD, SIP and Xen
I have Asterisk running in Xen virtual machines. Unfortunately, this
kind of virtualization makes a real time clock impossible, which in turn
makes ztdummy or a Zaptel driver impossible to load, which also makes
MeetMe conferences impossible.
As an alternative, I have downloaded, patched, compiled and installed
the app_conference source code against the headers in Asterisk CVS HEAD.
I can load
2006 May 29
4
app_conference DTMFs?
We need to conference together a call center agent, a customer and a third
party IVR and send DTMF tones from the agent to the IVR.
MeetMe has been eating our DTMFs so we'd like to try app_conference.
Has anybody setup such a configuration in app_conference and how did you
configure it?
-HJC
2005 Aug 02
1
stale nonce
I just updated one of my stable asterisk systems to head to test it
out.. and I'm receiving a interesting log message now in asterisk..
Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce
received from '<sip:3034585725@voip.livewirenet.com;user=phone>'
(one line per registration)
I'm using an AudioCodes mp108.. it worked fine with the latest stable..
2006 Jan 31
0
app_conference(Asterisk) with Speex
jonathan blais wrote:
> I'm using Linphone. I tested with Asterisk and Speex only, I created a
> channel with echo and it worked. It seems to have problem when using
> app_conference.
If you just use app_echo, then asterisk won't be trying to decode your
frames; it will just be sending them back to you. Therefore, if your
client is using an incompatible packing of the
2006 Jan 31
0
app_conference(Asterisk) with Speex
Jean-Marc Valin wrote:
>Just curious, how does Asterisk pack Speex frames in a packet. AFAIK,
>Linphone just sends raw packets, as specified in the RTP draft.
>
>
Asterisk expects speex frames to have a terminator. The phone I was
referring to was the X-Ten/X-Lite phones, which seemed to be adding
something _before_ the speex data to indicate the length of the frames.
2006 Jan 31
2
app_conference(Asterisk) with Speex
I'm using Linphone. I tested with Asterisk and Speex only, I created a
channel with echo and it worked. It seems to have problem when using
app_conference.
Jonathan
2006/1/31, Steve Kann <stevek@stevek.com>:
>
> jonathan blais wrote:
>
> > Hi,
> >
> > Does anyone ever used Speex with app_conference in Asterisk ? I'm
> > having a hard time to figure
2006 Jun 03
4
Meetme versus app_conference
As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.
Does the app_conference application works the same way?
Or if i have SIP/g729 users and i create a conference with other users
also at g729 asterisk will not transcode (when using app_conference)?
2005 Jul 06
2
app_conference and AGI
Hi,
i was successful in compiling app_conference and setting up an
conference was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from
inside the conference. So, if i dialed into an conference i want to be
able to press '*' and then the actual discussion is muted for me and i
and menu is read to me. Something like the ${MEETME_AGI_BACKGROUND} in
MeetMe.
2006 Jan 31
2
app_conference(Asterisk) with Speex
Hi,
Does anyone ever used Speex with app_conference in Asterisk ? I'm having a
hard time to figure why I always get this error "warning: Invalid mode
encountered: corrupted stream?".
Jonathan Blais
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2006 Jan 31
2
app_conference(Asterisk) with Speex
Just curious, how does Asterisk pack Speex frames in a packet. AFAIK,
Linphone just sends raw packets, as specified in the RTP draft.
Jean-Marc
Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit :
> jonathan blais wrote:
> > I'm using Linphone. I tested with Asterisk and Speex only, I created
> > a channel with echo and it worked. It seems to have problem when
>
2008 Sep 13
0
app_conference
Dear,
I am using app_conference, 2.0.1, with asterisk 1.4.
only a problem, if one of callers, disconnects the line, all of callers will be disconnected.
and conference room will be removed.
where is the problem ?
best
Mani
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