Displaying 20 results from an estimated 700 matches similar to: "video via IAX or SIP"
2017 May 12
3
pjsip: asterisk can't decide which codec to use
Hello!
I'm facing completely choppy sound. The wireshark trace shows, that
there are a lot of codec changes without any trigger (means no options
or reinvite or any other package).
Background:
The call is initiated by asterisk and is received by the same asterisk
conference room via
Phone extension -> asterisk -> provider A -> provider B -> asterisk.
Asterisk initially sends
2005 Oct 05
0
call transfer problem - something strange
Hi,
I try to set up planet VIP-050 with asterisk. Everything works fine
instead of the call transfer. When I press # console says something
like this:
>Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh,
format changed >to 1024
>Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein:
Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP
2004 Jul 08
0
Problem SIP no audio just noise
I'm trying to call from XLite phone to PSTN
(I've tried this from internet and from local network the same)
The Xlite doesn't write that it is connected but receives excelent audio.
At the other end comes only noise. Some times only for a second you can
here the
caller voice , but this was only one time :)
I saw with ethereal that UDP packets are coming and going to the
asterisk
2013 Sep 10
1
No remote address on RTP instance - On Ringing
Hello Everyone,
I have a new problem where when placing the call, asterisk will
automatically go into music on hold until the call is connected (ie,
no ringing). It was kind of confusing because sometimes `SESSION
PROGRESS` takes longer than others, during this time we are in MOH.
The call does eventually connect and the MOH stops. When debugging I
saw the following debug message:
[Sep 10
2010 Aug 04
1
Asterisk not working with Festival
Hello,
I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk
1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine
with SIP channels without Festival. I have written following context in
extension.conf:
[connect-to-me]
exten => s,1,Answer
Exten => s,n,SayDigits(?1?)
exten => s,n,Festival(hello john)
exten => s,n,Hangup
I use call files to
2003 May 27
1
chan_h323 + Ericsson Webswitch 100
I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk.
Has anyone gotten a Webswitch running? When I try to connect asterisk
thinks everything works fine, while the webswitch just rings. I belive
chan_h323 is picking the wrong port to talk at the webswitch on, however
I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump
is attached to show the session.
2005 Jan 25
0
coredumping on MusicOnHold
Hello,
I have upgraded to 1.0.4 version of asterisk. After that asterisk crash
every time
On receiving an call from iax2 trunk to musiconhold application. SIP
calls to
MusicOnHold is however working. I already upgraded to 1.0.5, but the
problem still
Remainig.
Any idea ?
Iax2 : call proceding :
Jan 25 17:29:40 DEBUG[9997]: pbx.c:1261 pbx_extension_helper: Launching
'WaitMusicOnHold'
2014 Mar 21
1
ast_writefile: No such format 'h261', yet h261 is the only video format that works.
Built asterisk 11.8.1 on a Debian VPS. Testing using ekiga.
If h261 is checked in ekiga's video format list I have video, and
mouse over the video window shows it to be using h261.
But then I get the following lines a dozen or more times in the CLI:
[Mar 21 16:25:32] WARNING[31818][C-00000010]: file.c:1241
ast_writefile: No such format 'h261'
The problem is that I can't seem to
2004 Oct 03
0
Call gets disconnected upon connect
Hi Everybody,
I am trying to use SIP (Sipura 2000) to connect to Asterisk which then
dials out a local number using the Digium E100P. We have purchased the
G729 codec licenses from Digium and loaded them into Asterisk
successfully. However, the call drops immediately after being answered
with the debug error message saying something like: "channel.c:2646
ast_channel_bridge: Didn't get a
2004 Aug 12
1
AgentLogin issue
Hi
i have an issue getting agentLogin working
/etc/asterisk/queues.conf
member => Agent/1001
member => Agent/1002
extension.conf
exten => 110,1,Wait,1
exten => 110,2,AgentLogin()
now, i call 110 by a firefly client, trying to login in as 1001 agent:
Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060>
--
2004 Aug 29
0
Asterisk H.323 channel...
Hi all,
I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2).
So far I have been using the H.323 channel included in the tarball (Nufone ?).
I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box :
=====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the
2010 Mar 26
1
problem with polarity reverse
Hi,
I have a problem with polarity reverse on answer
I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and analog card is Sangoma a400 with fxo ports
this is my config
[trunkgroups]
2005 Aug 02
0
Hang up as soon as other party picks up call
Hello,
I have an Asterisk box with a TE410P connected to a PRI line and agents with
X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I
make outbound calls it hangs up as soon as other party tries to picks up the
call. Does someone ever experienced this situation? On X-Lite, only
G711-ulaw is enabled and here is what i put in sip.conf:
[4001]
type=friend
username=4001
2010 Dec 06
1
Asterisk 1.6.2.10 video call
Hello list,
I'm trying to set up a video call from my Ekiga client to a Grandstream
GXV3140 IP-phone. The call succeeds but there is no video.
I have in sip.conf :
videosupport=yes
disallow=all
allow=alaw
allow=g726
allow=g729
allow=gsm
allow=h261
allow=h263
allow=h263p
allow=h264
The Grandstream peer has codecs (sip.conf) :
gsm;alaw;g729;h261;h263;h263p;h264
The Ekiga peer has codecs
2006 Mar 21
1
SIP video voicemail problem
Hello all,
I am trying to leave a video voicemail but am unable to do so. I am using
Ekiga (formerly Gnomemeeting) to make a SIP connection to Asterisk 1.2.4.
Ekiga supports h261 for video.
The call connects and negotiation seems okay. When I leave a message,
however, only the audio is recorded. Looking in the log file afterwards I see
many messages like this:
Mar 21 22:02:34 WARNING[2418]
2010 Oct 11
1
Unable to find a codec translation path from ulaw|h261 to slin
I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've
encountered a problem playing back a .wav file to an Ekiga client:
My dialplan looks like:
exten => 730,1,answer
exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign)
exten => 730,n,hangup
Sovereign.wav is a .wav file that plays nicely on my 1.4 server.
Here is what the console displays:
2003 Jun 18
0
MP3Player and Ringing (long)
[I'm reposting this to the asterisk-users list, since it seems to be a
bit more active.]
Hello,
I started messing with Asterisk few days ago, so my overall knoledge
about it is still fairy superficial.
I think I found an issue with MP3Player; it can be reproducted with this
extension:
exten => 6001,1,Answer
exten => 6001,2,Background(blahblah)
exten => 6001,3,Ringing
exten =>
2004 Aug 24
2
Voicepulse incoming / dial extension
All:
I am trying to use Voicepulse as my incoming line and want the caller to
simply dial the extension of the party they want to reach.
Here is my problem:
- the first time they dial it works fine and I see the
following on my console
Aug 24 23:14:31 DEBUG[-1126876240]: chan_sip.c:4408 build_route:
build_route: Contact hop: <sip:6035057098@66.234.228.137>
--
2004 Aug 26
0
Asterisk media problem behind NAT
Hello All,
I have a media problem while using sip communicator
user agent with asterisk behind NAT.I had enabled the
debug mode in asterisk and capture the results.I have
attached the results with this mail.Can any one help
me to fix the problem?
Thanks in advance,
Partha
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2004 Dec 14
1
SIP and Windows Messenger
I'm trying to get two Windows Messenger clients to communicate with
video and audio though asterisk. I'm running into one of two problems.
I get garbled audio under the current config. I had another config
where I could get a voice call to work but using video would cause the
caller to get music on hold. (very odd)
Calling a phone hanging off of an TDM the audio works great. This is