similar to: Using Windows Messenger+Video in *

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

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]

Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --------------- Wanna buy a duck?

Hi all, I tried making a video SIP call using Asterisk .... But it didnt work....only voice call works? Regards Faraj Khasib

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

I have success register polycom in to asterisk and it can called by other extension. But why it can't calling other extension ? and i have warning from asterisk chan.sip.c:3602 process_sdp: Unknown SDP media type in offer: application 49200 RTP/AVP 100 anyone undertand this warning ?

I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based

i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133333 at 216.82.224.202