similar to: Video support SIP and IAX2

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 am trying to set up an IAX2 trunk between two servers. Server A has the following in iax.conf: [general] ... [ServerB] type=friend trunk=yes host=dynamic secret=myPassword context=myContext Server B has the following in extensions.conf: [outgoing] exten=>_40X,1,Dial,IAX2/ServerB:myPassword@x.x.x.x/${EXTEN} I am using bmtools to monitor the bandwidth usage, and I am not seeing a difference.

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

Hi Nicolas, There doesn't seem to be any interest in using asterisk and video. I posted a $1,000 bounty to get video meet me working without a single reply. I have now just bumped this to $2000 http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid eo+conferencing This is a legitimate commercially binding bounty, I hope this might inspire some people to develop at least

Does IAX2 support a video call ? Jerry

Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the "v" flag on my extension for the meetme app? Thanks, Tim

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

Hi, I'm still hassling with the consultative/attended transfer stuff. Someone please help me identify this A lot has already been said about the ATA186. Some report it works fine, others say it doesn't. Lets get clarity on this. My scenario is reasonably simple (I think) Phone A: SIP/video1 Phone B: SIP/werkkamer Phone C: IAX2/provider Phone A calls phone B, they chat: *CLI> show

Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user

I'd like to cobble together a videophone from an analog phone, connected to an Asterisk FXS channel, and a co-located video camera, connected to a video grabber card on the Asterisk server (so I have a Linux video device providing the video stream). When a call is made from the phone, I'd like to somehow add the video and produce a SIP video call. I don't want to use any sort of

I'm using DUNDi. My lookup returns 'IAX2' for the tech, and 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for the destination. How do I dial this? I've tried dialling it with: "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" passed from my AGI script, but the other endpoint (xxx.187.142.204) is returning: May 11 09:23:41

Hi, I came back from a quiet weekend today and found my E400P box to have gone astray. Asterisk is loaded from inittab, and started crashing and reloading a couple of thousands of times, each time notifying my monitoring service :-P I remember there would be issues on old cvs stuff since the crash at digium so I made a clean checkout just now. Here is what happens when I load manually:

Hi all, DIAX 0.9.4 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. What's new in 0.9.4: - IAX2 support (new DLL); - selectable DSP: Echo cancellation, AGC, Denoise; - plaintext and md5 authentication supported; - the phonebook is now in a separate

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

I've added an important new application: app_transfer. This application is designed to allow Asterisk to request the transfer of an incoming call to a different extension. Consider the following diagram: Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App A caller calls an extension on PBX1 which forwards to PBX2. PBX2 executes app_transfer, which requests that hte

Apparently we've been having calls sporadically drop. We're using an IAX outbound trunk and SIP adapters on the inside. Below is a log excerpt detailing one of the calls which dropped, and it looks largely normal to me except for this: Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

I have a couple of Asterisk 13.4 servers with an IAX2 trunk between them: [phone1] <--pjsip--> [server1] <--iax2 trunk--> [server2] <--pjsip--> [phone2] With this setup, audio calls work fine, but video doesn't work: I get a black window and if I remember correctly, I was getting white noise in one direction (not sure if the noise thing is asterisk's fault or the

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Hi! I'm using asterisk 1.4.17 with twinkle and a custom phone based on iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf. While the twinkle client is able to initiate an attended transfer using *2 (as configured in features.conf), the iax client is not. I can see the DTMF messages showing up on the asterisk console, but asterisk does not invoke the features