similar to: What is voice format 8

A few weeks ago, I lost my Trixbox that was all set up with Hylafax and IAXmodem. I am trying to set it up for email > procmail > faxmail > iaxmodem > asterisk >sipext > ATA (with attached fax). I have followed all the instructions on creating the IAX extension and configuring the IAXmodem config file. I can see the connection in Asterisk: iax2 show peers Name/Username

Can someone explain me this message: "chan_iax2.c: Ooh, voice format changed to ..." Where can I find a list of numeric codes used to identify voice format? Then, sometime I get an infinite loop of messages like these: DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1 WARNING[15015] channel.c: Unable to find a codec translation path from g723 to alaw DEBUG[15015]

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

We are using the following to record conversations. exten => _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP}) exten => _1XXX.,2,Monitor(wav,${CALLFILENAME},m) exten => _1XXX.,3,Dial(IAX2/4506:zj5S3A5a@nl.voipgate.nl/${EXTEN:1}) exten => _1XXX.,4,Congestion exten => _1XXX.,104,Congestion This was working previously to record both sides of the conversation but now

When I have HylaFAX answer a call redirected to the fax extension in Asterisk when it detects CNG, Asterisk hangs up: Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 is ringing Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 answered Zap/23-1 Jan 30 14:32:59 DEBUG[1098]: Ooh, voice format changed to 8 Jan 30 14:33:01 VERBOSE[1098]: -- Channel 0/23, span 1 got hangup Jan 30

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101

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> --

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

Hi there, do I have a local problem, or is registration at IAXTEL impossible at the moment? "iax2 show registry" permanently shows a TIMEOUT for 69.73.19.178. Philipp

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

Hello, I have anolog lines coming throug Dahdi to Asterisk Server, one of the anolog lines is used for fax line. I received fax fine without any problems using Iaxmodem with Hylafax Server. Outgoint fax is the problem, when IAXMODEM dial out using Dahdi channel, dahdi answers and start to dial the outside number however Iaxmodem thinks that dahdi is the remote fax machine and starts sending fax

Hi All, I'm having a strange problem and I'm not able to understand what's happening. I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine. They are linked together through localhost. I've turned qualify on for the iax peer. Periodically I've this message: [Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer: Peer 'iaxfax' is now

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message "Didn't get a frame from channel: SIP/3805-df43", but I can't figure why. asterisk logs:

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, I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7 running on the same box. I wonder how to setup the iax account correctly so that I may initiate outbound calls via iaxmodem? registration upon iaxmodem startup is okay and I can direct calls to it. -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874 But upon an outbound call setup request from

Hello. I'm having a strange problem with the IAX2 channel and IAXmodem and I was hoping to get some light from someone in these lists. On my logs and on the console I'm getting a bunch of lines with: [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now UNREACHABLE! Time: 3 [Oct 23 14:26:28] NOTICE[4413] chan_iax2.c: Peer 'XXX' is now REACHABLE!

Hi, we have a faxserver with Asterisk, IAXModem and Hylafax. Faxes come from a SIP trunk to Asterisk, then are forwarded throught 5 IAXModems managed with Hylafax. Hylafax users can also send faxes to these modems and Asterisk send them throught the SIP trunk. We also have a dedicated modem used only for sending faxes coming from an Hylafax dedicated user. Sometimes Hylafax reports that a modem

Has anyone seen an issue with SIP phone (polycom 500) dropping calls at irregular intervals with no errors in the asterisk log files? I am having this issue as described and it is a complete pain in my rear to trouble shoot because when I call my cell phone I can get a call to last over 30 minutes yet when I call another office that uses a standard pbx I can't get past 10 minutes. For some