similar to: IAX timeout if no ringing

Hello, I've got some SIP clients, and an IAX2 long distance provider. Ideally, when a the dialed number is busy I will hear a busy signal. Instead, I get Congestion even though * knows it's busy. Is this a bug or am I missing something? The dial plan, in basically this Dial(IAX2/user@provider/19995551234,,) Goto(failedcall-${DIALSTATUS}) failedcall-CONGESTION plays congestion

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring (starts sounding normal, then sounds "weird" -- like two rings played at

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

This message was just bounced back to me. I am not sure if it made it to the list originally or not, as I received no responses. Since this message was written, I have installed Zap hardware into this server. The Zap channels can be transferred to the Meetme conference. The IAX2 calls still cannot. Any suggestions will be greatly appreciated. Sincerely, Trevor Hammonds Trevor G.

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Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

Hi, We have to servers set up in two different networks. We are able to connect calls via IAX and they work perfectly. We do not see caller ID from clients on either side. Our Grandstream phones say Eri and our XTen phones say Asterisk. We did a debug and I am pasting the output from both servers below. We tried setCallerId in several different ways. We see the value get passed to the

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

I have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David

Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Format for call is ulaw Feb

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

Hi - I'm running CVS-HEAD from 2005-08-11 20:17:17 UTC, and I'm trying to set up some redundancy on IAX connections between locations. I have two IAX peers set up that work correctly by themselves: "ast551-out" and "ast551-out-backup": [ast551-out] type=peer secret=secret username=ast551 host=X.X.X.X qualify=1000 disallow=all allow=gsm allow=ulaw trunk=no

Russell, What kind of zap cards do you have?? If T1, is it PRI or RBS -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Russell Horn Sent: Wednesday, December 22, 2004 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Problem ringing simultaneous channels I have a

Hi, testing a new fwd account, dialling from sip4030 to my FWD number, sip4021 rings as defined in extensions conf. Why is this happening twice? -- Executing SetCallerID("SIP/4030-a7f2", ""HTCAS"") in new stack -- Executing Dial("SIP/4030-a7f2", "IAX2/617533:xxxxxx@iax2.fwdnet.net/617533|60|r") in new stack -- Called

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

The setup: Inc SIP Call -> Asterisk 1 -- IAX --> Asterisk 2 -- SIP --> phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto("SIP/XX.XX.XX.XX-0819f590", "cytel-internal|3044|1") in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial("SIP/XX.XX.XX.XX-0819f590",

When Asterisk dials an IAX destination with no registration, it very quickly comes to the conclusion that it can't make the call -- Executing [500@default:2] Dial("Zap/1-1", "IAX2/guest@misery.digium.com/s@default") in new stack -- Called guest@misery.digium.com/s@default [Jan 29 21:43:15] NOTICE[1957]: chan_iax2.c:2686 __auto_congest: Auto-congesting call due to

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows: