similar to: Enhanced Voicemail Features + IAX

We have a running Asterisk on a small server (RH 9.0) und were able to make calls via SIP. Since the quality was rather poor (one of us has only a 150/75 kbps DSL connection, iLBC did not work(?)) we tried to setup clients (FireFly) and Asterisk for connections via IAX. In some point seems to be a mistake. Asterisk says: -- Executing Dial("IAX2[83022777@83022777]/2",

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Hello List, On my cvs-head (29-Dec) asterisk, my sip phones can use *78 for DND and *79 to turn it off. With my firefly (iax) client, I am getting the following errors if I try these feature codes: Jan 10 13:26:18 NOTICE[11702]: chan_iax2.c:5792 socket_read: Rejected connect attempt from xx.xxx.xxx.xxx, request '*78@internal' does not exist Jan 10 13:26:23 NOTICE[11702]:

Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register => FWDNUMBER:PASSWORD@fwd.pulver.com/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when

I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hi.... I'm having a extrange problem.... I cant register with Iaxtel or call to digium... But i cant make or recive IAX calls... ( I made some one with irc users ) Any idea why? At my logs i have this from iaxtel: NOTICE[196621]: File chan_iax2.c, Line 2832 (register_verify): No registration for peer 'xmarts' (from 192.168.0.11) NOTICE[196621]: File chan_iax2.c, Line 4389

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes

Hello, I updated CVS and nobody can call me any more with my IAX number 17007591228. I can only call other number but nobody can call me. This is what I get on debug when I call myself: -- Executing Dial("SIP/1011-7424", "IAX/bartosz:password@iaxtel.com/17007591228@iaxtel") in new stack -- Calling using options

Hi, I am problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server it says It with C in iax and Server B it speaks with D in iax, but Server it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you authenticate sipspo

FYI from my putting SVN-trunk briefly into production for stress-testing. Fresh checkouts from the last two days, most recently SVN-trunk-r30465, begin to show lots of errors like those appended below, after about an hour of operation. > May 26 00:29:51 NOTICE[4418]: chan_iax2.c:5001 register_verify: No registration for peer 'hackley' (from 10.17.1.2) > May 26 00:30:41

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

Hi guys, I'm using asterisk 1.2.1 since a week ago or so. today I found it crashed when making a call through teliax. This is how it looks: -- Called xxxxxxxxx@teliax/17075471770 -- Call accepted by 208.139.204.245 (format ulaw) -- Format for call is ulaw Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

Hello all I have looked in Google for a fix for this issue but I have had no luck, so I am posting it here. Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting causecode to be single byte but was 35 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting samplerate to be 2 bytes long but was 40 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625

The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below

To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian

hello all... newbie question: I have FireFly setup on my laptop and I would like to test this out using IAX in this scenario: FireFly Softphone > Asterisk > TDM Gateway i do not wish to use this on the firefly network, but simply within my own "3rd party" network as the website and setup of FireFly defines it... does anyone have a sample iax.conf and extensions.conf i

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------: