similar to: IAX Native Transfer

Hi; I've just had a couple of discoveries in the learning process that are really making me impressed with this software. 1) IAX transfer - I'm running Asterisk boxes A, B, and C. B is in the middle and has a dialplan that points to extensions on C. When a client on A in the proper context on B tries dialling a client of C, B is smart enough to release itself somehow from playing

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

Hello; I'm curious where I can find a good document describing how to weave together some servers in different places. Trying to keep things as simple as possible here, I don't understand how to get 2 way calling going on between clients connected to separate servers. First, I have 3 asterisk servers running. One is my firewall here in Buenos Aires, Argentina. I have two others in

I am having problems getting any form of call transfer working. I have reconfigured blind transfers to be #1 and assisted transfers to be *2 but these are not working. Looking at the wiki (http://www.voip-info.org/wiki-Asterisk+cmd+Transfer) it it does not mention IAX so I assume I have to use the native IAX transfer supported by Diax? I have tried using Diax but am getting a problem that after

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

Dear all, I've been taking my baby-steps toward setting up an Asterisk phone system in my office, as also between my home and office (connected by DSL). I'm have a rough time getting two * boxes talk IAX over a LAN. I don't know what I am doing wrong, but am attaching my iax.conf and extensions.conf on both the boxes. Does anyone see it? ------config files start------ site-0

Hi; I've been playing around with various report writers on the cdr_mysql table and I'm getting rather irritated. I started a thread a couple weeks ago on "Billing Fun" and got a lot of good responses on how to get an up-to-date copy of the NDA/NXX database, which I managed to retrieve and bring into MySQL in a script from the various pieces of code people posted... very nice.

I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server

I donwloaded two IAX Clients (firefly and IAX phone) and they did register with *. It would make authenticated calls, but wouldn't actually register with the server. When I start the IAX Client the CLI show me the message: -- Registered '2004' (AUTHENTICATED) at 192.168.199.69:4569 After 5s: May 21 17:24:41 NOTICE[1133742896]: chan_iax2.c:5035 iax2_poke_noanswer: Peer

Hello, I have asterisk setup with Broadvoice. It works great as PBX and Outgoing calling server for all local clients withing 192.168.1.0 network. My asterisk is running over NAT. I use linksys router. Now, I am trying to connect from outside to my asterisk server. I use Diax as iax client. For some reason I cannot connect to my server from outside. On my router I forward those ports to my

Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29

Hi, I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: >Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh, format changed >to 1024 >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP

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 have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable

The Applied Research Lab (ARL) of Washington University and the ATOMIC-2 Project at USC/ISI are pleased to announce the availability of a BSD device driver for the Adaptec 590x series of PCI ATM host adaptors (eg ANA-5940) Written by Chuck Cranor of Washington University's ARL (chuck@ccrc.wustl.edu), the "MIDWAY" ATM device driver originally supported Efficient Network's

Trixbox 2.2 Beta with freePBX 2.2.0rc1 I have a setup that looks something like this in ASCII art: Teliax IAX Trunk ------+ | V Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+ +--------------> Lima Office Server -----+|

Skipped content of type multipart/alternative-------------- next part -------------- ############ # amd BOX # ############ ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup("SIP/6202-d193", "IAX") in new stack -- Executing

Hi there. I am trying to connect Asterisk to a local danish ip-telephony provider. But is having some difficulties. First I thougt they were related to the provider. But then i started debugging on the Asterisk (aix2 debug) When I make a call using AIX to the provider everything seems to work just fine: *CLI> -- Accepting AUTHENTICATED call from 192.168.1.150, requested format = 1024,

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no