similar to: Duplicated DTMF with bridged IAX channels maybe?

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

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

Hi Guys, I have already tried this one on the developers list. I have not been successful getting much back there and I have notified them that I will post this on the users list instead. Hopefully somebody have tried something similar and can help out. I am developing AGI scripts on Asterisk and have run into some very strange behaviour and I think this is a bug, but I am not completely sure.

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

I had a working 1.0.9 asterisk installation and tried to get a Firefly IAX phone to register, but it was failing. I upgraded to asterisk 1.2.5 and the PBX is working fine, but the IAX phone still won't connect. Below is my iax.conf and the output from setting iax2 debug while the phone tries to connect. Could somebody please give me some pointers? This doesn't seem to be a normal

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

Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register => username:password@speex.dyndns.org and I cant get it to work. Maybe someone who already got this to work will help... When dialing my speex extension I see the next output from consol: IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:

I'm taking inbound calls on an * server, then transferring them to a second * server where they join a MeetMe conference. If I have 'notransfer=yes' set on the first * server it works fine, but if I allow the transfer (call then shifts to be between the DID provider and the second server), the call is dropped 3-5 minutes later. There is no firewall on my end, and the two

I downloaded iaxComm and get up my iax.conf file and the extensions.conf. Here is the out but from CLI in iax debug. What did I forget to do??? Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00001ms SCall: 10489 DCall: 00000 [192.168.50.66:4569] USERNAME : 100 REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:

Hi all, I'm using following software and equipment and I have very strange behavior: Asterisk CVS-NHEAD-05/30/05-16:42:41 H323 gatekeeper - GnuGK 2.2.2 IAX2 client - Firefly 1.9.8 build 3945 H323 client - SJPhone Build 1.50.271d H323 gateway - Welltech Wellgate 3504A When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected. When I dial from SJPhone (H323) ->

Can any one help? Thanks. we have two * servers (Version 1.2.1) and one 1.09 server. Calls between these two 1.2.1 servers have odd behavior. But call from 1.09 to 1.21 server working fine in either situations. See below pls: Local server iax.conf [tosyd] username=mel type=peer secret=xxxx host=198.168.2.66 remote server iax.conf [mel] type=user secret=xxxx host=198.168.2.67

Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an "iax2 debug" output from a DIAX call to a local * server, dialing the extension that goes directly to the "demo" application. AsteriskHouse*CLI> iax2

Hi, I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it into production. Ive done this by installing 1.4.18 onto the VM, putting my config files in place and then installing 1.4.37 over the top (which is what I'd have to do on production). I've found a few issues in the config files, but nothing I couldn't handle until... I hit inbound IAX issues. My

I would like to ask, if someone observe also problem with peer qualify problems, my asterisk log is full with UNREACHABLE/REACHABLE messages, even when two asterisks are in LAN environment, please take a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here...

For some reason, starting just today, 1 out 3 of my asterisk servers is having issues calling 1 other server. The only issue I see is that when it registers with the problem server it is using port 1027, not 4569. ie: Registered to 'Server 1', who sees us as 'Server 2':1027 Server 1 then proceeds to timeout trying to register with Server 2. The way I have each server registering

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

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

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and