similar to: Intertex IX66 incoming IAX

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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

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

Hi! I'm trying to build an HA system using heartbeat for failover. Everything works fine with SIP, but I cannot connect my IAX phone to the asterisk server using the managed IP address. Here is the configuration of the server (asterisk and the IP address are up, 'ip addr' and 'netstat' output): 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast

Hi, Anyone used this service, any comments on reliability/support? Thanks John

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

Hi, I'm aware that incoming and outgoing calls are going fine when isdn channels are involved - caller id properly identifies calling party, so user can call back.... But how to properly handle this for iax, sip calls.... I have few questions : - BTW, what to type for instance in remote firefly to make standalone calls to Asterisk default context or particular extension ? - If I receive

I have (more than 1) provider that I receive calls from using IAX, and I have 2 IAX deskphones, all work fine except for some reason with 1 provider, when the call comes in, it doesn't match up with the incomingcall context. (A bit worrying, since I don't want people to be able to relay calls off me.) in iax.conf I have: [ipcomms] type=user nat=yes dtmfmode=rfc2833 host=71.16.179.149

Good day, The unusual thing about this problem is that it doesn't occur just during a CAPI call, or just during an IAX/SIP call. Only during IAX/CAPI I'm having some trouble with the CAPI interface and it only occurs when a call comes in on an IAX channel and goes out the CAPI interface. The capi debug in the asterisk console is below as well as the relevent parts of .conf files from

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960

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iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome.

I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss

Hi *, Does anyone have a lead on a Nagios plugin that speaks IAX or a small app to do so? I'm trying to set up remote monitoring for my Asterisk server and only IAX2 traffic is allowed through the firewall. Simply using check_udp to port 4569 yields no usable answer and Asterisk complains about receiving a midget packet or something like that :) jens

Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the

I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Hi! I'm trying to set up the possibility for users to call my Asterisk from the net. The Asterisk is behind a Intertex IX66 in which a have set "Static domains" so it forward all calls for my hostname and external IP to the * box. when somenone calls at lars@<external-ip> it all works, but if they call lars@<hostnmame> it wont work. At my Asterisk console I get the

Hello R users, Imagine the data frame DATA with the variable x which is composed by numeric and alpha caracters. > DATA x 1 12F 2 13 AD 3 356PO 4 1D 5 GRT 6 PO52 7 LN4Z Is there a way to separarate x in 2 variables: y: only numeric caracters z: only alpha caracters For exemple: x y z 1 12F 12 F 2 13 AD 13 AD 3 356PO

This thread got me thinking of other servers that would run asterisk. The obvious question comes up if Xebian (the xbox version of Debian) would run as a SIP only server? Asterisk on an XBox would be a small box! Cheap too. Tom Schaefer