similar to: IAX user identification.

I feel I'm doing something obviously wrong here and will kick myself when I see the answer!!! The scenario: SIP phone -> Asterisk1 -> IAX -> Asterisk2 -> IAX -> Asterisk3 -> PSTN So I place a call from the SIP phone. A1 picks it up and forwards it to A2 which forwards it to A3. A3 sends the call to the PSTN. I control A1 and A2, but not A3. When a call fails (for

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

Hmm. By the way, please don't post bugs to asterisk-dev as I've been told :> That list if for on-going development. That sounds like a bug I encountered in 1.0.5. There is a division by zero bug in chan_iax2.c introduced somewhere after 1.0.4 I believe and currently fixed in HEAD. (They've given me enough shit for posting the bug while it was fixed in HEAD already. No need to

i had a sip invitation problem with my voip provider and here the message that was shown : Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:XXXXX@195.112.214.99>;tag=as7b43dfbd' -- SIP/callshop-3fcc is circuit-busy == Everyone is busy/congested at this time -- Got SIP

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

I have a problem with my asterisk@home installation (configured with AMP) My question is this, can you have different ports for different contexts within IAX? [Faktortel] port = 5036 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=all ; Allow all codecs register => XXXXX:XXXXX@iax.faktotel.com/EXTEN

Hi. Probably been asked before, but my IAX provider assures me its not their problem.... I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them. IAX debug output is: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1 Timestamp: 02608ms SCall: 00016 DCall: 00003

I have a problem with IAX call transfer. The call goes successful but consumes lot of BW in the middle tier. The actual connection is like this (NAT) DIAX(IAX2) -----> *1 ------> *2 *1 & *2 were public IP with asterisk. It consumes around 120kbps in total to forward a single GSM call. I have the following configuration in my iax.conf [general] ... disallow=all allow=gsm [provider]

I seem to be having a problem that I have narrowed down to a disagreement on codec negotiation or codec setup of some kind in an IAX peering arrangement. Here's a non-ASCII art version of the setup: DID origination provider via SIP/gsm to Call routing asterisk server via IAX/gsm to Client asterisk server via SIP/ulaw to Polycom 501 UA The problem that occurs

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

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

Hi I have a system with one asterisk box in front and a few PSTN gateways in the back. When a call comes from PSTN, it's directly forwarded to the edge/user asterisk box. Now if a number is dialled, and that number is not in use, blocked due to lack of paying etc, I want to signal that back to the PSTN gateway, making this playback or playtones to avoid picking up the phone. Can someone

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})

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]

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

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

So I thought I had SIP and NAT cracked a long time ago, but something's just happened that's sort of upset the cart )-: I have an * box behind a NAT firewall. Nothing unusual there, this is something I've done many times - sip.conf has the correct nat= localnet= externip= settings, the router has ports 5060-5069 and 10000-20000 forwarded to the internal IP address of the

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

Hi, I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7 running on the same box. I wonder how to setup the iax account correctly so that I may initiate outbound calls via iaxmodem? registration upon iaxmodem startup is okay and I can direct calls to it. -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874 But upon an outbound call setup request from

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