similar to: iax2 notransfer=yes ignored

This comment comes up fairly regularly and is confusing people. Why doesn't it say that it failed so we know? The way it is now it kind of leaves you hanging there and you don't know if the transfer happened or not. And why was it even attempted if it is obvious that transfer is off? (I know it can depend on the remote side.) -----Original Message----- From:

Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of IAX2[2109@2109]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

I have this in my dialplan: exten => 6,1,Dial(IAX2/rious@NuFone/16162180431,30,rTt) A call comes in via NuFone over IAX2. The caller presses 6. The dial application calls my cell phone. If I press # on either the calling phone or the called number, nothing happens. I have notransfer=yes set in my iax.conf. I can see both calls with show channels the entire call. If I do an iax2 debug, I can see

Hello, MOH always is choppy when someone calls from a cell phone to my pots or nufone 866. It sounds fine when it originates from a land line. I use zaptel hardware, and plenty of resources. I have tried to use different songs. None have the id3 tags, I tried the custom settings with -q -r 8000 -f 8192 -b 2048 --mono -s. Tried permanent resampling to 8khz, 16bit, filterd with lame -q1. I

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

When I look at my CDR data for calls to NuFone, the billsec for each call is 14 seconds or less. When I look at my NuFone account, the billsec has normal call lengths. So it seems that the billing on the Asterisk system terminates after about 14 seconds. The calls come in on an IAX connection and go out to NuFone on IAX. Are these calls bridging away from the Asterisk server? How can I

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all

Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>

Hi list, I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2. To cut a long story short, IAX2 is not tx-ing hangup... Scenario is composed of two asterisk systems A and B. A receives calls from IAX users X, Y, Z, etc, does some validation and forwards them to B, also over IAX. When B hangs up, it transmits IAX hangup which A receives who, in turn, does not transmit the IAX hangup to its

Evening everyone (obviously depends on when you're readin this, but hey). I'm trying to set up a multi * server situation, and am falling over at the second server, and after a day of google etc, have come up against somewhat of a brick wall. I can make calls each way between the two servers no problem, and can include the required extension at the remote * server as part of my main

I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html