similar to: IAX2 attempts native bridge when notransfer=yes

Hello, I have been getting outbound nufone calls dropped after about 70 seconds. CLI shows "Attempting native bridge of IAX2". I have put "notransfer=yes" in iax.conf in the main section and all identifier sections. I tried a Tt in the dialstring, but it still tries the bridge. a cvs update didn't help. internal *server1 tdm fxs pci card <--> iax2 trunk -

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

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.

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

Hi all, I'm experincing IAX2 call drops for about 20% of calls. I tried 'notransfer=yes' and 'jitterbuffer=yes' but to fail. My system configuration is like this. PSTN<========>Asterisk(TDM/Fxo 4port*3)<=====LAN(IAX2)=====>Iaxclient library And iax.con is... ----------------------------------------------- [general] port=5036 disallow=all allow=gsm

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

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

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

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

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

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 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

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 =>

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

Hi, I posted this on the Asterisk forum months back with no real answer() so i'll try here :o) Details: There is 3 asterisk boxes called X, Y and Z.. all boxes peer with each other via IAX2 and have dialplans setup... etc etc There will be asterisk based clients connecting via IAX2, and for example i'll call them A, B and C The clients only peer directly with one of the X, Y or Z

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

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

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

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