similar to: Ticket: 12775 Multiple IAX client behind a NAT

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

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.

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

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, My problem is related to a bug in the Asterisk bug database: (Bug 2773) http://bugs.digium.com/view.php?id=2773 Essentially, when one has a call going over more than one IAX legs, the audio is not transferred *sometimes*. This is quite randomly observed. With "notransfer=yes", the problem goes away. In my situation, the IAX channels originate from 2 Asterisk servers themselves -

Hello asterisk users, here is a little problem pulling out iax peers from real time database I have the following peer configured in my database mysql> select name,username,secret,type,context,host,disallow,allow,defaultip,deny,permit, ipaddr,port from iax_users where name='iaxtermination'; +----------------+----------+----------------------------------+------+-----

Hello, Right now there is not a table build script at: http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Therefore I have taken the SIP build script and added a few fields that I use from my iax.conf (could be more out there, please see the complete build script below): `dbsecret` varchar(100) default '', `notransfer` varchar(100) default '', `inkeys` varchar(100)

We have a problem with our iaxclients. Our asterisk runs on a public host with debian and many of our IAX2 clients are natted. The iax.conf looks like: [23456] accountcode=123 type=friend context=user auth=md5 secret=xxxx username=23456 callerid=Testuser 1 <23456> notransfer=yes host=dynamic The cli command IAX2 show peers shows all clients as unmonitored CLI> IAX2

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

Hi, I have a load of IAX extensions, which I'm trying to set up a standard macro to dial them, which gives unavailable or busy voicemail if there is no answer or the phone is in use respectively. The macro I have at the moment is: ; std-exten macro, ${ARG1} = Device to call, ${ARG2} = voicemail box [macro-std-exten] ; Call the user for 20 seconds exten => s,1,Dial(${ARG1},20,tr) exten

Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users

Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",

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

Hello. Just wanted to report this sysOID to you. The UPSC output is below. Network UPS Tools - Generic SNMP UPS driver 0.70 (2.7.1) kill: No such process No matching MIB found for sysOID '.1.3.6.1.4.1.318.1.3.2.11'! Please report it to NUT developers, with an 'upsc' output for your device. Going back to the classic MIB detection method. Detected SMART-UPS 2200 RM XL on host

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'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I

I donwloaded two IAX Clients (firefly and IAX phone) and they did register with *. It would make authenticated calls, but wouldn't actually register with the server. When I start the IAX Client the CLI show me the message: -- Registered '2004' (AUTHENTICATED) at 192.168.199.69:4569 After 5s: May 21 17:24:41 NOTICE[1133742896]: chan_iax2.c:5035 iax2_poke_noanswer: Peer

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

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, 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 -