similar to: IAX can be used on a different UDP port?

Hi, Which is the correct syntax to call using IAX? I have two Asterisk boxes behind a NAT and one of them use the default port 5036 for IAX, the second one use 5038. To call an extension of the first one, the line in extensions.conf is: exten => _9XXX,1,Dial(IAX/user:pass@195.3.32.191/${EXTEN:1}) and for the second one: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1})

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

Hi all, this is my actuel setup [SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900] Linux CentOS 3.3 (2.4.21-20.EL.c0) asterisk-1.0.1 asterisk-oh323-0.6.3b openh323_1.12.2 pwlib_1.5.2 Calling from SIPphone to the extension 8900 works always. Calling from 8900 to SIPphone works only sporadicly without any recognizeable pattern. Find below the output of the debug command: asterisk

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

The setup: Inc SIP Call -> Asterisk 1 -- IAX --> Asterisk 2 -- SIP --> phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto("SIP/XX.XX.XX.XX-0819f590", "cytel-internal|3044|1") in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial("SIP/XX.XX.XX.XX-0819f590",

Hi, I have a big problem with EXEC in AGI scripts: I do, for example, "EXEC Dial SIP/phone1", Asterisk says -- AGI Script Executing Application: (dial) Options: (sip/phone1) Jan 10 14:33:20 WARNING[10567]: chan_sip.c:1389 create_addr: No such host: phone1 Jan 10 14:33:20 NOTICE[10567]: app_dial.c:743 dial_exec: Unable to create channel of type 'sip' I do "EXEC

I have several DIDs (working well) with LiveVoip and I just signed up for some outbound minutes. Unfortunately they did not send connection instructions. I tried: exten => _1NXXNXXXXXX,2,Dial(IAX2/userid:password@217.160.244.186/${EXTEN}|60|s) but I get the error Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found --

Hello, I'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1",

i try to place a call exten => _X.,1,Dial(SIP/${EXTEN}@foo:5061,60,Ttr) where sip.conf has an entry [foo] secret=torture callerid="local ext 103" <19146665555> type=friend fromuser=asterisk auth=both host=dynamic canreinvite=yes context=in-914 mailbox=001 i get May 22 23:11:31 WARNING[140400128]: chan_sip.c:902 create_addr: \

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Hello, I have been trying to setup a Voxee Sip termination. If anyone has extensions.conf different than Voxee suggestion. Can you please send me a copy? Thanks! Jerry Voxee web site advises to use: [voxee] exten => _1NXXNXXXXXX,1,Dial,SIP/${EXTEN}voxee exten => _1NXXNXXXXXX,2,Hangup exten => _011.,1,Dial,SIP/${EXTEN}voxee exten => _011.,2,Hangup

Sorry to bug the entire list with this as this is really a question for those who have been sucessful in configuring [*] to place and receive a SIPCALL call. Everying looks right in my config, I can see it registered etc but when I try to place the call I get: -- Executing Dial("SIP/100-2371", "SIP/8703409095@sipcall/04") in new stack Apr 29 22:50:34 WARNING[27089840]:

Hello, I am attempting to use IAXComm as an end-point for my Asterisk instance. I have setup an entry in MySQL (using RealTime configuration) and am able to dial-out with no problem, although I notice this notice on the console of Asterisk: Jan 18 21:01:53 NOTICE[22491]: chan_iax2.c:4307 register_verify: No registration for peer '10000' (from 27.21.26.2) I then issue this Dial cmd:

Hi all, I am new to asterisk and trying to setup clients over LAN to enable voice-chat between them. I have got two clients(IAX-phone) having extensions 4061 and 4082. I am able to call extension 600(provided with sample configuration) from both of them but when I try to call 4061 from 4082 or vice-versa I get error message at server... chan_iax2.c:2215 create_addr: No such host: 4082

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call: