similar to: No Such host - IAX2 channel problem

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:

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

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: \

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

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

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, I'm back with my question, maybe someone can help me: I want to use IAX on another UDP port (not the default 5036), because I have 2 Asterisks behind the same NAT. Changing the default port in iax.conf file from 5036 to 5038 and then calling using the syntax: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1}) I get the follwing error in the Asterisk console: --

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]:

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",

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:

Hi. It is propbably a really naive problem, but I really couldn't find answer how to connect two Astrisks via SIP. I managed to do it via IAX without any problem. But this is a test installation and I would like to connect them via SIP. So I have two computers: pbx1 - 10.1.3.207 pbx2 - 10.1.3.204 pbx1 handles extensions 1xx and pbx2 for extensions 2xx. I would like to call user from pbx2 to

Hi everyone, I just got myself a IAXy device and am trying to integrate it to our asterisk server. I configured the IAXy and it is registering and I get a dial-tone. If I try calling another SIP device, and I get "can't translate IAX2 to SIP" How can I make my IAX device communicate with a SIP device (and vice-versa)? Here's what the log says: -- Executing

ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800450 at pstn2,,S(20)) thanks ________________________________ De : Adil Zaaraoui <adilzeaaraoui at yahoo.fr> ? : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Envoy? le : Jeu 8 juillet 2010, 19h 41min 15s Objet : Re :

Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a

Am getting netsock error like this when using IAX2, Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid = 4270) == Using SIP RTP CoS mark 5 -- Executing [2001 at Test:1] Dial("SIP/4090-00000005", "SIP/2001 at IAX2/IND-MAN,30") in new stack [Jun 23 06:31:36] NOTICE[4383][C-00000005]: chan_sip.c:29491 sip_request_call: Conflicting extension values