similar to: 1.6.1: unable to create channel IAX2 to Junction

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Hi Samba List, I believe I need to create a "Junction Point" on a Windows XP Pro machine that will seamlessly link up to a Samba Share on a Linux Server. >From my understanding, the creation of a "Junction Point" will hide from an application the fact that the "space referenced by that junction point" is actually located someplace else. While it is

I'm running DeltaCopy (essentially a Windows friendly GUI wrapper around RSync 2.6.6) on Microsoft Vista. I'm trying to make a backup of a user's profile directory. One big difference between Vista and earlier Windows versions is that Vista makes extensive use of Junction Points (sort of like Unix symlinks) for compatibility. For instance, in a profile directory there are 16

I''m looking into how to implement a relationship between some models, and I think I need to use a junction table. From the documentation I''ve read, a junction table doesn''t require a model. But from my controller, I''m not sure how to get the data I need. Here''s a description of my models and what I need in my controller: Models 1. Event 2. Site 3.

I am running 3.4.3 on a windows server and I ran the code in a new session. I get a warning when running packrat::init() on a project that is located on a windows network drive. The warning I get is Warning message: cannot set reparse point 'U:/packrat5/packrat/lib-R/base', reason 'Access is denied' The error is created based inside the function .Internal(mkjunction(fr,

Here is some Ruby code for creating junctions on NTFS volumes. These are the main routines: Dir.junction?(dir) => true if dir is a junction Dir.reparse_target(dir) => returns the target of a junction, or dir Dir.create_junction(junctName, existingTarget) => creates a junctName junction pointing to existingTarget (also used (Dan''s?) GetLastError code and did some simple

> -----Original Message----- > From: win32utils-devel-bounces@rubyforge.org > [mailto:win32utils-devel-bounces@rubyforge.org] On Behalf Of > Zach Dennis > Sent: Thursday, May 05, 2005 4:25 PM > To: Development and ideas for win32utils projects > Subject: Re: [Win32utils-devel] Ruby code to create junctions > on NTFS volumes. > > > Timothy Byrd wrote: > >

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

Anyone have experience with these guys? If so, good, bad, average? http://www.junctionnetworks.com Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/4222e3e7/attachment.htm

I?m trying to use rsync with windows vista. Vista?s odd form of ?links? confuses rsync and I get a recursive infinite loop at every such junction (until failure when the filename size limit is exceeded) . Could someone please point me to the solution or is there one? The std don?t follow links switch doesn?t seem to work. Eric -------------- next part -------------- HTML attachment scrubbed

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 :

We are looking for SIP trunks for our * pbx for our business. Being able to port our numbers is an absolute requirement. teliax can do it, but I am unsure of the others. Anyone have experiences (good, bad) with the above mentioned providers to share? Eg reliability, quality, etc. -Dan

I am having this same exact issue. Any info would be appreciated. -- Kurt Nelson (407)512-0689 http://claimid.com/kurtisnelson/ -------------- next part -------------- HTML attachment scrubbed and removed

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten =>

Hi All, I've got a realtime queue in place (strategy is "wrandom"), and have added a member dynamically via "queue add member ". My agent shows in the queue, but when he gets the call is not recognized as "In Use". Here is the output from "queue show" prior to the call: *CLI> queue show QUEUE_3 has 0 calls (max unlimited) in 'wrandom'

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

Guys.. Im trying to setup a fotphone using iaxcomm and when I dial that softphones extension, * complains of this: Mar 5 01:54:54 NOTICE[9962]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) Any hints?

I try to connect to voipjet, but I get always busy, ... it worked yesterday, ... no changes on my side.... -- Executing SetGroup("SIP/615-829b", "iax-voipjet") in new stack -- Executing Dial("SIP/615-829b", "IAX2/17567@voipjet/011886228357765") in new stack May 20 18:16:26 NOTICE[9733]: app_dial.c:973 dial_exec_full: Unable to create channel