similar to: IAX2 dropping call?

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Is it possible to allow distinctive rings work for FXS ports as well? I need a certain FXS extension to ring a distinctive double ring. I modified zapata.conf appropriately for dring1,dring2 and it just Seems to ignore my updates. Do distinctive rings only work for FXO ports? Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

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

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

I am attempting installation of spandsp on to an Asterisk installation on Linux RH9 the distribution i am using is that are URL http://ftp2.tootai.net - the README for which i have followed verbatim - my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h files in the 'headers' folder of the distribtion i put these in the /usr/include folder based simply on the

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

> -----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 >

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

Two * servers: *a and *b. Outside call comes in *b, and is automatically routed to *a. Someone on a sip phone connected to *a then decides to transfer the call to someone on a sip phone connected to *b. The transfer works. At this point, is *a still in the converstation? Or is * smart enough to see where the data stream is going/coming from? Thanks for any help in advanced, and sorry if

Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter > Sent: Thursday, August 11, 2005 12:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped > betweenpstn & norstar > > > I poured over my logs most of

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

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

Helo to all, I am attempting to establish an IPSEC tunnel to a remote freeswan G/W with my laptop. My laptop sits in behind shorewall at home. From the documentation, this is what I Modified in Shorewall: /etc/shorewall/tunnels: ipsec loc 24.65.x.x /etc/shorewall/policy vpn loc ACCEPT loc vpn ACCEPT My question is, have I left anything out?

Hello all, I am trying to setup a local asterisk server with SIP & ZAP extentions with And IAX2 'switch' to another Asterisk gateway with a PRI. I have managed to Get it working correctly for calls coming in and out. However, CallerID only Seems to work when I dial out. However, everytime I receive a call, it shows Up as 'IAX Guest User' on a SIP client. How can I