[asterisk-users] ${EXTEN} is limited to 17 characters under IAX ?
Hi list.
We have a problem when dialing over IAX to another Asterisk server:
we've setup an extension named
'f19dffb971b93746d73ec46d5f1d4b36c199f48c-g1' in a specific context (its
large because it needs to be unique). I've read in past discussions on
asterisk-dev list that the extension length is limited to 79 characters
- which I though should be more then enough.
Now were doing a DUNDi lookup on that extension and dialing to it from a
second Asterisk server. The dial address looks like this:
IAX2/dundi-context:[EMAIL PROTECTED]/f19dffb971b93746d73ec46d5f1d4b36c199f48c-g1
The problem is that on the local server, we try to read ${EXTEN} and
parse it (specifically - I want to get at the 'g1' at the end. for this
I use the CUT function):
[mydundictx]
exten = _[0-9a-fA-f_].,1,Set(lastpart=${CUT(EXTEN,,2)})
exten = _[0-9a-fA-f_].,2,Set(firstpart=${CUT(EXTEN,,1)})
and then we get this (in the console):
-- Accepting AUTHENTICATED call from 192.118.54.135: [...]
-- Executing [EMAIL PROTECTED]:2]
Set(IAX2/192.118.54.135:4569-1, lastpart=) in new stack
-- Executing [EMAIL PROTECTED]:3]
Set(IAX2/192.118.54.135:4569-1, firstpart=f19dffb971b93746d) in new
stack
I understand that Asterisk truncates the extension in the display (in
this case - to 17 characters), but I was under the impression that this
is for display only. Apparently this is not the case - the as evidently
at least CUT sees only the first 17 characters ?!?
Then we changed the setup to dial from server to server using SIP
instead of IAX2 - using this method, the entire extension is passed
correctly. Any idea whats going on here ?
We're using Asterisk 1.4.0.
--
Oded Arbel
Atelis
[EMAIL PROTECTED]
Tel: +972-54-7340014
::..
After a number of decimal places, nobody gives a damn.
___
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Re: [asterisk-users] Re: How to exit from console?
On Thu, 2007-01-25 at 18:01 +0200, Tzafrir Cohen wrote: On Thu, Jan 25, 2007 at 01:37:50PM +0100, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Try safe_asterisk , for an easy way to start asterisk in background, a plain 'asterisk' is even better and safer. asterisk -U asterisk . is better. /etc/init.d/asterisk start is similar. Why is this better than safe_asterisk? the init.d init scripts bundled with asterisk are using safe_asterisk and not calling the asterisk binary directly. E.g: because you have a valid PID file of the controlling process. If you actually want to kill it, you can. Granted, its a good idea. the init.d scripts bundled with asterisk kill safe_asterisk, which apparently works just as well (haven't looked at safe_asterisk code, but its probably killing its child when it is being killed, which should work well for any situation other then kill -9). And you don't need physical access to the system to get to the one and only real console. OTOH, if you do have physical access, you have full control of Asterisk, as you may inject custom dialplan. I wasn't aware that running asterisk -r on a physical tty has any advantages over running asterisk -r on a remote shell. -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. In this world, truth can wait; she's used to it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 problem with ztdummy and MeetMe()
On Thu, 2007-01-25 at 18:40 +0100, Stefan Wintermeyer wrote: Hi, when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and start asterisk to be able to use MeetMe(). When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and start asterisk but I am not able to use MeetMe(). What do I miss? I'm not sure, because we missed the entire problem description, which I would imaging would have included log snippets and/or error message reports, but it was apparently removed from your e-mail. http://www.catb.org/~esr/faqs/smart-questions.html -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. He's dead, Jim. You grab his wallet, I'll grab his tricorder. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rxfax and Txfax on Asterisk 1.4
On Fri, 2007-01-26 at 07:06 -0500, Remzi Semsettin Turer wrote: Has anyone successfully installed spandsp and rxfax and txfax applications on 1.4.0 release of Asterisk? I tried the latest snapshot of spandsp, as well as couple other previous versions. I compiled it fine, downloaded the asterisk.patch, manually patched the asterisk files, run .configure, make clean, make menuselect and it shows app_txfax and app_rxfax as XX (unavailable). Each time I made sure no other spandsp versions are installed and put the proper path in /etc/ld.so.conf and run ldconfig, prior to compiling Asterisk. Still no luck. I don't remember how I got rxfax/txfax to be available in menuselect, but they won't compile - I couldn't get the app_rxfax.c and app_txfaxt.c to compile against 1.4 and I didn't have time to figure out how to get them running. AFAIK the current recommendation is to use HylaFax with something called iaxmodem. -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. A man with one watch will always know the time, A man with two watches will always be in doubt. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel name
On Wed, 2007-01-24 at 11:26 -0800, Serge Blazhievsky wrote: Hello everybody, I was wondering if anybody knows how to make channel IDs different if all call are coming from the same host: core show channels Channel Location State Application(Data) SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback() SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback() The channel names are different - its just that 'core show channels' has a limited width in the display to show the full channel name and truncates the relevant parts. try typing 'core show channel ' and then hit TAB to see a list of possible channel names. -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. The true measure of a man is how he treats someone who can do him absolutely no good. -- Samuel Johnson (1709-1784) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On what distribution is www.asterisknow.com based on ?
On Sat, 2007-01-20 at 17:33 +0200, Maxim Veksler wrote: Hello Asteriskies, Has someone tried www.asterisknow.com ? What is the package manager used? I haven't tested it fully, but from first look it seems like a Fedora/RHEL/CentOS derivative - it uses the anaconda installer, and I think it uses yum over RPM as the package manager. And what is the added value compared to the well maintained debian based asterisk ? As well maintained the debian based asterisk is, AsteriskNOW is maintained by Digium itself. I would think that that counts for something. In addition, AsteriskNOW attempts to delivery an out-of-the-box usable Asterisk PBX, which isn't what the debian packages Asterisk is doing. If you want a simple straight forward install that gives you an easy to use Asterisk box with little other software involved, then you get AsteriskNOW. If you want a flexible server that can do a lot of other things, then you probably want to use something else - like Debian. -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. The universe is a big place, perhaps the biggest. -- Kilgore Trout ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parked calls with Asterisk 1.4.0
Hi List. We have a small issue with making parked calls work with the new Asterisk 1.4. I have an impression that this used to work with 1.2, so its either I'm doing something wrong, or a regression. I hope its not the latter and you can tell me what I'm doing wrong. The setup is an Asterisk with sip users in mysql realtime and dialplan in mysql static (mostly - some stuff is built-in). We have Linksys hardware voip phones connected to it, and a small dundi setup (I don't think its important in this case). Here's the SIP users' default context: [local-priv-incoming] exten = 910,1,Goto(parkedcalls,700,1) parked calls looks like this, of course: CLI dialplan show parkedcalls [ Context 'parkedcalls' created by 'res_features' ] '700' = 1. Park() [res_features] So , supposedly someone calls me (in this case through the dundi setup, but I don't think its a problem - we can reproduce this with local calls as well) and I do attendant transfer to '900'. I then hear the parked call number (701 in my case) and so I complete the transfer (in the Linksys phones, that means hittint XFer again). The caller now, instead of being parked, disconnects. In the asterisk CLI it looks like this: ## the remote DUNDi user goes through some stuff and eventually dials to my local SIP extension: -- Executing [EMAIL PROTECTED]:11] Dial(IAX2/192.118.54.135:4569-2, SIP/2006||L()) in new stack -- Called 2006 -- SIP/2006-009e9e10 is ringing -- SIP/2006-009e9e10 answered IAX2/192.118.54.135:4569-2 ## I'm putting the caller on hold while I start the transfer -- Started music on hold, class 'default', on IAX2/192.118.54.135:4569-2 ## dialling 910 -- Executing [EMAIL PROTECTED]:3] Goto(SIP/2006-009eccc0, local-priv-incoming|910|1) in new stack -- Goto (local-priv-incoming,910,1) -- Executing [EMAIL PROTECTED]:1] Goto(SIP/2006-009eccc0, parkedcalls|700|1) in new stack -- Goto (parkedcalls,700,1) -- Executing [EMAIL PROTECTED]:1] Park(SIP/2006-009eccc0, ) in new stack == Parked SIP/2006-009eccc0 on [EMAIL PROTECTED] Will timeout back to extension [parkedcalls] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls # this I'm hearing -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on SIP/2006-009eccc0 # now I complete the transfer == Spawn extension (parkedcalls, s, 1) exited KEEPALIVE on 'SIP/2006-009eccc0' -- Stopped music on hold on IAX2/192.118.54.135:4569-2 [Jan 15 15:22:17] WARNING[10582]: chan_sip.c:12310 handle_response: Notify answer on an owned channel? == Spawn extension (dundi-priv-lookup, 2006, 11) exited non-zero on 'IAX2/192.118.54.135:4569-2' -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/192.118.54.135:4569-2, -- Done with call --) in new stack -- Hungup 'IAX2/192.118.54.135:4569-2' -- Stopped music on hold on SIP/2006-009eccc0 == SIP/2006-009eccc0 got tired of being parked # and this is where the remote caller disconnects Can you please tell me what I'm missing ? -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. The gates in my computer are AND, OR and NOT; they are not Bill. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
