similar to: Trouble with regexten

Displaying 20 results from an estimated 1000 matches similar to: "Trouble with regexten"

2005 Dec 11
14
Regexten
Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.
2007 Aug 05
1
How does one use sip_autoreg
I've RTFM and Googled but can't seem to get sip_autoreg to work (or perhaps I'm just completely missing the point of it). (what I'd like to do is avoid having to put explicit entries for every SIP phone into extensions.conf). Asterisk is creating entries in the (virtual) context sip_autoreg: asterisk*CLI> dialplan show sip_autoreg [ Context 'sip_autoreg'
2006 Oct 06
3
regexten & regcontext broken for SIP?
Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no
2011 Dec 16
2
Which device auto-registered an extension?
Hi all, In sip.conf: [general] regcontext = autoreg [devabc] regexten = 543 creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the dialplan, because there's no device SIP/543. Now I know I can add a line like "exten=> 543,2,Dial(SIP/devabc)" for each and
2006 May 11
4
'extensions reload' clears Regextens
I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created from regexten in sip.conf, and I do an 'extensions reload', I lose all the priority 1 NoOps! This can't be right... this means that in a production environment, if you make a change to your dialplan and do an 'extensions reload', you lose your ability to terminate calls to phones on this system.
2007 Jun 06
1
Reload in 1.4 clears regexten
Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2006 Nov 01
2
Realtime, DUNDi and regexten
It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>
2009 Aug 07
1
regcontext regexten
Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm
2006 Dec 05
2
regcontext, NoOp extension vanishes when extension reload and doesn't come back
Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a >reload extension is performed, those dynamically created extensions in the regcontext vanish. Now
2006 Dec 05
0
RE: regcontext, NoOp extension vanishes when extension reload
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > JR Richardson > Sent: Tuesday, December 05, 2006 3:49 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] RE: regcontext,NoOp extension > vanishes when extension reload > > > > > Let me guess: The
2006 Mar 17
1
RE: DUNDi .... Halfway and CLUSTERING
At the moment I'm out of the office, but when I return I'll be certain to do that. Note that my solution is different from what you are working on with regexten, though I suspect some of the challenges that I've faced and overcome are not. I'm actually using UltraMonkey for load-balancing and failover of the Asterisk boxes, and my dialplan is set up so that it need not be changed
2008 Oct 07
1
regcontext
hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked
2006 Jun 08
1
Using regcontext
Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using
2006 Feb 23
6
username as extension
Is there a way to have extensions automatically created for registered sip users ? I did some investigation and found some hope in chan_sip with relation to the somewhat undocumented usereqphone option but i may be totally off track. All i want to be able to do is send a call to number@ip_address where the number is the username configured on the phone that has registered with asterisk
2005 Jan 05
5
Asterisk with MySQL
You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >
2005 Oct 08
0
Regcontext/regexten broken??
Recently I've noticed two bits of odd behavior with respect to regcontext/regexten in CVS HEAD & 1.2 Beta1, and I was wondering if anyone could shed some light on this. I've set up a regcontext in sip.conf. I've set up two users with regexten entries, one in sip.conf and one in a mysql realtime table. The first bit of oddness is that regexten seems to work somewhat as described
2006 Mar 13
0
Re: Regexten & Regcontext, working now
Just figured it out, I think. I put regcontext=mycontext into the [general] section in sip.conf instead of the the [user] section and when the sip user registered, the NoOp extension priority 1 came right up in the dial plan. All is well again, so far. Clarity of sight becomes infinitely greater with head removed from rectum. >> > Hi All, > > I've been trying to get
2011 Apr 20
1
[IAX] Everyone is busy/congested at this time (1:0/0/1)
Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from
2006 Mar 13
0
Regexten & Regcontext
Hi All, I've been trying to get regexten and regcontext going for some sip peers but following the examples on the wiki is not working, as far as I can tell, nothing is happening. the phone registers, sip show peers is ok, but the NoOp priority 1 extension never gets created or added to the dialplan. Has anyone got this working? Thanks. JR JR Richardson Engineering for the Masses
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang