similar to: Regexten & Regcontext

Displaying 20 results from an estimated 30000 matches similar to: "Regexten & Regcontext"

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
2006 Mar 16
0
Regcontext, only 1 context available?
Hi All, I'm working with regcontext and sip users/peers. In the wiki, the example shows you can put this parameter in the [sipuser] context, like so: [general] lots of general parameters [sipuser] regcontext=siptest regexten=1234 Now this does not create the Noop exten priority 1 in the dial plan when the sip user registers. Now if I put regcontext in the [general] section, the sip user
2006 Dec 05
0
Re: regcontext, NoOp extension vanishes when extension reload, WORKING
OK this was an easy one to fix. All I had to do is RTFM. Note on the wiki: ATTENTION: Make sure you take a look at bug report 7144 Just do what Kevin said, include the regcontext in whatever static context you have the priority 2 extension and don't make a static regcontext in extension.conf. Let sip module do the rest. Works great. Thanks Guys. JR On 12/5/06, JR Richardson
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
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 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
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 Jun 08
0
SV: Using regcontext
Hello Thanks for the answer... Just realized it myself, as your mail arrived :) Could be a nice feature though. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olle E Johansson Sendt: 8. juni 2006 12:09 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Using
2007 Oct 06
1
DUNDi, regcontext, softphones.. fail.
> I'm having an issue deploying softphones into my DUNDi/regcontext > setup. My current design is that all SIP users get registered into a > sipregistration context in the sip.conf. I then have a dialplan > function that includes that and does the dial: > > include => sipregistration > exten => _XXXX,2,Answer() > exten => _XXXX,3,Wait(1) > exten =>
2006 Dec 05
0
RE: regcontext, NoOp extension vanishes when extension reload
> > Let me guess: The context in which you have the 2 thru n priorities is > the same one as you're using for regcontext right? > > Don't do that, bad things will happen (as you've noticed). > > I'd have to review the code again, but I think what you're seeing is as > a result of this. > > Regards, > - Brad > No, not exactly, I have a
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 Nov 30
4
Trouble with regexten
Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]
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
2006 Mar 17
1
Re: DUNDi .... Halfway and CLUSTERING
Do you mean the peristence of connecting a specific phone to a specific server? If so, then it's relatively easy. The ldirectord has a persistence setting that does that. If I'm misunderstanding you, then could you explain further what you mean? Regards, - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On
2006 Oct 12
1
AccountCode set in sip.conf but not showing up in CDR
Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup
2008 Mar 01
2
Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
JR Richardson Engineering for the Masses> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of asterisk-users- > request at lists.digium.com > Sent: Saturday, March 01, 2008 12:00 PM > To: asterisk-users at lists.digium.com > Subject: asterisk-users Digest, Vol 44, Issue 1 > >
2005 Aug 27
0
Newbie :SIP ETXTN to SIP EXTN calls
I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,
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
2005 May 12
0
Asterisk, SIP and NAT: Help needed!
I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP