similar to: RE: regcontext, NoOp extension vanishes when extension reload

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

> -----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

> 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 =>

All, 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 => _XXXX,4,NoOp(sipregistration call - Name:

Hi guys, I'm building a PoC Asterisk 12 cluster based on a number of guides I've found on the net. The basic concept is using ARA in conjunction with DUNDi. I have set up ARA with pjsip according to this excellent guide here: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime This is working nicely, so now I am turning my attention to DUNDi, as per this guide here:

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

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

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

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

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

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

Does anyone have * talking to a Syndeo switch? I am new to * and trying to get this working. It appears to have a problem authenticating, but I do not see an "unauthorized" message. Then shows the * as a registered end point with an ip of 0.0.0.0 @1073435488.955240 las-cms-a :366 ResControllerSta 3 ResController::submit - Submiting resource Res 371658 to waiting queue

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

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

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

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

Hi, Is it possible to write a function in Asterisk, that returns a value? Sort of like any programming language allows? For example, I`d like function ReturnSipReg to return the right SipRegistration to dial, based on some value so that I could use it in my dial plan: i.e: exten => 1234,1,Dial(SIP/ReturnSipReg(John)) ; would dial John's extension, which I don't know at this

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

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings