similar to: Questions about contexts

I have to design a dialplan for mulitple contexts (multiple companies) and I'm not sure how to go about it and I thought someone may offer help. Here is some background. There are three separate companies, let's say A, B and C. Each has their own context and each has their own set of numbers (these are just examples, not the actual config): [ContextA] exten =>

Hello; we are looking to replace our current PBX with a *-box; it is connected to ONE ppp isdn connection that is terminated by the NC. We got on this box 4 msn's configured. currently we are working with pstn fxo's behind the PBX; it works but we can't use the CSID information behind it. We want to migrate and keep the MSN's to decide routing in combination with the CID.

Hello list, I have this strange problem whenever I try to make an "ael reload" from the Asterisk CLI. The command gives the following result and crashes: voip-1*CLI> ael reload Disconnected from Asterisk server Executing last minute cleanups Asterisk ending (0). root at voip-1:/etc/asterisk# As far as I can see, aelparse can't find any errors in my configuration, following

hi, how can I add or remove this line "include => context" by the command CLI ? regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030827/979ddd76/attachment.htm

Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

Hi, Is there some more thorough documentation of this change that has happened in 1.6? The upgrade.txt and changes.txt files mention it, but I have already seen details of this change that do not appear to be documented except in conversations on the mailing list... 1) It appears that it is no longer legal to have: [macro-contaxtA] ...stuff... [contextA] ...stuff... Is this true? Or have I

> -----Original Messages----- > From: "Steven Tardy" <sjt5atra at gmail.com> > Sent Time: 2018-02-26 10:48:48 (Monday) > To: "CentOS mailing list" <centos at centos.org> > Cc: > Subject: Re: [CentOS] How to update modules in iniramfs fastly > > On Sun, Feb 25, 2018 at 8:29 PM wuzhouhui <wuzhouhui14 at mails.ucas.ac.cn> > wrote:

I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and

Hello, When using R_tryEval (from JRI in my case), is there a way to setup error recovery strategy and more generally calling handlers. From my reading of context.c, R_tryEval calls R_ToplevelExec which creates a context like this: begincontext(&thiscontext, CTXT_TOPLEVEL, R_NilValue, R_GlobalEnv, R_BaseEnv, R_NilValue, R_NilValue); so I guess what I am trying to do is add

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

hi: Just a litle update: 41/8 allocated to AfriNIC (APR 2005). 73/8 allocated to ARIN (MAR 2005). hope it helps.

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hi All, i dont what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS

I know dracut can update modules in initramfs, but I think it is too slow. So I'm wondering what is the fastest way to update modules in initramfs of CentOS 7? Thanks!

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6