similar to: Newbie question - sip.conf incoming contexts

Hi all In my application I''ve some named routes defined this way... map.label_context1 '':context1/label'', :controller => ''mycontroller'' map.label_context2 '':context1/:context2/label'', :controller => ''mycontroller'' map.label_context3 '':context1/:context2/:context3/label'', :controller =>

Hi, I have a little situation with my dialplan, and I am wondering if what I want is even possible. Here it is: I have three contexts, context1 includes contexts2, and context2 includes context3. In other words, in context1 all extensions of context2 and context3 are valid (and actually working, so that's good). I am using those context for the sake of code clarity and reuse, and for

when i wnat to working with realtime and mysql for any context i have to insert (switch => Realtiem/context at extensions) statment into extensions.conf for example if i want to have 10 context, i have to insert these lines into extension.conf : [context1] switch => Realtiem/context1 at extensions [context2] switch => Realtiem/context2 at extensions [context3] switch =>

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

Hello, Is it possible to use the switch => statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions => mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch => Realtime/includecontext at

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

Hi all, is there an easy way to get a 'copy' of a type living in another context into the local context? Background: when calling a function residing in a different module (context2) from a module (context1), we first need to introduce a function declaration of the function with empty body. However, in order to do so, we need the function type. pFuncInContext2->getType gives us the

Calls that come in on DAHDI FXO ports are routed to [context], extension 's' INSTEAD, I would like to route specific ports to specific extensions, For example: I want DAHDI/1-1 to go to 1234 I want DAHDI/1-2 to go to 2345 I want DAHDI/1-3 to go to 3456 ...etc What is the CLEANEST way to do this? Yes, I can create a private context for each DAHDI channel but that seems messy and

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 -> voicemail -> Secretary 1 555-1235 -> voicemail -> Secretary 2 Any help would be greatly

Hi, I had the following odd behaviour in Asterisk 1.2 - We are migrating to 1.6, and I will re-test ASAP, though it is quite hard to replicate, but I am curious to know whether it is a known IAX issue in 1.2. We had 2 users in iax.conf: [user1] username=user1 secret=secret1 context=context1 host=iax.hostname.com [user2] username=user2 secret= context=context2 host=dynamic deny=0.0.0.0/0.0.0.0

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register => nnnnnnn:ppppp@sip.provider.net -or- register => nnnnnnn:ppppp@sip.provider.net/nnn to come directly into an extension in the dialplan It seems that

Hello! I've got a problem to select the correct trunk if there is one provider and different numbers with different configurations for this same provider. Example: trunk-prov1-2345 trunk-prov1-2346 trunk-prov1-2347 Each trunk registers an own number (at the same provider) and provides own configuration: they have different allowed codecs e.g.. What I'm experiencing now, is, that each

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

sip.conf.sample: ;register => 2345:password at sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no

Hello Joshua, thank you very much for your extremely quick answer! I really appreciate your work and your friendly and your patient support! On 06/07/2017 at 10:33 PM, Joshua Colp wrote: > On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote: >> Hello! >> >> I've got a problem to select the correct trunk if there is one provider >> and different numbers with

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

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 >

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No