similar to: Digit timeouts vs includes in diaplan

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 =>

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 =>

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 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

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

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

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

Hello! I've been struggling with the documentation for months on this simple subject... I still have not been able to get this concept down... I have 3 sip accounts (PSTN DID's) that come into my asterisk box and give me phone service from my itsp via SIP. I for the life of me have not been able to figure out how to get them to come in to 3 seperate contexts! This must be simple but I

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

I have a question about contexts calling each other. We have one * box that is setup for multiple companies. Calls come into the default context and that hands them out to the context for each company. For example, 1x goes to context1, 2x goes to context2, etc. Each context includes "outbound" which says that if you dial 1+ or a local number, you are sent out to the Cisco

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

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks???

Have, i want to create a sip extension to a context in my dialplan. how i can do that?

Are there any configuration options to allow certain sip/iax accounts to dial out over specific trunks, and also to stop them dialing out over other trunks. Thanks in advance Bails

Hi, I'm trying to configure Asterisk as SIP-H.323 Gateway via ooh323, but I have an error relatively to the GK Confirmation message. >From the log: "H323 RAS channel creation - succesful Sent GRQ message Gatekeeper Confirmed (GCF) message received ERROR:No Gatekeeper ID present in received GKconfirmed message Ignoring message and will retransmit GRQ after timeout Error: Failed to

Greetings list, A quick question regarding extensions.conf #include behaviour if I may. I'm sure someone will know the answer off the top of their head... How does asterisk handle "overloading" of contexts. For example, say an extension exists in extensions.conf as follows: [incoming] <some stuff> Then one includes a, b and c.conf, each of which also contains: [incoming]

Hi all, I'm struggling some time now with this problem. Googling and searching on this topic did not deliver the answer yet, so my last hope is this list. Analogue to the things which are possible with modem.conf, where I can configure the MSN's to act on, I would like to have similar functionality. This is the idea: I have 1 ISDN line, it can be reached by 4 different MSN's. I have

I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the "avoided deadlock" message below. *CLI> == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on 'SIP/3211-1-081c40a8' -- Executing NoOp("SIP/3211-1-081c40a8", "") in new stack -- Executing AGI("SIP/3211-1-081c40a8", "diallocal.agi") in new