similar to: Dialplan Design Q

I'm trying to clarify contexts and their uses. I do have a good general understanding of them. My question is about "undeclared" and "non-existant" contexts. If I have a block somewhere (in sip.conf, for example), and it has no "context=thiscontext" field, does it just automatically use the "default" context? Or is this settable? (I see there is an

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, 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

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

I know this has been asked before, but I've checked the archives and I haven't found anybody that has given a definitive yes or no, just "yeah, it should work.....". If I have a T.38 gateway like a Cisco 5300 and a T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work? I have it setup and it doesn't work, so I want to know if I am doing something wrong,

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

Is it possible to show CallerID names for dialplan applications? When I call from phone-to-phone, it shows the CallerID from sip.conf or iax.conf, but I don't know of any way to show CallerID Name when I call the extension for an application (voicemail for example): exten => 1000,1,Answer exten => 1000,n,VoicemailMain I'd like the display to read "VOICE MAIL" when I

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

I'm trying to setup a primitive announcement-paging system in my house using the line-out from my * box to a cheap amplifier that runs to speakers on our first and second floors from the basement. I have a extension that connects to Console, and console is set to auto-pickup. I'm using alsa drivers. This all works great, except for one thing. I want to play a tone over the console after

Scenario: Both boxes are behind firewall, port udp 4569 is open. If I don't want the username and password in dialing string do I have to use register statement in IAX.CONF. Can anybody post some working samples; I have a hard time making it to work with the samples posted on wiki. -- #Joseph

Estimada comunidad tengo un problema del que no encuentro datos que me ayuden mucho en la web. Estoy haciendo calculos por grupos con data,table. Tengo un archivo (zp.res) con tres columnas que clasifican los datos (sol, con, dia) y una columna de datos numericos (media), de la siguiente forma: sol con dia media 1: con 0 1 -22.6 2: con 0 1 -36.6 3: con 0 1 -35.6 y

I'm trying to setup a primitive announcement-paging system in my house using the line-out from my * box to a cheap amplifier that runs to speakers on our first and second floors from the basement. I have a extension that connects to Console, and console is set to auto-pickup. I'm using alsa drivers. This all works great, except for one thing. I want to play a tone over the console after