search for: internalextensions

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

...=> s,1,Answer > exten => s,2,Wait(1) > exten => s,3,ResponseTimeout(2) > exten => s,4,DigitTimeout(6) > exten => s,5,Background(bgw-ThanksForCallingBgw) > exten => s,6,Background(bgw-IfYouKnowExt) > exten => s,7,Background(bgw-OtherwiseHold) > include => internalextensions > > exten => t,1,Dial(ZAP/3,15) > exten => t,2,Background(bgw-ThankYou) > exten => t,3,Hangup > > exten => i,1,Background(bgw-SorryInvalidEntry) > exten => i,2,Goto(s,5) > > exten => h,1,Hangup > > [fromzap] > ignorepat => 9 > include =&gt...

First off, please, for the love of God, don't cremate me, if I should already know the answer to this! I've installed a small setup for an office who wanted to be able to talk to each other instead of having to rely on MSN to communicate. Weird request, I know, but hey, we do what we need to do to get paid. I installed soft phones, gave everyone an extension, and bingo, they can call and

...; Address to bind to context = incomingcall ; Default for incoming calls tos=lowdelay disallow=all allow=alaw allow=gsm allow=ulaw [104] accountcode=vsllc type=friend context=employee username=104 secret=**redacted** host=dynamic qualify=yes reinvite=no canreinvite=no mailbox=104@internalextensions,750@internalextensions callgroup=1 pickupgroup=1 dtmfmode=rfc2833 ;nat=no -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/d29e2a97/attachment.htm

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

Hi I am having alot of difficulty connecting to SIP providers (I am trying 3) and can't seem to find anything similar in the wiki or on the lists.....I can receive inbound calls fine however on placing an outbound call the calling phone never gets 'connected' but 2 way audio is passed for about 20secs before some sort of timeout. Anything suggestions as to what I could try

...Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') pbx*CLI> Sip read: 0 headers, 0 lines -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/internalextensions/230/INBOX/msg0008 format: wav49, 0x8144680 Jun 24 00:56:31 WARNING[7507]: app.c:619 ast_play_and_record: No audio available on SIP/233-a3ba?? -- User hung up -- Executing GotoIf("SIP/233-a3ba", "1?menuinternal|t|2") in new stack -- Goto (menuinternal,t,2) -- Exec...