similar to: automated dialout detect forward

hi community, I'm new to this list & asterisk in general, so let me first say thx to everybody involved in providing such great tools & ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk.

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are

G'day. I am having reasonable success getting Asterisk 1.4.2 running and doing what I want, but I can't figure out one particular idiom that I want: There are a few situations where I want to have Asterisk push a call through to the first available transport on a list, such as: I have two SIP ports attached to one local (two port) analog phone system. I want to ring line 1 for the

Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707

I've a big problem with SIP forwarding back into 'ringing groups' creating what can only be described as call storms :-( I have a 'ringing groups' of SIP phones with an effective dialplan (much simplified) like so: ; Purchase ledger [ptsn_inbound] exten => _846061,1,Dial(Local/6061 at groups) .... [groups] exten =>

Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour

Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open!

Hi, Anyone know what the settings in SIPDefault.cnf should be for Cisco 7940 phones this year? Came in today to find they'd all moved one hour ahead (NTP server is correct and ok). Found the "day" was set to "26", but on trying to change the settings to the below, my test phone isn't changing back: dst_start_month: March ; Month in which DST starts dst_start_day:

All, I'm trying to configure queue agents w/ a DUNDi setup so that an agent can login to whatever server they please w/o any custom setup. In general this seems to work, agents login w/ AgentCallbackLogin into the incoming context (not a special queue context) and can receive queue calls. The problem is that since the incoming context is the same context as the normal incoming call context,

I am trying to build a simple queue for the receptionist phone. In other words, there is only 1 agent and that is the receptionist phone. I just defined a few lines in queues.conf [console] strategy = ringall member => SIP/4000 ;4000 is the console extension In extensions.conf, it is: exten => 4000,1,Answer() exten => 4000,n,Queue(console) exten => 4000,n,HangUp() I pressed

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

Hello, I'm glad to announce that DeStar 0.2.2 version has been released. This release contains a large number of bugfixes and new features, see CHANGELOG.txt for the full list. You can find it in the usual place: http://developer.berlios.de/project/showfiles.php?group_id=2112 Thanks for using DeStar, Santiago Ruano Rinc?n http://destar.berlios.de -------------- next part -------------- A

Hi, in the earlier version there was a sounds.txt with the transcript of the soundfiles. Does this still exist somewhere? Is there a plan to make speech synthesis available the same way as soundfiles, ie. instead of playing language/soundfile.wav, send the text to the speechengine and play the output...? Jay... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call.

Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

Hi, Been playing with Call parking, and I can`t help but wonder if I am doing something incorrectly. The way I understand it (using default config in features.conf), is I would transfer a call to extension 700, which would park the call, tell me "701". I could then hang up, go fetch the fright person and tell him "call 701 you have a call waiting for you". The way I

Hi folks, i'm trying to configure my extensions.conf as small as posible and for that reason i'm using macros. The problem is that maybe I have a misunderstood the concept for the directive "mailbox" in sip.conf. Under my knowledge configuring the mailbox directive to the mailbox I want would be enought to leave an retreive messages in that voicemail box. Of course it seems to

I was wondering if it is possible (in 1.2.x) to get the SIP response code back after doing Dial(). Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and some are NOANSWER, but I want to know the SIP response code, so I could return the right tones to the user, not just a congestion tone for every fault. Anyone know a way to find out that information, so I want the