similar to: Passing Digits between ISDN PBX and Asterisk

Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics & Computer Science) ----------------- Mobile: +27 (0)83 791 6662 Email:

Breeze to set up, too. To monitor and transfer to SIP/1000 / ext 1000: 1. Insert exten => 1000,hint,SIP/1000 into your default context (the context the extension is in) 2. In the monitoring phone's web interface, click Function Keys, pick a key, change it to Destination and type in SIP/1000. Once you submit the form it will change to a SIP URL, that's OK. 3. There is no step 3.

How do you set the CallerID? Have you checked with your provider that they've enabled callerid? If yes, are you using a correct number that the provider allows? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Garth van Sittert Skickat: den 2 februari 2006 12:37 Till: Asterisk Users Mailing

Hi, I have just installed a Sirrux PC140 card for the first time. Managed to build the drivers and get * to load them on FC4, but it does not work. It seems that layer1 in the ISDN is not even activated. The same ISDN lines connected to a Samsung DCS works so it is not the lines. I am including my sirrix.conf and the output of some of the * Srx commands below. Any pointers would be

Hi All Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in the query? I have: exten => s,5,Set(query=SELECT name from contacts where tel like %${number}) exten => s,6,MySQL(Connect connid hostname username password dbname) exten => s,7,MySQL(Query resultid ${connid} ${query}) But there seems to be a problem with the % sign and I don't know how to

Hi All Has anyone come across a handset that can somehow replace FOP? Some users don't like FOP unless it is on a dedicated PC. Thanks Garth

Hi All I have noticed that when calling over mISDN channels the call itself is made over the channels listed in misdn.conf eg: 1,2,3,4 as misdn/1, misdn/2, misdn/3, misdn/4 etc. However, when looking through the CDR's the call gets logged as misdn/0, misdn/1, misdn/2, misdn/3, misdn/4, misdn/5 etc. I also noted that busy, failed, congested calls link mainly to misdn/0. The only logic I

Hello all! Sirrix AG, Saarbr?cken (manufacturer of the Sirrix.PCI4S0 4-port ISDN card for Asterisk) has released the new Asterisk dialplan command PPPD (app_pppd). It allows to connect a Linux PPP daemon to an arbitrary digital (ISDN) Asterisk channel to provide RAS dialin and dialout. The PPPD command has successfully been tested with Sirrix.PCI4S0 cards and a standard ISDN4Linux ipppd "on

Hi All Does anyone have any ideas around what processing power is needed when bridging PRI channels and recording? I am not sure how the bridging takes place with and without recording? I basically have a situation like this: Telko <----> Asterisk <-----> Legacy PBX Where the lines are PRI's between Telko and Asterisk and Asterisk and the Legacy PBX. At what level does

Hi All Has anyone experienced a crash specific to asterisk 1.2 and Centos 5 when using the misdn hfcpci module that comes with zaptel? I have an asterisk pack based on asterisk-1.2.17 that I have been using on dozens of machines that are rock solid and stable. Today when I tried moving it to Centos 5 I experienced a complete OS crash when calling over HFC misdn channels. Didn't really

Anybody tried one with Asterisk yet ? Views ? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration

Hi guys, How's things going ? Got a bit of a weird one here that I've been unable to solve. I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box - this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says

Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404

Hello! I am selling hardware I have been using in various Asterisk test installations. If you are interested in buying all or any of the following items please reply off list. I got a few * X100P clones * Sirrix PCI4S0 ISDN BRI cards (+ 1 Sirrix S0 voltage/power supply) * 1 Polycom Soundpoint IP 500 (+ Plantronics headset) * Conrad HFC-S cards to give away. Some of the pieces are unused,

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of "call center". That is, we want to get a few land-lines from our telco in different countys and "bridge" all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP

Hi list! Does anyone have experiences with the updated model of the Kirk IP600? The V3 model is supposed to support SIP instead of only SCCP or H323 which would make the use with Asterisk a lot easier. I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is still giving me severe headaches : - the standard Skinny driver in * doesn't work, only the version of Sergio

Hello, I had had some echo issues. I purchased a digium echo canceling card, and the echo issue seems to be reduced but not eliminated as I hoped it would be. I currently have it set to 128 'yes'. I've tried 256, but when I try 256 what happens is that instead of getting better echo canceling I get AWEFUL echo. Can anyone enlighten me? I am running 1.2.6 with a 4 port PRI card.

I am trying to set up a linux based faxing solution for a client, and have found that the modem they have (ancient dataplex external unit) just isn't up to the job. It talks to some remote fax machines but not others. A new external modem ranges from AUD$75 to AUD$400, which got me thinking of other possibilities... #1 FXO PCI card (more expensive for 1 port, probably cheaper for 2+) #2

Hi All Does anyone know how the hints in asterisk works? How does a SIP phone interact with the hints? I am having a problem with certain phone models that do not set the hints correctly when I list the hints with a 'show hints'. Thanks Garth