similar to: ATA's and faxing

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.

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

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

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 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

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

Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney

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()

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

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.

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.

As the subject line says: Is PING a good indicator of network latency? If not, how can I measure latency? Thanks, Cosmin Prund

Hi All According to http://bugs.digium.com/view.php?id=6457 this has been resolved since 04-11-2006 and I have seen mentioned since 1.2.7. I have tried using mixmonitor on asterisk 1.2.13 and 1.2.17 with the exact same results. The WAV files are recorded but are cut short. I am using a b410p card on the box. When using HFC based cards I have no problem with the recording. Does anyone

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

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator