similar to: SV: Outbound Caller ID number on E1

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:

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

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

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 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

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

Im curious. Does anyone have experienced echo-problems that later where solved by buying a hardware-echo canceller such as the Wildcard TE411P? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r James Harper Skickat: den 6 februari 2006 11:46 Till: Asterisk Users Mailing List - Non-Commercial

I don't quite follow you? There are USB headsets that don't require a soundcard at all. They have a built in soundcard which (I suppose) could be better than the crap they build into most laptops. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r El Flynn Skickat: den 24 maj 2006 10:17 Till:

>From what I understand it means that the *hardware* in your computer *acknowledges* the call as soon as it is recieved and then sends it to asterisk dialplan for processing. You would essentially need to put the delay before the call ever reaches asterisk. So this problem isn't asterisk related... if I've understood your question and the answer I found correctly. Regards, Jan

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

What kind of help do you need then? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Help on queues There is no good help on wiki and voip-info.org, I've

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

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

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

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 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.

Thanks for the quick response! I read about logrotate at voip-info.org but I didn't quite understand it. I'm no asterisk/linux expert unfortunately. First of all. What exactly does happen when I run: /usr/sbin/asterisk -rx 'logger rotate' Does it clear the file and create a new one? Can I run this manually without any interruption in the system? And what does the script do? I

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

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