similar to: Oz ISDN

I need clarification as to DID in T1 connection. T1 provides 24 channels for voice/data. Do it assign each channel to particular DID. Or you can have unlimited DID to share the 24 channel as an example. ie. Outgoing/incoming traffic is not bound to particular channel. Whatever is available will be used according to the grouping in zapata.conf. -- David Kwok, CISSP Tel: 612 82315701 ext 1002

I'm planning a system, just need to know if it works with Telstra's network. Cheers, Clint Sydney, Australia

On 11/22/05, Sharon Kimble <sharon04 at gmail.com> wrote: > Does anybody know the gnome program that you can write in latex with please? > > I'm not sure but I think its in the base repo. Its not winefish as it > (winefish) doesn't compile the source text like the program that I'm looking > > for does. > > Hopefully thanks > Sharon. > -- I am

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

Does anybody know the gnome program that you can write in latex with please? I'm not sure but I think its in the base repo. Its not winefish as it (winefish) doesn't compile the source text like the program that I'm looking for does. Hopefully thanks Sharon. -- 16:41:46 up 1 day, 16:56, 2 users, load average: 1.54, 1.45, 1.51 A taste of linux

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

I'm looking at getting a T1 into a location in Melbourne, Australia and was wondering if anyone has a good source and pricing for this. Cheers, Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061019/deded6a9/attachment.htm

I'm experiencing a very unusual problem with a few of my sipuras. They keep over heating after minimal usage. I unplug them and let them sit for 15 minutes and it starts working fine. The other way around is that I put a fan on it or something to keep it cooling. Has anyone else experienced this or are they just defective units? Regards, Mohammed Salim EZZI Telecom, Inc.

Hi all! I am intrested in the following scheme My mobile phone -> SMS to SOMETHING -> Redirect to FWD number -> FDW redirect to my * -> My * doing smtg There are companies like calluk.com that provide DIDs for free, but they do not support SMS. In http://www.voip-info.org/wiki-Asterisk+cmd+Sms they say "Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK"

Hi, I'm after some ISDN card resellers in Australia. I've noticed the AVM cards should work well here, however I've found the reseller here has no website, is listed in Yellow pages as management consultants (??) and when I called quoted almost $400. Does anyone know where I can get the AVMs in Australia a bit cheaper or another card (not the Netjet) that works with Asterisk.

I have, but I will be happy to review it once again. On 12/18/18 2:14 PM, admin wrote: > Am Dienstag, den 18.12.2018, 14:06 -0500 schrieb Albert E. Whale, CEH > CHS CISA CISSP: >> >> I have two servers pointing to an NFS mounted mail spool with >> dovecot.? Since I recently switched from using Dovecot v1.X, I have >> been experiencing high CPU use with the two

Is anyone aware of an milliwat/echo test number for telstra or similar? I want to fiddle a bit with gains but can't seem to get a hold of a miliwatt test number Im aware of a number they distribute for line quality testing for modems for BigPond, but I can't track it down .... -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part

What is the simplest OS I can install in Xen, preferably Linux based, and don''t tell me MS-DOS :). I have installed a minimal version of Fedora, but I want something that is even less cluttered. I have looked at Minix, but it is not fully supported. I need something small, lightweight (doesn''t have to be multi-threaded or anything), that will run small C and Assembler programs

I have 1 * box having x100p installed and the other has no zaptel card at all. both of the * box has compiled in ztdummy module and both have been activated by modprobe ztdummy. When using trunk to connect the 2 *box. The one without zaptel card complaint about unable to accept trunked packet: no matching peer. On the one has zaptel card I have tried to remove the ztdummy module and connect

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

is it possible to do with asterisk something like this how ? maybe some copy of extensions and some other file ..pls. call to asterisk server -> user 100 is busy -> recorded msg say "all lines are busy , pls wait" ,"you are second caller in 'queue', pls wait" -> caller is on hold till user 100 busy then ring user 100 (ext100) another call to asterisk

I have two servers pointing to an NFS mounted mail spool with dovecot.? Since I recently switched from using Dovecot v1.X, I have been experiencing high CPU use with the two Dovecot servers. I am not certain why they are not well behaved.? Here is the configuration information. This configuration is currently running at a load average of 17. /usr/sbin/dovecot -n # 2.1.15:

Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060

Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and