similar to: 32 bit server is ok?

hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have two Asterisk servers running on the same LAN. One starts fine, but when I start the other I get: pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use and Asterisk does not start. OK I thought, I'll just change the port in dundi.conf. I changed it to 4521 and indeed it started just fine, until I reboot that is. WHen I reboot, I get the same

hi i am using asterisk 1.4.22 ubuntu 8.4 i have two Ethernet one for ssh and other one only for voip calls when i start a call using originate in the manager or the cli in the voip Ethernet i get something like 4Mbits/sec of traffic only 1 G711 call. if i start the call using a soft phone everything is normal. any idea? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak <michiel at vanbaak.info> wrote: > On 22:46, Mon 15 Sep 08, hugolivude wrote: >> I have two Asterisk servers running on the same LAN. One starts fine, >> but when I start the other I get: >> >> pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: >> Address already in use >> >>

hi wich predicitive dialer are you using and wich one do you recomend? a link to the project/product and a link to a how to will be VERY apreciated. Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi this is mi first email and just for say hello. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/375a91e5/attachment.htm

hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part --------------

hi i am thinking to use some xorcom hardware i only used digium hard... what do you think? it work ok? what about echo? it is plug and play only reload or i need to restar asterisk ? i understand the drivers are in the trunk of asterisk, is this rigth? have any one a conf file of a xorcom device? any other important comments are welcome. Thanks David -- (\__/) (='.'=)This is

hi there are a lot of virtualization solution out there and every one "is the best" and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your

Hi all! Im looking for 1ezphone to use as a web softphone but I?cant access to 1ezphone.com. Anyone knows what happened with this site?. Thanks! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Modified httf.conf file and added : ------------------------------------------------------ Alias /recordings/ "/var/spool/asterisk/monitorDONE/" <Directory "/var/spool/asterisk/monitorDONE"> Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all </Directory> Created a folder under vicidial as recordings. FULL_RECORDING is also enabled.

Hi all.... I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk release all calls...I checked the log file and found.. [Feb 28 11:34:14] NOTICE[19197] chan_sip.c:

hi i want to ask for your opinion what is better for a call center 100 current calls and other 200 current calls make the server step by step or use a auto install cd like asterisk now, druid elastix ....? and why? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part

hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi where i should load the module for the trasncoder wctc4XX (lspci shows TC400P) thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi sorry about the urgent but it is urgent i have problems configuring a connection between asterisk and avaya using H323. the module i am usign is ooh323 what do you need to help me? and any tip or hint? thanks!!! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998