Re: [asterisk-users] FW: Redundancy

session border controller can be a solution. Thanks Khaled [EMAIL PROTECTED] wrote: Dears Do any one have an idea to make a redundant plan for asterisk ,if a call established between two points and the server interface became down ,do we you have an idea how to let the

[asterisk-users] Looking for toll free in Italy

Hi, I am looking for tollfree number in italy. Anybody providing that? Charge per minute? It will connect to my asterisk pbx box. Thanks CM __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

[asterisk-users] DiD for less then $4

Anybody selling DID flat rate for less then $4 with sip or iax incoming? Let me know Thanks CM __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

[Asterisk-Users] Error on AAHome Beta 4

Hi, I have installed AAH beta 4 and I am getting this error. I have installed it from aahbeta.tar.gz so I can make the server dual boot. this is what I am getting in error, any clue how I can fix this? Thanks Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission

Re: [Asterisk-Users] Error on AAHome Beta 4

CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] WiFi Phones

I am also interested to find out about wifi handset tha voipsupply.com sells. is it any good? Thanks James H Thompson wrote: Anyone have good words to say about any of the WiFi handsets currently available? Thanks. Jim

[Asterisk-Users] Asterisk at home and Asterisk 1.2 beta

Any chance anybody has asterisk at home with asterisk 1.2 beta? any problem if I reinstall the beta on top of asterisk at home? Thanks CM ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] oh323 and IAX2

Anybody here using iax2 for one call leg and other call leg for oh323? I am getting broken sounds from Iax2 call get. Can somebody here help? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] OH323 call leg and IAX call leg

will be the correct set? I am using Codec 729 for both call legs. Anybody has any idea on this issue? Thanks CM Rahman Jr. CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)

-users -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- CM Rahman Jr. CTO CCS Internet www.ccsi.com

[Asterisk-Users] CDR Server

does anybody know any CDR server in public domain or low cost? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Comments on Areski Calling Card Solution plz

/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] oh323 version 0.6.6.

Hi, I have downloaded asterisk-oh323-0.6.6.tar pwlib-Janus_patch4-src-tar openh323-Janus_patch4-src-tar pwlib and openh323 compiled fine as instructed. When I tried to compile asterisk-oh323 I am getting this and anybody know howto fix this? [EMAIL PROTECTED] oh323]# cd

RE: [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show

/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Asterisk and 2 line MGCP phone

HI, Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to figure out if there are such device available and if so, how does it differenciate between the lines that is associated with extention number. Thanks ___ Asterisk-Users

RE: [Asterisk-Users] Asterisk and 2 line MGCP phone

-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[Asterisk-Users] matching sip connection under sip.conf

Anybody know how to match under sip.conf and cisco 53xx ? It looks like due to dynamic port number, it is not able to authorize it. Here is what I get under debug Using latest request as basis request Sending to 216.236.160.15 : 5060 (non-NAT) Found no matching peer or user for

[Asterisk-Users] dynamic port problem !!

Is there a way to fix this problem? I am using cisco 5300 to connect to asterisk but it is failing to recognize by IP due to port number changing. Unfortunately, Cisco send sip request with changing port number. What can be done? No matter what I put on sip.conf, I couldn't get it to match. Any

Re: [Asterisk-Users] G723.1 and G729 on Athlon 64

://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] PSTN-5300-asterisk(sip)

I know there was a posting regarding how to configure 5300 and asterisk so I can dial pstn and get connected to asterisk. Can somebody share the sip.conf and dial-peer config with me? Thanks ___ Asterisk-Users mailing list

[Asterisk-Users] 5300 to asterisk

Hi, Anybody here configured 53xx to connect to asterisk ? So, if the pstn call is made it will go through autoattendant on asterisk? If you have it, please share your sip.conf and dial-peer of cisco. Thanks ** C.M. Rahman Jr.

[Asterisk-Users] callback broken?

HI, Can somebody tell me how to get callback working? I have put a script in /var/spool/asterisk/outgoing but nothing happens. even the debug shows nothing is happening. But the file gets erase. How to check what is happening? Thanks ___

RE: [Asterisk-Users] callback broken?

PROTECTED] On Behalf Of CM Rahman Jr. Sent: Monday, April 18, 2005 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] callback broken? HI, Can somebody tell me how to get callback working? I have put a script in /var/spool/asterisk/outgoing but nothing

[Asterisk-Users] IPP g723 and getting error when starting asterisk

The compilation of codec g723.1 was fine. After I have copied to /usr/lib/asterisk/modules and started the asterisk -c .. I get this below error before asterisk quit. Anybody had any idea on Intel codec 723.1 ? [codec_g723.so] = (G723.1/PCM16 (signed linear) Codec Translator, based on IPP)

[Asterisk-Users] oh323 on @homeasterisk

Anybody here added oh323 to @homeasterisk? I have compiled and add the oh323. I am wondering if anybody able to add the oh323 under web interface AMP? If anybody did it or know how to do it, please let me know. It has option for sip, IAX.. why not add h323 !! Thanks

RE: [Asterisk-Users] Help with simple callback application from newbie

Discussion Subject: Re: [Asterisk-Users] Help with simple callback application from newbie On Apr 6, 2005, at 8:31 PM, CM Rahman Jr. wrote: I am looking for same type of solution. Anybody here can help? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf

RE: [Asterisk-Users] Help with simple callback application from newbie

I am looking for same type of solution. Anybody here can help? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herbert Chan Sent: Wednesday, April 06, 2005 12:06 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help with simple callback

[Asterisk-Users] SRV and Asterisk

HI, Anybody here used SRV feature in Asterisk to route sip calls? If anybody were able to get this working, I like to know how the sip configuration was set. I am testing this in a lab with two different domain. I was hoping Asterisk will be able to route the call by doing DNS look up