[Asterisk-Users] Lucent TNT / Asterisk Help

Title: Message Anyone out there have any experience with setting up asterisk and a Lucent TNT...I need some help with setup and configuration...I have no experience with Lucent...I have referenced the wiki on voip-info.org but does not do much good since i also need help configuring the

[Asterisk-Users] Lucent TNT / Asterisk Help

be appreciated... thanks, Jeromy Global reach, local touch... Jeromy Grimmett CEO Comuniquémonos, Inc. / SA 1803 Swathmore Street Alexandria, LA 71301 [EMAIL PROTECTED] IM: MSN: [EMAIL PROTECTED] http://www.comuniquemonos.com tel: fax: mobile: +593 (4) 390 0022 +1 (501) 646-0680 +593 (9) 366

[Asterisk-Users] TNT / Asterisk Help

be appreciated... thanks, Jeromy Comuniquémonos, Inc. / SA Jeromy Grimmett CEO [EMAIL PROTECTED] 1803 Swathmore Street Alexandria, LA 71301 tel: +593 (4) 390 0022 fax: +1 (501) 646-0680 mobile: +593 (9) 366 6521 IM: MSN: [EMAIL PROTECTED] http

[Asterisk-Users] Lucent TNT / Asterisk help

Title: Message Anyone out there have any experience with setting up asterisk and a Lucent TNT...I need some help with setup and configuration...I have no experience with Lucent...I have referenced the wiki on voip-info.org but does not do much good since i also need help configuring the

[Asterisk-Users] Quintum Asterisk w/ Realtime

Title: Message Hi all, I am currently using the latest CVS head to do realtime and it is working just fine so farhowever, I was asked to use a Quintum Tenor AX gateway for testing out the SIP send/receive call abilities in Asterisk and unfortunately I have no experience with Quintum

[Asterisk-Users] AGI/PHP errors

Title: Message hi all, im trying to get a php file to run thru AGI...but when i run it through *this is what i get i have the latest php installed ver. 5...had ver 4.3.11...but had this problem so i figured an upgrade to the highest would resolve it... im running CVS Head and

[Asterisk-Users] NAT RealTime

Title: Message I have read the info, and its not clear... does RealTime support NAT traversal and message indicators or not? below is the information from the RealTime page on voip-info.org... Olle explains the world In the new RealTime architecture, all database specific code is moved

[Asterisk-Users] OH323 problem

Title: Message May 28 01:58:32 WARNING[6821]: chan_sip.c:1830 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)May 28 01:58:33 WARNING[6821]: channel.c:2126 ast_channel_make_compatible: No path to translate from SIP/3901506-efd7(4) to

[Asterisk-Users] Another OH323 Problem

Title: Message anyone got any ideas on this? TDM H323 Gateway SIP Inbound H.323 call 'ip$200.93.237.82:12984/2853' detected.Channel OH323/R2853 created and attached for inbound H.323 call 'ip$200.93.237.82:12984/2853'.Setting channel 'OH323/R2853' (ip$200.93.237.82:12984/2853) native

[Asterisk-Users] TDM * rings once and goes fast busy

Title: Message Hi all, Any help with this would be appreciated: we are running 0.7.1 OH323 for asterisk...here is the scenario... TDM H323 Gateway Asterisk SIP Endpoint the SIP phone rings one time and then the origination TDM phone goes fast busy now...on the other

[Asterisk-Users] IAX Firefly config

Title: Message hello all... newbie question: I have FireFly setup on my laptop and I would like to test this out using IAX in this scenario: FireFly Softphone Asterisk TDM Gateway i do not wish to use this on the firefly network, but simply within my own "3rd party" network as the

[Asterisk-Users] Windows IAX Softphone

Title: Message Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users

RE: [Asterisk-Users] LOOKING TO HIRE

'Can't we all just get along' and move to a pure Microsoft environment?!?!?! HAHAA Comuniquémonos, Inc. / SA Jeromy Grimmett CEO [EMAIL PROTECTED] 1212 South Hampton Drive Alexandria, LA 71301 tel: +593 (4) 287 3854 fax: (501) 646-0680 mobile: +593 (9

RE: [Asterisk-Users] LOOKING TO HIRE

PROGRAMMING LANGUAGE DEBATE!!! KICK THE RECRUITER OFF THE LIST!!! DOWN WITH THE HEAD HUNTER REGIME!!! Off with their heads!!! Comuniquémonos, Inc. / SA Jeromy Grimmett CEO [EMAIL PROTECTED] 1212 South Hampton Drive Alexandria, LA 71301 tel: +593 (4) 287 3854

RE: [Asterisk-Users] LOOKING TO HIRE

Just when I thought the horse was shot dead...I suppose we should take shifts and start beating the hell out of it again...Any voluteers? Damned horse just doesn’t want to die... Comuniquémonos, Inc. / SA Jeromy Grimmett CEO [EMAIL PROTECTED] 1212 South

RE: [Asterisk-Users] H323 to SIP

Comuniquémonos, Inc. / SA Jeromy Grimmett CEO [EMAIL PROTECTED] 1212 South Hampton Drive Alexandria, LA 71301 tel: +593 (4) 287 3854 fax: (501) 646-0680 mobile: +593 (9) 366 6521 IM: MSN: [EMAIL PROTECTED] http://www.comuniquemonos.com -Original Message

[Asterisk-Users] H323 to SIP

Title: Message Hi all, Of course I am a newbie, so please bear with me... I'm having a lot of trouble getting things to work properly and I am sure it is a configuration issue somewhere, I'm just not sure where...I have been all through my extensions.conf and cannot seem to see a

RE: [Asterisk-Users] Is SKYPE a threat or should we do something(together)

I am very interested in helping with this...I am currently working on a project that may be very useful in making this happen... Comuniquémonos, Inc. / SA Jeromy Grimmett CEO [EMAIL PROTECTED] 1212 South Hampton Drive Alexandria, LA 71301 tel: +593 (4) 287

[Asterisk-Users] H323 to SIP

Title: Message Hi all, Of course I am a newbie, so please bear with me... I'm having a lot of trouble getting things to work properly and I am sure it is a configuration issue somewhere, I'm just not sure where...I have been all through my extensions.conf and cannot seem to see a

[Asterisk-Users] H323 to SIP

Title: Message Hi all, All outbound calls work perfect from my SIP ATA 186... SIP ATA 186 (w/ private IP) Asterisk H323 (public IP) TDM works perfect TDM H323 (public IP) Asterisk SIP ATA 186 (w/ private IP) fast busy with this error message in the CLI of Asterisk: May 17