similar to: chan_h323: Different ports for both media channels (in, out)

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

Hello all, I just installed chan_h323 by Jeremy (nufone) with asterisk-head. well, I have some questions if someone please briefly answer. 1 - can we configure h323 to work in g729 pass-thru mode, like we do in SIP. becaz I had to installed g729+intel libraries to work with g729. where as we don't need to install g729 to work in pass-thru mode. 2 - now most important question is

hello, i have problem with oh323 channel driver (tryied differnet versions). when i try to make call between oh323 - sip, oh323-isdn, oh323-capi asterisk crash with segmentation fault. Channel driver was compiled with pwlib 1.5.0 and openh323 1.12.0 libs. Does anybody know solution ? WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Dial("H323:31119",

Hello, I think I have the same problem as Martin Bene mentioned in http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html Since I found no further information about this I'd like to ask wether you know what the reason for this problem is and how one can get around this. * is registered to the innovaphone gatekeeper. Trunk connection is done with an AVM-B1 and chan_capi.

Hi thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service. I have drafted a mini-how-to which is available at http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf This is a first draft, I will amend this further, in particular the "verify and debug" section

Sir, Here is the working Audiocodes MP-11X FXO configurations to work with Asterisk. ;************** ;** Ini File ** ;************** ;Board: MP-118 FXO ;Serial Number: 905371 ;Slot Number: 1 ;Software Version: 5.00A.024 ;DSP Software Version: 204IM => 209.13 ;Board IP Address: 192.168.0.195 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 192.168.0.1 ;Ram size: 32M Flash size:

Hi, My setup is: IPPhone1 --- Asterisk1 with B410P ---- Patton 4638 --- Asterisk2 --- IPPhone2 I want to evaluate Asterisk1 in TE/PtmP mode. So, Patton box is configured in NT/PtmP (with 3 BRI links between both systems). Anyway, asterisk -rx "pri show spans" keeps replying : PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned,

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk