similar to: Oh323 and Asterisk with MERA

Hi, We seem to be having some teething issues with a new Hylafax - happy to pay someone to complete the installation. Please contact offlist. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499

Hi, I'd like to know how I can playback a pre-recorded message to a user using our system without answering the call. I want to do the above in the scenario where the user dials a number and the number has been dialled incorrectly. Regards, Sahil Gupta VoiceValley

Hi, I have a Gateway running in "TE" (terminal equipment mode as "slave" that I need to connect to my asterisk server using a TE100P card. Can anybody give a quick run up of how to run the TE100P's in Network Termination mode to have this working sucessfully? Cheers! Regards, Sahil Gupta VoiceValley

Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley

I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background("OH323/RXXXXX", "vm-extension") in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing

Hello. I'm try to set up asterisk for making outgoing calls with oh323 channel driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37. Our provider uses Mera MVTS softswitch and supports only H.323. We don't use gatekeeper for connection but provider requires SOURCE PHONE NUMBER for route out calls and I don't know how I can specify this number. Call with this string exten

Hi, We seem to be having an interesting issue with Asterisk whereby, it keeps routing calls coming in to the 'default' context.... regardless of what changes occur to h323.conf. <SNIP> [POP-A] type=user host=1.2.3.4 context=international </SNIP> == Starting H323/ip$1.2.3.4:12914/16313 at default,12126599878,1 failed so falling back to exten 's' == Starting

Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sip----ASTERISK----h323-----GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

dear all, we need help for integration asterisk (sip) with mera we have configure for sip.conf and extentions.conf sip.conf [mvts] context=mvts type=friend host=10.10.0.2 dtmf=rfc2833 in extentions.conf [mvts] ; ; mvts exten => _01162.,1,SetCallerID(mvts) exten => _01162.,2,SetCIDName(to mvts) exten => _01162.,3,Dial(SIP/${EXTEN:3}@mvts) i need if i dial 01162 in mera replace with

I'm having trouble to send calls to a Mera MVTS softswitch (with SIPHIT) when the asterisk box has a dynamic IP address. If the Asterisk box has a fixed IP, everything is OK. Any ideas? I'm looking for a working sample of the sip.conf in this case... user.cfg (for MVTS) is also appreciated if any special setting should be done there also.

Hi guys, I've been trying to update my chan_oh323 from 6.1 to 6.3b. I built asterisk from cvs-head on the date Micheal said he made it compatible, pwlib-1.6.6 and openh323-1.13.5 (both with nothing more than the ./configure, make, well aplied patch on openh323) When I start * with my normal config I get this: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing

I've complied the OH323 .so successfully and can easily receive calls from my H323 gatekeeper (using 711u), however it seems that all outgoing calls are refused and I'm getting "reason 23 (Temporary failure)" as an error code which I can't find documented everywhere. My H323 gatekeeper needs a 001NXXNXXXXXX to dial out to the PSTN even if I'm in north america (Montreal)

Hi there. I thy to compile asterisk-oh323-0.6.3a but it fail in the make command. I have the pwlib-v1_6_6-1 and openh323-v1_13_5-1 as saying in the README file of the packet asterisk-oh323-0.6.3a I do make and this is the error: # make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target

i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

I've got problem with connections h323 -> sip and sip -> h323. I've Cisco 7940 phone with sip soft and Netmeeting as h323 node. As gatekeeper I've gnugk and brand new asterisk from cvs + chan_oh323 0.5.5 When I call from Cisco (SIP) to h323 node by alias registered on gatekeeper and h323 node will answer the phone... I have on my Cisco still Ringing. Call termination, no

What is the difference between H323 and OH323 in Asterisk? I need Asterisk to have basic H.323 support so we can offer some simple H323 termination for some of our Cisco and Quintim hardware. Our upstream provider uses SIP, so I figured I'd use Asterisk as the go-between. I already setup Asterisk so it can push calls out through our providers via SIP. I just need a good/solid/very simple H323

Try these guys: http://easyspeedy.com/ Haven't tried them, but when I was looking into a while back they responded quickly. -- Bjorn -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Sahil Gupta Sent: Wednesday, May 03, 2006 1:47 PM To: asterisk-users@lists.digium.com Cc: asterisk-biz@lists.digium.com Subject:

I am trying oh323(version 0.67) , make call from sip UA to h323 gateway, can't get Call-id pass from sip UA to h323 gateway, h323 always gets call-ID sent from Asterisk as *. are there any configure to pass the correct call-id from sip UA to h323 gateway? or this is a bug in oh323 0.67? how about oh323 0.73 ? Mario On 9/29/05, Kanishka Somaratne <kani@technoportal.biz>