similar to: sip over h323

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &

Hi ! I have a small problem. When switching a call (pstn -> sip user), I get the sip phone ringing - ie. everything is OK, but I do not get a ringtone in the handset on the pstn side. Can anyone help me out in how to make * play tones ? My setup: E1 IP pstn ------ Asterisk ------ sip phone Regards, Dave

Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. Thanks Neil -------------- next part -------------- An

with a h323 client over my gatekepper a call comes over asrerisk to my sip endpoint: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial("H323/ip$10.0.0.124:49638/18690", "SIP/0699073201") in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing -- SIP/0699073201-dc61 answered

Using 'Samba version 2.2.1a' Please could someone offer advice in order to configure the following. I need domain level authentication for most shares, however 1 share needs to be made available to a none domained NT client, such that perhaps share or user level security would be appropriate. How can this be configured on a single samba server. Thanks in advance to anyone who responds

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Hi there I am pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2

Hi Developers, I have a 16-bit PCM data buffer, I want to write that to ogg file. Could you help me to understand how to write pcm data to the ogg? Thanks in advance. Arun balaji -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20150518/6c8c9a63/attachment.htm

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

I've got two patterns I want to match on making an outgoing call... (one day - to do Least Cost Routing for Cell/Mobile calls) Firstly - I prefer '0' rather than '9' to get an outside line... Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084) or its just another number to dial... I added the following... the playback just advises me which 'route' is

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

I've downloaded rsept31.zip, unzipped it and have been able to run it under OS/2. When I start it I get an error (error no. 6) but the package seems to run just fine using OS/2 Warp 4's Win-OS/2 version of Windows 3.1. (I always have to tell it wherre the rprofile is. Is that normal?) I've tried to run the same Windows 3.1 version of R on a computer running Windows 3.1 (actually

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

hi, I am new to * and setting up a test system. here my setup : - debian (from knoppix 3.3) - Asterisk 0.7.1 (from the debian package) - AVM Fritz card used with i4l - softphone I use for testing SJphone on windows - I can make great softphone - softphone calls - I can call from an outside line * and get connected to a softphone here my problem: I can not make outbound calls. I place a call

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the

I am having dial out issues and was hoping someone could shed some light. The problem is Intermittent: extensions.conf [globals] ; Trunk Info for outbound calls via PSTN - See the zapata.conf file in /etc/asterisk TRUNK=ZAP/G1 ;Trunk Interface ;MSD digits to strip (usually 1 or 0), 1 = remove a leading 9 TRUNKMSD=1 ; -------------------------------------------------- ; [trunklocal] - Defines

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24