similar to: Updated: No audio when dialing in via PRI with Q.SIG

When inserting Ringing() before MeetMe()-conference picked up the call, everything works like a charm. I guess the PRI needed to see the ringing status before the call was answered. This is however never needed when dialing a SIP-extension or similar. I have also an update considering bad PRI b-channel numbering. It seems that only my first 15 channels actually work. Then our PBX tells Asterisk

I've been trying lots of configurations now. And the problem that I can't solve is this: I have a Digium T205P card. I have connected one of the connections to our internal PBX (NEC 2000 IPS). The Asterisk is configured as pri_cpe, and the NEC is configured to be the network side of the connection. Both ends are using b-channels 1-15 and 17-31, the d-channel is on 16. When I start

Add an Answer() as your first step in your dialplan and see if that help. snip

Hi, I'm using samba in RedHat Linux 5.0. It works almost ok... First of all, I think everything is MUCH to slow. I am running it on a Pentium Pro 200MHz with 64Mb RAM. And there is only 5 clients using it. I'm using version 1.8.19p3 on an RedHat 5.0 Server. Is there something special I need to do about the filelocking?? Or have I just done something wrong during setup...? Before this

Hi: I was wondering whether there's a way to bridge two conference bridges using Asterisk. I want to allow a meetme conference to join an external conference over the PSTN. One way of doing it, in theory, would be to use Omniis' TAPI driver and place a CAPI call to the ISDN line (external conf.), while the driver calls an internal meetme conf on asterisk over SIP. Didn't have luck

Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a

Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is "caller id name" transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ

Hello, I am using a Cisco-2,811 as a gateway between the Asterisk PBX and our Nortel TX-1 university's PBX. It is working but no names are exchanged. From the debug mode I see that the Cisco sends the display name (which does not appear on the Nortel's phones) and the Nortel does not bother to send it at all. I recall that when I had a pilot with Cisco CCM two years ago we had to set

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrated via CAPI and the chan-capi channel driver into my Asterisk). MWI works just fine there. I

>-----Original Message----- >From: Larry Shields [mailto:LJ.Shields@Verizon.net] >Sent: Friday, August 27, 2004 12:20 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() >If I assign the DID to ring extension SIP/2000 and then after time-out send >it to MeetMe() or Playback() it works and the caller

Hi, To make a long story short, I can't install any TAPI driver on my XP platform. A. Within Config Panel|Modems and Telephony options|Advanced parameters, I've got a list of 7 TAPI drivers. Among them is Omniis TAPI driver for Asterisk. B. I can properly configure this driver (line, context, ...). C. When I open Outlook 2002 Contacts panel, I can select "Call this contact"

Hi, Is there any softphone or TAPI plug-in that allows one to dial from a web page? As you may know, Skype has a mechanism that converts phone numbers on a web page to a click-to-dial application. I'd like to use this but on a normal softphone (Bria, Xlite, other). Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been using asterisk tapi from omni for a few days now with outlook and it rocks, thanks guys (you should set up a paypal account, even if you don't want the cash nominate a charity or something). Anyway long story short, I'm using asterisk@home and I would like from time to time to be able to transfer people directly into a meetme room extension 800. Is there a way

Had I have been smart originally I would have done this to start. Some rudimentary documentation above and beyond Asttapi 0.10's poor documentation is available along with the download at http://www.kirkhamsystems.com/asttapi. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent:

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

Hi, Is there a reliable way to auto-dial SIP phones (specifically Polycom) with some sort of TAPI driver in Windows? I am aware of SIPTAPI, which makes the user's phone ring, and when picked up dials the desired number, but I (and more to the point, many of my customers) find this annoying. I'd like the phone to autodial on speakerphone (or headset if there is one), without any human

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

<div dir='auto'>Maybe LLD is lagging behind, but is the TAPI stuff that's landed close enough to the original such that Apple cctools could be modified to use it?<div dir="auto"><br></div><div dir="auto">Cheers,</div><div dir="auto"><br></div><div

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

Oof, I didn't think about Clang not being in the same place. Perhaps we could put this in clang-tools-extra to solve that? As for the unification of the code bases. I was assuming we didn't want to just throw a ton of code over the wall anyway so the merge was going to need to be reviewed chunk by chunk anyhow. Support for the two formats should be possible to add in parallel (although, I