similar to: Sangoma and Digium same machine?

Anyone know how (or if it's possible) to get a logo on the screen of a Polycom 501? I've been looking around for hints on how to do it but so far nothing would indicate Polycom supports doing it. -bill

I've got a Zhone 24 port FXS to configure. The configuration is beyond stupid. The people that designed this unit should be chased down and fired. I'm going around in circles frigging with all the options. Does anyone have a config file for this unit that I can use as a starting point? -bill wlloyd@slap.net

One of the suggestions I got for dealing with my analog FXO woes was to look into using a Voicetronix card instead of the Digium. They're more expensive but if it worked better it may be reasonable. They indicated they were compatable with the CVS HEAD version of *. Can someone share with me their experiences with Voicetronix OpenLine and/or OpenSwitch cards with *? I'm specifically

I too have been having SIP problems the last couple of days, I get the same message as Louis-David but in this setup only: PSTN >-ISDN-> AS5300 >-SIP-> * I can make outbound calls no problem but inbound calls seem to stall, according to 'sip debug' it just says 'Ignoring this request' but I cant establish why .... > -----Original Message----- > From: William

I'm in search of useful ACD type statistics from the queues. Ie talk time, ratio's, dropped calls etc. The flat file queue_log is nice, but more useful would be the data in Postgres or Mysql. Unfortunately the queue module does not yet support ODBC DB logging (yet). In the meantime this quick and dirty hack gets the job done. Replace the flat file with a unix named pipe.

Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 "Not Found" error. Here are the relevant sections from my configs. sip.conf: context=broadvoice-in register =>

Any one has documented how-tos for making voicetronix openline 4 to work with Asterisk. I have been contacting Australian Digium resellers and Digium cards are not approved in Australia. So I suppose Australian users are interested into putting Voicetronix in use. Any expereience to share will be most appreciated. David Kwok -------------- next part -------------- A non-text attachment was

I'm trying to get a patch tested for inclusion in CVS. Anyone that is running E&M on a T1 and had to fool around with emdigitwait could you please try this. This patch removes the need for the emdigitwait parameter and speeds up dialing. This situation is mostly interfacing a legacy PBX/Key system with Asterisk. I've been running a couple of systems with this patch for

I'm sure this is easy and I'm missing something, but how can I add an extension into the directory command (ie get's it's list of extensions from voicemail.conf) yet have no voicemailbox for the extension? Basically I have an extension that gets forwarded onto somebody cellphone where they use their own voicemail. I'd like to be able to list them in the company

Scenario is as follows. Caller comes in over ZAP channel connects to handset on another ZAP channel. Call is bridged. I'd like the callee to be able to hangup on the caller and then be presented with a agi application. Basically the agent that answered the call has to enter a few responses to questions asterisk asks. On some ACD phone systems this is called a "wrap code".

I've been trying to work with the dynamic feature support.. IE adding codes like *2 to features.conf that can trigger a dialplan application to run. I've been unable to get "goto" to work properly. "AGI" also seems to not function correctly if called as a feature. Anyone else playing around with this feature might have some insight? -bill wlloyd@slap.net

Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE

Hi to All, I dont know much about PCI express slots in newer Servers, my doubt is if the Digium and Sangoma PCI express cards, are compatible with the x8 PCI express slots that come in the HP Proliant ML150 G5 server. Thanks in advance. Ricardo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij leon at scarlet-internet.nl

Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pau?p

Has anyone had experience with one or both of these cards? I'm in a position where I might need to recommend one over the other. I've read everything that I can find online, so now I'd like to hear of personal experiences. Everything I read on both cards is "5 stars! Awesome! It Rocks!" They both seem to have similar capabilities, similar pricing, etc. Could those of

Evening all. I'm looking at building an Asterisk system for one of the projects of the Charity where I'm the SysAdmin. The project has two analogue phone lines - BT Featureline Compact, we're in the UK - that I'd like Asterisk to handle. My current quandary is which FXO interface to use. I've been looking at the Digium TDM400P and Sangoma A200, which are similar prices for

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a

Hi all, I’m running Xen 4.1 on a couple of NetBSD dom0s and NetBSD’s pkgsrc provides both the xl and xm tools for working with Xen 4.1. I understand that xl is the new way of doing things, but I can’t get it to create my guests the way I want and the main symptom of this is that the text console isn’t available. When I create a guest with xl it starts up qemu-dm (which xm doesn’t do) and I get