Re: [asterisk-users] Audio dropping

+0200, Mark Scholten wrote: What could the reason be audio in 1 direction is dropping? (Normally from the Asterisk server to the mentioned SIP clients.) No clear information is in the logs (it is like the call ended normally) and not all calls are having problem (most not, but it happens to often

Re: [asterisk-users] Audio dropping

-27 at 10:31 +0200, Mark Scholten wrote: Hello, We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. SIP clients (all behind NAT at different locations, so not a single NAT solution is used): - x-lite - linksys pap2t - polycom kirk (multiple type numbers) - polycom

[asterisk-users] Audio dropping

if it was the firewall we disabled the firewall on the Asterisk server and moved the Asterisk server before the other firewalls we have. What could the problem be? And even more important what could solve it (and/or explain it)? Kind regards, Mark Scholten

Re: [asterisk-users] A few questions regarding Asterisk 1.8.0

Good luck as with any new version there may be some bugs so if you bump up against ones report them so they can be fixed. Also don't just drop it into production with out testing it on a box for a bit. 1.8 has a lot of changes. Most appear to be for the better. The only important difference I

Re: [asterisk-users] Door Contacts via Asterisk?

Hello, We did something like that in the past (but for 1 company, but it shouldn't be really different). The easiest solution for us was to use a door opener that could work with almost any normall phone connection and use a Linksys pap2t or something similar. With kind regards, Mark

[asterisk-users] A few questions regarding Asterisk 1.8.0

Hello, I have a few questions regarding Asterisk 1.8.0. If you can answer a question, please do so. Is Asterisk 1.8.0 stable enough for production environments? Is it possible (and if yes what is the best option) to use CDR MySQL with Asterisk 1.8.0? With 1.6.x we use the add-on package for

Re: [asterisk-users] FAX Options

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Alejandro Imass Sent: Monday, August 02, 2010 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FAX Options

Re: [asterisk-users] # -key not to be 'transfer'

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Sunday, August 01, 2010 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] # -key not to be 'transfer' Hello list,

[asterisk-users] Confirm answering a call

Hello, I am working on getting the following to work and I couldn't find it in the documentation I did read. Where should I look or does someone have an example how I can do it? Current situation: Incoming call - 3 SIP phones + 2 mobile phones ring - if mobile phone goes to voicemail the call is

Re: [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of John Novack Sent: Tuesday, May 04, 2010 12:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridging old

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Saturday, May 01, 2010 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ATA shootout: