similar to: Multiple SIP registrations for one account?

Hi, I've been asked to see whether it is possible to do call logging for call center environments at a lower budget than the usual $1000 per channel. Afaik, with PRI this is possible through a high-impendance Y connection, but I wonder whether this would work with the Zapata cards. Anyone ever tried this? Regards, Cees -- XP SP2 can cause cancer in rats

I have updated the voice prompts (mostly small fixes, but still work left to be done - however, it's usable now, I'd say) and the patch file (to current CVS) for Asterisk-in-Dutch. As far as I can tell, all of the grammar work is now in - if anyone has feedback to share, please do so before I spend lots of time cleaning up the patches so they can be integrated back.

Hi there, here is my attempt to initiate a "restart when convenient" from a software SIP phone. exten => 588,1,Answer exten => 588,2,Wait(1) exten => 588,3,Playback(restart-convenient) exten => 588,4,Wait(1) exten => 588,5,Authenticate(00000) exten => 588,6,System(/usr/sbin/asterisk -rx "restart when convenient") exten => 588,7,Hangup The problem: We

Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel

Does anybody have useful tips on creating good quality recordings for use with prompts in asterisk? I'm interested in hearing input on hardware (mics, dats, sound cards, etc) and software (recording software, dsp) as well as recording techniques. Jerimiah Tularosa Communications

I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

I have experienced lots of apparently dropped packets (in other words, lots of short interruptions of what the other party tries to tell me) with a GS102 and chan_capi. The GS102 is connected through a lightly-loaded switch directly connected to the * server, so bandwidth/latency shouldn't pose a problem. Funny thing is that the switch indicates 10mbit on the GS102 port - is that correct?

Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no

Hi! I'm new here, and I was wondering if there is any newsgroup gateway to the Asterisk lists? Thanks! Testa

I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI (chan_capi 0.3.0), and have a couple of funny things - I wonder if anyone else has seen them: - Now and then, * just exits. Until now I had lowish-level verbosity on, so all I saw was 'Executing last minute cleanups'. What can trigger * exits? (in other words, what should I pay attention to when attempting to

I have also noticed that sip.conf doesnt get updated without a restart..... was thinking I am doing something wrong, but maybe not now...... Chris -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, 8 January 2004 8:42 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Another concern I have on this

Hi, I am registering my Asterisk boxes to a SIP provider for outgoing calls. My "outgoing" dialplan context tries to dial out in sequence, starting with the SIP provider then ISDN lines and finally analog lines. So the idea is that if the SIP trunk fails then all calls are dialed out via ISDN and analog. I noticed however that if I switch my DSL connection off (ie. no internet access

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is

Hi Guys, i have recently connected my (working) asterisk 1.2 server, with two 1.4 asterisk servers (one using SIP the other using IAX), since then (i believe) people starts complaining about a high background noise when using the handset on Polycom phones (but when using the speaker it's fine, and i noticed that my self), my question is, can anybody tell me any step to begin diagnosing the

I have just completed a set of voice files in Dutch, plus a patch that forces Asterisk to sane (i.e. Dutch ;-)) behaviour when composing dates, times, numbers, etcetera. The current release, 0.0.1, is a sort of pre-release - some known issues have been identified, but I nevertheless would like to have some feedback so we can do a minimum number of polishing rounds. The patch is a bit

Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil

Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. We configured 4 SIP account on Patton (1001, 1002, 1003, 1004). The system is fully functional, but we have a problem to recognize incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. I have contacted Patton

Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp