similar to: Updated Asterisk-NL

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

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

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

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

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

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

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

Hi, For various reasons a customer of mine is moving from a SER-based to an Asterisk-based installation, mostly because of problems with SIP devices behind NAT trying to reach each other and because it's easier to do accounting when all calls go through Asterisk (canreinvite=no is the idea). The database-based SIP registration mechanism of Asterisk seems to have one shortcoming - it

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

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

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

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?

I have a rule that sends all mail from root to a mailbox, but I want it to NOT send mail from mailing-lists there. if allof (address :contains :localpart "From" "root", not anyof(exists ["List-Id","List-ID","Mailing-List", "X-List-Name","List-Post"])) { fileinto "root-mail"; stop; }

Forgot to mention, sometimes I have to do force start other volumes as well, its hard to determine which brick process is locked up from the logs. Status of volume: rhev_vms_primary Gluster process TCP Port RDMA Port Online Pid ------------------------------------------------------------------------------ Brick spidey.ib.runlevelone.lan:/gluster/brick/rhev_vms_primary

[It seems the last time this came up was in march, under the heading of "Sony Cybershot should be in hardware notes". This message is intended mainly to document what I've managed to track down.] The m500s still will not sync with pilot-link 0.11.7 in -STABLE. An easily triggerable panic is another issue [1]. The pilot-link code first opens /dev/ugenX and then switches to

Dear Dan, thanks for the quick reply! I actually tried restarting all processes (and even rebooting all servers), but the error persists. I can also confirm that all birck processes are running. My volume is a distrubute-only volume (not dispersed, no sharding). I also tried mounting with use_readdirp=no, because the error seems to be connected to readdirp, but this option does not change

Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to <sip:bla@voiphost> from "Bla

Hi, I've been on the list for slightly under a month now and noticed; a) a fairly high amount of traffic, b) a lot of questions which come up more than once, c) the archives at lists.digium.com are not searchable. I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done

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

Concur ... I reckon that you need a bunch of how tos, that you can sent these rude people. I think that you are a victim of your own success. Samba has hit the mainstream, and now you are paying the price :-) Still ........ I prefer these messages to those filled with hypertext ... or worse.... 2 Meg of mime formatted drivel :-) Has anybody thanked you recently for providing so much help on