similar to: music on hold file

Good day all I got my Music on hold to work but can I/how do i get music on transfer? Please help Thanks

Good day all I added the music on hold entry in vpb.conf and commented out default line in musiconhold.conf. Asterisk starts up with the default mp3 but as soon as I remove it and add my mp3 it just doenst start up and gives a broken pipe error? Please Help or advice Thanks ALtus

Goo day all I'm going to put a asterisk server running sip in at a client.The server is going to have a public ip so that it can talk to another server. My question is how do I secure asterisk/sip. I got a firewall only allowing tcp/udp 5060? I got sip to work with md5 What more? Please Advice Thanks a million

Good day all How is asterisk version I'm having problems with 1.0.0 and 1.0.1 If I'm starting asterisk it give problems with the modules saying unresolved symbols Please give some input Thaks Altus

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

I'm trying to install spandsp But when I try to patch the Makefile it gives this error [root@pbxmilkshake apps]# patch < apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on

Hello all. Has anyone managed to get SIPCALL.co.uk's service working with the [*] box? I've managed to register with other SIP providers but not SIPcall. The debug just show's [*] attempting to register. But receiving a 401 error everytime. Cheers Matt

Hi all ! Let's say you have about 6 small different companies sharing the same E1 / Asterisk server, and every company needs its own fax number. Since they don't really need fax machines, what would be the most cost-effective way to handle this (keeping fax-privacy at its best) ? Is there a way to configure Hylafax or sth & one modem behind an ATA-186 to email faxes to different

Hi, I used to have a Grandstream phone connected to Asterisk a few months ago. Worked just great! Then today I do a new install, rather than an upgrade, and all of a sudden I cannot check voicemail with it. No problem calling or receiving call. It simply speeds through the vm greetings but I cannot hear them. If I check the same VM with an analog phone it works fine. So I wanted to check

Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301

How to change available options (behaving) during listening of voice mail? (They are unnecessarily complicated) For example, I don't want to press 3 (advanced options) and again 3 for envelope. I just want to play envelope. Also, when saving message, I do not want to choose folder, I want that message as default be saved in old messages. And, I don't want to press 6 for next message, I do

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Running slackware 9.1 with compiled kernel from source 2.6.5 running ok. I have 2 HFC-S chipbased Billion Bipac PCI ISDN BRI cards installed in PC. Would like to use one card as in TE and one in NT mode. System works fine running pbx4linux.But want to use SIP functionality, so I would like to try out the Asterisk. Trying to install the bri-stuff.0.0.2.tar.gz (May 10 2004)package, getting the

Just idly watching * in console mode and saw that someone from 50.49.54.102 tried to register with my *. whois gives:- OrgName: Internet Assigned Numbers Authority OrgID: IANA Address: 4676 Admiralty Way, Suite 330 City: Marina del Rey StateProv: CA PostalCode: 90292-6695 Country: US NetRange: 50.0.0.0 - 50.255.255.255 CIDR: 50.0.0.0/8 NetName: RESERVED-50

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or

Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S 09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S 11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S

Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold

Hello, I'm writing because we have problems with an asterisk installation (Trixbox ver. 1.2.3). We have a customer which is receiving a lot o telephony traffic (more or less 1 call/2 min.); we are using a TDM400 board, with 3 PSTN lines configured and we have two big issues: - Calls are dropped during conversation (I have a busycount=8 from the initial value that was 4) -