similar to: Error Message.

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

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

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, 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>

Hello, I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I

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

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

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'

Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070117/93bc7fdb/attachment.htm

Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070126/a49e3bdb/attachment.htm

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 cannot use CVS, is there anoyher way to use direct pickup ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Alexander Lopez Inviato: venerd? 16 settembre 2005 17.53 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: RE: [Asterisk-Users] direct sip call

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

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 just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql.

it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c greg Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When

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

hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from '<sip:@172.16.4.249>' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Rodney