similar to: Get older CVS version

While compiling Asterisk from one month ago cvs checkout -D "last month" asterisk I got compiling error: term.c:55: conflicting types for `term_color' include/asterisk/term.h:47: previous declaration of `term_color' term.c:98: conflicting types for `term_prompt' include/asterisk/term.h:49: previous declaration of `term_prompt' make: *** [term.o] Error 1 --------------

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

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

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

I used: cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds iaxyprov astcc and in the same order I try to compile it. Asterisk ends with the lines below. It complains of a newer libpri, but I just did it a step before! What do I miss? chan_zap.c:62:2: #error "You need newer libpri" chan_zap.c:128: error: parse error before '<<' token chan_zap.c:133:1:

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>

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'

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

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

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

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) -

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 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

Sorry if this show up twice, yesterdays post did not show up. We are having trouble setting up two ext's on one phone. We have it the point where the first two lines are ext 4000 then the third line is ext 4013. We can receive calls to both ext's but we can't make out going calls on ext 4013. The other thing that is strange is we need to have ext 4013 at the top of the config for ext

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

Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G