similar to: SIP and error talking to voicemail

I just got an email from SIPphone advising that there have been problems with the above firmware and advising to reload from their server. This does in fact reload 1.0.4.35 into the phone. And now voicemail has gone AWOL again. -- Dave Cotton <dcotton@linuxautrement.com>

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 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

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",

Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ

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

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

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

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 inadvertently unplugged my Budgetone phone tonight and I think it went out and upgraded its firmware. It is now at 1.0.3.81. Does anyone know how new this might be? Suddenly things that have been working wonderfully before don't work--basically it seems to do the SIP stuff just fine but then all the RTP stuff breaks, even something as simple as retrieving my voicemail from a server on

Hi All, Is anyone use the sipcall.co.uk 'professional' account with a UK geographic number? What do you think of the service? Alternatively, who else are you using to terminate a UK geographic number on asterisk? Thanks, Nathan. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.529 / Virus Database: 324 - Release Date:

Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does Asterisk support

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

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire underground Sent: Friday, January 09, 2004 1:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Screen Pop & Remote Agents > can I put a .csv file in the sql DB and have it dial from there? and will I be able to set a > Dial Plan to

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) -

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