similar to: Re: sip change? (Rich Adamson)

Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for seqno 102

Kind of off topic but I know CVS is the "prefered" way of upgrading, however are there such things as "stable" CVS upgrades? It seems a lot of the CVS's have a lot of devel bugs in this that I would be scared to put even near production. Just IMHO. :-) Matt -----Original Message----- From: Rich Adamson [mailto:radamson@routers.com] Sent: Friday, August 27, 2004 9:15 AM

Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta > From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to >

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Interesting story. I was following up any wild goose chase i could to find a resolution including cleaning dust out of the machine. At 4 pm I plugged the line in and called for the sake of luck, and it was back to working. In the mean time, thanks for the troubleshooting advice. I will keep this in my log book so I have the steps for next time something goes awry. ------------------------------

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

> Date: Sun, 19 Sep 2004 12:59:52 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: [Asterisk-Users] Re: [Asterisk-Dev] X100p on VIA EPIA-V > problems > To: asterisk-users@lists.digium.com > Message-ID: <Chameleon.1095617288.adar0@vegas> > Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > > > I'm running Asterisk on my (new) VIA EPIA-V

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing

I have quite an old version of Chameleon Mail, currently the prompts played when leaving a message are ? -- Executing VoiceMail("SIP/209-3b0e", "u5") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

More than one hundred messages related to *8 or call pickup problem in last 6 months!! Please someone in the development team could clarify this and make himself responsible for the response. By now It seems a bad joke. We have spent thousand dollars with hardware, sip phones, working men hours, and with digium stuff (E1, fxo, fxs cards etc) and we have had the *8 problem (sip callee ringing

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

There is an option that you can add to your dhcp server option 150 IIRC. -----Original Message----- From: "Rich Adamson" <radamson@routers.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: 2/27/06 4:37 PM > Has anyone done any integration work with Covad's hosted solution ? I > am considering

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello, I've got dovecot 1.2.11 running on Fedora 12 (x86_64) with the managesieve protocol and sieve plugins enabled, with deliver handling local delivery. I've set disable_plaintext_auth = yes ssl = yes ssl_cert_file = /etc/pki/dovecot/certs/dovecot.pem ssl_key_file = /etc/pki/dovecot/private/dovecot.pem mail_debug = yes protocol managesieve { listen = *:2000 login_executable =

I am trying to set up Icecast and IceS. I have icecast 2.0.1 up and running. but when I try to start IceS 0.3 I get a seg fault. Here is the output: # /path/ices -r -c /confpath/ices.conf -F /playlistpath/playlist.pl -P password Logfile opened DEBUG: Sending following information to libshout: DEBUG: Stream: 0 DEBUG: Host: localhost:8001 (protocol: xaudiocast) DEBUG: Mount: /example1.mp3,

No, DON'T use mpg321, it will not work. Make sure you're using mpg123. See the wiki. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Rich Adamson > Sent: Friday, October 01, 2004 12:13 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: