similar to: Internal FXS to SIP problem

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

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Hi, I got back from two weeks away and appear to have lost audio on my tdm411 fxo. Everything was working properly when I left. I checked the logs, config files and can't see any problems, the zap channels and modules are all loaded properly, asterisk starts without probs and everything looks sweet on the colsole with -vvvvvvvvc when I make calls, but I just don't hear a dialtone or

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

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

Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion >

It would be nice if R ignored more unicode white space characters. For example, if I have "\u2028" in a command (which I get from a line-break in keynote) I get the following error: > qplot(carat, price, data = diamonds, colour=clarity) Error: unexpected input in "qplot(carat, price, data = diamonds, ?" And occasionally have such problems when copying and pasting from

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

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 All, I've spent a day trying to get a Polycom IP500 wokring with my Asterisk box. I have several others that are working fine, but this one is getting by me. Can someone on-list tell from the following SIP debug what I've missed? Sip read: INVITE sip:2000@192.168.1.30:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E From: "2004"

Dear all: In fact, even I add the user in the command, the rsync`s error is still appearing: >./bin/rsync -avz tong@192.168.1.30:/BGI/UCSC-mirror/bin/rsync-2.5.4/man ./backup-3000/ rshd: 0826-813 Permission is denied. rsync: connection unexpectedly closed (0 bytes read so far) rsync error: error in rsync protocol data stream (code 12) at io.c(151) another try: >./bin/rsync -avz

Hello list, I was able to get passive mode worked out. I'm really glad I was able to do this. I'm able to log into the ftp server, list directories, enter subdirectories and upload/download files. However my next task is to enable virtual users using mysql. I have installed proftpd-mysql and enabled the sql modules in the config. I found a good article on how to do this here:

Dear all: I installed the rsync 2.5.4 in my two machine (192.168.1.30 and 192.168.1.120, both are AIX OS) to backup data each other. The software`s installation is ok, and I can copy local files. But when I try to backup data between two machines, there reports some errors: >./bin/rsync -avz 192.168.1.30::web ./backup-3000/ rsync: failed to connect to 192.168.1.30: Connection refused

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of

I'm replacing my old Fedora 7 mail server with a new one running CentOS6.3. The old server uses plain-text logins and password for pop3 and IMAP. I'm unable to get dovecot to authenticate. It's failing the password check. Trying 192.168.1.50... Connected to orion (192.168.1.50). Escape character is '^]'. +OK orion.crucis.net Dovecot ready. user joe +OK pass abcdefg -ERR

Hi all! I''m new to the list :-) I''m having a weird problem in which I''m a little bit lost right now. I''ve got a machine (Debian) connected to 2 different networks (A and B, with 2 different net cards), and a web server that is listening on the second one (B). I think I''ve configured correctly the rules and routes, but I''m not being able to

Hi there. I have just installed DOVECOT 1.0.1 for the first time. After launching it, I can verify that ps -ax | grep -i dovecot 1052 ?? Ss 0:00.03 /usr/local/dovecot/sbin/dovecot -c /etc/dovecot/dovecot.conf 1053 ?? S 0:00.06 dovecot-auth 1058 p2 R+ 0:00.01 grep -i dovecot ps -ax | grep -i imap 1054 ?? S 0:00.08 imap-login 1055 ?? S 0:00.08 imap-login

I have a Cisco ATA-188 with two POTS phones and latest stable cvs. In any situations with call waiting (existing connection and calling again) the second call cause both calls to drop. This is the same for "internal" extensions and from external (ZAP and SIP). It seems to be a "502 - The transaction could not be executed, because the endpoint does not have sufficient

Hi list, I install dahdi-linux successfully with the module of oslec for the echo, but when I specify it in the system.conf the echo canceller oslec it shows me errors: DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22) I see that the echo cancellers is supported: mg2, kb1, sec2, and sec because oslec is not supported?, but he has support to compile it with dahdi_linux! best