similar to: More complicated huntgroups / delayed ringing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

Hmm. By the way, please don't post bugs to asterisk-dev as I've been told :> That list if for on-going development. That sounds like a bug I encountered in 1.0.5. There is a division by zero bug in chan_iax2.c introduced somewhere after 1.0.4 I believe and currently fixed in HEAD. (They've given me enough shit for posting the bug while it was fixed in HEAD already. No need to

You could try to use the native mp3 support for MOH if you really want mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept doing nasty things to my system :) See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon hold.conf there is a section about the native support. Guillaume > -----Original Message----- > From: Stefan Gofferje

On Tue, 15 Feb 2005 03:02:45 +0100, Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I have some trouble with the FOP and would appreciate if anyone could > point me into the right direction. There is a FOP user list, although not too active. http://www.asternic.org/ > Is there a way to define a button like Zap/g1/6000 and have it light up > when

I'm developing asterisk to work in a small Call Center for a Mobile Communications Provider. I'm looking for references for Agent Logins and Huntgroups. If any of you have already configured Asterisk for such a Task and would be willing to let me take a look at your config files could you please email them to me at mailto:jhelmich@bluesky.as Thanks, Jason Helmich MIS, Blue Sky

I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Thanks much. BTY: Does anyone have a How-To on getting the 7960 fully configured

Is anyone successfully using this phone with Asterisk? There is a lot mentioned about CISCO but nothing about Nortel... Alex.

Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply

Hi! I run a mongrel cluster with 6 mongrels in it. I want to monitor them individually for process hangs (and then restart them) and this is the solution I came up with: Here''s my configuration file for monit (/usr/local/etc/monitrc): [snipped relevant bits] ------ #check lighttpd process check process lighttpd with pidfile /var/run/lighttpd.pid start program =

Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. Thanks in advance. Regards Dave

Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>: > > Do you really want to detect "ChallengeSent"? That should occur also on > legitimate login processes... > Hi , strange thing is that I still have not this asterisk in production and I see many attempts Connection. Now keep in mind that when a connection of authentication is successful the

I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a