similar to: skinny problem

Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEP<MACADDRESSHERE>.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24

I have recently played with and did get a working copy of Asterisk functioning in the lab environment. Then when I moved it out of the lab and onto the outside of our firewall it functions as before except that I get a Connected to Asterisk CVS-08/21/04-12:16:04 currently running on 63 (pid = 1478) ??? -- Starting Skinny session from ??.???.114.131 (PIX (6.3) our firewall) Aug 26 14:48:34

Hi everyone, I want to have a group and dial multiple phones/lines simultaneously. If I use this Dial command: exten => 222,2,Dial(${TRUNKBP}/246&SIP/258&${TRUNKBP}/243,20,tTr) ... all phones ring just once, after that only the first one continues ringing and only that one can answer. Can anyone tell me why? thanks! Tomica -------------- next part -------------- An HTML

Can someone tell me what is wrong here: Feb 2 19:45:44 ERROR[1074441696]: cdr_addon_mysql.c:381 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database is created, cdr table also, the username and password is right. I have tried configuring cdr_mysql.conf to connect via localhost mysql.sock or via tcp port, but in both cases I got this error. Thanks!

Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a

Hi all: I've got some core dumps with chan_skinny. The client is ATA186 with v2.16.1.ms ata18x (Build 030814a). The * version is CVS-10/05/03-16:03:26. When I make a call, the phone connected with ATA rings only 1 time and * dies. Maybe I have some errores in ATA config. If someone has proven configs for ATA, please send me the details. Thanks in advance, Gus The logs: *CLI> Version

Hi everyone, Maybe someone could help me. I have Asterisk in production with TE410P connected to PSTN. When I call from internal phones, either voip or connected via other PRI trunk, to PSTN and if the called phone is busy I don't hear anything!?! I should hear tone indicating that called number is busy. I have played with busydetect and callprogress in zapata.conf, but I didn't find

Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica ---- This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr

Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone

Hi all, Can any one please help me in intergrating PHP/Mysql with my running asterisk server to configure IAX or SIP users? I will highly appreciate any help in this regard. Regards Nawaz. --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, >

Need an example or help with dialing a PBX with SIP and then entering an extension number. I know how to use SIP to get to the auto attendant on the PBX. If I send or type an extension number during the prompt, it will go to that extension. I don't know how to send the extension when SIP has obtained the PBX. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, Skinny gurus. (btw, I'm super pleased to see development happen on this). Thoughts on this?? I added this context to my skinny.conf: [ppc] device=SEP00022D494F2A context=employees line => 50 ; Dial(Skinny/1@ppc) I've downloaded the 30 day Window eval of VTGO! (PPC) from www.ipblue.com and it hangs on Registering. *CLI> skinny debug Skinny Debugging Enabled

Hi to everybody my system is be attack, but I dont know what this means [May 27 15:12:24] WARNING[26018] chan_skinny.c: Partial data received, waiting (76 bytes read of 786) [chan_skinny.c] skinny_session[0][C-00000000] skinny_session: WARNING[May 27 15:52:32] Asterisk 13.8.0 built by root @ asterisk on a x86_64 running Linux on 2016-04-04 19:02:51 UTC [May 27 15:52:32] NOTICE[2306] cdr.c: CDR

This is the core trace (gdb) bt #0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6 #1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828 "default", interpclass=0x0) at res_musiconhold.c:646 #2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of bounds>, interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614 #3

Hi, I am having some problems mentioned below, the box is in production live environment with traffic around 30 - 100 calls. I am running T/E410P in a Dual P4 xeon with HT disabled. I am using zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql, perl (small script) and asterisk. System runs very smoothly if the calls are around 40-50 and comes one by one , however sometimes

I am working with someone that has 3 locations in the state in three different cities, connected by routers, with the main office using a Linux server with a WanPipe board replacing the router. It looks a bit like this: OFFICE #1: Internal IP: 192.168.7.??? Linux (Internal): 192.168.7.100 Linux (To OFFICE #2): 192.168.8.101 Linux (To OFFICE #3): 192.168.9.101 OFFICE #2: Internal IP:

Hi, the last 2 days i was working on getting the 7920 Phones to work with Skinny & Asterisk; however no luck (yet). Does anybody has a SEPDefault.CNF.xml and a SEP<mac>.CNF.xml handy for me ? it should be documented at the cisco page, but it isn't :-( I still have the issue that the 7920 spits out "No Service - IP Config failed" but Asterisk is giving me sign that the

Dear Skinny/SCCP lovers :-) I've just completed & uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... --jan (*1) http://chan-sscp.sf.net (*2) yes, bugtracker is down at the moment, will fix