similar to: wireless

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

has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto:

When setting include => daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

At the present time you have to have a VALID ip address in bindaddr for Skinny to work. If bindaddr is either 0.0.0.0 or simply commented out all packets requiring the IP address contain 127.0.0.1. I forgot their nick, but someone in IRC recommended we make Asterisk be smart enough not to pick that interface, but I'm not sure of that is the problem or not. I simply have not had the

Dear Guys, One of my customers ask me for a Wireless IP PBX, well the Asterisk fits IP PBX, But what about Wireless, ans also portable, a very special cutomer :) 1.-one possible solution is a PDA with one of those SIp softphones but VERY EXPENSIVE for the PDA 2.- a fxs channel bank and lots of Panasonic like wireless phone but the question is 1.- can my customer use all the features

When trying to dial out 982420173 our main number I get the engaged signal before I finish entering the phone number Any ideas ???? Regards Mick

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 list! I'm trying to configure the Kirk IP 600 wireless (DECT) system. The wireless phones are regular DECT phones, the Kirk IP 600 is doing the voip part by registering/gatewaying the phones to a callmanager server. The phones do not work and I think the problem is that they do not register at asterisk but I'm not sure because I don't understand a thing of the sccp/skinny

Hi list! Does anyone have experiences with the updated model of the Kirk IP600? The V3 model is supposed to support SIP instead of only SCCP or H323 which would make the use with Asterisk a lot easier. I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is still giving me severe headaches : - the standard Skinny driver in * doesn't work, only the version of Sergio

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

Is there a place where I can find documentation on the Skinny protocol in Asterisk? I did a search, but I cannot seem to find much... I'm looking for information on how to get a Cisco 7960 with the CM image on it to work with asterisk.. Is it support for Skinny in asterisk even mature enough to be able to do this? Thanks, - Andre

My thanks for previous help on fixing IXJ issues in 1.2.40; I now have problems with a just-built 1.4.35 on the same host: [Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) O/S: Linux 2.4.27; IXJ driver for Linux Rev. 3.5, gcc 3.01 I applied the patch for

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --