similar to: splitting sip.conf to two files

hi: In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i added polarity reversal property but for fxo number 4 i didnt add polarity reversal property but it still giving me on cosole that fxo number 4 is polarized (because the line on fxo number 4 is not polarized). what i want to do is to not let polarity reversal take effect on fxo number 4. that what i have in my

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi pals, Pardon me if this question sound basic please. This is the first installation where I have to use the analogue card and therefore a little lusty. I have googled a lot, but though there is a lot of information about the above file, none indicate where the file lives. I have a installed asterisk and zaptel software on a fresh installation of CentOS 5. This all from source and following

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

The setvar command below works fine in iax.conf and in sip.conf but not here in zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated, can't seem to find an answer. ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid="John French" <103> mailbox="101" callwaiting=yes threewaycalling=yes

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

From: Malcom Kemp Sent: Wednesday, April 25, 2007 11:19 AM To: 'asterisk-users@lists.digium.com' Subject: AsteriskNOW generation of zapata.conf file I am a new user of Asterisk, and an trying to use AsteriskNOW. The test system is dual processor, so I am using the beta 4 version. I am currently trying to manually configure, as the GUI does not seem to let me accomplish what I need.

I have had a TDM400 with 2 FXO and 2 FXS working for ages (>12 months). It has stopped working. All four green lights are still lit. I have rebuilt zaptel and asterisk and restarted but the problem persists. /sbin/ztcfg -vvvv Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03:

I am having trouble with chan_zap.so not loading. When I load it from modules.conf, Asterisk bails out without any error message. When I load it from the console, it just says "Unable to load module chan_zap.so" no matter what verbose level I am using. dmesg says: Zaptel Version: 1.4.4 Zaptel Echo Canceller: MG2 Freshmaker version: 73 Freshmaker passed register test Module 0:

> -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] > Sent: Thursday, March 16, 2006 8:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on > *HAandPolycomphone!! > > > > > > "Q: What are the plans for HA? > > That's BS. Last time I

Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 > >> - downloaded the file opvxa1200.c > >> - copied in zaptel-1.4.7.1/ > >> - edited makefile adding opvxa1200 in the modules and the voice > >> opvxa1200.o : zaptel.h wctdm.h > >> - edited zaptel.sysconfig adding MODULES="$MODULES

Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

Hi all, I bought a Sangoma A200 card from an online supplier and explained exactly what I wanted, 3 incoming phone lines to PBX and a life line (some where to connect a standard BT phone to the PBX incase the power goes, making the BT phone ring). I was told to order 1 x FXS module (2 FXS ports) 2 x FXO modules (4 FXO ports) However being a complete noob, I have connected the 3 lines to the