similar to: Asterisk slow transferring calls

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hi, We have a Ericsson BP250 Phone system setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next

Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten => s,1,Set(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,Set(EMERGENCY=1,g) exten => s,n,Set(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})

I am having problems getting a Nicherons TDM400P wildcard clone to dial out. Everything appears to be configured correctly, but although I see call progress, it never seems to actually pick up the phone. (The following is a test of 911 emergency, where I substitute 811 [repair service] as the actual number dialed.) *CLI> -- Executing [911 at from-internal:1]

Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these extensions.conf lines: exten => s,1,SetVar(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,SetGlobalVar(EMERGENCY=1) exten => s,n,SetVar(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} =

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

Dear; In the extensions. conf, I have the following: exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}@Internal) So, I am writing the arguements of the Voicemail ( ) wrong? Regards Bilal > > Dear; > > > > Where I can find a new documentation for Asterisk > 1.8? > > > > Where is the wrong in that line? I see it is as 1.8 > version ! > > > >

Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>

Hi All; Already in the voicemail.conf file, I added the extension 500 and kindly find below my voicemail configuration: [Internal] 0 => 1234,Gama Operator,Operator at gama.com 500 => 1234,Operator,Operator at gama.com 501 => 1234,Employer Name,employer_email at gama.com 502 => 1234,Employer Name,employer_email at gama.com Asterisk version is 1.8 and currently I am getting this

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect). When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Everybody, I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8 build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on LAN (without NAT). With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can have a very fine secure conversation in both directions. When I want to do the same with my iPhone,

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

I don?t know if the last message was with content. So, I sent again. I have installed a Digium card TE210P and unicall for use MFC/R2. I think that it?s all right but I can?t make and receive calls. I?m using asterisk 2.1 with the patch made by Jos? P. Leit?o and the follow libs: libsupertone-0.0.2 libunicall-0.0.3 libmfcr2-0.0.3 zaptel 2.1 My number is 34318300. The Telco send me only 8300.

Hi All, I'm having some troubles with using call files. I'm trying to establish the following: - want to use call files to connect two (outside) extensions - want to use the outbound routes set in FreePBX - want to set the outgoing callerid for both calls - want to set a custom CDR field in MySQL ( field name 'azonosito' ) Asterisk is version 1.8.2.3 with freepbx 2.8.1. What

Hi! Trying to figure out how to rewrite calling number of an incoming call... A cell phone (0733025975) dials a X-Lite (977). X-Lite "shows" 733025975 at the display, but I want it to be 0317998975. I thought i could do something like: exten => 977/733025975,1,Set(CALLERID(number)=0317998975) exten => 977,n,Dial(SIP/0317998977) [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272