similar to: Dial Problem

I have the following in extentions.conf: exten => 2111,1,Dial(SIP/2111 at gw1.langley) exten => 2111,2,Voicemail(u2111) exten => 2111,3,Hangup exten => 2111,100,Voicemail(b2111) exten => 2111,101,Hangup I have the following in sip.conf: ; Cisco 1750 [gw1.langley] type=friend host=172.16.17.1 context=default canreinvite=no Like the ATA, lots of stuff doesn't work on the 1750

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing this and should I expect smooth operation? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.

Everyone, Just thought I would drop a line telling everyone here I have the software RxFAX/TxFAX up and running without any real problems. I did have to..... RH 9.0 1) Install an audio devel rpm 1) install libtiff from source, and copy over a bunch of include files to /usr/local/include 2) build/install spandsp 3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree. 4)

We're still having problems with DTMF detection on our X100P cards. Incoming callers that hold down the "1" button for too long are being connected to extension 11. One would think fat fingers were uncommon, but it happens to alot of people. I suspected this was related to our having to increase the txgain, but I tried turning it down with no effect. I also tried disabling the

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hello, Does anyone know what may be causing this? Asterisk was built from cvs tonight. Ztcfg also says us is an invalid tone zone. Anyone got some information on what this is, why is it happening, and possibly some solutions? [root@anistonetech zapata]# asterisk -f -d DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/asterisk.conf DEBUG[1024]: File config.c, Line 653

Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone -> Asterisk -> X100P -> PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and

I'm having problems with my dial plan. I have the following in my extentions.conf: [incoming] ; ; Incoming calls via the PSTN land here ; exten => s,1,Answer exten => s,2,DigitTimeout(5) exten => s,3,ResponseTimeout(10) exten => s,4,BackGround(dial-exten) exten => s,5,Wait(15) exten => s,6,Congestion exten => s,7,Wait(10) exten => s,8,Hangup ; ; Let incoming calls

I'm wondering if anyone knows of ISPs with service that has QoS features that would be good to use with VoIP stuff. Granted, the QoS would only be supported as long as you stayed within their network, but it might be better than nothing. --Eric

I'm looking at this document to convert RHEL4 systems to Yum and setup an updates repository. http://sial.org/howto/yum/ Which version of yum would you recommend for that? What would be the dependencies that need to be installed? Any guidelines for installing Yum on RHEL4 without breaking the RPM dependencies and stuff (and with minimal changes to the system)? The document recommends

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI20830061xxxx at 127.0.0.1>' failed for '127.0.0.1' - No matching

Hello, In order to overcome a known performance bug in perl-5.8.8-10 in centos 5 (see https://bugzilla.redhat.com/show_bug.cgi?id=196836) I downloaded the perl package from fedora 8 (http://mirror.internode.on.net/pub/fedora/linux/releases/8/Fedora/source/SRPMS/perl-5.8.8-30.fc8.src.rpm) and mod_perl

I have a data.frame which I believe is melted already and am having trouble casting it to 'wide' format. It looks something like > (x <- data.frame(ticker=c(rep("A",5),rep("B",6)), date=c(1:5, 1:6), value=c(NA,100*exp(rnorm(10,0,.1))))) > cast(x, date ~ ticker) # this does what I want with toy data But when I use my real data frame >

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing