Re: [Asterisk-Users] OT: Polycom IP501 and Speed Dials

Mojo with Horan Company, LLC wrote: if you reboot your phones from the asterisk server ie via cron or so, that reboot script could potentially delete the phone-specific directory xml before the sip message is sent Sadly, that doesn't work -- the Polycoms store their directories locally as

RE: [Asterisk-Users] Asterisk Referral - Cleanup on Aisle 7

My experience is that many asterisk vendors are one or two man shows that are often between real gigs and disappear like fruit files. Another reason for due dilligence on the customer's part and a reason why vendors should stop selling at low profit margins. -Original

[Asterisk-Users] How to use Sendtxt?

I tried the example I found: exten = 123, 1, Answer exten = 123, 2, SendText(hello world) exten = 123, 3, HangUp However there was nothing on the display! Any hints? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] kernel recompilation on a asterisk server

i've tried make clean make make install in zaptel...but i still get errors... particularry i get: wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table

[Asterisk-Users] Dial cmd has delay for the last dialed number on FXO

Hi list The * server one TDM04B card and my dialplan: exten = 080.,1,Dial(Zap/g1/${EXTEN}) All four FXO ports has group=1 in zapata.conf After dialing 0800012345 from a FXS extension, with one DTMF detector tapped on the line, I found * dialed 0,8,0,0,0,1,2,3,4 in even interval, delay

RE: [Asterisk-Users] 1.2.6 doesn't use mpg123?

I use mpg123 for streaming but I can't get it to compile under SuSe10 and x86_64 CPU. Does anyone have any recommendations for other programs that allow streaming and will compile on this arch? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf

RE: [Asterisk-Users] kernel recompilation on a asterisk server

Is it a 2.6 kernel? Did you included CRC_CCITT and RTC support when you made the make menuconfig? -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de nik600 Enviada: sábado, 1 de Abril de 2006 10:28 Para: [EMAIL PROTECTED]; Asterisk Users Mailing List

Re: [Asterisk-Users] kernel recompilation on a asterisk server

ok now it works! thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How to check if a phone / line is used?

Jerry Jones wrote: Show channels? Yes, on Linux bye Ronald Wiplinger On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote: In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single

[Asterisk-Users] INX (Internationalnumber.com)

Unable to make outgoing calls from my asteriskusing INX (international.com) but incoming works fine. FYI! my asterisk is working fine for Vbuzzer.com for incoming and outgoing calls. Please reply with extension.conf and sip.conf section related to INX, if anyone out there using it successfully.

[Asterisk-Users] Newbie question - sip.conf incoming contexts

Hello! I've been struggling with the documentation for months on this simple subject... I still have not been able to get this concept down... I have 3 sip accounts (PSTN DID's) that come into my asterisk box and give me phone service from my itsp via SIP. I for the life of me have not been

Re: [Asterisk-Users] Echo cancellation problem

Avi Miller ha scritto: Giuseppe wrote: Can anybody tell me if there is some error or something missing in this configuration please? I have the same card in a few of my servers and the echo canceller works just fine. I'm not 100% sure, but something does jump out at me: Mar 31 16:40:21

Re: [Asterisk-Users] Panasonic KXTD 1232 6

charles napisa�(a): I want to replace a Telebutler software It's Telebutler software some simple IVR/CC solution? auto attendent system that used a 4 port Dialogic board connected to a Panasonic KXTD 1232 6 line system. We have spare computers here. How do I connect asterisk to this

[Asterisk-Users] Asterisk box with unreliable ping/latency

Hello! Over the last couple of days Ive been trouble shooting a strange problem with Asterisk. First of all, I should say that Ive been running Asterisk on a Fedora Core 3 box since last May, but decided to do a reinstallation of everything due to some problems weve had with echos

Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

You don't have to use it in newer versions. Get your mp3, ant convert to slin format with sox. Ex: sox -V file.mp3 [-c1] file.slin -V: just to show you what's going on -c1: convert to 1 channel, if your mp3 is stereo Then edit your musiconhold.conf like this: [native] mode=files

Re: [Asterisk-Users] Building Asterisk embedded device

Jim Houser wrote: http://gumstix.com/waysmalls.html Thanks for your link. how to build asterisk into this hardware? Thanks Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sam Sent: Friday, March 31, 2006 8:01 AM To:

Re: [Asterisk-Users] Re: How is Teliax ?

I would not ride on a tracert too much. We use Teliax also and our ISP that we have at the data center switched there backbones around the same time Teliax where doing there upgrades. For those that have not analyzed how tracert actually works, you can't depend on its output to give you

Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Ok this is great... but I just noticed this morning while doing some tests that asterisk seems to start a new stream for every caller With mpg123 it would just start one and all calls would hear the same stream.Unless something was seriously lagging, my test calls this morning all were in

RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

I want to stream shoutcast etc. but mpg123 won't compile. I use native moh with files but it won't work with streams. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Melcon Moraes Sent: 01 April 2006 14:33 To: Asterisk Users Mailing List -

Re: [Asterisk-Users] Confused on Agents and Queues

However, anyone have a good way to log the agent out without having them enter their agent ID and then have to hit # for the new extension? There are a couple of ways listed here in the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin

[Asterisk-Users] Refer Accountcode Bug

I would like to know if there are any turn around for accountcode missing in a refer request, I see the bug still marked as avaible and I would like to know if in mean time if there are any solution avaible for it. Best Regards, Elton ___

RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Has anyone else had a problem with asterisk creating multiple threads? I'm still testing but I've move from native to mpg123 for the machine with the problem and the problem hasn't come back. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt

Re: [Asterisk-Users] Asterisk box with unreliable ping/latency

First of all, I should say that I’ve been running Asterisk on a Fedora Core 3 box since last May, but decided to do a reinstallation of everything due to some problems we’ve had with echos during conversations (100% SIP based, so no ZAP echos). We are talking about a low-volume installation

Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts

I've been struggling with the documentation for months on this simple subject... I still have not been able to get this concept down... I have 3 sip accounts (PSTN DID's) that come into my asterisk box and give me phone service from my itsp via SIP. I for the life of me have not been able to

Re: [Asterisk-Users] Dial cmd has delay for the last dialed number on FXO

The * server one TDM04B card and my dialplan: exten = 080.,1,Dial(Zap/g1/${EXTEN}) All four FXO ports has group=1 in zapata.conf After dialing 0800012345 from a FXS extension, with one DTMF detector tapped on the line, I found * dialed 0,8,0,0,0,1,2,3,4 in even interval, delay longer, and

Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

How did you switch from native to mpg123 on 1.2.x? That's what I can't figure out. On 4/1/06, Lee Archer [EMAIL PROTECTED] wrote: Has anyone else had a problem with asterisk creating multiple threads? I'm still testing but I've move from native to mpg123 for the machine with the problem and

RE: Re: [Asterisk-Users] Routing SIP calls via URI

Dear Group, I was able to fix this problem; The solution was to use a prefix to dial out. The next challenge was to send the SIP Domain over IAX2!. I found that if I included @SIPDOMAIN it would break the IAX2 communications. exten = _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/[EMAIL

[Asterisk-Users] Free Software/Open Source Telephony-Summit 2006

Free Software/Open Source Telephony-Summit 2006 Tuesday, May 2nd 2006 Wiesbaden, Germany For the third time the German Unix User Group (GUUG - www.guug.de) organizes the Free Software/Open Source Telephony-Summit, an international workshop

[Asterisk-Users] AGI hangup problem

I have problem with Asterisk. [sendCommand]=EXEC DIAL IAX2/somehost/somenumber|10 [readReply]=200 result=-1 [sendCommand]=GET VARIABLE ANSWEREDTIME 1086422 [Thread-7] ERROR - establishConnection: exec encountered exception net.sf.asterisk.fastagi.AGIHangupException: Channel was hung up. This means

[Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed

Hello Everyone. I usually find my own solutions for problems but this time, after several months, Ive given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call,

[Asterisk-Users] TO have ringing tone instead MOH

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed

Try adding 'r' to the dial options. According to show application dial: r- Indicate ringing to the calling party. Pass no audio to the calling party until the called channel has answered. exten = 3058472194,1,Dial(SIP/1035SIP/[EMAIL PROTECTED],50, r) Julian. On 4/1/06,

Re: [Asterisk-Users] AGI hangup problem

Branko Samardzic wrote: I have problem with Asterisk. [sendCommand]=EXEC DIAL IAX2/somehost/somenumber|10 [readReply]=200 result=-1 [sendCommand]=GET VARIABLE ANSWEREDTIME 1086422 [Thread-7] ERROR - establishConnection: exec encountered exception net.sf.asterisk.fastagi.AGIHangupException:

[Asterisk-Users] Incorrect CDR results

When I look at my CDR data for calls to NuFone, the billsec for each call is 14 seconds or less. When I look at my NuFone account, the billsec has normal call lengths. So it seems that the billing on the Asterisk system terminates after about 14 seconds. The calls come in on an IAX

[Asterisk-Users] *.conf generator

Is there a good free *.conf generator out there. Manual configuration is just too tedious. I run Astlinux so a lot of the GUI's such as AMP (FreePBX)are not an option either. I used to use IPManager which did a great job but that project has been discontinued :(.

Re: [Asterisk-Users] Incorrect CDR results

On Saturday 01 April 2006 14:09, Michael Welter wrote: When I look at my CDR data for calls to NuFone, the billsec for each call is 14 seconds or less. When I look at my NuFone account, the billsec has normal call lengths. Are you transfering off of your asterisk box? IAX2 by default will

Re: [Asterisk-Users] Iaxmodem speed limit?

Carlos Chavez wrote: I just installed Hylafax with Iaxmodem and I am not getting good results when receiving faxes. I can see that the modem is reporting the following: Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 2400 bit/s Mar 31 16:19:08 pbxoficina FaxGetty[5377]:

RE: [Asterisk-Users] AGI hangup problem

Well that's fine but then I don't have any channel variable info I might have from channel otherwise. I am catchning exception now but I am wondering if there is some more appropriate way for Asterisk to handle hangup (to stick around for some more time just in case someone wants to pull any

Re: [Asterisk-Users] TO have ringing tone instead MOH

Use r instead of m in your transfer contexts.. and umm if you really want 'ringing' when someone is on hold... just put a sound file with 'ringing' in it in the moh directory. Otherwise, you can disable MOH and you will just have silence when someone is on hold. On 4/1/06, Alberto Sagredo

Re: [Asterisk-Users] *.conf generator

I don't understand your question. You don't want to generate the config files by hand, but yet you can't use FreePBX? Why can FreePBX not generate the conf files and then you go on to use astlinux? FreePBX should run on any linux distribution. On 4/1/06, mustardman29 [EMAIL PROTECTED]

Re: [Asterisk-Users] *.conf generator

Other thing I'm thinking... why are you running astlinux? Asterisk really isn't that hard to install.. make; make install; make config On 4/1/06, Matt [EMAIL PROTECTED] wrote: I don't understand your question. You don't want to generate the config files by hand, but yet you can't use

RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Check the musiconhold.conf.sample in the asterisk/configs directory. That will tell you what you need to know. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: 01 April 2006 16:41 To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [Asterisk-Users] TO have ringing tone instead MOH

Can I ask you why? []'s MM -Original Message- From: Alberto Sagredo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Sat, 01 Apr 2006 20:54:33 +0200 Delivered: Sat, 01 Apr 2006 15:56:25

[Asterisk-Users] no audio

Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with phones outside my network. I call the extensions without a problem, it rings but when they answer I can't hear them and they can hear me. I set up in the SIP.CONF nat=yes I'm I missing any other setting or do I need a special

[Asterisk-Users] chan-capi: Sending digits on a bri (isdn) d-channel

Dear asterisk users! I want to control a hardware pbx with asterisk. What I need to do this is being able to press hold which can be done with capicommand(hold) and then send digits on a bri card which connects to my asterisk computer. So far I use Dial(CAPI/ISDN1/27:digits/bo,15) to do this. Are

[Asterisk-Users] Re: no audio

Wierd timing - I'm struggling with exactly the same issue. My problem was with ZAP - ZAP. The phones ring, but no audio. Turns out there's a bug with the version I'm running. It has to do w/ the system date. When I changed my system date to 1-Jan-06, everything worked!! Here's what I found

[Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)

Just updated two fc3 systems running svn trunk. One updated, installed properly, and is working fine. The second box failed during the 'make install' process with: /usr/lib/libnetsnmp.a(parse.o)(.text+0x275a): In function `unload_module': : multiple definition of `unload_module'

RE: [Asterisk-Users] OT: Polycom IP501 and Speed Dials

http://www.voip-info.org/wiki/view/Polycom+reboot+hardphone+script -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Avi Miller Sent: Saturday, April 01, 2006 4:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT:

Re: [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)

Rich Adamson wrote: Is this worthy of opening a bug assuming the above comment is still valid? Would the individual(s) maintaining res_snmp want to log into either of these internet accessible boxes to identify the root cause? The module loader in trunk is undergoing changes that will

[Asterisk-Users] channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!

I never so this error.I am using H323 with Asterisk 1.2.6 Any idea what can be the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] vmail access problem

Hi everybody..I have the follow problem with my vmail access: http://voicemail.cheaphone.com/cgi-bin/vmail.cgi?action=""> For example this is the address to access the voice mail of one customer. If that customer change the number for :

Re: [Asterisk-Users] G729 codec problems

Hi, all Sorry, I was out of action for some time. I am using Voipstnt to plae calls to USA/Canada and I bouhj 1 copy og G.729. This was mainly to get one of the local Australians VoIP providers working. Anyway, when I am trying to place calls to USA, it tries to use G.729 and I am getting

Re: [Asterisk-Users] G729 codec problems

RumaTech wrote: And it keeps running like that. Call usually come through OK. If i try to use show g729 command, it shows that all codecs are in use. Well, this is fine, I am using one, but I do not want to see those warnings. Once is quite enough. Those continuos warnings make it impossible

Re: [Asterisk-Users] G729 codec problems

I am not. I have one license and use i channel. It seems to detect the fact there are no more channels left and keeps warning me about it in case I want to use more. It is fine, but the warning is constant. All you see on Asterisk console is running warning message. Rudolf On 4/2/06, Kevin P.

Re: [Asterisk-Users] Re: no audio

yes, it was fixed immediately following this bug. Update your version of Asterisk and you will be fine. Also in AAH you should do the following in sip.conf externip=your ISP external IP address localnet=your internal network (ie) 192.168.1.0/255.255.255.0 remove nat=yes in sip.conf and add it

[Asterisk-Users] FreePBX on Debian

Hi I got Asterix running on Debian Etch a few days ago. Version 1.2.6. Today i tried to install Freepbx and I got really confused. Do I really need that Zaptel stuff? It always prompted errors so i am now using mISDN -without errors, is there a module for freePBX for mISDN? Then I tried

[Asterisk-Users] H323 on way voice

Hi,I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing.The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts?

Re: [Asterisk-Users] H323 on way voice

Good luck. Try to switch between channel drivers. Chan_oh323, chan_h323 and ooh323. and remember to install the *exact* lib versions recommended on the readmes May the force be with you... Isamar On Sat, 1 Apr 2006, Il Neofita wrote: Hi, I installed H323, however when I make a call

Re: [Asterisk-Users] Problem: ringtones stop unexpectedly

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone.

Re: [Asterisk-Users] Zap channels - help

On Fri, Mar 31, 2006 at 10:36:02PM -0300, Josué Conti wrote: I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the

Re: [Asterisk-Users] H323 on way voice

Il Neofita wrote: Hi, I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts?

Re: [Asterisk-Users] Panasonic KXTD 1232 6

FYI, the Panasonic 1232 system is discontinued. On 4/1/06, Krzysztof Drewicz [EMAIL PROTECTED] wrote: charles napisa�(a): I want to replace a Telebutler software It's Telebutler software some simple IVR/CC solution? auto attendent system that used a 4 port Dialogic board connected to a

Re: [Asterisk-Users] voicemail to email sending problems

What does your /var/spool/mqueue look like? The messages in there should give you a clue what's wrong. For some reasone I'm suspecting that your exchange server is behind the same NAT as the sendmail machine, sendmail is resloving yourdomain.com to the public MX record, and your NAT device (like

RE: [Asterisk-Users] *.conf generator

Because Aslinux is an embedded solution just like most PBX's. -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Saturday, April 01, 2006 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] *.conf generator Other thing

Re: [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)

Kevin P. Fleming wrote: Rich Adamson wrote: Is this worthy of opening a bug assuming the above comment is still valid? Would the individual(s) maintaining res_snmp want to log into either of these internet accessible boxes to identify the root cause? The module loader in trunk is

Re: [Asterisk-Users] Problem: ringtones stop unexpectedly

Have you tryed phoning a fixed line instead of a cell phone?is this giving the same result?I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result.Alyed Return-Path: [EMAIL PROTECTED] Sat Apr 01

Re: [Asterisk-Users] G729 codec problems

I used g729 couple of times in the past and got the warning messages ONLY when I was trying to use more channels than the total amount of licenses I'd got.If you are sure you are using only one device that needs the license, I would suggest to check out how it is communicating with Asterisk.

re: [Asterisk-Users] Re: no audio

That was a bug fixed in Asterisk version 1.2.3 recently version 1.2.6 was released, so don't worry you can try the latest one without timing fears :DAlyed Return-Path: [EMAIL PROTECTED] Sat Apr 01 15:42:39 2006Received: from digium-69-16-138-164.phx1.puregig.net

[Asterisk-Users] G729 Passthrough question

hi group, is there a way that SIP phones be allowed to use G.729 passthrough when calling each other and when calling PSTN through Zap that asterisk force the phones to use ulaw. thanks, ultor ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] G729 codec problems

Rudolf Ladyzhenskii wrote: I am not. I have one license and use i channel. It seems to detect the fact there are no more channels left and keeps warning me about it in case I want to use more. I reviewed the code for that module after reading your original message, and confirmed that it will

Re: [Asterisk-Users] H323 on way voice

On Sat, 1 Apr 2006 20:09:35 -0500, Il Neofita [EMAIL PROTECTED] wrote: Hi, I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect

Re: [Asterisk-Users] Re: How is Teliax ?

On Sat, 1 Apr 2006, Rich Adamson wrote: end-to-end path. Each step through the tracert process does nothing more then issue an icmp echo request, measuring the response time and displaying it. maybe on windows it does icmp echo but no unix does this (at least not by default). i recommend you

Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts

What version of asterisk? (been lots of changes happening to the sip code over the last year) SVN-branch-1.2-r9156 I think what I am trying to do is pretty basic and should not have changed much in the past year. I got started in July of 2005 and I upgrade about once per month. In all this

[Asterisk-Users] morcdr v0.1 released

CDR Stats Analyzer and Report generator It's a rework of famous Asterisk Stats written by Areski. The main goal for this project is to concentrate more on PDF reports (managers love them!). Later more functions will be added. Please test it and send suggestions how to improve it.