Re: [Asterisk-Users] How to configure LineJack

As far I remember. I had one Linejack, ISA bus... Line jack is unable to place calls, only to recibe calls. The channel is a phone channel, you need to use the telephony driver from the kernel for linejack, and configure phone.conf of asterisk and then use it as any other channel. Ganbaa w

Re: [Asterisk-Users] How to configure LineJack

Ganbaa wrote: Hi all, We are testing LineJack (Quicknet) phone card with asterisk. Does anybody know how to configure LineJack on the Asterisk? (Incoming and outgoing call). Would anybody have any advice on what I should do? Sell it on Ebay and use the proceeds to buy a more capable TDM c

[Asterisk-Users] How to configure LineJack

Hi all,   We are testing LineJack (Quicknet) phone card with asterisk. Does anybody know how to configure LineJack on the Asterisk? (Incoming and outgoing call). Would anybody have any advice on what I should do?   Thanks & regards,   Ganbaa ___ --B

RE: [Asterisk-Users] ericsson pabx and digium card TE110P

Did you verify with the pbx engineer on how many digits the pbx are sending? Usually this should be the setting in the pbx.   CCF -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of vador loupeSent: Sunday, October 30, 2005 10:23To: Asterisk-

Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

Thanks Tad. This might turn out the be the clue I was looking for. It appears AMP has a macro-dial which has a comment about dealing with CFWD, DND etc. It actually dials using a script: exten => s,4,AGI,dialparties.agi I'm still trying to figure out what it does exactly because the code is not

[Asterisk-Users] ad hoc conferencing-reg

Hi all How to configure adhoc conferencing in asterisk for sip phones.pls give me if any document for that. regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

[Asterisk-Users] Re: Cisco 7970

I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on yo

[Asterisk-Users] How to make write and read formats equal to native format?

I am playing around with different codecs between 2 * servers. However I don't seem to have any impact on bandwith. I always get something like this: Name: IAX2/ds02-1 Type: IAX2 UniqueID: 1131421484.2 Caller ID: s Caller ID Name: (N/A) DNID Digits: (N/A)

RE: [Asterisk-Users] libmfcr2 - spandsp.h: present but cannot becompiled

Steve, have you tested r2mfc under 1.2beta2 with latest spandsp? I compiled the latest spandsp with 1.2beta2 and works great but wanted to know if you have tested r2mfc under that. Thx! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Steve Underwoo

RE: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold(1.2.0-b2)

We had problems with music on hold and finally decided to move to option 2 on the faking it document. We have not had any trouble since. Good luck. http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] O

[Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)

I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages f

Re: [Asterisk-Users]how to send fax using Spandsp

Compile CVS HEAD and it's all built in. Andy Kuo wrote: Hi, I've been trying to get fax going for the last few days. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but when I tried sending the received fax file to a fax machine, I either get "line error" or just a bl

Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

Sounds to me like that you want to log the phones into a queue, then simply logout the phones that you don't want to receive calls. If you were tricky, you could write a macro to log them in/out as they divert/undivert to/from voicemail. Eg. Dial an extension number to divert to VM (and log th

[Asterisk-Users] FW: Error building res_perl

I am getting the following error when I try to make res_perl.  With 1.2 beta2, centos 4.2 x86_64.      Anyone have any idea what the problem is?   Am I missing something?   Thanks.. Doug       bash# make Phew, You have the right perl. /usr/local/bin/perl -MExtUtils::Embed -e xsinit g

Re: [Asterisk-Users] Can't make calls from Asterisk IAX to other IAX

can you paste you iax.conf On 11/8/05, chawki hammoud <[EMAIL PROTECTED]> wrote: > Hi: > > I have been having this problem for sometime that I am > not able to solve and I hope someone can help. > > I can make VOIP calls between my Asterisk box and my > VOIP provider using sip channel without a pr

Re: [Asterisk-Users] MP3 or OGG

Wasn't aware of it, but if quality is good, it makes sense since all I'm archiving is speech. Will evaluate further. Thanks, Waldo On Nov 7, 2005, at 7:14 PM, Mark Edwards wrote: I would recommend vorbis speex for this. You can get windows drivers to read speex files directly. Vorbis are t

[Asterisk-Users] several beginner questions

Actually 1 beginner w/ multiple questions... I'm getting ready to make my first jump into VoIP and the Asterisk PBX - Katrina has forced my hand much earlier than expected. My phone and ISP (Eatel) is leaving New Orleans so I've got just a couple weeks to get this done. I read some good revi

[Asterisk-Users]how to send fax using Spandsp

Hi,   I've been trying to get fax going for the last few days. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but when I tried sending the received fax file to a fax machine, I either get "line error" or just a blank page.   Is anyone using Spandsp to send fax to fax machi

RE: [Asterisk-Users] MP3 or OGG

I would recommend vorbis speex for this. You can get windows drivers to read speex files directly. Vorbis are the same bunch that develops ogg. Ogg and mp3 are more suited to music rather than speech. Speex is a much better fit for speech archiving. Mark -Original Message- From: BJ We

Re: [Asterisk-Users] libmfcr2 - spandsp.h: present but cannot be compiled

Jesus Mogollon wrote: Hi all When I try compiling libmfcr2 I get: spandsp.h: present but cannot be compiled Any ideas? Either: a) Ignore that message, and carry on. It works anyway. b) Use a newer version of spandsp (pre21b eight now) which should no longer be generating that message.

Re: [Asterisk-Users] Missing audio from Zaptel channels - SOLVED!?

For those who are interested, the problem appears to NOT exist in 1.2Beta2. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 I

[Asterisk-Users] Very basic switching application -- bounty?

Eric Lyons wrote: The basic function is to take an incoming DNIS/exten on one port, look it up in the db, then dial out to another number on another port. This is just basic dialplan work... why you would need a custom application?Hi, Kevin. Yes, it *is* the most basic of dialplan configurat

[Asterisk-Users] SIP domain support for authentication and virtual hosting

Hello, Where may i find documentation about SIP domain support and dnsmgr.conf , Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette v

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

First off, if they are on the same network without any nat, then it is not needed at all. Since this works well with pre 1.2b2 I would say you should open up a but on the bug tracker at: bugs.digium.com. I did not yet update to 1.2bx so I have no way of confirming this. Thank You. On 11/7/05, Wald

Re: [Asterisk-Users] zaphfc not generally compatible with kernels >= 2.6.13

On Mon, 7 Nov 2005 23:06:24 +0100 Gerald Dachs <[EMAIL PROTECTED]> wrote: > Hi, > > I am very new to asterisk so forgive me if I tell something stupid. It has happend, my post was stupid > I am investigating currently a problem with zaphfc. I get only very few > interrupts, > they don't get lo

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

Ok. The keepalives work for other phones, but not the UIP200. I have a bunch of X-Lites, X-Pros, SPA-841s, and UIP200s. It works fine in all but the UIP200 (only in 1.2b2). As far as your questions: 1) They are on the same network and same netmask 2) They are not natted. Let me know what

[Asterisk-Users] sill looking for a provider

OOPPS! Looks like someone just broke voipjet's tos gw at adcomcorp.com gw at adcomcorp.com wrote on Sat Nov 5 11:36:46 CST 2005 I tend to agree with you, my experience with Teliax has been decent, and get

[Asterisk-Users] zaphfc not generally compatible with kernels >= 2.6.13

Hi, I am very new to asterisk so forgive me if I tell something stupid. I am investigating currently a problem with zaphfc. I get only very few interrupts, they don't get lost, the interrupt count increases only very slowly. I really don't know where to look for the problem, so I looked here an

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

If this is the case. then we now know what the problem is. The keepalives from asterisk to the phones were not working in 1.2b2. The question now is why? Please work with this so that we can troubleshoot this to see if it's a bug with 1.2b2 or not. 1. Is the UIP200 on the same subnet as asterisk? 2

Re: [Asterisk-Users] Problems with DTMF on Polycomm Phones

At 15:36 11/7/2005, Krishna Sumanth Chava wrote: hi, Would like to have help in fixing the DTMF problem i am facing on Polycomm Soundpoint IP Phones I am having the following network setup.. I have my Asterisk PBX server connected to the Cisco 3620 Router with an ethernet cable which intur

[Asterisk-Users] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.

Ich werde ab 07.11.2005 nicht im Büro sein. Ich kehre zurück am 04.12.2005. Ich bin vom 8.11.05 bis 4.12.05 nicht erreichbar und werde die Emails sobald als möglich bearbeiten. Dringende Anfragen bitte an Andreas Widrig/CZWIAN/CH/Ascom oder Ralf Knobel/CZKNOR/AScom senden.

[Asterisk-Users] libmfcr2 - spandsp.h: present but cannot be compiled

Hi all When I try compiling libmfcr2 I get: spandsp.h: present but cannot be compiled Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/li

[Asterisk-Users] new sip domain support and REGISTER requests

hello, since beta2 there is the new sip domain support - but somehow this feature is still a bit unclear for me. routing to different contexts based on the domain in the extension.conf seems to be rather trivial, but is it possible to do the following things: a) allow two users with the sam

[Asterisk-Users] Problems with DTMF on Polycomm Phones

hi,   Would like to have help in fixing the DTMF problem i am facing on Polycomm Soundpoint IP Phones   I am having the following network setup..   I have my Asterisk PBX server connected to the Cisco 3620 Router with an ethernet cable  which inturn is connected with a T1 circuit to my SIP Provider

Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

A workaround (bvut slighty messy) would be to set up two lines on each phone. Standard calls (which can be forwarded) can go the first (main) line/extension. Group calls go to line 2. PaulH - Original Message - From: "John Lange" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non

[Asterisk-Users] [OTAnn] Feedback

I was interested in getting feedback from current mail group users.We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband.Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast vide

Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance

Considering that I am a user 'down under' that's even funnier. PaulH - Original Message - From: "Kevin P. Fleming" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Tuesday, November 08, 2005 2:12 AM Subject: Re: [Asterisk-Users]

[Asterisk-Users] Can't make calls from Asterisk IAX to other IAX

Hi: I have been having this problem for sometime that I am not able to solve and I hope someone can help. I can make VOIP calls between my Asterisk box and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hang

Re: [Asterisk-Users] asterisks talking to asterisks

Indeed. They prefer to talk IAX protocol and can do everything you would expect them to do. I'm a consultant in NJ. Contact me off list and I'll discuss with you how to do it. Mark 973 828 1625 Jason Brashear wrote: I have a request. I have a server in Texas And one in NJ. Is it possible

Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

I use [EMAIL PROTECTED], and if I have one of my Cisco phones forwarded to my cell phone, when the phones ring in a ring group, it never forwards. You may want to look at the latest configs that comes with [EMAIL PROTECTED] and see if theres some special dialplans thats doing what your looking for.

Re: [Asterisk-Users] asterisks talking to asterisks

Technically, yes. On 11/7/05, Rob Lith <[EMAIL PROTECTED]> wrote: > Wouldn't IAX be more efficient as you can trunk simultaneous calls and save > bandwidth? > > Rob > > > On 11/7/05, Andy Kuo < [EMAIL PROTECTED]> wrote: > > > > I do that through SIP. > > > > Assuming your TX extensions are 10XX,

RE: [Asterisk-Users] Re: Help with dialplan to allow breakout to DISA

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Frank Tarczynski > Sent: Monday, November 07, 2005 2:26 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Re: Help with dialplan to allow > breakout to DISA > > Yes, I know. > > BUT

Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

John Lange wrote: Reading the source code I see there are two parameters for channels, allowredir_in & allowredir_out. These offer me some hope that Asterisk has the ability but I couldn't figure out what these do or how to make use of them (I'm not a C programmer so maybe its just a red herring

Re: [Asterisk-Users] asterisks talking to asterisks

There is a Step-by-Step HOW-TO on the Brilliant voip-info.org site about connecting * servers.. The HOW-TO is Titled "Asterisk Connect 2 servers" and can be found @ http://www.voip-info.org/tiki-index.php?page=Asterisk+Connect+2+servers Hope This Helps... Best Regards Gavin Spurgeon Assistant

Re: [Asterisk-Users] asterisks talking to asterisks

Wouldn't IAX be more efficient as you can trunk simultaneous calls and save bandwidth?RobOn 11/7/05, Andy Kuo < [EMAIL PROTECTED]> wrote:I do that through SIP.   Assuming your TX extensions are 10XX, and NJ extensons are 20XX On your NJ box... sip.conf [gwtx] type=friend secret=x host=10.11.12.

Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 16, Issue 44

I'm 15,000kms away and 9 hours time zone away yet I get superb same day response from the folk at Digium. Bending over backwards to help in anyway.I'm in South Africa.Met the Digium team at Astricon in Anaheim and I can say that while our business is probably a rounding error compared to what is do

RE: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

Some ATAs do not like the qualify, I have some MTA102 and that's the case with those, if I enable qualify, the ata doesn't work with asterisk, if I disable qualify, the ATA works without problems. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of C F |S

Re: [Asterisk-Users] DNS Server Failure wreaks havoc

I installed a caching dns server on the * box itself 'cause when the external dns stopped resolving guess where my emailed voicemails went? Ya, I don't know either. :P They weren't in the mailq but showed up just a little while later when the names began resolving again :) Brian Capouch wrot

Re: [Asterisk-Users] MP3 or OGG

Thanks - Waldo On Nov 7, 2005, at 1:52 PM, BJ Weschke wrote: You're probably not going to be violating any patent protections by using OGG instead of MP3. As far as compression goes, I've found the difference between the two of them to be negligible. I've always used OGG when possible to stay

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

I do have qualify=yes pretty much in all my sip entries. I just changed all the entries where I have a UIP200 to qualify=no and now they all work. The funny thing is that it worked with qualify=yes in 1.0.9 and 1.2b1 Thanks, Waldo On Nov 7, 2005, at 1:29 PM, C F wrote: I guess that somew

RE: [Asterisk-Users] asterisks talking to asterisks

Standard IAX link found on wiki > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason Brashear > Sent: Sunday, November 06, 2005 11:13 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] ast

Re: [Asterisk-Users] asterisks talking to asterisks

I do that through SIP.   Assuming your TX extensions are 10XX, and NJ extensons are 20XX On your NJ box... sip.conf [gwtx] type=friend secret=x host=10.11.12.13(your TX IP)   extensions.conf [toTX] exten => _10XX,1,Dial(SIP/[EMAIL PROTECTED])   On your TX box sip.conf [gwnj] type=friend secret=

[Asterisk-Users] Re: Help with dialplan to allow breakout to DISA

Yes, I know. BUT, I want the line to work as normal for incoming calls AND allow the user to breakout. So how do I merge: [incoming] exten => 1000,1,Ringing exten => 1000,2,Answer exten => 1000,n,Dial(IAX,iaxy/20) exten => 1000,n,Voicemail() exten => 1000,n,Hangup AND exten => *, 1, Authent

[Asterisk-Users] asterisk 1.2b2 compiling problem

I just checked out asterisk 1.2b2 for zaptel, libpri, asterisk and asterisk-sounds. Zaptel and libpri compile fine with a 'make clean' and 'make install'. However even after a make clean, the asterisk 'make install' does not finish on my redhat 7.3 system. CVS-D2005.09.12.05.00.00-09/14/05-02:0

RE: [Asterisk-Users] CentOS vs. Vanilla Kernel

Ryan Amos <[EMAIL PROTECTED]> wrote: > The default CentOS kernel has worked fine for me. > Just an FYI; CentOS uses the RedHat EL kernel source to > build... It's pretty heavily patched so if you want to use > the latest stable, download the SRPMs from RedHat/CentOS > and patch in the kernel.org pa

[Asterisk-Users] Stopping Asterisk from forwarding calls?

The first time I asked this to the list I didn't do a great job of it so I'm posting again with more details. Problem: when ringing multiple extensions, if one user has their phone forwarded directly to voicemail, it stops the whole group from ringing because the voicemail picks up immediately. A

RE: [Asterisk-Users] Change Asterisk User

Use group permissions. Add the apache user to the asterisk group and give the group the appropriate read and/or write access. IMO this is the easiest way to get around the apache permissions thing, and probably the Right Way (tm) -Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[E

Re: [Asterisk-Users] Getting ztdummy to load on startup for X100P

I'm not sure where in your startup process asterisk gets loaded. I load my asterisk from my rc.local file, so I can of course control when ztdummy would be loaded in relation to asterisk. Tzafrir Cohen wrote: On Fri, Nov 04, 2005 at 11:43:37AM -0900, Mojo with Horan & Company, LLC wrote: Tr

Re: [Asterisk-Users] MP3 or OGG

You're probably not going to be violating any patent protections by using OGG instead of MP3. As far as compression goes, I've found the difference between the two of them to be negligible. I've always used OGG when possible to stay "IP safe". On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrot

RE: [Asterisk-Users] CentOS vs. Vanilla Kernel

The default CentOS kernel has worked fine for me. Just an FYI; CentOS uses the RedHat EL kernel source to build... It's pretty heavily patched so if you want to use the latest stable, download the SRPMs from RedHat/CentOS and patch in the kernel.org patches. But yeah, stick with the CentOS kernel

Re: [Asterisk-Users] asterisk-1.2-bêta2 | presen ce/subscription support in the SIP channel driver

There could be 1 of 100 reasons that's causing this not to work. Let's start out by you posting your relevant sections of sip.conf and extensions.conf and then do a "sip show subscriptions" from the CLI and give us the results of that as well. On 11/7/05, harry gaillac <[EMAIL PROTECTED]> wrote

RE: [Asterisk-Users] Change Asterisk User

Thanks for your answer, I am working on Debian Sarge but I have compiled Astersik 1.0.9 myself without .deb Packages. I need to access to voicemail and sound files from my web-interface (php and cgi/perl) but I can't change Apache user because of others applications. Asterisk creates files under As

Re: [Asterisk-Users] Re: Double DTMF with tdm card

Just wanted to let the group know this problem is fixed (for me). Mark log-on to my system and found a "bug" in chan_zap.c on Saturday night and made the correction - I believe the change is available for download by now at zaptel 1.0.9.2, or CVS Head. He stated that recent changes "unmask" th

Re: [Asterisk-Users] CentOS vs. Vanilla Kernel

On Mon, 2005-11-07 at 18:17 +, Julian Lyndon-Smith wrote: > > What is the opinion of this fine list - should I use the default CentOS > kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable > (2.6.14) > > Anyone got any clues / hints / tips on what should go into the kerne

Re: [Asterisk-Users] CentOS vs. Vanilla Kernel

my $0.02 if you are going w/ RHEL use one of the kernel rpms provided. You can always add a module rpm to supplement it. Once you roll your own there might be better distros for you, since you are going to break the rpm/up2date features that make RHEL a desirable product. On Mon, 7 Nov 2005, J

Re: [Asterisk-Users] asterisk as SIP gateway

Miloš Kocbek wrote: I want to enable access to some context in asterisk without authentication. In sip.conf: [username] type=friend host=x.x.x.x context=context_for_this_user -- Regards, Peter Petrov [EMAIL PROTECTED] ___ --Bandwidth and Colocati

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

I guess that somewhere in your settings you have a qualify on, or that 1.2 has it on by default. Do the following: cd /etc/asterisk grep ".*qualify.*" ./* and see the output, if the only line that has qualify is that qualify=no, then this looks like a bug to me. Please report back. On 11/7/05, Wal

[Asterisk-Users] Re: Help with dialplan to allow breakout to DISA

I have my dialplan setup the same, only with 0 instead of * as the extension. What would the reason be, after authenticating, that I get a dialtone, as expected, but no response to any DTMF tones I input? It is as if the DISA works, gives me tone, but is unresponsive? The destination context is exa

[Asterisk-Users] asterisk-1.2-bêta2 | pres ence/subscription support in the SIP channel driver

Hello, I configure Polycom ip300 for presence but when status change notify is no sent to subscriber !? Why ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouvea

[Asterisk-Users] Speex codec problems

I am trying to tweak my Asterisk servers to talk to each other using Speex codec. I downloaded and installed speex and speex devel libraries, recompiled asterisk (including make clean), did set up speex codec as only one allowed on both sides. Sounds enough. However, conversations are not Speex enc

Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance

On Monday 07 November 2005 12:57, George Gardiner wrote: > Most modern installations/buildings are wired with RJ45, as are the patch > panels.  RJ12 is a real pain - I had to chop up patch leads and put RJ12 > sockets on the end.  Very messy and a waste of time.   We just moved in to a new buildi

[Asterisk-Users] AGI environment dump callerid

Hi, Since * 1.2-beta1 (incl CVS HEAD) there is a change in the callerid's output to STDERR when an AGI environment dump is requested: Asterisk CVS HEAD built by root @ chick on a i686 running Linux on 2005-11-06 16:35:14 UTC AGI Environment Dump: -- accountcode = -- callerid = 1234689 -- caller

[Asterisk-Users] CentOS vs. Vanilla Kernel

HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or downloa

Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance

  Most modern installations/buildings are wired with RJ45, as are the patch panels.  RJ12 is a real pain - I had to chop up patch leads and put RJ12 sockets on the end.  Very messy and a waste of time.      On Sun, 6 Nov 2005 22:04:48 -0500, Andrew Kohlsmith wrote:> On Sunday 06 November 2005 21:46

Re: [Asterisk-Users] sill looking for a provider

On 11/06/05 02:31 Dustin Goodwin said the following: Of course it's hard for me to see the return route with traceroute. I assume the return path from their host takes on some bizarre route that adds a lot of latency. try a traceroute with lft. lft gives you the different AS/BGP routers you

Re: [Asterisk-Users] Help with dialplan to allow breakout to DISA

I do it this way: exten => *, 1, Authenticate(PASSWORD) exten => *, 2, DISA(no-password|DESTINATION_CONTEXT) exten => *, 3, Hangup It seems to work fine... -Rusty On 11/7/05, Frank Tarczynski <[EMAIL PROTECTED]> wrote: I'm trying to set-up a dialplan for incoming calls that allows a breakoutby

[Asterisk-Users] Help with dialplan to allow breakout to DISA

I'm trying to set-up a dialplan for incoming calls that allows a breakout by pressing something like "*". Users would then be able to get an inside dial tone for voicemail, outgoing calls, etc. I've been struggling with Waitexten(), Disa() in the dialplan but not having much luck. Are there any

Re: [Asterisk-Users] References?

Hrm. Perhaps I should have actually responded off-list... DOH! :D On Nov 7, 2005, at 9:11 AM, Chad Scott wrote: Matt, Sorry for the response off-list... Would you be willing to talk to the "powers that be" for about 30 minutes about your experiences with Asterisk? I don't know what que

Re: [Asterisk-Users] References?

Matt, Sorry for the response off-list... Would you be willing to talk to the "powers that be" for about 30 minutes about your experiences with Asterisk? I don't know what questions they're planning to ask, but they're likely to be centered around reliability and supportability as those ar

Re: [Asterisk-Users] Change Asterisk User

On Mon, Nov 07, 2005 at 05:27:00PM +0100, Amaury BOSSE wrote: > Hi all, > > I would like to start asterisk with a different user than "asterisk" in > order to use the same than my apache server. > Hmmm, you basically need to run apache's user to the Asterisk group. -- Tzafrir Cohen | [

Re: [Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic

On Mon, Nov 07, 2005 at 03:43:03PM +0100, gincantalupo wrote: > Hi, > > I had some problems to with a quadBRI with a 2.6 kernel debian distro. > Have you tried to insmod the zaptel.ko module instead of modprobing? > It worked for me, hope it will work for you too. > > Giorgio Incantalupo Could y

Re: [Asterisk-Users] Change Asterisk User

If you give us more info it is easier to help. For example, if you are using a standard debian sarge setup I could help you and be sure to give you the right advice. However you might want to think carefully about this type of change. There are other approaches such as setting ownership and pe

Re: [Asterisk-Users] compiling problems

Some problems happened with precompiled kernels. If you compile your own vanilla kernel, I'm sure that you haven't this issues. Remember, if you use 2.6 kernel, you can need udev and hotplug systems to better performance. I allways use Debian with vanilla kernel that I compile, and I haven't

Re: [Asterisk-Users] Change Asterisk User

there is a lot more to changing the user than just su'ing you need to change the permissions on a lot of files too. On Mon, 7 Nov 2005, Amaury BOSSE wrote: Hi all, I would like to start asterisk with a different user than "asterisk" in order to use the same than my apache server. I have t

[Asterisk-Users] Change Asterisk User

Hi all, I would like to start asterisk with a different user than “asterisk” in order to use the same than my apache server.   I have tried to change it in /etc/init.d/asterisk but when I change USER, asterisk doesn’t start.   Has someone already start asterisk under other user that “a

Re: [Asterisk-Users] asterisks talking to asterisks

Yes. Most certainly. Take a look at IAX (Inter Asterisk eXchange) protocol to enable this functionality for you with minimal impact on your firewall/NAT setups. On 11/6/05, Jason Brashear <[EMAIL PROTECTED]> wrote: > I have a request. I have a server in Texas > And one in NJ. > Is it possible for

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1. Very strange. Anyway, thanks. - Waldo On Nov 7, 2005, at 10:57 AM, C F wrote: The unreachable is the problem. Try adding a qualify=no to that sip entry. On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: Additionall

[Asterisk-Users] asterisks talking to asterisks

I have a request. I have a server in Texas And one in NJ. Is it possible for the system in Texas to log into the system in NJ so that Extensions can call each other? -J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

The unreachable is the problem. Try adding a qualify=no to that sip entry. On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > Additionally: > > *CLI> sip show peer 100074 > > * Name : 100074 > Secret : > MD5Secret: > Context : qa > Subscr.Cont. : > Langua

Re: [Asterisk-Users] Dropping last digit from phone number

Here is what I do: ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} that should give you for the following exten => 123456789,1,Noop(${EXTEN:0:$[${LEN(${EXTEN})} - 1]}) 12345678 Hope this helps. On 11/7/05, Bartosz Piec <[EMAIL PROTECTED]> wrote: > Erik napisał(a): > > exten => _XX*,1,NoOp(${EXTEN:0:-1}) > > e

Re: Re: [Asterisk-Users] call from asterisk to SIP cisco 5300

sorry, i didnt write i have voip peer so i have sloved thy problem, nubder like #00#7091222 *00*7091222 *777 doesnt work Cisco says dpMatchPeersMoreArg: Match Dest. pattern; called () and when i tries to dial *777*777 it says dpMatchPeersMoreArg: Match Dest. pattern; called (777) But

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

Additionally: *CLI> sip show peer 100074 * Name : 100074 Secret : MD5Secret: Context : qa Subscr.Cont. : Language : en AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PR

[Asterisk-Users] Help needed for Onhold calls

Good Day list, I have read wiki pages I have googled to death and am getting no closer to understanding the methodology of onhold music. Maybe I am trying to do something that is just not possible: Here is my desire. 1) Call comes in to the asterisk box via Zap

RE: [Asterisk-Users] Cisco 7970

Hi Greg, Would you mind a telephone call to help me with the final steps? - Dan Levine [EMAIL PROTECTED] 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mail

Re: [Asterisk-Users] FXS problems

Andrew Kohlsmith wrote: On Monday 07 November 2005 08:03, bails wrote: I seem to be having some problems with the FXS modules on i, for example when i dial 90044117XX Nov 7 13:01:01 VERBOSE[2516]: -- Starting simple switch on 'Zap/1-1' Nov 7 13:01:05 DEBUG[2516]: DTMF d

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 16, Issue 44

The 104d has been available for a few weeks. I've had one for 4 weeks working with Sangoma on the driver side. My echo issues are a thing of the past. I had a few issues configuring, but it turned out to be Asterisk configuration- not Sangoma configuration. You must download their latest drivers. T

Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance

[EMAIL PROTECTED] wrote: I was pretty unhappy to see that the new cards had RJ12 sockets - you can put RJ12 into RJ45, but not the other way round... But I do know that a lot of people would ask if RJ12 would fit, so it might have been to cut down on support calls. Definitely not :-) It was do

RE: [Asterisk-Users] Voicemail

The text sent on this notificationscan be found in voicemail.conf Hope this helps. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Andrew Nowrot |Sent: Monday, November 07, 2005 5:54 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users

Re: [Asterisk-Users] upgrade to 1.2 beta 2 issue

[EMAIL PROTECTED] wrote: Ever since I upgraded to beta2, the console is littered with these kind of messages: NOTICE[206]: chan_iax2.c:5654 update_registry: Restricting registration for peer 'kkai13' to 60 seconds (requested 0) Any way to suppress this? Of course! Fix your IAX2 client to

Re: [Asterisk-Users] One Touch Record in 1.2

BJ Weschke wrote: You're going to need to do more than just putting the recorded media file into the voicemail folder hierarchy if you want the apps to recognize them. You will need to accompany them with their respective .txt file so the voicemail system and various web interface tools recogniz

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