[Asterisk-Users] asterisk CVS-HEAD is nutzo!

I probably dont know what I am doingthats all But from a clean install I installed zaptel, libpri, asterisk CVS-HEAD and asterisk-addons. All went ok...but asterisk is acting crazy. It wont let me register any SIP channels, otherwise it hangs when I start it at the SIP part. When I start as

Re: [Asterisk-Users] how to debug frame slips?

On Tue, 2004-12-28 at 06:10, Michael Welter wrote: > Try 'lspci -v' and look at the latency timer for your Digium card(s). > You can set it higher with 'setpci -v -s xx:yy.0 LATENCY=TIMER=ff' (xx > is the bus number and yy is the slot). > Shouldn't you decrease the latency? ie, to something low

Re: [Asterisk-Users] PRI & CPU Usage

On Wed, 2004-12-29 at 11:22, Derek Conniffe wrote: > Hi everyone, > > I'm a bit unclear about how PRI voice channels use CPU usage in an asterisk > box - it is like a codec conversion or does a PRI channel have a generic > codec itself? AFAIK, a PRI will use either ALAW or ULAW, depending on yo

Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

Nevermind, it looks like "Asterisk cmd Read" is my lucky command :) Thanks! Start Your Own Internet Service! http://www.YourOwnISP.com - Original Message - From: "Me" <[EMAIL PROTECTED]> To: "C F" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wedn

Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

I was trying this logic before, I got as far as going into the Macro, playing a message and then.. Well... I got lost, I am not sure how to go about require them to press a button. Normally I can make someone press an extension but from what I read about Macros in * you have to stay within the "s"

[Asterisk-Users] external Radius Server integration with asterisk

Hi every body   I am bit new with asterisk . i have configured it to get it working for SIP calls through xlite dialer from xten. for this i used sip.cof and the user were defined there ,i want to put the user in db and want the raduis to authenticate the user through it . can any body let me

Re: [Asterisk-Users] Invalid Extension

I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? [inbound-sip] exten => 533990,1,Answer exten => s,2,ResponseTimeout(5) exten => s,3,Background(mymenu) Bearing in mind that the extensions are => e

Re: [Asterisk-Users] Callmanager 4.1 and Asterisk

You need to create a SIP trunk in CCM and in Asterisk a peer in sip.conf with the IP address of the CCM (trunk) In the trunk configuration change the transport to UDP. Enter the IP of Asterisk. And create a route pattern with gateway the SIP trunk   In Asterisk in extensions.conf create the route t

[Asterisk-Users] Mediatrix 1204 DialPlan and Delay

Hi everybody, I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing. The problem here is the delay. When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call. For OUTGOING My Dialplan for the Mediatrix box

Re: [Asterisk-Users] Hardware opinions?

> What sort of chipset is your SATA controller interface? Intel > ICH6R? Adaptec ICH5R SATA controller according to SuperMicro which makes the Mobo. The board has an Intel® E7501 main chipset. Start Your Own Internet Service! http://www.YourOwnISP.com - Original Message - From: "Dorn H

Re: [Asterisk-Users] Sending call to analog then to Vmail aftertimeout?

-- Forwarded message -- From: C F <[EMAIL PROTECTED]> Date: Wed, 29 Dec 2004 00:34:28 -0500 Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail aftertimeout? To: Me <[EMAIL PROTECTED]> try the M option which will do a macro and will not connect the caller unless s/h

Re: [Asterisk-Users] OT: Linux routing with T100P problems

Matthew Boehm wrote: Hey gang, I was successful in recompiling my 2.4.20 kernel to support HDLC. I was successful in hooking up our T1 line into the zap card. I was successful in being able to ping equipment on the other end of the T1. I was unsuccessful in pinging the outside world from the other

Re: [Asterisk-Users] OT: Linux routing with T100P problems

On Tue, 2004-12-28 at 23:18 -0600, Matthew Boehm wrote: > Hey gang, > I was successful in recompiling my 2.4.20 kernel to support HDLC. I was > successful in hooking up our T1 line into the zap card. I was successful in > being able to ping equipment on the other end of the T1. I was unsuccessful

Re: [Asterisk-Users] OT: Linux routing with T100P problems

What can I use to find out why packets destined for the outside world (via 65.78.109.2) are not being routed? Check with your ISP and make sure they have you set up correctly. I have had issues in the past with that. Fact is, if you can ping the far end, *and packets are returned*, then the prob

Re: [Asterisk-Users] Sending call to analog then to Vmail aftertimeout?

I was aware of the "c" option but it's a pain for people to have to press the # sign plus they have to know they are suppose to do that. In addition, I tried to use the "A" option to play a sound to them when they answer reminding them to press pound at the end of the message but the sound doesn't

[Asterisk-Users] OT: Linux routing with T100P problems

Hey gang, I was successful in recompiling my 2.4.20 kernel to support HDLC. I was successful in hooking up our T1 line into the zap card. I was successful in being able to ping equipment on the other end of the T1. I was unsuccessful in pinging the outside world from the other end of the T1. I've

Re: [Asterisk-Users] Hardware opinions?

So are you saying that if I have one of the supported controllers, FC1 will work out of the box with the SATA drives attached? Also, what about FC2 or 3? Is there a patch for any of these three builds that will support the SATA controllers? Thanks! -- Start Your Own ISP! http://www.YourOwnISP.c

Re: [Asterisk-Users] Dialplan variables

Norman Zhang wrote: Hi, May I ask what does exten => s,1,Answer exten => s,2,ResponseTimeout(5) exten => i,1,Playback(pbx-invalid) s, t, i stands for? It says it is someexten but I still don't get it. Predefined Extension Names Asterisk uses some extension names for special purposes: * *i

Re: [Asterisk-Users] Realtime extension problem

Well if you got 100% CPU usage right after you insterted the rows, it seems more like a database issue. What did you use to determine 100% usage? Did you use top and it said asterisk was using 100%? -Matthw - Original Message - From: "VoIP" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailin

Re: [Asterisk-Users] MySQL Realtime Driver

There is info on the wiki. Matthew - Original Message - From: "Chris Tooley" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, December 28, 2004 5:38 PM Subject: Re: [Asterisk-Users] MySQL Realtime Driver > Is there any documentation or in

Re: [Asterisk-Users] Fedora Core 3 app_curl compile error?

At 03:55 PM 12/29/2004 +1300, you wrote: Michael Swan wrote: Hi, I'm making the latest CVS asterisk source on a newly installed Fedora Core 3 distribution. However, when the makefile for asterisk/apps runs, it generates an error when trying to link app_curl.so complaining about not finding -lidn. H

RE: [Asterisk-Users] Realtime extension problem

Actually I only inserted these two records to my database and got CPU 100% load. The AGI script didn't run because I didn't make any call. I am using RH9 and checkout the CVS data 12/09/04 version. - Original Message - From: "VoIP" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List

[Asterisk-Users] 500 "Internal Server Error"

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways.  I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "In

Re: [Asterisk-Users] Invalid Extension

Norman Zhang wrote: Hi, I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? Regards, Norman Zhang [inbound-sip] exten => 533990,1,Answer exten => s,2,ResponseTimeout(5) exten => s,3,Background(mymenu

Re: [Asterisk-Users] Fedora Core 3 app_curl compile error?

Michael Swan wrote: Hi, I'm making the latest CVS asterisk source on a newly installed Fedora Core 3 distribution. However, when the makefile for asterisk/apps runs, it generates an error when trying to link app_curl.so complaining about not finding -lidn. Has anyone else run into this problem? I c

Re: [Asterisk-Users] Invalid Extension

I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? I had to change all occurance of s to 533990 in order for this to work. 533990 is my FWD #. May I ask how can I genearlize this using s? Regards,

[Asterisk-Users] Fedora Core 3 app_curl compile error?

Hi, I'm making the latest CVS asterisk source on a newly installed Fedora Core 3 distribution. However, when the makefile for asterisk/apps runs, it generates an error when trying to link app_curl.so complaining about not finding -lidn. Has anyone else run into this problem? I can chase down libidn

[Asterisk-Users] Sending e-mail from dialplan

I would like help with a “dial plan” that will do the following: I feel pretty good about each of the elements except; how to e-mail the recorded file to an e-mail address.   Allow a caller to call into the system:   Answer play a short pre defined greeting Allow caller to ente

[Asterisk-Users] Invalid Extension

Hi, I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? Regards, Norman Zhang [inbound-sip] exten => 533990,1,Answer exten => s,2,ResponseTimeout(5) exten => s,3,Background(mymenu) exten => t,1,Goto(

Re: [Asterisk-Users] Dialplan variables

1/3 down the page are your answers. http://www.voip-info.org/wiki-Asterisk+config+extensions.conf - Original Message - From: "Norman Zhang" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, December 28, 2004 7:20 PM Subject: [Asterisk-User

Re: [Asterisk-Users] ASTCC Expiration

I told somebody I would get it done over Christmas. I'm still planning on getting the expiration date stuff up and running before Jan 1st but... If a few people would like to try out my last patches for astcc found at http://bugs.digium.com/bug_view_page.php?bug_id=0002796 I would bump it up

Re: [Asterisk-Users] How to connect two Asterisks as secure as possiblewithout too much additional bandwidth ?

Check this out: http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/ There's an article on how to use openvpn to encrypt data between two Asterisk Boxes. Should help, looks easy enough. Erik On Tue, 28 Dec 2004 16:57:11 -0800, Christopher Dobbs <[EMAIL PROTECTED]> wrote: > This

Re: [Asterisk-Users] Hardware opinions?

On Tue, Dec 28, 2004 at 04:12:26PM -0600, Me wrote: > Dorn, > > Can you give me some details on this linux md driver you mentioned? > > Also, you say not to scrap the SATA drives, is this because you think I can > use them with FC1 or because you think I should try Debian? I really don't > want

Re: [Asterisk-Users] How to connect two Asterisks as secure as possiblewithout too much additional bandwidth ?

This problem is being solved. See http://lists.digium.com/pipermail/asterisk-users/2004-November/073666.html I am currently in pre-testing phase of development. Features include: Optional Secondary Compression Selectable Encryption Level, from 32bit to 1024bit Uses UDP Voi

RE: [Asterisk-Users] Meetme scalable to 300 people?

[EMAIL PROTECTED] <> scribbled on : > What is the difference between meetme and app_conference? I > am looking to > conference a mere 10 people. Just getting into some testing > here. I have > ordered a $229 dell (on sale now) 2.5ghz celery/512mb > ram/80gb hd/cdrom/etc. > for a server and was

RE: [Asterisk-Users] How to connect two Asterisks as secure as po ssiblewithout too much additional bandwidth ?

SSH tunnel is the way to go. Here is a little tid bit about setting up SSH keys, a simple keep alive script, and creating the SSH tunnel I use to tunnel my SMTP traffic to a reliable SMTP server since my ISP blocks all traffic incoming/outgoing on port 25. http://xj.cdevco.net/comp/smtptunnel/

RE: [Asterisk-Users] SuperValetParkCall Application Unableto Re-ParkCall

The thread discussed being unable to re-park a call with app_supervaletparking -Original Message- From: Paul Zimm [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 28, 2004 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SuperValetParkCal

RE: [Asterisk-Users] Meetme scalable to 300 people?

What is the difference between meetme and app_conference? I am looking to conference a mere 10 people. Just getting into some testing here. I have ordered a $229 dell (on sale now) 2.5ghz celery/512mb ram/80gb hd/cdrom/etc. for a server and was thinking about ordering three PAP2-NA's for testing

Re: [Asterisk-Users] Sending call to analog then to Vmail after timeout?

Follow these: http://www.voip-info.org/wiki-Asterisk+zap+channels looks like this would work: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1c/551212,20,rTt) ;note the c after the channel number exten => 1200,3,VoiceMail([EMAIL PROTECTED]) exten => 1200,4,Goto,t|1

[Asterisk-Users] PRI & CPU Usage

Hi everyone, I'm a bit unclear about how PRI voice channels use CPU usage in an asterisk box - it is like a codec conversion or does a PRI channel have a generic codec itself? I am wondering what the CPU requirements are to, for example, handle all 120 channels on a TE405p and then trunk all th

[Asterisk-Users] Dialplan variables

Hi, May I ask what does exten => s,1,Answer exten => s,2,ResponseTimeout(5) exten => i,1,Playback(pbx-invalid) s, t, i stands for? It says it is someexten but I still don't get it. Regards, Norman Zhang ___ Asterisk-Users mailing list Asterisk-Users@lists

[Asterisk-Users] Meetme scalable to 300 people?

Hi everyone. I am looking at providing a conference for up to 300 people and was wondering if anyone has scaled meetme to 300 people. Here are some points: 1) I am using an IAX2 gateway hosted on a VOIP service provider. 2) The machine is hosted at the providers site so one has to assume tha

RE: [Asterisk-Users] Wildcard remote looping

We usually put an Adtran CSU ACE inline with the asterisk box and the Voice T1 that way if the LEC wants to loop a CSU they can. An Adtran CSU ACE just passes the traffic as it receives it, there is no setup involved. Henry Devito Telephone Connection, Inc Network Design / Implementation Phone: 4

Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

It would be nice if the table schema and basic usage would sketched out. I'm happy to add it to the wiki, I just have to get a usable configuration to do so. I have the following as my create table statement. I appreciate it if it was corrected. CREATE TABLE iax ( uniqueid int(11) NOT NULL au

RE: [Asterisk-Users] How to connect two Asterisks as secure as possiblewithout too much additional bandwidth ?

Couldn't you just tunnel the involved ports over SSH? As far as bandwidth is concerned you could enable compression and may even end up with a smaller data stream. You could generate both keys before hand and very simply do this on a *nix box. This would probably require both peers to have an ad

Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

- Original Message - From: "Nick Bachmann" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, December 28, 2004 3:33 PM Subject: Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime' > Gabriel Afana wrote: > > >Ahhh, and I've read e

[Asterisk-Users] ASTCC Expiration

How do you set the expiration date in ASTCC? DO you have to customize the CGI script? A maintenance fee field would be nice as well. Anybody? -- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 479.273.9992 Voice 479.464.8998 Fax ___

Re: [Asterisk-Users] MySQL Realtime Driver

Is there any documentation or insight on figuring out how to get RealTime IAX set up? I'm trying to do just that. Also can do separate peer/user configurations or just friends? And how do you configure the rest of the iax.conf information? On Fri, 10 Dec 2004 10:01:00 -0600, Matthew Boehm <[EM

Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

Gabriel Afana wrote: Ahhh, and I've read every message telling everybody they dont have the lastest version...thats why I went to asterisk.org and downloaded the highest-number version I could find in the FTP Okdownloaded latest CVS but now asterisk wont compile. I had it working before. d

Re: [Asterisk-Users] Wildcard remote looping

Mark Farver wrote: Is there something special that needs to be done to allow a T100P/T400 to respond to a remote loop request? For a T1 the phone company would expect to see their Network Interface Unit (NIU) which can be looped with a special repeating code. Using a different repeating code they

Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

> > > > /usr/bin/ld: /usr/lib/mysql/libmysqlclient.a(libmysql.o): relocation > > > > R_X86_64_32 can not be used when making a shared object; recompile > > > > with -fPIC > > > > /usr/lib/mysql/libmysqlclient.a: could not read symbols: Bad value > > > > collect2: ld returned 1 exit status > > > > >

[Asterisk-Users] Intercom System with Asterisk and Cisco 7960

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time.  I have 4 phones I would like all of then to answer when I dial x300.   Any help would be great Thanks   Tuska   extensions.conf [conference]exten => 300,1,AGI(callall)

Re: [Asterisk-Users] Hardware opinions?

On Tue, 2004-12-28 at 16:12 -0600, Me wrote: > Dorn, > > Can you give me some details on this linux md driver you mentioned? > > Also, you say not to scrap the SATA drives, is this because you think I can > use them with FC1 or because you think I should try Debian? I really don't > want to vent

Re: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco 79xx phones

Paul Rodan wrote: No clue. I know 150 worked though. I now have dhcp-option 66 and 150 defined as the IP and the phone took it, so whichever the phone wants it'll be there. Cisco IP phones prefer option 150 and will use it when it is available. If option 150 is not provided, they fall back to

RE: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco 79xx phones

No clue. I know 150 worked though. I now have dhcp-option 66 and 150 defined as the IP and the phone took it, so whichever the phone wants it'll be there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Tuesday, December 28, 2004 5:18 PM

Re: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco 79xx phones

> > > > But the Cisco phones are ignoring it. According to RFC2132, DHCP > > Option/Code 66 is the TFTP server name. But the Cisco 79xx phones I’ve > > tested are ignoring this. Code 66 works just fine for me on a Win32 dhcp server and multiple 7960's. Never misses. Rich

Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

> > > /usr/bin/ld: /usr/lib/mysql/libmysqlclient.a(libmysql.o): relocation > > > R_X86_64_32 can not be used when making a shared object; recompile > > > with -fPIC > > > /usr/lib/mysql/libmysqlclient.a: could not read symbols: Bad value > > > collect2: ld returned 1 exit status > > > > Never s

Re: [Asterisk-Users] Hardware opinions?

Dorn, Can you give me some details on this linux md driver you mentioned? Also, you say not to scrap the SATA drives, is this because you think I can use them with FC1 or because you think I should try Debian? I really don't want to venture away from Fedora at the moment for a few reasons. Than

Re: [Asterisk-Users] Hardware opinions?

On Tue, Dec 28, 2004 at 02:17:47PM -0600, Me wrote: > Hello, I am trying to build up a pretty meaty Asterisk box after doing our > initial testing and playing on a 1ghz system. > > Right now I have decided on a prebuilt system which I normally don't do but > thought it seemed like a good deal. >

Re: [Asterisk-Users] Compile Error

On Tue, Dec 28, 2004 at 10:02:12PM +0200, David Norton wrote: > Hi, > > I have been running asterisk for about a week though on a debian system > through apt-get. I am now trying to compile it use the CVS and im getting > this error. > > /usr/bin/ld: cannot find -lssl > > What do I need to in

Re: [Asterisk-Users] Zaptel ISDN BRI settings for The Netherlands KPN

On Tue, 28 Dec 2004, Remco Barende wrote: > signalling = bri_cpe_ptmp Use this to terminate a Point To Multipoint isdn line > ; p2p TE mode > ;signalling = bri_cpe And this if the line is point-to-point. This is likely if you have did:s or multiple isdn lines grouped together. > ; p2mp NT mode

Re: [Asterisk-Users] how to debug frame slips?

On Tue, 28 Dec 2004, Michael Welter wrote: > Joe Presto wrote: > > > >>Check 'vmstat 1'. With a "quiet" system you should see mostly 100% idle > >>time. How many interrupts are you seeing per one second interval? It > >>should be +/- 1000 for the system timer and +/- 1000 for each Digium car

Re: [Asterisk-Users] Music instead of Tunes

On Tue, 28 Dec 2004, Marc Storck wrote: > more and more operators in Europe offer music instead of ring tunes. > E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, > or Mozart Currently I will have to answer the line to do that. Is > there a way to do this with asteris

Re: [Asterisk-Users] Dialtone for Software phone?

Lane wrote: Hi, Is it possible with asterisk to deliver a dialtone to a software phone, such as kphone? I'm able to dial, but the silence seems to confuse my users :) SIP phones provide their own local dialtone. If you can get the SIP phone to call a predefined extension when it goes offhook yo

Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

> > /usr/bin/ld: /usr/lib/mysql/libmysqlclient.a(libmysql.o): relocation > > R_X86_64_32 can not be used when making a shared object; recompile > > with -fPIC > > /usr/lib/mysql/libmysqlclient.a: could not read symbols: Bad value > > collect2: ld returned 1 exit status > > Never seen that befor

[Asterisk-Users] Calling Card question

I am new to the list, so if this question is redundant, please point me in the right direction for reading. I want to setup some calling cards for fundraising. I have ASTCC installed and working, but I am wondering how things might work once in production. A customer calls an 800 number (sixTel)

[Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

Im pulling my hair out over here playing with this all night last night andall morningwhat am I missing!?!?I tried getting the Realtime extensions working...but I've been running intolots of problems and im getting frustrated.  first problem was theasterisk-addons wasn't compiling right

Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

> /usr/bin/ld: /usr/lib/mysql/libmysqlclient.a(libmysql.o): relocation > R_X86_64_32 can not be used when making a shared object; recompile > with -fPIC > /usr/lib/mysql/libmysqlclient.a: could not read symbols: Bad value > collect2: ld returned 1 exit status Never seen that before. What OS ar

[Asterisk-Users] Music instead of Tunes

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO

Re: [Asterisk-Users] Dialtone for Software phone?

On Tue, 28 Dec 2004, Lane wrote: > Hi, > > Is it possible with asterisk to deliver a dialtone to a software phone, such > as kphone? > > I'm able to dial, but the silence seems to confuse my users :) Tell them to think of it as a cellphone... -- Rick Green "They that can give up essential lib

FW: [Asterisk-Users] Compile Error

Sorry about that, I was been an idiot again!   I had openssl and libssl installed, but had the wrong version of libssl-dev.       From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Norton Sent: Tuesday, December 28, 2004 10:02 PM To: 'Asterisk Users Mailing

Re: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco79xx phones

Paul Rodan wrote: Curious, has anybody tried the Asterisk ipkg for OpenWRT on a WRT54G? http://nthill.free.fr/openwrt/ipkg/testing/Packages I'm running it in a variety of configurations on a number of WRT54GS models. There's no reason to think that it wouldn't work on the plain Gs as well, esp

Re: [Asterisk-Users] Sending call to analog then to Vmail after timeout?

Sorry about the HTML emails, on my laptop and forgot to change the sending format from the default. - Original Message - From: Me To: asterisk-users@lists.digium.com Sent: Tuesday, December 28, 2004 2:01 PM Subject: [Asterisk-Users] Sending call to analog then to Vmail after timeout? I

[Asterisk-Users] Hardware opinions?

Hello, I am trying to build up a pretty meaty Asterisk box after doing our initial testing and playing on a 1ghz system.   Right now I have decided on a prebuilt system which I normally don't do but thought it seemed like a good deal.   I have included the initial specs below, I will be addi

RE: [Asterisk-Users] Dialtone for Software phone?

Lol. It seems as simple as telling them to hit "9" and then dial. You can then get Asterisk to generate a dialtone, you could use something like DISA (I'm not too familiar with it [but I want to be]) or maybe just a background sound of a dialtone, and have it in a context where the outbound rules

[Asterisk-Users] Two problems with the Perl AGI

Hi, I have a * 1.0.3 running on a Gentoo box and I installed Perl AGi from http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz. When I write this : >#!/usr/bin/perl >use Asterisk::AGI; >my $AGI = new Asterisk::AGI; >$AGI->exec ('Dial SIP/kphone1|30|tTr'); >my $duration = $AGI->get_varia

Re: [Asterisk-Users] Dialtone for Software phone?

Lane wrote: Hi, Is it possible with asterisk to deliver a dialtone to a software phone, such as kphone? I'm able to dial, but the silence seems to confuse my users :) Tell it's a software cell phone. NEXT!!! Steve ___ Asterisk-Users mailing list Asteris

[Asterisk-Users] Sending call to analog then to Vmail after timeout?

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number).   When I do this in my extensions.conf:   exten => 1200,1,playback(pls-wait-connect-call)exten => 1200,2,Dial(Zap/1/551212,20,rTt)exten => 1200,3,VoiceMail([EMAIL PROTECTED])exten => 1200,4

[Asterisk-Users] Compile Error

Hi,   I have been running asterisk for about a week though on a debian system through apt-get. I am now trying to compile it use the CVS and im getting this error.   /usr/bin/ld: cannot find –lssl   What do I need to install to get rid of this message?   Regards   David Norton

[Asterisk-Users] Asterisk users manual

Hi List! All Asterisk info I found so far is relating to installing hardware or configuring asterisk. Does anybody know of a simple (small) manual for the users of *? Especially where details like transferring calls, 3way conversations, setting voicemail, diverting calls etc. are discussed? (I

[Asterisk-Users] Dialtone for Software phone?

Hi, Is it possible with asterisk to deliver a dialtone to a software phone, such as kphone? I'm able to dial, but the silence seems to confuse my users :) thanks, lane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digiu

RE: [Asterisk-Users] rejected calls from IAX provider

A register line simply tells the provider where to send your calls. It is still up to you to setup a user entry in your iax.conf that they will use to send the call. This is simply a case of you not properly configuring iax. NEXT!!! - Joshua Colp. -Original Message- From: [EMAIL PROTECTE

RE: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco79xx phones

So that's what the parameter is for. Jeesh. Live and learn. Anyways, I'm going to try DHCP Option 150 as well to see if that does the trick. Thanks! I think I have Wonder Shaper working on my WRT54G as well, so that'll do the QOS I need. It's also a TFTP Server to hold the phones config files an

[Asterisk-Users] rejected calls from IAX provider

Hello, I am register to IAX provider. in iax.conf: register => user:[EMAIL PROTECTED] When the user is trying to call me my asterisk rejected all calls. I get on debug: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type

RE: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco 79xx phones

> The thing I dislike the most about the 79xx phones is that in > DHCP mode, they expect the DHCP server to tell them their > TFTP server address. They won't let you set it manually. Ummm, yes they do. The 7960 I had previously used DHCP to get it's internal IP from my router but allowed me to spe

Re: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco 79xx phones

Paul Rodan wrote: The thing I dislike the most about the 79xx phones is that in DHCP mode, they expect the DHCP server to tell them their TFTP server address. They won't let you set it manually. So if I don't have DHCP server that gives TFTP server info, which is most of the DHCP servers at out th

Re: [Asterisk-Users] DHCP, the TFTP Server setting and the Cisco 79xx phones

Paul Rodan wrote: The thing I dislike the most about the 79xx phones is that in DHCP mode, they expect the DHCP server to tell them their TFTP server address. They won’t let you set it manually. So if I don’t have DHCP server that gives TFTP server info, which is most of the DHCP servers at out

[Asterisk-Users] Polycom phone stops working

my polycom 300 works fine when i first turn it on but after five minutes or so it stops accepting calls but i can still make calls from it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-

Re: [Asterisk-Users] Linux Distribution

On Tue, Dec 28, 2004 at 09:13:02AM -0800, Geoff Nordli wrote: > [EMAIL PROTECTED] <> scribbled on : > > > On Sat, Dec 25, 2004 at 12:29:21PM +, Jean-Michel Hiver wrote: > >> Seth Ueland Chancy wrote: > >> > >> Probably your best bet is Debian + 2.4 kernel + X100P card + apt-get > >> install a

[Asterisk-Users] Callmanager 4.1 and asterisk

I have a similar setup.   To make it easy and get the best of both worlds, have the Linux softphones (SIP or IAX) register to Asterisk.   Keep the physical phones registered to CM.   From there setup a dialplan on both Call Manager and Asterisk to relay calls between the two systems.   For

[Asterisk-Users] DHCP, the TFTP Server setting and the Cisco 79xx phones

The thing I dislike the most about the 79xx phones is that in DHCP mode, they expect the DHCP server to tell them their TFTP server address. They won’t let you set it manually. So if I don’t have  DHCP server that gives TFTP server info, which is most of the DHCP servers at out there, then

Re: [Asterisk-Users] Incoming Calls

-- Forwarded message -- From: C F <[EMAIL PROTECTED]> Date: Tue, 28 Dec 2004 13:21:45 -0500 Subject: Re: [Asterisk-Users] Incoming Calls To: Rich Adamson <[EMAIL PROTECTED]> I didn't try Dial but I did try wait and it didn't help. I'll try dial and see what happens. It might take

[Asterisk-Users] Asterisk and ISDN via RemoteCapi

Hi folks, I was looking quite unsuccessfully for some info or experiences using * with chan_capi or isdn4linux using a RemoteCapi client (distributed Capi client), based on ISDN-DCP based ISDN-routers like a Zyxel Prestige 202 or CISCO 740 or BINTEC BIANCA RCapi ... Anybody out there who could

Re: [Asterisk-Users] turn on/off auto/attendant by dialing an extension

I think you're misunderstanding DBget and DBput. Asterisk has a built in database where it will store this information. No need for MySQL or any other DB software. This solution will provide you with copy/paste installations and is by far the simplest solution. You can read about the Asterisk d

[Asterisk-Users] VoIP Equipment

Anyone interested in a lot of gently used IP500’s with SIP Image? Also, anyone in need of large quantities of SPA-2000’s at an extreme discount?   If so, please contact me off list.   Thanks,   Garrett Smith [EMAIL PROTECTED]   B2 Technologies/ VoIPSupply.com 454 Sonwil Drive Buf

Re: [Asterisk-Users] ZtDummy vs Hardware

Kristian Kielhofner wrote: Jean-Michel Hiver wrote: I was wondering what could be pros and cons of ztdummy vs proper timer device (i.e. X100P). I am going to set up an asterisk server in europe (to do trunking, to save bandwith) and I was wondering if it'll be OK to get it going with ztdummy.

Re: [Asterisk-Users] FC3, TDM11B (DEVPCI) and asterisk

I had the same problem, the way I worked it around was that I added the follwoing to the script: /etc/rc.d/init.d/zaptel case "$1" in start) # Load drivers rmmod wcusb >& /dev/null rmmod wcfxsusb >& /dev/null rmmod audio >& /dev/null action "Loading zaptel

[Asterisk-Users] ztdummy necessary?

I have got my first * server set up and serving users in three different locations over the Internet. This is currently a test setup so I am experimenting with the different features of *. When I set up asterisk, I only checked out the Stable source of Asterisk from CVS, and compiled it. I did not

[Asterisk-Users] Wildcard remote looping

Is there something special that needs to be done to allow a T100P/T400 to respond to a remote loop request? I am having some issues with a Point to point T1 line using zaptel ppp. This line gave us small problems when we had a pair of Cisco 2600's on either end but now with the zaptel ppp it is go

RE: [Asterisk-Users] Mysql and Voicemail

Hello try to enable mysql debug: "log=/var/log/mysqlfull.log" in your /etc/my.cnf and off course reload mysql then tail -f /var/log/mysqlfull.log it will show you if your asterisk connects to the DB... if not, it's a makefile problem... re-read tutorial... PS: don't forget to try if your "ful

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