Re: [Asterisk-Users] Two AVM Fritz Card (hack does not work) what I am doing wrong?

hi Jakob¸ i see you have installed 2 fritz card in PC .. I have a lot of problems already with one card .. When i type capiinfo computer freeze. I have shared irq of fritz card with some motherboard resources ? You think this is a problem ? what kernel you use? what drivers version of drivers yo

[Asterisk-Users] Read this if you use CVS HEAD (Unstable) and chan_capi and jitter buffering in IAX

Hi, IF: you are using chan_capi in combination with IAX, and you are using the IAX jitter buffer (see iax.conf) and you run with the CVS HEAD version of Asterisk, THEN: You must make sure that you are running the most current chan_capi version - which is 0.3.4 at the time of writing. Ol

R: [Asterisk-Users] Which Linux ?

> Based on th wiki, avoid kernel 2.6 unless you know what you are doing. > Likewise with fedora, which seems to work but needs kernel thread turned off. Just my experience: I have installed Asterisk twice on Fedora Core 1 with kernel 2.4.22-1.2188.nptlsmp on Dual Xeon systems. It has worked perfe

re: [Asterisk-Users] Which Linux ?

>From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thre

[Asterisk-Users] false hangups

Hello, We are using a TDM400p with 4 FXOs and SIP phones in a high call-volume environment. At least twice a day there are complaints of 'dropped calls'. Examining the debug logs, I see that in each case, an "on hook" event is detected, followed by the zap channel being hung-up and * saying

Re: -- [Asterisk-Users] Serious issues with current CVS?

For me chan_capi set up the call but no sound available. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://list

RE: [Asterisk-Users] CDRs, Conferencing, and MeetMe

On 07:42 PM 6/23/2004, Dan Austin wrote: >I started to think about such a system for internal use at work. I >never >got past the brainstorming phase, but I'd suggest having an AGI script >answer the call, query and verify the conference/pin numbers and then >transfer the call to the appropriate M

Re: -- [Asterisk-Users] Serious issues with current CVS?

Its been fixed... wait for the mirrors to get the update... bkw - Original Message - From: "TC" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 23, 2004 10:59 PM Subject: Re: -- [Asterisk-Users] Serious issues with current CVS? > OK at least i have it narrowed down he

Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!

I'm excited. At the moment we're using aggressive echo cancellation - which is nice, and works - but it would be great to not have to complete kill the other person's voice when speak. Looking forward to trying it a bit later... Andrew _ On 24/06/2004, at 11:54 AM, Isamar

Re: -- [Asterisk-Users] Serious issues with current CVS?

OK at least i have it narrowed down here if you revert channel.c to r1.21 and channels/chan_sip.c to r1.422 life seems better i can't do a diff any more since digiium cvs just vanished on me - Original Message - From: "TC" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 2

[Asterisk-Users] Which Linux ?

Hi there, linux got so many distro, but which one that have more compability with the Asterisk? Regards, Freddys ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] cdr_mysql: Unknown connection error

Usually we get this error "cdr_mysql: Unknown connection error" early in the morning when the first call tries to get logged into the database. The result of course is that the call does not get logged. But all calls after that get logged fine for the rest of the day. This happens on ALL our

Re: -- [Asterisk-Users] Serious issues with current CVS?

I just updated to current. My SIP clients, Xlite and a Sipura-2000, no longer connect. There's nothing listeed under "sip registry", and with "sip debug" enabled, it shows no activity as Xlite is trying to login. On Wed, 23 Jun 2004 20:46:49 -0500, John Baker <[EMAIL PROTECTED]> wrote: > > I've

Re: [Asterisk-Users] Channel bank problem via long cable

On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote: > > Nah, a true analyzer will detect framing errors do loopback echo tests, etc. After much pfutzing around and talking with CAC's tech support, I'm finally coming around to your original suggestion and am in the process of finding someo

RE: [Asterisk-Users] CDRs, Conferencing, and MeetMe

I started to think about such a system for internal use at work. I never got past the brainstorming phase, but I'd suggest having an AGI script answer the call, query and verify the conference/pin numbers and then transfer the call to the appropriate Meetme. Dan -Original Message- From:

RE: [Asterisk-Users] Really basic stuff :(

Gavin Hamill wrote: > Hi :) > > I've had all this working before, but I'm revisiting it, and in > short, I currently have huge problems receiving incoming calls. I've > been trying with both FWD and voiptalk.org. I'm running CVS HEAD of > asterisk, zaptel and libpri as of yesterday afternoon. >

[Asterisk-Users] #1 Asterisk and Locustworld

I currently working on a project using Locustworld's wireless mesh network and running SIP through the network.  I want to install Asterisk as server running sip and Digium FXO FXS dev lite card.  Unfortunately I having the time of my life to install Asterisk.  The astinstaller does everythi

RE: FW: [Asterisk-Users] No dial tone after installation

The stuff I suggested you put in /etc/asterisk/modules.conf would prevent either of those modules from loading, so asterisk is not reading that file correctly On Wed, 2004-06-23 at 21:12, Don Moskaluk wrote: > So far so good, vi came back to me after 5 years of being idle! > > I'm getting error m

RE: FW: [Asterisk-Users] No dial tone after installation

So far so good, vi came back to me after 5 years of being idle! I'm getting error messages on the following: Chan_skinny.c 2569 reload config Chan_oss.c sound card init unable to load And yes no dial tone. Don Moskaluk [EMAIL PROTECTED] www.moskaluk.com

Re: [Asterisk-Users] Dell 400SC and X100P

Is your kernel ACPI enabled ? The motherboard in the PE400SC is basically the Dimension 8300, which i use for my development box with 1 X101P, 1 TDM400P and two ISDN cards here at home and that works without problems. One thing to make sure with these boards is that ACPI is enabled, since they ar

Re: FW: [Asterisk-Users] No dial tone after installation

I forgot to add noload => chan_alsa.so On Wed, 2004-06-23 at 20:43, Eric Wieling wrote: > /etc/asterisk/modules.conf put in: > > noload => chan_modem.so > noload => chan_modem_i4l.so > noload => chan_modem_bestdata.so > noload => chan_modem_aopen.so > noload => chan_phone.so > noload => chan_oss.

[Asterisk-Users] Dell 400SC and X100P

I have a Dell PowerEdge 400SC with a X100P and a TDM01b. The board works wonderfully in another machine but in this brand new one, it just get in nuts. The problem is: 1) Zaptel recognizes it perfectly 2) No IRQ conflicts, two-wire new cable. 3) Asterisk starts up and listen the ring and answer

Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!

Beside of problem of CallerID not working in several countries, what specially I still don't know if it's a X100P's limitation or Zap driver limitation, the major problem with this devices is no doubt the echo. If you guys can solve this with tdms and x100ps, it's gonna be a big step. On Wed, 23

[Asterisk-Users] Video/H323/SIP

It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To U

Re: -- [Asterisk-Users] Serious issues with current CVS?

I've got exactly the same results also. John Baker Wojciech Tryc wrote: Same here, I also lost DTMF on some SIP devices (Grandstream phones) and fax detection on Zap devices. W. Is anybody else having serious issues with the current version from CVS? I just compiled and installed it and: 1) I was

Re: -- [Asterisk-Users] Serious issues with current CVS?

Damn... I just upgraded my Fedora Core 1 from .2188 to .2194, I thought that was the problem so I backtracked to .2188 only to find my kernel-source garbled so I'm downloading it again... I could have saved myself the trouble.. :-) Oh well... Something good came out of it, I learned how to back

Re: FW: [Asterisk-Users] No dial tone after installation

/etc/asterisk/modules.conf put in: noload => chan_modem.so noload => chan_modem_i4l.so noload => chan_modem_bestdata.so noload => chan_modem_aopen.so noload => chan_phone.so noload => chan_oss.so noload => chan_iax.so noload => chan_skinny.so On Wed, 2004-06-23 at 20:09, Don Moskaluk wrote: > Oh

RE: [Asterisk-Users] help needed with read()

Hi! > > Back to my question, lets say I want to use the digits collected by read() > > to dial out an extension. how do I do that ? exten => 1234,1,Read(testvar) exten => 1234,2,Dial(SIP/${testvar}) Cheers, Philipp ___ Asterisk-Users mailing list [EM

Re: -- [Asterisk-Users] Serious issues with current CVS?

YUP lots of total weridness i am trying to track down 1) on sip devices no DTMF, and endless msg reting to get a lock 2) ast_masq deadlocking ..stay tuned but stuff is seriously broken or I am on best drugs i have had in years - Original Message - From: "Wojciech Tryc" <[EMAIL PROTECTED]>

FW: [Asterisk-Users] No dial tone after installation

Oh geez nobody respond, I guess I have to be more descriptive.   I'm install Asterisk and FXO FXS Lite dev kit from Digium everything has installed and I am getting an error message that my sound card is not enabled.  I'm using a Dell Optix (whatever).  Everything seems to be in order.  I c

Re: -- [Asterisk-Users] Serious issues with current CVS?

Same here, I also lost DTMF on some SIP devices (Grandstream phones) and fax detection on Zap devices. W. > Is anybody else having serious issues with the current version from CVS? I > just compiled and installed it and: > > 1) I was able to establish one and only one call before things went weird

[Asterisk-Users] Serious issues with current CVS?

Is anybody else having serious issues with the current version from CVS? I just compiled and installed it and: 1) I was able to establish one and only one call before things went weird. 2) It stopped responding to IAX calls after the first. Completely ignored any subsequent commands, including h

[Asterisk-Users] latest CVS DTMF with Grandstream broken

Just installed the latest CVS and the * does not recognize DTMF from grandstream devices. The Sipura 2k ports are fine. Also, it seems that I lost fax receiving functionality. The incoming fax just rings and gets my IVR it doesn't get detected I downgraded to resolve DTMF with Grandstream and e

[Asterisk-Users] Problem with Unavailable Message Creation

I've changed the spool directory in asterisk.conf to point to a different directory. Everything works/gets created just fine with the exception of the unavailable messages. When a user tries to create one, I get this on the console (below). I changed the directory to /vm in asterisk.conf. Any he

[Asterisk-Users] tdm (and x100p?) echo - fix is coming!

Good news! FYI, worked with Mark this afternoon to test changes needed to reduce or eliminate echo involving pstn calls on the new tdm fxo card (bug #1902). Three simple source code changes (testing-only at this point) resulted in zero detectable echo on all incoming and outgoing tdm pstn calls,

[Asterisk-Users] Swissvoice ip10s

I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be grea

[Asterisk-Users] Asterisk and ISDN

Hi,   I'm trying to use Asterisk with one ISDN TELES 16.03 c PnP Card (ISA) Now i can call Asterisk with the Modem i4l driver etc But need more information to make a better config and also know how call this card to make outgoing calls and receive incoming calls. I know how use with a Zap card, but

RE: [Asterisk-Users] Skype 4 Linux

On the same topic, the SDK for the Siemens cordless USB phone that runs Skype was released to developers this week (I'm in Australia so I don't have it yet) but would be interesting to see if someone can make these M34 handsets work with asterisk. Cheers, Dean -Original Message- From: [E

[Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] Asterisk answering only one (dialed-) Number on a PTMP (German "Mehrgeräteanschluss")?

Thank you! I am using AVM Fritzcard (fcclassic) with chan_capi and the tip of using capi.conf in /etc/asterisk/, to solve my problem, was great! I didn't notice this capi.conf cause I thought it is the same one as the capi.conf in /etc. ;-) Thanks to all you helpful people! Now I got my "toy" wi

[Asterisk-Users] New VM feature: broadcast and delete=yes

New in CVS - taken from bug 1361: Voicemail broadcasts Description: Add a flag to the last field of voicemail boxes that allows deletions: delete=yes Also permits voicemail to be entered in the following manner, to allow multiple recipients for a single recording: Voicemail([EMAIL PROTECTED]

Re: [Asterisk-Users] Sipura config

Jay, The steps are: 1. Set up a web server 2. Create a flat text file with all the parameters you want to load in a format like: Display_Name[1] "bla bla" ; User_ID[1]"12345" ; Password[1] "45667" ; ..etc... 3. Feed that file thou

[Asterisk-Users] Asterisk info needed for new application development.

As some of you know we are developing some client software, CallerID, Receptionist programs, Call management etc. If you can answer some questions it would really be helpful. Please respone off list if you feel appropriate. 1) How many extension do you have on your systems? 2) How many inco

RE: [Asterisk-Users] SIP and audio delay

Kubat, Philip <> scribbled on Wednesday, June 23, 2004 2:32 PM: > I have a SIP connection to Broadvoice and sometimes when I make > outgoing calls from a SIP ATA-188 (could be the same number) (the > ATA-188, is currently the only extension), there is no audio passed > for 5-10 secs. I have set a

RE: [Asterisk-Users] Asterisk inbound assistance

Ata186 are already registered with asterisk and I can call them ext to ext   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kubat, Philip Sent: Wednesday, June 23, 2004 4:47 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk inbound assistance   Do

[Asterisk-Users] UPDATE Patch for postgres enabled app_voicemail.c

I forgot to take out the portion that would read in the voicemail boxes from the text file. If you want to leave it in then you could have some voicemail boxes defined in the text voicemail.conf. I do not, so I have removed it. Below is the new patch: *** app_voicemail.c 2004-06-23 07:55:54.0

[Asterisk-Users] Really basic stuff :(

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My

[Asterisk-Users] Patch for postgres enabled app_voicemail.c

Hello all, I am just getting going on building my system, but I thought I'd send you all a patch that I wrote so the command: show voicemail users issued from the CLI works properly when there is a postgres backend for the voicemail. The current version of the app does not display the voicemail

RE: [Asterisk-Users] Asterisk inbound assistance

Do you have the ATA “registering” with Asterisk?   SIPRegOn set to 1.   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Kenney Sent: Wednesday, June 23, 2004 4:34 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk inbound assistance   I have a te

[Asterisk-Users] Asterisk inbound assistance

I have a telephone number that points to my asterisk service and I want it to ring to my cisco 186 when I call it but for some reason when I dial it I get a busy signal and my asterisk server rejects it what am I doing wrong?     Sip.conf [dstech4] type=friend username=dstech4 secret

[Asterisk-Users] SIP and audio delay

I have a SIP connection to Broadvoice and sometimes when I make outgoing calls from a SIP ATA-188 (could be the same number) (the ATA-188, is currently the only extension), there is no audio passed for 5-10 secs.  I have set all the codec the same to 711u and also ensured canreinvite is set

[Asterisk-Users] Works for a while and then rings off hook

I have a X100P that works great for a couple days maybe even a week and then outside callers say my phone just rings and rings. When I try to dial out during this period, it waits and then Allison says "Goodbye" and hangs up. I have to stop *, modprobe -r wcfxo, modprobe wcfxo, and ztcfg then r

Re: [Asterisk-Users] Codec G729 Registration problem

You don't use that codec anymore.  Its not the one thats being sold by digium anymore... get the new one from ftp.asterisk.org/pub and the new register program and then register the new one.   bkw - Original Message - From: Carlos Medina To: [EMAIL PROTECTED] Sent: We

RE: [Asterisk-Users] help needed with read()

On Wed, 2004-06-23 at 15:10, Sathya wrote: > Thanks Steve. Please do not group reply or reply all. > I thought it is better that agi be run without any user interactions. When > called, It will do its thing and give back the control to extension.conf. Is > it not ? Better than what? It is easy,

RE: [Asterisk-Users] help needed with read()

Thanks Steve. I thought it is better that agi be run without any user interactions. When called, It will do its thing and give back the control to extension.conf. Is it not ? Back to my question, lets say I want to use the digits collected by read() to dial out an extension. how do I do that ? c

Re: [Asterisk-Users] sidetone noticeably loud on analog handsets on T100P

I just received some new analog phones that sound great, the sidetone isn't noticeable. They are Easy Touch 76510's by NorthWestern Bell Phones. The problem still exists for my other analog phones. :( I figure it probably must have to do with my Rhino Equipment FXS channel bank guys... I'm in

[Asterisk-Users] Cisco ata-186 port died

This might be the problem. I remember that i turned of the ringer (its an older style telephone with a switch on the back to switch to pulse, and turn the ringer off) so maybe the ATA had to much resistance and blew something. Anyone have expirience with this?

[Asterisk-Users] Codec G729 Registration problem

Hi, i have a problem trying to register the codec G729, my licence is valid but when i try to Register i got the following error:   "Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate wit

[Asterisk-Users] CDRs, Conferencing, and MeetMe

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this inform

[Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart

The problem is resolved... I updated to the very latest CVS and it seems to work now... Thanks to everyone who commented... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCR

Re: [Asterisk-Users] Fax with SPA-2000's?

On 2004.06.19 05:51 mattf wrote: We have it working reliably on calls coming in over a T1, I don't remember the exact settings we used, but I will send on Monday when I'm back in the office. Make sure that everything is 711Ulaw only and that your fax or modem operates no faster than 9600, data thro

Re: [Asterisk-Users] Fax with SPA-2000's?

On 2004.06.18 23:15 Seth Mattinen wrote: I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get

[Asterisk-Users] problems compiling zaptel X100P on Redhat Fedora 2.6.5-1.358

Ok I did see that, hoever I have never recompiled, or even compiled the kernal.. Didn't want to make a hash of it.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update optio

[Asterisk-Users] Digium/Asterisk in Paris

I will be coming to Paris and then Bordeaux for the Libre Software Meeting (LSM) 2004. We will have a small get together for Asterisk users in France or Europe on Friday July 2, 2004 at Le Lateral, 4 Avenue Macmahon (Metro Charles de Gaulle Etoille, sortie Avenue de Wagram) at 12:00 p.m. So, in su

Re: [Asterisk-Users] X100P Noise

Brent Franks wrote: How does one prevent the interrupts from being shared? Check your BIOS settings. You should be able to assign from there. Do you mean like setting up the ISA slots? I've got a built in ethernet and USB which both sit on IRQ 10 (drives me nuts) and have no idea how to set eithe

Re: [Asterisk-Users] zttool CLI

On Wed, Jun 23, 2004 at 12:45:59PM -0400, [EMAIL PROTECTED] wrote: > I need to check red alarms status from the script, but asterisk CLI "zap > show channel 1" or "pri show span 1" does not tell me this. > zttool does, but I can run it only in interactive curses mode. > Is there any ready solution

[Asterisk-Users] Three Way Calling and External Flash Hook

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and th

[Asterisk-Users] Compiling zaptel under 9.1 Suse

Have some errors with the above. I have tried make and make linux26 Anyone got any clues ? I've googled but only got the "make linux26" help Asterisk compiles and runs great, libpri compiles with no problems. TIA Julian. pbx:~ # cd /usr/src/zaptel pbx:/usr/src/zaptel # make linux26 make -C /u

Re: [Asterisk-Users] sidetone noticeably loud on analog handsets on T100P

I figure it probably must have to do with my Rhino Equipment FXS channel bank guys... I'm in the process of researching if there's anyway to adjust it on the equipment. I also agree with what some of you guys are reporting re. the rx/txgain not making any difference. I think that's for the s

[Asterisk-Users] no sounds from sample config

I have installed a default * installation with the samples. I can run * and it seems to be working correctly but I can not get the samples to play. When I launch the console and type dial 1000 I get a single beep from the sound card and the following on the console window but I do not hear the demo

RE: [Asterisk-Users] Polycom IP 600 Programmability

I would also be interested in similar functionality. We have agents using Polycom IP 600s that would like some sort of notification that they are logged into Asterisk Queues - either a flashing LED or perhaps some sort of graphic on the display. I know that there are numerous configuration options

Re: [Asterisk-Users] Re: Polycom IP 600

I would also be interested in this functionality once Polycom responds as we have many phones which require more lines. Thanks, Erik On Thu, 2004-06-17 at 15:10, Eric Mandel wrote: > Hi Tor, Thanks for trying. I opened up a case with Polycom today to look > into this. I spoke with a tier 2 eng

Re: [Asterisk-Users] problems compiling zaptel X100P on Redhat Fedora 2.6.5-1.358

On Wed, 2004-06-23 at 12:11, Tucker wrote: > Hi can you help ? > > I have been trying to compile the zaptel modules all > day for installation with out success, see messages > below > > Setup as follows > X100P card > Intel Celeron (686) 1.7ghz processor > Redhat Fedora Core 2 (Linux-2.6.5-1.

Re: [Asterisk-Users] zttool CLI

On Wed, 2004-06-23 at 11:45, [EMAIL PROTECTED] wrote: > Hello, > > I need to check red alarms status from the script, but asterisk CLI "zap > show channel 1" or "pri show span 1" does not tell me this. > zttool does, but I can run it only in interactive curses mode. > Is there any ready solution?

Re: [Asterisk-Users] X100P Noise

On Wednesday 23 June 2004 10:48, Neil Cherry wrote: > Ryan Courtnage wrote: > > On Wednesday 23 June 2004 08:17, Lee Norvall wrote: > >>I have 2 x X100P on UK BT, both have been working fine for a while, but > >>now I have started to get a beeping sound my end every 8/10 sec, and > >>break-up in th

Re: [Asterisk-Users] X100P Noise

> > How does one prevent the interrupts from being shared? > Check your BIOS settings. You should be able to assign from there. Thanks, - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-user

Re: [Asterisk-Users] Conference application !

Sergio Galeotti wrote: Hi, I´m just compiling the app_conference but I can´t locate the common.h file , those it´s requered. Someone help me to locate Common.h file Thanks What's wrong with app_meetme? Jeremy McNamara ___ Asterisk-Users mailing list

[Asterisk-Users] tdm fxo users - new bug tracker entries

For those using the new tdm fxo modules, please review and add your comments to: Bug Description 1898 tdm fxo does not sense broken pstn line 1899 tdm fxo senses pstn line disturbances as ringing 1902 tdm fxo echo when calling from sip phone There is at least a suspicion on my part that

Re: [Asterisk-Users] zttool CLI

nothing ready, but is pretty simple to strip away all newt stuff from zttool and make the output go to stdout... On Wed, 2004-06-23 at 18:45, [EMAIL PROTECTED] wrote: > Hello, > > I need to check red alarms status from the script, but asterisk CLI "zap > show channel 1" or "pri show span 1" does

Re: [Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart

Same here, everything running as root. Extensions.conf seems to update fine when I do changes from the CLI Please tell me you don't need to set asterisk u+s... - Original Message - From: "Nik Martin" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 23, 2004 9:41 AM S

Re: [Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart

Same here... very strange - Original Message - From: "Shaun Ewing" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 23, 2004 9:37 AM Subject: Re: [Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart > On Wed, 23 Jun 2004 09:22:40 -0700, Chris Shaw <[E

[Asterisk-Users] SNOM 200 using GSM Codec dtmf problem

Hello All, I'm trying to get my SNOM 200 to work using the GSM codec. The problem is it wont' pass dtmf digits. I have it dtmf outband on, inband off. In my sip.conf I have dtmfmode=rfc2833. Whenever I press a digit on the snom phone I get "invalid gsm data" so I think it's a configuration p

[Asterisk-Users] problems compiling zaptel X100P on Redhat Fedora 2.6.5-1.358

Hi can you help ? I have been trying to compile the zaptel modules all day for installation with out success, see messages below Setup as follows X100P card Intel Celeron (686) 1.7ghz processor Redhat Fedora Core 2 (Linux-2.6.5-1.358) When I do the make I get the following errors Any commen

RE: [Asterisk-Users] FXO impedance matching

From: Nik Martin <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] FXO impedance matching Date: Wed, 23 Jun 2004 11:02:00 -0500 To: [EMAIL PROTECTED] > Michael Welter wrote: > > Jason A. Pattie wrote: > >> Robert Hajime Lanning wrote: > >> > >>> Echo echo ech

Re: [Asterisk-Users] X100P Noise

Ryan Courtnage wrote: On Wednesday 23 June 2004 08:17, Lee Norvall wrote: I have 2 x X100P on UK BT, both have been working fine for a while, but now I have started to get a beeping sound my end every 8/10 sec, and break-up in the voice call inbound/outbound. Any ideas??? Sounds like your x100p c

[Asterisk-Users] zttool CLI

Hello, I need to check red alarms status from the script, but asterisk CLI "zap show channel 1" or "pri show span 1" does not tell me this. zttool does, but I can run it only in interactive curses mode. Is there any ready solution? Best Regards, Ivan __

Re: [Asterisk-Users] Skype 4 Linux

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 23 June 2004 03:56 am, Stefan de Konink wrote: > Hi All, > > Since 21 june skype is available to be used on Linux, with a static > binary, which includes QT, of 8 meg its big. > > http://www.skype.com/help_linux_faq.html I downloaded it t

RE: [Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart

Chris Shaw wrote: > Ok I have googled and googled and combed through the wiki for an > answer to this and have come up empty. What I'm finding is that when > a user changes their VM password, it is saved somewhere like maybe > the CSV database or something because when you log in, the new > passwor

Re: [Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart

On Wed, 23 Jun 2004 09:22:40 -0700, Chris Shaw <[EMAIL PROTECTED]> wrote: > Is this true? Or can asterisk save to the voicemail.conf and I just need to > update my CVS? > >-Chris I don't know about the rest, but my Asterisk install certainly updates voicemail.conf when changing passwords.

[Asterisk-Users] Conference application !

Hi, I´m just compiling the app_conference but I can´t locate the common.h file , those it´s requered. Someone help me to locate Common.h file Thanks

RE: [Asterisk-Users] asterisk + appradius & freeradius

> [EMAIL PROTECTED] wrote: > Here is the jist: Freeradius is up running and > functional using SIP Express radius how to. My > asterisk box has app radius installed. Is there > any documents on how-to link asterisk to freeradius? > documentation is lacking on app radius, at least > not as detailed

RE: [Asterisk-Users] X100P Noise

Hi I remembered that I had disabled USB 2.0 on the motherboard last week. I rebooted the server, enabled USB 2.0 and all seems a lot better. I guess this may have cleared any conflicts.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage Sent

[Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart

Ok I have googled and googled and combed through the wiki for an answer to this and have come up empty. What I'm finding is that when a user changes their VM password, it is saved somewhere like maybe the CSV database or something because when you log in, the new password works fine, but it's not s

Re: [Asterisk-Users] help needed with read()

On Wed, 2004-06-23 at 10:12, Sathya wrote: > asterisk*CLI> > -= Info about application 'Read' =- > > [Synopsis]: > Read a variable > > [Description]: > Read(variable[|filename]): Reads a '#' terminated string of digits from > the user, optionally playing a given filename first. Returns -1

[Asterisk-Users] asterisk + appradius & freeradius

Here is the jist: Freeradius is up running and functional using SIP Express radius how to. My asterisk box has app radius installed. Is there any documents on how-to link asterisk to freeradius? documentation is lacking on app radius, at least not as detailed as I need. Anyone know of a how-to or a

RE: [Asterisk-Users] FXO impedance matching

Michael Welter wrote: > Jason A. Pattie wrote: >> Robert Hajime Lanning wrote: >> >>> Echo echo ech ech ec ec e e . . >>> >>> :) >>> >>> >>> What's the importance of the impedance matching in a FXO interface ? >> >> >> > My experience is with excessive buzz and hum on the line. W

RE: [Asterisk-Users] Call generator

GIBERT Frédéric wrote: > Hello Adam, > > I'm interested by this solution, but can you please give me more info > because I don't know how to generate calls with asterisk and the > spool directory. How don't know wich files do I need to use. > > Thanks. > Fred Look in your ./asterisk directory,

[Asterisk-Users] Asterisk as a SIP UA and voicemail with SER not working anymore

Hi,    I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine.  I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering t

Re: [Asterisk-Users] Failover of IAX or Spillover as the case may be

On Wed, 23 Jun 2004, Andrew Kohlsmith wrote: > On Wednesday 23 June 2004 06:01, Philipp von Klitzing wrote: > > Secondly you could also use the exit codes of Dial() for failover action, > > but better prevent the necessity for that with the help of SetGroup(). > > IAX2 does not return the disco

[Asterisk-Users] Conference calling

Hello all, I have an ISDN card lspci: 07:06.0 Network controller: Eicon Technology Corporation DIVA Server PRI-30M 2.0 which has 8 channels active. I am wondering if a:, this card is supported/can be made to work with Asterisk, and b:, if it is possible to make Asterisk initiate 2 outgoing v

Re: [Asterisk-Users] FXO impedance matching

Jason A. Pattie wrote: Robert Hajime Lanning wrote: Echo echo ech ech ec ec e e . . :) What's the importance of the impedance matching in a FXO interface ? If impedance matching is that important, then how is it accomplished? I'm fairly sure our X101P is not impedance matching properly. I've n

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