[Asterisk-Users] Cameron Palmer / voiceglo

I found a message in the archives from Cameron Palmer on 23 Dec regarding his voiceglo SIP configuration. Unfortunately (for me), the archive has his email address removed. So, Cameron -- or anybody else using voiceglo with their * box -- please reply to me so that I can get your email address and

Re: [Asterisk-Users] Internal Lines Dialing Out

Oops! I forgot the leading underscore. Use this version below. Steve. [always-out-pots] ;as generic as possible to allow all calls out other than local extensions which loads first above exten =>_ NXX,1,Dial(Zap/1/$EXTEN) exten => _NXX,2,Goto(102) exten => _NXX,102,Congestion e

Re: [Asterisk-Users] Internal Lines Dialing Out

Try replacing these lines: > [always-out-pots] > > ;as generic as possible to allow all calls out other than local extensions > which loads first above > exten => _.,1,Dial(Zap/1/$EXTEN) > exten => _.,2,Goto(102) > exten => _.,102,Congestion > exten => _.,103,Hangup with these: [always-out-pots

RE: [Asterisk-Users] Re: MeetMe Video option

Few pieces of info that might help us from re-inventing the wheel. What we are all taking about has a name in video conferencing circles. It is called an MCU, Multi Conferencing Unit. The OpenH323 project has an MCU called OpenMCU. Since the H323 support in * is based on OpenH323 perhaps it would b

[Asterisk-Users] Call Queues

I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent "agent 1001 hang up on customers. they must be pissed off". I agreed. My queues

Re: [Asterisk-Users] How to delay dialing

On Friday 30 January 2004 20:05, Terence Parker wrote: > > 1) The 'W' character is only for the zaptel channel. > > 2) It's case insensitive (i.e. it does NOT need to be uppercase). > > See line 2387 of zaptel.c if you'd like to confirm this for > > yourself. 3) There is no current way within As

Re: [Asterisk-Users] Max messages in VoiceMailMain

On Friday 30 January 2004 23:18, Derek wrote: > On Jan 30, 2004, at 11:24 PM, Tilghman Lesher wrote: > > On Friday 30 January 2004 22:14, Derek wrote: > >> We just switched the company over to * today and have run up > >> against a problem I didn't foresee. Comedian voicemail has a limit > >> of 99

Re: [Asterisk-Users] Max messages in VoiceMailMain

Thanks Tilghman, I was actually recompiling with that set to 999 as I read this:) Maybe the default setting should be higher than 100 in the source code in case someone else runs into this? I don't think it would hurt. Or better yet add a setting in voicemail.conf? -D On Jan 30, 2004, at 11:2

Re: [Asterisk-Users] Introducing Firefly

- Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> >Hi, > >I downloaded this the other day and finally got it to stop crashing. It appears that any response from asterisk >that implies an error (for example dialing a non-existant number, using the wrong password, selecting a cod

[Asterisk-Users] Internal Lines Dialing Out

* Gurus, I have been trying, with mixed results, to setup an * server as a pbx in my home. Internal dialing works great, sip phone to sip phone and 1 fxs phone to sip phones, as well as inward dialing ringing all extensions then going to vmail. All great. But, when I try to dial out I run into is

Re: [Asterisk-Users] Max messages in VoiceMailMain

On Friday 30 January 2004 22:14, Derek wrote: > We just switched the company over to * today and have run up against > a problem I didn't foresee. Comedian voicemail has a limit of 99 > voice-mails per mailbox, even though the naming scheme would allow > for 10,000. Our company accepts orders by vo

[Asterisk-Users] Max messages in VoiceMailMain

We just switched the company over to * today and have run up against a problem I didn't foresee. Comedian voicemail has a limit of 99 voice-mails per mailbox, even though the naming scheme would allow for 10,000. Our company accepts orders by voicemail during the night with a usual night consis

Re: [Asterisk-Users] Words for Allison(?)

At 5:51 PM -0600 1/30/04, Rob Fugina wrote: I've been looking at the weather vocabulary in asterisk-sounds in CVS. I've run into a few hitches with words I can't seem to find. So far, I'm looking for 'point' (for constructing floating point numbers) and 'around' as in "high around 70" (don't I wis

[Asterisk-Users] Question on setting up asterisk with hunting lines

*My apologies if this message is posted 3 times, I was trying to sent it to the list once before I am a list-member, the second time before I was approved. Can anyone point me to some resources on using hunting lines with Asterisk? Sales support of my telco have no idea what I am trying to do. T

[Asterisk-Users] Voicemail not receiving password or audio

Hi,   I'm using * with the latest CVS image as of this afternoon on Fedora1. I am using snom 200's with the latest firmware.   When I dial the voicemail extension and try to enter my password, * does not receive the password. The console complains about not receiving the password. Also, when

[Asterisk-Users] recorder

I just got Asterisk Developer's Kit (TDM) from digium. and thanks to the great support at digium i have got it working in no time, so now i want more;). what configuration file can i change so all calls made from the analog phone pluged in the tdm400 port #1 are recorded in mp3or any other format

Re: [Asterisk-Users] IAX1 vs IAX2 for IAXtel

G'day Tilghman, On Fri, 30 Jan 2004, Tilghman Lesher wrote: > bash# touch /etc/asterisk/iax1.conf > bash# asterisk -rx reload Thanks, this worked! Thanks also to Mark and Philipp; I think I might give that a try also (as it is probably not sensible to load a module that does nothing :-) Cheer

Re: [Asterisk-Users] How do you turn on the 7960 msg waiting light?

I can only speak for the SIP IOS load on the 7960's (We're running 6.1 ) but if you add: [EMAIL PROTECTED] It "should" work Note: 7188 being the mail box number and "ContextInVoicemailConf " being the context in the "voicemail.conf" file where the mail box 7188 exists. Example: [7188] type=fri

Re: [Asterisk-Users] How to delay dialing

Hi Tilghman, Thanks for your reply - really appreciated! I have tried applying your patch, but unfortunately get the following compilation errors: chan_vpb.c: In function `int vpb_call(ast_channel*, char*, int)': chan_vpb.c:683: syntax error before `char' chan_vpb.c:696: `t' undeclared (first us

[Asterisk-Users] Re: DISA and authcodes (was: t410p)

[moved from -dev, as the thread is better suited for -users] At 5:10 PM -0600 1/30/04, James Sharp wrote: > I've pretty much got the routing covered at this point, I'm just not sure how to get the Asterisk system to answer and give me dialtone immediately. Any ideas or recommendations would be

RE: [Asterisk-Users] Call quality questions

Thanks... -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Fri 1/30/2004 5:16 PM To: '[EMAIL PROTECTED]' Cc: Subject: RE: [Asterisk-Users] Call quality questions Hello, Did you se

Re: [Asterisk-Users] test

OMG I want to say If I find the person that has the mail server Never mind its fixed now. But please check your mail filters to make sure they bounce mail properly. bkw On Fri, 30 Jan 2004, Brian West wrote: > test > ___ > Asterisk-Users mailing l

Re: [Asterisk-Users] Firefly and asterisk*

- Original Message - From: "FastJack" <[EMAIL PROTECTED]> > GREAT!!! Just got my asterisk* calling firefly users. Setup was really easy: > > Anyone knows how to receive calls on my asterisk*-box from the > firefly-network? > I'll fix this soon, then you should be able to connect to fire

RE: [Asterisk-Users] Re: Adtran 750 DID question.

Yes. immediate=no is in zapata.conf before the channel declaration. This makes absolutely no sense at all. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Don Pobanz Sent: Friday, January 30, 2004 5:11 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asteris

[Asterisk-Users] test

test ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] determining legal VoIP service

Actually I believe this is one of the few things that can be done without worrying about the state(s) PUC coming down on your head. Since your users are in another country the state PUC cannot consider you providing a telephone service in their jurisdiction. - Dustin - Walker Haddock wrote: C

Re: [Asterisk-Users] Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c

On Fri, 2004-01-30 at 17:17, Brancaleoni Matteo wrote: > Hi > > > > No need for the profanities...(I'm not offended) > > > > This is a FINE mailing list. Works just the same. > you're right. I've been too rude > sometimes I get mad about questions answered only > few posts before I wasn't tr

Re: [Asterisk-Users] Hangup

Eduardo Goncalves wrote: Hi list, I'm with a little problem on my E1 (E&M signaling) link. Every call a make hangs up after 2 or 3 seconds of conversation. I got the fowling messages from cli: : Zap/1-1 answered SIP/atapd-238e Urgent handler Urgent handler -- Hungup 'Zap/1-1' Urgent handler

[Asterisk-Users] Voicemail/Playback Questions

We are using a SCSI based IBM eServer x300 for our PBX. In setting this unit up, we used a backup machine, which was IDE only. The problem that we are currently experiencing is that the voicemail prompts are coming out the system so fast that the words overlap each other, and sometimes are unin

Re: [Asterisk-Users] Re: Grandstream Firmware ?

Do both the budgetone and the handytone use the same firmware? On Fri, 2004-01-30 at 06:26, Stephen R. Besch wrote: > Greg Boehnlein wrote: > > > On Thu, 29 Jan 2004, Michael Welter wrote: > > > > > >>I have 1.0.4.45 (beta) on my tftp server. Try it at 66.250.23.58. > >> > >>Cheers, > >>Micha

Re: [Asterisk-Users] Compiling while * is running

On 30 Jan 2004, Joe Phillips wrote: > On Fri, 2004-01-30 at 14:26, David Gomillion wrote: > > Rob Fugina wrote: > > > > > Seg faulting compiles usually indicate a memory problem on the > > > machine. Not lack of size, but bad memory, badly seated memory, > > > etc... There's no reason asterisk ru

[Asterisk-Users] Words for Allison(?)

I've been looking at the weather vocabulary in asterisk-sounds in CVS. I've run into a few hitches with words I can't seem to find. So far, I'm looking for 'point' (for constructing floating point numbers) and 'around' as in "high around 70" (don't I wish). Any chance of getting these? While I'm

Re: [Asterisk-Users] Adtran 750 DID question.

Shouldn't they be FXO cards for CO lines? Bisker, Scott (7805) wrote: Yes. Adtran FXS cards. Did you say you were using Adtran FXS cards? Bisker, Scott (7805) wrote: Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now he

RE: [Asterisk-Users] Re: Asterisk Manager Interface notes

Hello, I was referring to the availability of the "ExtensionState" Action (see the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API ), even though I don't actually use it. For my purposes of status of an extension I wrote an updater script, that runs outside of Astrisk

Re: [Asterisk-Users] Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c

Hi > > No need for the profanities...(I'm not offended) > > This is a FINE mailing list. Works just the same. you're right. I've been too rude sometimes I get mad about questions answered only few posts before matteo -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl ___

[Asterisk-Users] Re: Asterisk Manager Interface notes

Hi to all, specially to Matf. When you say you can "- Get the status of an extension" , can you expand a little? I've been batling with it without getting anything usefull whatever the parameters I put in. Julio Anjos Portugal ___ Asterisk-Users mail

[Asterisk-Users] determining legal VoIP service

Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a car

RE: [Asterisk-Users] Re: MeetMe Video option

Tim - I'm actually quite fond of the 2b solution in the video conference tools I've used (notably, Polycomm) where the video switches or camera pans depending on audio energy. This could work quite well with the existing features of "m" and "t". A combination of blending an audio-energy an

Re: [Asterisk-Users] How do you turn on the 7960 msg waiting light?

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:[EMAIL PROTECTED] Paul - Most people want to know how to turn it OFF. :-) The quick answer is

RE: [Asterisk-Users] Auto dial in Off Hook situation.

Thanks John, Found it. The Multitech's are part of a legacy system used by a new customer of mine. I just latched onto it for ease of communications, it's been in for some years now. Regards Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

RE: [Asterisk-Users] How do you turn on the 7960 msg waiting light?

Title: Message If you have it configured to use * and the voicemail is configured, it should just light up.     -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul MahlerSent: Friday, January 30, 2004 3:34 PMTo: [EMAIL PROTECTED]Subje

Re: [Asterisk-Users] Extension Questions

> ;All US Calls > > exten => > _9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) > > > ;Dial 9 for outgoing numbers > > exten =>_9.,1,Dial(Zap/g1/${EXTEN:1}) > > > > ;include Brunswick > > switch => IAX2/dornoch:[EMAIL PROTECTED]/sip > Try this: [us-out] _

RE: [Asterisk-Users] Call quality questions

Hello, Did you set the flag in the makefile for zaptel for SMP kernels? 1. I have a couple Snom200 phones on my system running redhat with a P4 HT and haven't had any issues with horrible sound quality using 711ulaw. 2. As for the speakerphone cutout, that's to be expected, The snom200s are just

RE: [Asterisk-Users] Re: Adtran 750 DID question.

On Friday, January 30, 2004 3:56 PM, Bisker, Scott (7805) [SMTP:[EMAIL PROTECTED] wrote: > I guess asterisk is winking properly then, because the line rings when > dialed. In zaptel.conf the lines are set to e&m and in zapata.conf > they are set to e&m_w. The FXS cards are series L1. > > What

[Asterisk-Users] Re: MeetMe Video option

No, there is no video output once the call goes into a meetme room. What I was talking about is a case where you have a regular video call between 2 video phones, then you try to send them to a conference room. The audio still works but the softphone's (Linphone in our case) behavior is to just

Re: [Asterisk-Users] Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c

On Fri, 2004-01-30 at 15:52, Brancaleoni Matteo wrote: > hi > > > > What is in the file IAX1.conf, why I don't have this > > file? Why I get warning on chan_ian2.c? How can I > > solve these problems? > Sorry if I'm rude, but RTFML (read the f*** mailing > list)... this has been widely discussed

[Asterisk-Users] Pictures of new multiport FXO/FXS from digum

Hello, I have heard that digium (mark) is going to introduce a new multiport FXO/FXS card next month. Does anyone have any details or some pictures of the card. The latest I know about the card is that it will have 16 ports and I wonder how these 16 ports will be connected to a PCI interface usin

RE: [Asterisk-Users] Re: Adtran 750 DID question.

I guess asterisk is winking properly then, because the line rings when dialed. In zaptel.conf the lines are set to e&m and in zapata.conf they are set to e&m_w. The FXS cards are series L1. What I'm seeing is that the DNIS info is not being passed through to asterisk. Since I get no DNIS,

Re: [Asterisk-Users] Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c

hi > > What is in the file IAX1.conf, why I don't have this > file? Why I get warning on chan_ian2.c? How can I > solve these problems? Sorry if I'm rude, but RTFML (read the f*** mailing list)... this has been widely discussed recently. then... first: don't crosspost to the -dev mailing list...

RE: [Asterisk-Users] Compiling zaptel

Dear sb I am running redhat 7.3 upgraded to this version of kernel. Yes, I have installed the kernel-sources but not the kernel-util rpms, I don't think I ever did install kernel-util with the original installation of redhat 7.3 or did I? But I was having no problem installing zap at that time tho

[Asterisk-Users] Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c

Hi, all Please help me. My platform is RedHat Linux 9.0. I have a wildcard x100p. I just installed asterisk by following step: # cd ../zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples When I test Asterisk

[Asterisk-Users] Call quality questions

Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc

[Asterisk-Users] How do you turn on the 7960 msg waiting light?

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting?   Thanks!   Paul     Paul Mahler mail:[EMAIL PROTECTED]  

[Asterisk-Users] SNOM 200 question

Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragme

RE: [Asterisk-Users] Compiling zaptel

On Fri, 2004-01-30 at 15:01, T. Chan wrote: > Dear all, > > I have been testing with Asterisk for a bit of time and yesterday I tried to > upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not > able to compile Zaptel. The kernel runs good and everything intact, I was > trying

RE: [Asterisk-Users] Compiling zaptel

I take it you are running RedHat 8 (or 9) since this is the most up-to-date kernel. Did you install the kernel-sources and kernel-util rpms as well? You'll need these in order to compile and install zaptel. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Beh

[Asterisk-Users] Re: Adtran 750 DID question.

I posted this below message few days ago, but some how it didn't show up in the mailing list. When some one calls into your DID Trunk line what symptom do you see on Adtran as well as on Asterisks console? Asterisk winks properly in FXO DPO mode, we had checked this things with our Telco instrum

Re: [Asterisk-Users] Re: Compiling while * is running

On Fri, 2004-01-30 at 14:33, Stephen R. Besch wrote: > Yes, that's what I had thought also, providing that you add in paging > file errors. And yet the machine in question is, and has been, running * > flawlessly for over a month now - and continues to do so during and > after the compile. Nev

RE: [Asterisk-Users] Re: MeetMe Video option

So you are actually getting the video to come out though? I am not getting any outbound video RTP traffic at all. What settings do you have? If I get a chance this weekend I will take a look at the implementation and see what I can see. The mosaic thing should be pretty easy actually (really, jus

RE: [Asterisk-Users] Auto dial in Off Hook situation.

James I would have to change several other units over from proprietary to h323 that are already in the loop. I added mine to the loop so they could call for support. I have started to play with h323 on the * but not got my head round it yet. Regards Dave -Original Message- From: [E

RE: [Asterisk-Users] Compiling zaptel

Dear all, I have been testing with Asterisk for a bit of time and yesterday I tried to upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not able to compile Zaptel. The kernel runs good and everything intact, I was trying to recompile Asterisk in order to make sure that every

Re: [Asterisk-Users] Compiling while * is running

> Yes, Nortel Meridian's can get 5 9's easily. They are very expensive, > but we have one running at a government site in Indiana that has been up > for 15 years without interruption. When you upgrade the 1 control unit, > the other 1 is servicing all the requests. There is a brief period of > t

Re: [Asterisk-Users] Compiling while * is running

> Anyway, thanks for bringing my bad math to my attention. So, here's the > question: has anyone worked on a phone system that DID have 5 9's? I'm > not talking about core services that AT&T Long Lines owns, I mean > customer-premises equipment. Is that an unrealistic goal? I've never seen a ph

RE: [Asterisk-Users] Compiling while * is running

Yes, Nortel Meridian's can get 5 9's easily.  They are very expensive, but we have one running at a government site in Indiana that has been up for 15 years without interruption.  When you upgrade the 1 control unit, the other 1 is servicing all the requests.  There is a brief period of time whe

[Asterisk-Users] Re: Compiling while * is running

Steven Critchfield wrote: You apparently still have quite a bit more to learn. If you read the first line quoted, you will see that it is the compiling that is a problem. At no time during compile is the application you are compiling actually executed. Only gcc and it's helpers should be executed

Re: [Asterisk-Users] Compiling while * is running

> This is a reason I argue for binary packages in production > environments. You can build the packages (eg. debs or RPMs) on a > development machine at your leisure and install the binary in minutes on > the production machine. If your packages use proper dependencies you > can also be much more

[Asterisk-Users] Re: Compiling while * is running

David Gomillion wrote: I don't agree. When first learning to program, my programs segfaulted all of the time, regarless of what machine I was on. Often, it was doing something stupid, like trying to replace a file that was in use, etc. I knew that was a problem in windoze, I did not think it wa

[Asterisk-Users] Hangup

Hi list, I'm with a little problem on my E1 (E&M signaling) link. Every call a make hangs up after 2 or 3 seconds of conversation. I got the fowling messages from cli: : Zap/1-1 answered SIP/atapd-238e Urgent handler Urgent handler -- Hungup 'Zap/1-1' Urgent handler Jan 30 18:46:17 W

[Asterisk-Users] Re: Compiling while * is running

Rob Fugina wrote: On Fri, Jan 30, 2004 at 12:21:49PM -0500, Stephen R. Besch wrote: I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the

Re: [Asterisk-Users] Compiling while * is running

Steven Critchfield wrote: > On Fri, 2004-01-30 at 13:26, David Gomillion wrote: >> Rob Fugina wrote: >> [snip] Is there a way to safely compile while * is running, so that I can minimize down time of the server? >>> >>> Seg faulting compiles usually indicate a memory problem on the >>> ma

RE: [Asterisk-Users] mediatrix, dtmf

Hi, Thanks to the lag, I have sorted this out myself. Out of band singalling, codec change fixed it. Regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik Sent: Friday, January 30, 2004 2:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk

Re: [Asterisk-Users] HANGUPCAUSE

It would to be good in any way... :) - Original Message - From: "Tais M. Hansen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 30, 2004 12:57 PM Subject: Re: [Asterisk-Users] HANGUPCAUSE -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 30 January 2004 15:59,

Re: [Asterisk-Users] Extension Questions

On Fri, 2004-01-30 at 14:00, Shad Mortazavi wrote: > Dear all, > > > > I have the following lines in my extentions.conf file; > > > ;All US Calls > > exten => > _9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) > > > ;Dial 9 for outgoing numbers >

[Asterisk-Users] Re: MeetMe Video option

That's one of the things that's been on our (1control, I have nothing to do with Digium) wishlist/"to do" list that just hasn't gotten done yet. Currently, video in meetme is not supported. What we experience is the audio will conference with the other audio streams but the video just freezes.

[Asterisk-Users] Extension Questions

Dear all,   I have the following lines in my extentions.conf file;   ;All US Calls exten => _9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])   ;Dial 9 for outgoing numbers exten =>_9.,1,Dial(Zap/g1/${EXTEN:1})   ;include Brunswick switch =>

[Asterisk-Users] X-Lite, X100P, and Speex

Title: X-Lite, X100P, and Speex I'm having a problem with using X-Lite to initiate a call via Asterisk out an X100P analog port, using the Speex codec.  I've put in the registry fix for X-Lite and Speex so that works OK, and calling the echo test extension works.  However, if I call out the an

[Asterisk-Users] error on IAX1.conf and warning on chan_iax2.c

Hi, I have a wildcard x100p. I just installed asterisk by following step: # cd ../zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples When I test Asterisk typing # asterisk –c I find one error and one wa

Re: [Asterisk-Users] Compiling while * is running

On Fri, 2004-01-30 at 13:26, David Gomillion wrote: > Rob Fugina wrote: > [snip] > >> Is there a way to safely compile while * is running, so that I can > >> minimize down time of the server? > > > > Seg faulting compiles usually indicate a memory problem on the > > machine. Not lack of size, but b

Re: [Asterisk-Users] Compiling while * is running

On Fri, 2004-01-30 at 14:26, David Gomillion wrote: > Rob Fugina wrote: > > > Seg faulting compiles usually indicate a memory problem on the > > machine. Not lack of size, but bad memory, badly seated memory, > > etc... There's no reason asterisk running, or the drivers being > > loaded, should >

RE: [Asterisk-Users] G729 license

I purchased a license from Digium, If you ask they will can also give you a trial license to test out.   Wes -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Jess MagnayeSent: Friday, January 30, 2004 10:29 AMTo: [EMAIL PROTECTED]Subject: [

Re: [Asterisk-Users] Compiling while * is running

Rob Fugina wrote: [snip] >> Is there a way to safely compile while * is running, so that I can >> minimize down time of the server? > > Seg faulting compiles usually indicate a memory problem on the > machine. Not lack of size, but bad memory, badly seated memory, > etc... There's no reason asteri

Re: [Asterisk-Users] IAX call problems

Hi Rattana, Do you have jitterbuffer enabled? Dan On Fri, 2004-01-30 at 13:40, Rattana BIV wrote: > hi, > > I use IAX softphone with asterisk and I notice that a call between two > IAX softphones end after 1 min. Then I can't hear anything but the > call still in progress. > I have this log in

RE: [Asterisk-Users] Adtran 750 DID question.

Yes. Adtran FXS cards. Did you say you were using Adtran FXS cards? Bisker, Scott (7805) wrote: > Hello All, > > I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't > configured properly. Now heres the next question. 12 E&M wink lines from telco. I > hav

RE: [Asterisk-Users] Auto dial in Off Hook situation.

Yes, it is that simple, but of course there is a precursor requirement that you need to 1) be able to configure your ATA, and 2) know how to configure your ATA. Google is your friend. Please use Google before you reply back with additional questions; it saves us all time and email bandwidth.

RE: [Asterisk-Users] MeetMe Video option

I was wondering if it was supported, and how. It seems to me that video conferencing is a different beast than audio conferencing because you cannot simply mix video like you can mix audio. The conferencing server would have to 1) "mix" the video by creating one aggregate outbound paneled type w

Re: [Asterisk-Users] Adtran 750 DID question.

Did you say you were using Adtran FXS cards? Bisker, Scott (7805) wrote: Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 w

RE: [Asterisk-Users] Auto dial in Off Hook situation.

> Thanks John, > > > I think it is not that simple. I am not using a phone but a Cisco ATA. > > The scenario: - > > User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100 > (FXO))--Cisco ATA--Asterisk--Any extension Any reason you can't use the H.323 load for the MVP200? I've not t

[Asterisk-Users] Newbridge Mainstreet 3624

I've got a Newbridge CB hanging on the wall not being used right now and I'd like to hear opinions on using it with Asterisk. If anyone has a manual for it I'd like to get a copy of it. I tried the googling approach but turned up nothing much except a Tech manual if I want to change out control boa

RE: [Asterisk-Users] MeetMe Video option

I would also be interested in this in regards to working with D-Link videophones. They use the same setup as netmeeting h.263, but with another rfc add on. I know current OpenH323 configs do not quite work with it, but I saw a post that it is in cvs working using a patch to ffmpeg. -- Jonathan M

Re: [Asterisk-Users] MeetMe Video option

Citeren "Regovich, Timothy" <[EMAIL PROTECTED]>: > Has anyone configured a meetme conference to use video? > I have successfully used video phones to talk through *, but I cannot seem > to get video when those phones dial into a meetme conference. Cool, what devices are you using ? Would love to

RE: [Asterisk-Users] MeetMe Video option

I have written my own. Java(JMF) based. It is pretty rudimentary, but does handle audio (gsm, ulaw) and video (jpeg and H263). Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Friday, January 30, 2004 1:30 PM To: [EMAIL PROTECTED] Subjec

RE: [Asterisk-Users] Auto dial in Off Hook situation.

Thanks John, I think it is not that simple. I am not using a phone but a Cisco ATA. The scenario: - User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100 (FXO))--Cisco ATA--Asterisk--Any extension The Multitech MVP100 used to connect to my old analogue switch which was set to a

Re: [Asterisk-Users] MeetMe Video option

Regovich, Timothy wrote: Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. What video phone did you use? __

Re: [Asterisk-Users] How to delay dialing

On Fri, Jan 30, 2004 at 10:58:15AM -0600, Eric Wieling wrote: > Are you sure this works for VoiceTronix Driver? It's not implemented in > app_dial, but in chan_zap. > I've only used it with a zap device. Sorry I didn't think this through. > On Fri, 2004-01-30 at 10:15, Walker Haddock wrote: >

RE: [Asterisk-Users] Adtran 750 DID question.

I tried both featd and em in zapata.conf, to no avail. I restarted in between all changes. Is it possible to signal the DPO ports on the 750 with fxo_ls or fxo_ks? This is the last piece to my DID puzzle. Anyone else with experience on this oddball config? Thanks, -sb -Original Message

Re: [Asterisk-Users] Compiling while * is running

On Fri, Jan 30, 2004 at 12:21:49PM -0500, Stephen R. Besch wrote: > I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my > test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my > production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the > zaptel and

[Asterisk-Users] MeetMe Video option

Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the "v" flag on my extension f

RE: [Asterisk-Users] List traffic

Michael, I have the same thing -- 1 to 4 posts a day ! regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Graves Sent: Thursday, January 29, 2004 12:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] List traffic All of a sudden my

Re: [Asterisk-Users] HANGUPCAUSE

See Bug Number 890 on bugs.digium.com. --Eric > From: "Tais M. Hansen" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, January 30, 2004 9:20 AM > Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1 > > > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > On Friday 30 January 200

Re: [Asterisk-Users] has Allison said this ?

Lance Arbuckle wrote: > Does anyone know if Allison has recorded anything along the lines of: > > "You don't have permission to dial that number." I think so... under tt-monkeys.gsm. > > Thanks. You're welcome. PS. Sorry, I couldn't resist on this one. __

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