Re: [Asterisk-Users] Teliax IAX problems -- Asterisk doesn't see answer

I think I ran in to this problem a while back as well. I'm also running a CVS version of Asterisk. I talked to David and he switched me to SIP from their gateway to their Asterisk proxy which solved the issue. On Mon, 2005-10-17 at 22:05 -0500, Rob Fugina wrote: > On 10/17/05, Rich Adamson <[EMAIL

RE: [Asterisk-Users] Middle Ground between POTS and T1?

Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of our telcos (DT T-com) we can get PRA in 10 increments: 10B, 20B and 30B >We have a partial T1 (5B + D, iirc) from Allstream - there may be a >provider in your area that does something similar. > >Regards, >-- >Anthony Rodg

Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

- Original Message - From: "Tzafrir Cohen" <[EMAIL PROTECTED]> To: Sent: Monday, October 17, 2005 7:18 PM Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk On Mon, Oct 17, 2005 at 06:56:39PM +0100, Steve Daniels wrote: Try a a good old netstat -a | grep 5038 ne

Re: [Asterisk-Users] Ask for config files of Nortell Meridian Op11 & Asterisk for PRI

Hi,  We have done it a little different...that said though, we have had periods of weird things happening, like digits dropping, (we blame the Nortel though!,  so not the gospel ;  ) : >>zaptata.conf<< [channels]     context=incoming     switchtype=euroisdn     pridialplan=local     signallin

[Asterisk-Users] How to use Use different ports to authenticate SIP/IAX users

Is there a way to config a sip user so that he appears to be connecting from a different IP address? I want to use different IP addresses to authenticate different accounts with service providers rather than the username/password combo. Are there SIP settings to allow that? /Obelix -

Re: [Asterisk-Users] Problem with compiling spandsp

Download the latest app_rxfax.c and app_txfax.c for pre21 (Dated 12 October 2005). For the first week or so pre21 was available the older versions were posted by mistake and caused exactly this compilation error. Craig - Original Message - From: "Administrator" <[EMAIL PROTECTED]> To

RE: [Asterisk-Users] Where can I find Polycom 600 config files?

Thanks all for help. I finally downloaded 1.6.2 version sip and everything seems to be working fine. robert > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Kristian Kielhofner > Sent: Monday, October 17, 2005 11:29 PM > To: Asterisk Users Mailin

Re: [Asterisk-Users] Polycom MWI

I, myself would be happy if you could configure the DND button to send back to the server, shall we say, more configurable options. Instead of just having it option to "indicate busy on DND active", it would be nice to enlist another state, maybe still allow calls through (DND override) if a s

Re: [Asterisk-Users] Can I use ANY port for SIP device?

> I wonder if I can use ANY port for a SIP device? Technically yes, however, it's UDP and not TCP. For SIP there is a control port (usually 5060) and then the RTP port pair so one port isn't sufficient to establish a connection. ___ --Bandwidth and Coloca

Re: [Asterisk-Users] Dial command in extensions

Patrick wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? > Try with "h" (for hangup): exten => 1234,1,Dial... exten => 1234,h,... He actually meant the 'h' exten and not priority: exten => h,1,blah but that wo

Re: [Asterisk-Users] Bizarre Echo Problem

Hi Mark, Citeren Mark Edwards <[EMAIL PROTECTED]>: > to add some fuel to the fire, I was monitoring one of the agents last night. > He made a call to a target and then had to call them straight back to > confirm some information. > > The first call was as echoey as the inside of a cathedral. > Th

Re: [Asterisk-Users] Where can I find Polycom 600 config files?

Robert Augustyn wrote: Yep, I tired these too ... After the reboot it freezes. The phone work ok with sip 1.4.2 though and previous versions of these configuration files. robert Robert, If you are using pre 1.5.2 firmware, you need to use the files in "/asterisk/pcom/", if you are using 1.5.

RE: [Asterisk-Users] Bizarre Echo Problem

> -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Edwards > Sent: Monday, October 17, 2005 4:50 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Bizarre Echo Problem > > Hey Kris, > > to add some fuel to

Re: [Asterisk-Users] Teliax IAX problems -- Asterisk doesn't see answer

On 10/17/05, Rich Adamson <[EMAIL PROTECTED]> wrote: I've been trying to diagnose the same problem with teliax, and it seemsto be a jitterbuffer problem. Since turning it off, we've not had a problem.My guess is that teliax servers are not current code, or they've modified the code for some reason.

Re: [Asterisk-Users] ACD calls to busy agents

JT, yes.    Here's how I've done it before for other clients:    On the dialout portion I've changed the dial plan to:   exten => _1NXXNXX,1,GotoIf($[${LEN(${$[AGENTBYCALLERID_${CALLERIDNUM}]})} > 2]?2:3)exten => _1NXXNXX,2,PauseQueueMember(|Agent/${$[AGENTBYCALLERID_${CALLERIDNUM}]})exten

[Asterisk-Users] Re: GROUP and GROUP_COUNT

On Sun, Oct 16, 2005 at 05:15:14PM -0600, Ryan exclaimed: >I have a macro and when I call it I have something like this: > >exten => s,1,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)}) >exten => s,n,Set(GROUP()=MYGROUP) ;Set Group >exten => s,n,NoOp(Group List: ${GROUP_LIST()}) >exten =>

Fwd: Re: [Asterisk-Users] SIP to SIP sadness

Note: forwarded message attached.--- Begin Message ---  SIP requires RTP connections in addition to the signaling connection which normally happens on UDP 5060. The RTP connections vary in port usage (the range is configurable through rtp.conf) and are nearly impossible to get going without some "m

Re: [Asterisk-Users] Dial command in extensions

Edwin Lam wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? Try "g": exten => 1234,1,dial(SIP/1234,,g) exten => 1234,2, g: When the called party hangs up, exit to execute more commands in the current context. http:

[Asterisk-Users] DID's

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 ca

RE: [Asterisk-Users] Where can I find Polycom 600 config files?

Yep, I tired these too ... After the reboot it freezes. The phone work ok with sip 1.4.2 though and previous versions of these configuration files. robert > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Anthony Rodgers > Sent: Monday, October 17,

[Asterisk-Users] Can I use ANY port for SIP device?

I wonder if I can use ANY port for a SIP device? Could I use port 80 - I know it is http, or 21? bye Ronald Wiplinger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lis

Re: [Asterisk-Users] Can Asterisk "proxy" a SIP phone to make it look like a Cisco skinny softphone?

On Oct 17, 2005, at 6:53 PM, Jason Haar wrote: Tom Rymes wrote: Why don't you connect to Cisco via Chan_sccp and use a soft or hardphone to connect to asterisk. Like this: Cisco<-(chan_sccp)->asterisk<-(SIP)->Your phone Just a thought. Do you mean "Cisco" as the actual p

Re: [Asterisk-Users] Dial command in extensions

On Mon, 2005-10-17 at 17:04 -0700, Edwin Lam wrote: > hi folks. > > is there anyway to make the dial command return and execute > the next line in the dial plan after the channel hangs up? > > suppose i want to do something like this: > > exten => 1234,1,dial(SIP/1234) > exten => 1234,2, > > bu

[Asterisk-Users] Asterisk 1.0.9 - PortaOne Radius

  Hi! I´m confused about setting up PortaOne Radius Client for Asterisk because information is about 1.02, and i don´t know if i need to patch 1.09 or what.   Could somebody help?   Thanks!   Gustavo Hernandez Baratta ___ --Bandwidth and Colocation

[Asterisk-Users] Dial command in extensions

hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten => 1234,1,dial(SIP/1234) exten => 1234,2, but when the dial command hangs up normally, line 2 won't get executed. -

Re: [Asterisk-Users] Bizarre Echo Problem

Hey Kris, to add some fuel to the fire, I was monitoring one of the agents last night. He made a call to a target and then had to call them straight back to confirm some information. The first call was as echoey as the inside of a cathedral. The second (next) call was as clear as a bell. Same nu

[Asterisk-Users] Argentina - Vontel - Asterisk

Any expert that configured Asterisk with Vontel in Argentina? Please give advice on the sip.conf configuration. Thank you!-- Leandro Rzezak[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asteris

Re: [Asterisk-Users] Where can I find Polycom 600 config files?

You should see a list of files and folders - click on the 152 folder, and the sample files should be listed . What are you seeing? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http:

[Asterisk-Users] Uniden UIP200 Issues

Phone won't register on LAN port registers but doesn't work on PC port. SIP to SIP works. Anyone have a Configuration that works out there? Phone has 4.63 Firmware [EMAIL PROTECTED] 2.0 Beta -- Jeff Herring / [EMAIL PROTECTED] Seacoast Laboratory Data Sys

Re: [Asterisk-Users] Problem with incoming calls

I'm sorry, someone very nicely pointed out that I should be using the insecure setting in sip.conf to solve this. I must have missed it in my new config files and figured it was no longer needed. Once I added the appropriate insecure flag, it began taking calls correctly again. Very sorry to have

RE: [Asterisk-Users] Where can I find Polycom 600 config files?

Anthony, If you referring to: http://www.krisk.org/asterisk/pcom/ Then it does not seem to work for me robert > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Anthony Rodgers > Sent: Monday, October 17, 2005 5:57 PM > To: Asterisk Users Mai

RE: [Asterisk-Users] Bizarre Echo Problem

> -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Edwards > Sent: Monday, October 17, 2005 3:28 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Bizarre Echo Problem > > Before I relate the actual problem, so

Re: [Asterisk-Users] huge problem compiling * on gcc4.x (SUSE 10.0)

On Sun, Oct 16, 2005 at 09:21:09PM +0200, [Ludwig IT-Services - GMAIL ] - Michael Ludwig said: > I'm very new to this list and to asterisk and stuff at all. > To build my asterisk server I installed a new machine running the new > SUSE Linux 10.0 (retail version on DVD). > I need asterisk (tried 1

Re: [Asterisk-Users] Can Asterisk "proxy" a SIP phone to make it look like a Cisco skinny softphone?

Tom Rymes wrote: > Why don't you connect to Cisco via Chan_sccp and use a soft or > hardphone to connect to asterisk. Like this: > > Cisco<-(chan_sccp)->asterisk<-(SIP)->Your phone > > Just a thought. > Do you mean "Cisco" as the actual phone - instead of the CallManager? I would

[Asterisk-Users] Bizarre Echo Problem

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initia

RE: [Asterisk-Users] Delayed ringing on some SIP phones

I am glad it helped, I hope the solution will not provide you any troubles with the CDR. Jurczak -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, October 17, 2005 5:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discuss

Re: [Asterisk-Users] Teliax IAX problems -- Asterisk doesn't see answer

I've been trying to diagnose the same problem with teliax, and it seems to be a jitterbuffer problem. Since turning it off, we've not had a problem. My guess is that teliax servers are not current code, or they've modified the code for some reason. The tech that I corresponded with suggested 'gsm

[Asterisk-Users] Teliax IAX problems -- Asterisk doesn't see answer

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time.  As of today, I'm running the latest code from CVS.     -- Called teliax/1314321     -- Call

[Asterisk-Users] Re: Problem with compiling spandsp

Use pre10 or pre11. > Date: Mon, 17 Oct 2005 12:53:43 -0700 > From: "Administrator" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Problem with compiling spandsp > > Actually I am using 0.0.2pre21, also tried pre20finally got a > different error after trying just about everything includi

Re: [Asterisk-Users] Where can I find Polycom 600 config files?

There is a link to excellent samples on the Polycom Phones page on voip-info.org. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 17-Oct-05, at 1:50 PM, Robert Augustyn wrote: Hi, I am trying to

Re: [Asterisk-Users] (no subject)

I am having the same problem, but on both PSTN and a Voicepulse Connect IAX line. PSTN rings clicks dead air, then rings and connects, IAX just clicks, has dead air, rings and connects. Don't have a clue on how to fix it though. Greg Roger Johnsen wrote: I have a Wildcard TDM400P card bei

[Asterisk-Users] (no subject)

I have a Wildcard TDM400P card being used with Asterisk. For some reason, incoming PSTN calls are getting delayed before they ring through on the Asterisk PBX to an extension. The calling party hears an initial ring tone and then a click noise, at which point it will then actually starts to ring

[Asterisk-Users] Where can I find Polycom 600 config files?

Hi, I am trying to configure my 600 phone using ftp. The phone loaded the boot rom 2.6.2 and sip 1.5.2 but I am having problems getting sip.cfg and phone1.cfg configuring Can someone send me examples of these two files, I would very much appreciate that. robert ___

[Asterisk-Users] FW: ISDN PRI and E1

Hi , I have a Siemens HICOM 350H PBX system with 500 lines . I want to buy another PBX system which is IP enabled and tie both of these in order to have a total of 1000 lines . One of the options is through ISDN line and the other through Fiber optic cable. what would be t

Re: [Asterisk-Users] Re: CDMA phone line for Asterisk?

thank you...! what is the best CDMA card or module for asterisk? so i can ask it from our CDMA provider.On 10/18/05, Justin Newman < [EMAIL PROTECTED]> wrote:It may be easier to buy a CDMA card or module made for Asterisk. There are several GSM products and at least one company that has CDMA produ

Re: [Asterisk-Users] ACD calls to busy agents

PauseQueueMember works fine, they just never get unpaused if they hang up the call... BJ Weschke wrote: You must specify to pqm and upqm what interface you're attempting to pause and unpause. It will not figure that out based on what interface the channel that called the app in the dial plan m

[Asterisk-Users] Re: ACD calls to busy agents

> Date: Mon, 17 Oct 2005 14:20:18 -0500 > From: Corey Frang <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] ACD calls to busy agents > > So, I'm looking into using PauseQueueMember and unpause queuemember > > How the heck to you get Unpause to run, no matter what, after the call > is over? > >

RE: [Asterisk-Users] iax invtation problem

-Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de jonny hashem Enviado el: Domingo, 16 de Octubre de 2005 02:44 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] iax invtation problem i had a sip invitation problem with my voip provider and h

Re: [Asterisk-Users] Problem with compiling spandsp

Administrator wrote: Maybe I'm not editing the makefile correctly? I am cutting/pasting from the patchfile so I know it's not a typo. -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Friday, October 14, 2005 6:14 PM To: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] Re: CDMA phone line for Asterisk?

It may be easier to buy a CDMA card or module made for Asterisk. There are several GSM products and at least one company that has CDMA products. Let me know if you need more information. -J > Date: Tue, 18 Oct 2005 01:49:16 +0600 > From: Widyachacra Rajapaksha <[EMAIL PROTECTED]> > Subject: [Ast

Re: [Asterisk-Users] Middle Ground between POTS and T1?

We have a partial T1 (5B + D, iirc) from Allstream - there may be a provider in your area that does something similar. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 17-Oct-05, at 12:48 PM, Matth

Re: [Asterisk-Users] Problem with incoming calls

 Can you open a bug please in bugs.digium.com and attach your sip.conf and then a full SIP debug/trace of a call attempt into the bug?    This will help us reproduce/diagnose the issue.    Thanks.   On 10/17/05, Sherwood McGowan <[EMAIL PROTECTED]> wrote: Gents, this concerns a CVS-HEAD downloade

RE: [Asterisk-Users] Re: Can't compile ast_*fax

You are talking on the app_rxfax.c & app_txfax.c? Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Newman Sent: Monday, October 17, 2005 3:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Can't compile ast_*fax Try add

[Asterisk-Users] Ztdummy is shutting down my sound

Anytime I start Asterisk I get an error about not being able to locate a timing device.  My basic extensions still execute, for the most part.  They include some text to speech with festival and some playbacks.   The performance has been a bit shaky so I thought lack of a timing device might be a c

Re: [Asterisk-Users] Ask for config files of Nortell Meridian Op11 & Asterisk for PRI

Wow - I thought we were the only ones doing this. OK - here goes. Our zaptel.conf looks like this: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #clear=1-24 loadzone = us defaultzone=us Our zapata.conf looks like this: [trunkgroups] [channels] context=incoming switchtype=5ess usecallingpres=yes ec

Re: [Asterisk-Users] ACD calls to busy agents

 It is possible. I do it here and at many other client installs.    Please post your configuration so we can see why it's not working for you.   On 17 Oct 2005 15:31:46 -0400, J Thomas <[EMAIL PROTECTED]> wrote: Given the current state of queues, it does not seem possible to stop ACDcalls coming t

RE: [Asterisk-Users] Problem with compiling spandsp

Actually I am using 0.0.2pre21, also tried pre20finally got a different error after trying just about everything including deleting the source dir and unpacking again, editing makefile again, etc. app_rxfax.c: In function `rxfax_exec': app_rxfax.c:265: error: structure has no member named `log

[Asterisk-Users] Middle Ground between POTS and T1?

I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together?

[Asterisk-Users] CDMA phone line for Asterisk?

Dear friends, im very new to asterisk, even diz z my 1st mail to the list. am working a small company & it has two main CDMA telepone connections. now they wants to deploy a pbx & get out 20 nods(telephone extension lines). so is it possible with Asterisk? & our both CDMA phones are HUAWEI ETS2000

[Asterisk-Users] Problem with incoming calls

Gents, this concerns a CVS-HEAD downloaded today. I configured my system as I usually do, including using allowguest=yes to attempt to correct the following problem, but to no avail. When any call comes in from an external server I get this: Oct 1715:36:43 NOTICE[4040]: chan_sip.c:10774 handle_re

Re: [Asterisk-Users] Multiple calls per phone

yes, thats correct.On 10/17/05, Ariel Batista <[EMAIL PROTECTED]> wrote: Asterisk wrote:> Hello,>> I am new to this list and to Asterisk.  I am using Asterisk @Home, but> have begun to be comfortable editing the scripts.>> I have a Grandstream GXP-2000 with 4 line buttons.  Is there any way I > can

Re: [Asterisk-Users] ACD calls to busy agents

You must specify to pqm and upqm what interface you're attempting to pause and unpause. It will not figure that out based on what interface the channel that called the app in the dial plan might have used. On 10/17/05, Corey Frang <[EMAIL PROTECTED]> wrote: So, I'm looking into using PauseQueueMemb

Re: [Asterisk-Users] SIP to SIP sadness

 SIP requires RTP connections in addition to the signaling connection which normally happens on UDP 5060. The RTP connections vary in port usage (the range is configurable through rtp.conf) and are nearly impossible to get going without some "man in the middle" help when you have two Asterisk serve

Re: [Asterisk-Users] ACD calls to busy agents

Given the current state of queues, it does not seem possible to stop ACD calls coming to a busy agent who has made an outgoing call. Looks like feature request is the right way to go for this. I am going to post this on dev mailing list too before making a feature request in case we have missed so

[Asterisk-Users] Re: Can't compile ast_*fax

Try adding "#define _GNU_SOURCE" before including lock.h and pthread.h. Justin > Date: Mon, 17 Oct 2005 14:29:12 -0400 > From: "Carlos Alperin" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Can't compile ast_*fax > > I tried to compile app_rxfax.c & app_txfax.c, and always fails. > > > > Ev

Re: [Asterisk-Users] SIP to SIP sadness

just 5060/udp and the rtp ports (beginning at 1 and incrementing) should be free and clear. I open 1-10500 for my inter-office phones (and update rtp.conf to match) Michael Furdyk wrote: Okay so it seems like it was the firewall, someone just suggested that we disable it (On Redhat se

RE: [Asterisk-Users] SIP to SIP sadness

Look in rtp.conf. You must have the same udp-ports open as the settings in rtp.conf   Anders   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Furdyk Sent: den 17 oktober 2005 21:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [

Re: [Asterisk-Users] ACD calls to busy agents

So, I'm looking into using PauseQueueMember and unpause queuemember How the heck to you get Unpause to run, no matter what, after the call is over? The "g" argument to Dial only works when the >called< party hangs up. Using the "h" extension appears to be doing nothing... Is there any way we

Re: [Asterisk-Users] How can I get a dialtone calling from outside...

I guess that's pretty true. When I needed to set it up, I lucked out with a google of "asterisk internal dialtone" but can't imagine what else I'd search for to hone in on it. Wilson Pickett wrote: Isn't that what you would call Direct Inward System Access? Probably FWIW, DISA is one of t

Re: [Asterisk-Users] Multiple calls per phone

Asterisk wrote: Hello, I am new to this list and to Asterisk. I am using Asterisk @Home, but have begun to be comfortable editing the scripts. I have a Grandstream GXP-2000 with 4 line buttons. Is there any way I can set Asterisk to send more than one call to the phone without setting up mult

Re: [Asterisk-Users] Multiple calls per phone

turn on callwaiting.  It will then ring one of the other lines automagically.On 10/17/05, Asterisk <[EMAIL PROTECTED] > wrote:Hello,I am new to this list and to Asterisk.  I am using Asterisk @Home, but have begun to be comfortable editing the scripts.I have a Grandstream GXP-2000 with 4 line butto

Re: [Asterisk-Users] Asterisk and Fedora

I have asterisk 1.20beta1 running under Fedora Core 4, and the only problem that I have had, was I had to remove my GigE because it was causing one heck of an echo problem.> Here is a link to get you going:> http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3> > Rudolf> > - Ori

RE: [Asterisk-Users] SIP to SIP sadness

Okay so it seems like it was the firewall, someone just suggested that we disable it (On Redhat server) and it's working fine... so does anyone know clearly what ports (other than 5060) SIP uses for these calls?   -- Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Micha

Re: [Asterisk-Users] How can I get a dialtone calling from outside...

Wilson Pickett wrote: Isn't that what you would call Direct Inward System Access? Probably FWIW, DISA is one of those concepts it is not easy to find without already knowing the name, and if you know the name, you prolly know what it is, etc. The snake that bites its own tail. The only reaso

[Asterisk-Users] Multiple calls per phone

Hello, I am new to this list and to Asterisk. I am using Asterisk @Home, but have begun to be comfortable editing the scripts. I have a Grandstream GXP-2000 with 4 line buttons. Is there any way I can set Asterisk to send more than one call to the phone without setting up multiple accounts

[Asterisk-Users] SIP to SIP sadness

Wow, after getting the O'Reilly book delivered last week along with two Digium TDM400P's, I'm really getting the hang of this. But the SIP to SIP issue is still a problem... and it seems silly because everything else (should have been?) so much harder but is working pretty flawlessly. Basica

Re: [Asterisk-Users] How can I get a dialtone calling from outside...

> Isn't that what you would call Direct Inward System Access? Probably FWIW, DISA is one of those concepts it is not easy to find without already knowing the name, and if you know the name, you prolly know what it is, etc. The snake that bites its own tail. The only reason I discovered it was be

RE: [Asterisk-Users] initiate call recording from phone.

Thanks everyone for the help so far. I figured it out using automon, howver my next question is this: Is there a way to make the recording go to the voicemail directory for the user that records it. This way, they can access it from their phone. -- H. Andy Goss Network Engineer Network Advocat

Re: [Asterisk-Users] Polycom MWI

> se.pat.misc.1.inst.1.type="silence" se.pat.misc.1.inst.1.value="1" > se.pat.misc.1.inst.2.type="silence" se.pat.misc.1.inst.2.value="2" > se.pat.misc.1.inst.3.type="silence" se.pat.misc.1.inst.3.value="1"/> > > I didn't bother taking out the unnecessary stuff, I just changed where > it said chor

Re: [Asterisk-Users] How can I get a dialtone calling from outside...

Exactly what I need! Thanks a million. Francois > Isn't that what you would call Direct Inward System Access? Probably > what you'd find at > http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DISA > > > > Francois Meehan wrote: >> Hi all, >> >> How can I configure, in extension.conf, to

[Asterisk-Users] can't compile ast_*fax

This is all the reference to PTHREAD_MUTEX_RECURSIVE on lock.h   #ifdef __APPLE__ /* Provide the Linux initializers for MacOS X */ #define PTHREAD_MUTEX_RECURSIVE_NP  PTHREAD_MUTEX_RECURSIVE #define PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP   { 0x4d555458, \  

RE: [Asterisk-Users] Can't compile ast_*fax

I already asked this and I get no answer, but may be is the same issue that others already have on this app.   I tried to compile app_rxfax.c & app_txfax.c, and always fails.   Even if I put this on a different directory and I do gcc –pipe app_rxfax.c I get the same   /usr/include/a

Re: [Asterisk-Users] How can I get a dialtone calling from outside...

Isn't that what you would call Direct Inward System Access? Probably what you'd find at http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DISA Francois Meehan wrote: Hi all, How can I configure, in extension.conf, to call and extension and have a dialtone so I can compose a number

Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

On Mon, Oct 17, 2005 at 06:56:39PM +0100, Steve Daniels wrote: > Try a a good old > netstat -a | grep 5038 netstat -lntup | grep 5038 -l is generally preferable than -a: only gives listening ports. -t: tcp sockets only. -p: also pid and name of the process that has the socket. > That will tell y

[Asterisk-Users] How can I get a dialtone calling from outside...

Hi all, How can I configure, in extension.conf, to call and extension and have a dialtone so I can compose a number to dialout? Basically, I want to be able, when I am out of the office, to call in my asterisk box and then dialout from it. Regards, Francois _

Re: [Asterisk-Users] Re: Modifying Voicemail App & Record App

Thank you for your comments. Yup, I kept running into limitations in the dialplan. There are some things agi apps can do, but not everything. I have also just hit the wall with the Record() app, it might be better for me to start in app_record.c for some of the things I want to do, it's much sm

Re: [Asterisk-Users] initiate call recording from phone.

And the w or W options must be specified in the Dial() cmd, as in: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial Moj Mojo with Horan & Company, LLC wrote: If you have httpd with php on the * server, you can do what I did: I set up the key combination *# in features.conf to mon

Re: [Asterisk-Users] initiate call recording from phone.

If you have httpd with php on the * server, you can do what I did: I set up the key combination *# in features.conf to monitor and created a few php files to interact with the results. Save the four php files at: http://horanappraisals.com/asterisk/ into a folder on the * web server, eg: /va

Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

Try a a good old netstat -a | grep 5038 That will tell you if * is listening and what it's listening on. Then if it show's * is listening, it must be a permit =, or a firewall issue. HTH Steve - Original Message - From: "Chuck Bunn" <[EMAIL PROTECTED]> To: "Asterisk - Users" Sent: M

Re: [Asterisk-Users] initiate call recording from phone.

On Mon, Oct 17, 2005 at 01:27:59PM -0400, Andy Goss exclaimed: >I am looking for a way to allow a user to record a call simply by >pressing a button or some combination of buttons on their phone. Is >this possible? > >I have read the stuff about the monitor command; however, this doesn't >seem to

[Asterisk-Users] initiate call recording from phone.

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Ne

Re: [Asterisk-Users] Callerid on t1 lines

How are you checking if CallerID is received? You should do at least a Noop(${CALLERIDNUM}) or if running head: Noop(${CALLERID(NUM)}) so that you can verify that. How do you know that your telco is giving you CID? If you live in the US then setup the Adit to do LSCPD and Asteisk as ks_fxs. and not

[Asterisk-Users] Interface with ability to originate call

Hi all, Is there an interface like Flash Operator Panel which allows to transfer or to originate calls from Outlook  contact database. I would also make transfer directly to voicemail and to transfer callers to music on hold.   Thanks Amaury   ___

[Asterisk-Users] Ruby module for the Asterisk Manager Interface

I have just released the first version of Rami, a ruby module for the Asterisk Manager Interface.  It includes a client library and proxy server for sending multiple simultaneous requests with just one open connection to asterisk. One of the unique features is that the proxy server stores interna

[Asterisk-Users] cmd SIPRedirect for loadbalancing

Hi folks, I've just been reading about the above command and wonder if anyone has made use of it for load-balancing or if doing so would be completely inappropriate!? I'm thinking of the scenario where there are a number of Asterisk gateways and incoming SIP traffic. From what I've read, wit

Re: [Asterisk-Users] Double Ringing for PRI Calls

Matt wrote: Yes, Go into sip.conf and add this line: progressinband=no Thank you!!! My Cisco 7960's started acting weird with SIP version 7.5, so I kept them at 7.4 for this reason. Works great now! Mark ___ --Bandwidth and Colocation sponsor

RE: [Asterisk-Users] Calibrating both RX and TX gain?

[EMAIL PROTECTED] wrote on 10/17/2005 12:45:13 PM: > > > Here's a couple of ways to determine levels... > > > > > > 1. using the model 4 transmission test set, attach the tone generator > > > to one analog pstn line and the transmission level test jacks to a > > > second pstn line. Dial from one

[Asterisk-Users] astcc missing to bill random calls?

Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of "hangup". I suppose that astcc.agi was not able

[Asterisk-Users] Connecting TIE trunk to Astericks

We have a Meridian Option 81C with a TIE trunk for our long distance. Anyone have any ideas/information on setting up this trunk for Astericks? Route: TYPE RDBCUST 00ROUT 1DES  SBC Long Distance TKTP TIENPID_TBL_NUM   0ESN  NOCNVT NOSAT  NORCLS EXTDTRK YESDGTP DTIISDN NODSEL 3VCEPTYP DTT  _

RE: [Asterisk-Users] Calibrating both RX and TX gain?

> > Here's a couple of ways to determine levels... > > > > 1. using the model 4 transmission test set, attach the tone generator > > to one analog pstn line and the transmission level test jacks to a > > second pstn line. Dial from one line to other and measure the tone. > > Divide by two, and the

  1   2   >