Re: [Asterisk-Users] SPA-3000 and incoming faxes

If you are willing to dedicate a fax line and forward faxes to a dedicated extension it work 100% for Incoming and Outgoing faxes with Sipura-3000 Though, you have to change in the Regional Tab: Ring Waveform: from Sinusoid to Trapezoid I've been using NVbackgroundDetect with Sipura-3000 and

[Asterisk-Users] Bad FCS nightmare to Nortel SL100 with TE410P

I have an * box connected to a Nortel SL100 through a PRI (US) using the Digium TE410P (quad-span T1 card). I don't have access to the SL100 - it is handled by another group. The span comes up OK (timing, framing fine). However, as soon as the D channel comes up, I get endless HDLC Bad FCS

[Asterisk-Users] Non-blocking Dial (and other commands): is there a way?

In order to use a with GrandStream BT-488 as pass-through gateway, I need a way of sending the FXO port off hook when I'm using the FXS port for VoIP communications, because I want to use the hunting line feature to let incoming call skip that FXO port and move on to the next free line. The only

RE: [Asterisk-Users] IBM x306 - some progress

Hi Marco, As far as I can recall, the IBM setup utility can enable you to change the IRQ of the SCSI controller. In addition, I've never seen a WildCard board bound to IRQ7 on any box, which is very weird in it self. I'm flying over to Ireland today (actually, at the airport right now), and

Re: [Asterisk-Users] Grandstream 496 not working on cordless phone

On Mon, 2005-09-26 at 17:24 -0400, Nana Tandoh wrote: Hi All, We are using SER/Asterisk, it works fine from X-lite to corded phones but have problems using a cordless phone on the Handytone 496. Has anyone experienced this problem. Well, if you told us what the problems are perhaps we

Re: [Asterisk-Users] What ISDN hardware would you recommend?

On Mon, 26 Sep 2005, Francesco Peeters wrote: Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I

Re: [Asterisk-Users] sip, call ransfer and call waiting

trixter http://www.0xdecafbad.com a écrit : On Mon, 2005-09-26 at 11:08 +0200, Daniel ANDRE wrote: Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on

Re: R: [Asterisk-Users] Problem setting up TDM22B card

Hi, I did dmesg | tail it says ... dmesg | tail f6 != 58 f7 != 59 f8 != 58 f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test This is small part of dmesg output which may help in diagnosing the problem. Registered Tormenta2 PCI No ISA tormenta card found at

Re: [Asterisk-Users] Re: Problem setting up TDM22B card

Hi, I know some bits about zaptel.conf and zapata.conf but problem is modules wctdm, wcfxo, wcfxs are not getting loaded. modprobe zaptel is successful! Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 11:50:06PM -0700, somesh s wrote: Hi,

Re: [Asterisk-Users] Re: Problem setting up TDM22B card

Hi, And I have the digium's hardware too in one of my PCI slots. Regards, Somesh S. Shanbhag --- somesh s [EMAIL PROTECTED] wrote: Hi, I know some bits about zaptel.conf and zapata.conf but problem is modules wctdm, wcfxo, wcfxs are not getting loaded. modprobe zaptel is

[Asterisk-Users] Termcap missing (compile error [editline/libedit.a] Error 1)

Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. The first error message happens by using the famous script from http://www.szmidt.org/asterisk/asterisk-update.sh : configure: error: termcap support not found

RE: [Asterisk-Users] Removing - (Dash) from Dialed Numbers

[EMAIL PROTECTED] wrote: I am trying to enable dial-by-email by using LDAPget to query an Active Directory server. I've got it retrieving the phone number fine. Unforunately, the numbers stored in active directory are either in the format: (xxx) xxx- or xxx-xxx-.

Re: [Asterisk-Users] Termcap missing (compile error [editline/libedit.a] Error 1)

On Tue, Sep 27, 2005 at 09:20:13AM +0200, [EMAIL PROTECTED] wrote: Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. Start with defining a standard deb-src of Sarge (I think it is defined by default. Maybe

RE: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)

You must install libncurses5-dev regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Enviado el: martes, 27 de septiembre de 2005 9:20 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Termcap missing (compile

[Asterisk-Users] pbx_wilcalu.so: undefined symbol:

Anyone run into this? This is from the latest 1.2.0 beta1 tarball. Got it all compiled, but this undefined symbol is stopping asterisk from loading. Can I savely bypass this module and if so, what does it actually do? Cheers ___ --Bandwidth and

Re: [Asterisk-Users] Early Media in 180 Ringing

Hi :) Am Montag, 26. September 2005 19:48 schrieb Ronald Voermans: Hello, As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? Well, I am not that expert but AFAIK your PSTN

[Asterisk-Users] Integration with NMS AG-E1/T1

I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want

RE: [Asterisk-Users] Initial release of AMPortal Debian/Xorcom-Rapidpackages

Tzafrir, I got these error messages installing AMP on your distribution rapid 1.1. Versions of Apache and PHP are the ones that come inside the package and nothing new has beeen added. Any idea about what went wrong? Regards, Jose M. Limeres /etc/apt apt-get install amportal Reading Package

RE: [Asterisk-Users] Early Media in 180 Ringing

If guess I figured it out already. I made some changes in chan_sip.c (when ringing was received, it didn't check for SDP), and recompiled. It's working now! Ronald - -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens

[Asterisk-Users] failed make install on Solaris 10

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p

Re: R: [Asterisk-Users] Problem setting up TDM22B card

Hi, I didn't get any solution in the mailing list. [http://asterisk.linkx.net/asteriskusers/200409/msg01167] What should be the next step? Changing the machine??? Is it machine dependent?... Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Sep 27, 2005 at

[Asterisk-Users] Listening for DTMF when dialling

Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message (Press 1 to leave a message) and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as Thanks, Peter Spikings This message has been

[Asterisk-Users] Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!)

Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message (Press 1 to leave a message) and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as the dial command doesn't have appropriate options and takes

RE: [Asterisk-Users] WRT54GP2 SIP server on LAN port

Some more information if that might give anyone some ide what can be wrong. WRTG54GP2 settings under Line1 that are set to something: User ID: 100 Authentication Password: xxx Registration / Proxy Server: 192.168.15.10 NAT Traversal: None Under the router ip/Voice_adminPage.htm secret page and

Re: [Asterisk-Users] I got 403, Forbidden... please help

Hi Harry, I tried your suggestion and it worked. But I don't hear any voice from the anonymous user. I don't hear the voice prompt? What should I do? Thanks, Ryan harry gaillac wrote: Hello, Try insecure=very in [sip.philonline.com] Harry --- Ryan Pagquil [EMAIL PROTECTED] a écrit

[Asterisk-Users] radius and *

any one know where to get a radius module to work with the * sip server so SIP auth and Call accountingcan also bedone by radius? thanks! Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

RE: [Asterisk-Users] Best drivers for HFC-S ISDN cards

I said: I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). Many thanks to those who replied. General consensus seems to be switching to mISDN or CAPI won't solve the intermittent echo

[Asterisk-Users] Turn off echo-cancellation when fax is detected?

How can I do this? Ive set faxdetect=both in zapata.conf. Does this cancel echo-cancellation (echo-training) when a fax is detected or is this just for using exten=fax, in extensions.conf.? Im having trouble getting spanDSP - RxFax to recieve faxes. I am using Asterisk 1.0.8 and

[Asterisk-Users] * Accounting with Oracle

Hello all, I use the asterisk with a oracle db in th ebackend. I want to use the db for accounting also. I saw that AMP has a mysql table with the accounting datas. Isit possible to por this to oracle or does anybody has a accounting agi or whatever which uses oracle? Regards Rene

R: [Asterisk-Users] Best drivers for HFC-S ISDN cards

Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with

Re: [Asterisk-Users] Teliax

Does anyone have any experience with Teliax for inbound IAX? Been working fine for me for over six months with multiple did's over iax. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

RE: [Asterisk-Users] Sangoma and Digium same machine?

I'm using a Sangoma A101 card alongside an older TDM400 and they seem to be playing nice. I've had it in production for a few months now with no problems. Thanks, Reid Forrest, CISSP Max-IS Inc. [EMAIL PROTECTED] Direct/Cell: 321-214- Main: 407-786-9600 -Original Message- From:

R: [Asterisk-Users] Best drivers for HFC-S ISDN cards

Well done Tim...could u post here your Zapata.conf ? :) I'm in Italy and have some issues with echo Thanks Giordano Grandis [EMAIL PROTECTED] -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Tim Robinson Inviato: lunedì 26 settembre 2005 22.30 A:

Re: [Asterisk-Users] iax problem

On Mon, Sep 26, 2005 at 11:02:47AM -0600, Rich Adamson wrote: For #2, incoming calls would be handled with: exten = 6789,1,Dial(SIP/1235) Besides that : *CLI iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569

[Asterisk-Users] wait before accepting the call

hello! i'm looking for a way to prolonge a pstn-call for 5 seconds before it enters the extensions.conf. this is for testing purposes, all numbers of a ddi should be received by asterisk before the call is walking through the extensions. how can i achive this? i've not seen a feature like this

Re: [Asterisk-Users] Integration with NMS AG-E1/T1

What is the line protocol you're using on this legacy PBX? Is it EM Wink? If so, then you'd just configure the Digium card for wink, plug in a T1 crossover cable and you should be ready to start testing. On 9/27/05, Exciting [EMAIL PROTECTED] wrote: I want to replace a custom PBX, that is infront

Re: [Asterisk-Users] Teliax

Jason Schafer wrote: Does anyone have any experience with Teliax for inbound IAX? Yes, have many accounts. Very good service and support. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED]

Re: [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server

I still find out how to let LCS 2005 accept SIP invite from Asterisk, Need more help. Hi jacky, can you please share your experience and explain how to let LCS accept SIP invite from Asterisk. I deseperate trying to place a call from asterisk to LCS. (calling from Asterisk to LCS using

[Asterisk-Users] Re: Dialtone problems with phpagi and asterisk

Does nobody know a solution or an approach to a solution? Michael Michael Häberle wrote: Hi there In our php-application we use phpagi to communicate with asterisk (as the voip-client we use x-pro) Sometimes it occurs that the dialtone is very choppy or not present. If we dial directly in

Re: [Asterisk-Users] Integration with NMS AG-E1/T1

I don't kwnow if between NMS and PBX there's a EM Wink protocol, do you have some info to retreive it from ag.cfg (NMS board config file) ? The PBX is connected to an E1,(Italy) (ITU-T G.703 E1), and also to 2 AG-E1 machines, the pbx route incoming calls to an proprietary ivr on these machines.

[Asterisk-Users] IAX2 hard phone

I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the

Re: [Asterisk-Users] Integration with NMS AG-E1/T1

What do your NetworkInterface.T1E1[X..X] and TCPFiles[X] lines in ag.cfg look like? Yes. I've worked with the AG series boards from NMS before. On 9/27/05, Exciting [EMAIL PROTECTED] wrote: I don't kwnow if between NMS and PBX there's a EM Wink protocol, do you have some info to retreive it from

Re: [Asterisk-Users] Teliax

Yes, and I posted the information on the Wiki. Regards, Chris - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 27, 2005 7:15 AM Subject: Re:

[Asterisk-Users] Unable to create channel of type 'Zap'

I installed Asterisk 1.0 CVS on a Debian Sarge System. I am using two ISDN-HFC-Cards and a point-to-point ISDN Connection. Everything seemed to work pefectly. But today I realized that I cannot use two lines at the same time. I get the error message: 3 active channel(s) asterisk*CLI show

Re: [Asterisk-Users] AsteriskJava - Queue

Thanks for your response. AFAIK I can redirect, bridge, drop and answer a call but I can't find the way to do, for example: - Get the call back from the queue, play a message and put it again in the queue. and - Get a linked call (caller to Agent), unlink it (releasing the agent) and play

[Asterisk-Users] Moaning dog...

Here's one for you phone people An elderly lady phoned her telephone company to report that her telephone failed to ring when her friends called - and that on the few occasions when it did ring, her pet dog always moaned right before the phone rang. The telephone repairman proceeded to the

Re: [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk

Without information about your dialplan and what the phpagi script does there is not much anyone can do. I do not know of any known issues that may account for the problem you are having. Update with further information and maybe someone will be able to provide some insight. --johann

RE: [Asterisk-Users] IBM x306 - some progress

I dug up my Netfinity ServeRaid readme: Power on your system and observe the screen. Press F1 when the Press F1 for Configuration/Setup and Press F2 for Diagnostics messages appear. The Configuration/Setup Utility main menu will appear. Select Advanced Setup using the Up or Down arrow key and

Re: [Asterisk-Users] iax problem

For #2, incoming calls would be handled with: exten = 6789,1,Dial(SIP/1235) Besides that : *CLI iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569 Username1 [MYIP]:456960 Registered

RE: [Asterisk-Users] IAX2 hard phone

Alberto, PA168 chip does not have Hold and Transfer features on it until firmware version 1.44. Atcom never claimed that these will work as the Pa168 firmware is still under development. Yesterday I met Peter Sun, President and owner of Atcom China, in New York. He is here toattend VON

Re: [Asterisk-Users] pbx_wilcalu.so: undefined symbol:

Pikoro wrote: Anyone run into this? This is from the latest 1.2.0 beta1 tarball. Got it all compiled, but this undefined symbol is stopping asterisk from loading. When you change major versions, before you install you should: rm -rf /usr/lib/asterisk/modules/* I also rm -rf

Re: [Asterisk-Users] Early Media in 180 Ringing

Ronald Voermans wrote: If guess I figured it out already. I made some changes in chan_sip.c (when ringing was received, it didn't check for SDP), and recompiled. I don't know what all of this means, but I'm sure it could be of value to others. Can you submit your patch to bugs.digium.com?

[Asterisk-Users] function LEN missing

I'm running asterisk 1.2b1 and all seems to be workingright in general. I load modules explicitly in modules.conf, and since my upgrade ast 1.09 I have only one problem: The LEN function (length of string). What module do I need to load to get this string handling function? Thanks MD

[Asterisk-Users] 405 Method Not Allowed error

Hi everybody, I'm curious to know what this message generally indicates. I have XLite softphones on two different machines accessing the Asterisk server behind a NAT. The server is able to find them just fine, but instead of registering when Asterisk reloads they return this message back to

[Asterisk-Users] Software only Asterisk PBX (commercial)

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing

RE: [Asterisk-Users] FW: channel offhook state

We had the same thing until we started using Voicetronix, it seems that this happens when calls collide i.e... incoming call with an outgoing? We added a script that did a soft hang-up after a call was ended and that seemed to work ok. -Original Message- From: [EMAIL

[Asterisk-Users] Asterisk European Digital CAS Help

Someone can give me more info about Asterisk European Digital CAS , I need to make talk asterisk with a AG-E1 card with this protocol. (TCP=euc0.tcp); Is built in supported or i need some patch ? Regards _ Get free infected, boring,

Re: [Asterisk-Users] Re: Dialtone problems with phpagi and asterisk

Ok :) the dialplan looks like that (mynumber is a tel-number): - [general] static=yes writeprotect=no [telout] exten = _X.,hint,SIP/41 exten = _X.,1,dial(SIP/${EXTEN}) exten = _X.,2,SetCIDName(anonymous) exten = _X.,3,dial(SIP/[EMAIL

RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Have a look at the AMP project http://sourceforge.net/projects/amportal

RE : [Asterisk-Users] IAX2 hard phone

Hello Alberto, You must upgrade the firmware by taking the last one at www.aredfox.com which is the PA168 manufacturer. Mine Ip-phones are running well with IAX2 and flash hook for transferts. Good luck. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL

[Asterisk-Users] VoIP Buster stopped working?

Hi, I was successfully using VoIP Buster via IAX2 for several weeks now. Yesterday/today it spontaneously stopped working. Using the real client the connection works well though. Anybody else experiencing this problem? Or asked differently: Is there anybody for whom it is still working? Can

[Asterisk-Users] How to change ${VM_DATE} in voicemail.conf

Hi all, I don't find where you can setup the date (${VM_DATE}) in french for the mail. Is anybody can help me? Amaury BOSSÉ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Extension availabilty

http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extensions http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20BristuffDevstate Can anyone say for certain what asterisk version introduced the hint priority? ___ --Bandwidth

[Asterisk-Users] analogue phone with asterisk

I am a newbee to asterisk. I recently installed [EMAIL PROTECTED]. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do

Re: [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server

Hi, Richard, I still try, but fail with rtp transfer. 2005/9/27, richard Coco [EMAIL PROTECTED]: I still find out how to let LCS 2005 accept SIP invite from Asterisk, Need more help. Hi jacky, can you please share your experience and explain how to let LCS accept SIP invite from

Re: [Asterisk-Users] failed make install on Solaris 10

I am sure you might have tried adding the current directory to the PATH variable. I never compiled asterisk on solaris, but it seems to be working for my other applications. regards, rajesh - Original Message - From: Joseph Rothstein [EMAIL PROTECTED] To: asterisk-users@lists.digium.com

[Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R

Hi I have looked around but I cant find an answer for this, I randomly get the error 'TDM PCI Master abort' and the system locks up. All I have found so far are a couple other posts on it but no solution. Running fedora core 3, asterisk stable, zaptel stable. Any help will be appreciated.

Re: R: [Asterisk-Users] Problem setting up TDM22B card

[EMAIL PROTECTED] wrote on 09/27/2005 03:13:21 AM: Hi, I did dmesg | tail it says ... dmesg | tail f6 != 58 f7 != 59 f8 != 58 f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test The only time I've seen this it has been on a PCI 2.1 computer.

RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

Also check out http://www.bicom.us pretty expensive but if that's your thing :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald Hartmann Sent: 27 September 2005 16:47 To: 'Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] Polycom IP 500 - problem dialing extra numbers

hi thereI'm setting up [EMAIL PROTECTED] and I'm using Polycom IP 500 phones.When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits.  When I try it with X-Lite things appear to work fine,

[Asterisk-Users] One-way audio with VPN

I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:[EMAIL PROTECTED]) B is connected to the Asterisk server via VPN B is

RE : [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)

Many thanks Tzafrir and Sergio, Now, I have another error when compiling zaptel : /lib/modules/2.6.8-2-686/build make -C /lib/modules/2.6.8-2-686/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernel-headers-2.6.8-2-686' CC [M] /usr/src/zaptel/zaptel.o In file

[Asterisk-Users] blindxfer atxfer not working?

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked Enter the number of packages, followed by the Pound key. I found on the wiki

Re: [Asterisk-Users] One-way audio with VPN

On Tue, September 27, 2005 20:22, Alex Lake said: I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:[EMAIL PROTECTED]) B

RE: [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers

Change your dtmf setting. Covered lots of times before, or info on voip-info.com Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jachin Rupe Sent: Tuesday, 27 September 2005 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [Asterisk-Users] analogue phone with asterisk

You need an Analog Terminal Adapter (ATA). Sipura makes some good ones. Check out http://www.voipsupply.com/product_info.php?cPath=96_118products_id=321 http://www.voipsupply.com/product_info.php?cPath=96_118products_id=713 That's what I use, and I love 'em. /edg --On Tuesday, September 27,

Re: [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers

Jachin Rupe wrote: hi there I'm setting up [EMAIL PROTECTED] and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. dial 1 or such and such, dial 2 for this and that...) the Polycom doesn't seem to send the extra digits. When I try it with X-Lite

Re: [Asterisk-Users] analogue phone with asterisk

On Tue, September 27, 2005 19:10, Rajesh Bhairampally said: I am a newbee to asterisk. I recently installed [EMAIL PROTECTED] Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk

RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

Don't you ever recommend Bicom as they take your money and will never deliver a product that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Morgan Gilroy Sent: Tuesday, September 27, 2005 1:20 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R

[EMAIL PROTECTED] wrote on 09/27/2005 01:18:35 PM: Hi I have looked around but I cant find an answer for this, I randomly get the error 'TDM PCI Master abort' and the system locks up. All I have found so far are a couple other posts on it but no solution. Running fedora core 3, asterisk

[Asterisk-Users] [EMAIL PROTECTED] inbound call problem to SIP trunk. (voipfone UK)

Hi all, I have recently installed [EMAIL PROTECTED] and outbound calling is working great. However I am strugglings with inbound calls. I have setup a trunk for my provider, voipfone and in the inbound area on AMP I have the following :- user context name = 3011 context=from-pstn

Re: [Asterisk-Users] FSX/UK analogue Phone rings all the time

Best thing is to get a 'Master' or PBX Master socket, cut one end off an RJ11-RJ11 lead, and connect the red/green pair (centre two pins or the RJ11) to pins 2 and 5 of a master socket. Rgds Tim John Crowhurst wrote: On Mon, September 26, 2005 20:35, Asterisk said: hi Asterisk users, I

[Asterisk-Users] SIP Tandem Inbound only.

Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this

Re: [Asterisk-Users] What ISDN hardware would you recommend?

Quad or octo-bri from www.junghanns.net We use a few of these and they are not cheap but they work without any hassle. Rgds Tim Robinson Francesco Peeters wrote: The machines themselves will not pose much of a problem, but what ISDN hardware would you recommend for this? (1 site

Re: [Asterisk-Users] SIP Tandem Inbound only.

On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I

Re: R: [Asterisk-Users] Best drivers for HFC-S ISDN cards

Hi As requested here are my configs. I have 3 zaphfc cards - 2 in NT mode and 1 in TE mode connected to the BT network. I have a variety of phones - a Cisco 7940, a Snom 190, a Grandsteam Budgie plus 2 cordless ISDN phones on one of the NT ports, and a Network Alchemy Cybergear Gold on the

Re: [Asterisk-Users] blindxfer atxfer not working?

double-check your usage of the t and T parameters to the Dial command, detailed here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial Mojo hugolivude wrote: I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to

Re: [Asterisk-Users] Best drivers for HFC-S ISDN cards

Chris Looking at this file below you need to move the stuff below the channel = 4-5 line to each span definition. Anything below the channel line gets completely ignored! I stand to be corrected but I think your current config will not have any echo cancellation at all. I have just posted

Re: [Asterisk-Users] Polycom Setup Questions

So while I'm waiting to see if anyone can help with those questions, I thought I would ask one more :-) All of the sudden 3 of my Polycom501 handsets started having a 1 way audio problem. My setup: 30 Polycom501 handsets all connected to Asterisk (CVS HEAD from a week or two ago) over a

[Asterisk-Users] Review: Digium TE405P v2

Hello, We have finished our tests of the new Digium firmware on the quad T1 cards(TE405P/TE410P). Overall it is a big improvement over the version 1 firmware. Here's the review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html MATT---

Re: [Asterisk-Users] SIP Tandem Inbound only.

On Tuesday 27 September 2005 3:12 pm, Peter Bowyer wrote: On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have

[Asterisk-Users] cgi-bin/vmail.cgi - - Invalid Context

Greetings: I have been playing around with vmail.cgi and am able to log into and listen to my message with no problem. I added the correct context to vmail.cgi so I don't have to enter the mailbox + context. However, when I try and delete a message or move to a different mailbox I get the

RE: [Asterisk-Users] SIP Tandem Inbound only.

Hi Scott, To do what you want to do you do indeed need to use a peer entry, with the IP address where INVITEs will come from specified as the host, and insecure=very. Your OPTIONS though is being caused by qualify being turned on somewhere. Joshua Colp -Original Message- From: [EMAIL

Re: [Asterisk-Users] Polycom Setup Questions

Matthew T. O'Connor wrote: The 3 phones in question were working morning yesterday, then for no apparent reason, the user could no longer talk. The polycom user could hear the person at the other end, but could not talk to them. Nothing has changed as far as I can tell, and I have no idea

Re: [Asterisk-Users] Polycom Setup Questions

At 14:40 9/27/2005, Matthew T. O'Connor, wrote: So while I'm waiting to see if anyone can help with those questions, I thought I would ask one more :-) All of the sudden 3 of my Polycom501 handsets started having a 1 way audio problem. Did you make certain canreinvite equals no?

Re: [Asterisk-Users] Software only Asterisk PBX (commercial)

I would take a look at Signate, too. Tom On Sep 27, 2005, at 11:12 AM, Matthew Crocker wrote: Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their

[Asterisk-Users] AstriCon 2005 - Now With Free Beer!

AstriCon Update: Only Two Weeks To Go! October 12 - 14, 2005 Anaheim, CA AstriCon 2005 starts two weeks from today. We now have a complete roster of speakers covering Asterisk from soho to carrier. We've added the Code Zone, a working lab with a full compliment of VoIP and TDM equipment. We

RE: [Asterisk-Users] wait before accepting the call

Why don't you write a couple of lines AGI scripts that will call asterisk command WAIT(5) Thankx -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 27 Sep 2005 13:42:31 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wait before accepting the call hello!

Re: Subject: [Asterisk-Users] Vonage-type service

On Monday 26 September 2005 10:41, Federico Alves wrote: We don't sell the system. We provide a full independent system for customers including co-location, for a setup fee and 1/2 cent per call, regardless of length. We also provide US termination via our own DS3 for 1.3 cents a minute, and

RE: [Asterisk-Users] Polycom Setup Questions

I had a loose headset cable doing that one day -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Tuesday, September 27, 2005 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom

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