[Asterisk-Users] 410P upgrade to 411P?

Does anyone know if the echo cancellation module can be retro-fitted to a 410P to turn it into a 411P? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650

Re: [Asterisk-Users] 410P upgrade to 411P?

Yes, it can. I just had one of my old TE405Pv1 cards upgraded to a TE406P(same process as TE410P to TE411P upgrade). The cost is quoted at $895US. You do need to send it to Digium though, not sure if they have a partner in AUS that is able to do upgrades or not. Just contact digium and request a

[Asterisk-Users] (no subject)

Hi, I m trying to install [EMAIL PROTECTED] after installing and logging in as root password i made network connections using netconfig command there i gave ip address as provided by my network provider it displays the ip address I m SORRY to ask that how can i access the net GUI if u can

Re: [Asterisk-Users] Not can call to PSTN

On Wed, Sep 07, 2005 at 10:38:20PM -0500, [EMAIL PROTECTED] wrote: Hello I have installed asterisk with a card X100P. receives calls but when doing the call to the PSTN. says that there are circuits no available... I have given to many returns but profit not to make work it I need that they

[Asterisk-Users] (OT) Dialplan Standards for Business/Offices

Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved fora specificuse ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are

[Asterisk-Users] Transfer calls from cellphone

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up

Re: [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?

Brian Capouch wrote: Olle E. Johansson wrote: Try setting _ALERT_INFO The reason for this is that if you set *any* variable with one underscore prefixing the name, that variable will be copied to the new channel created by dial() - without the underscore. If you create a variable called

[Asterisk-Users] Setting up multiple trunk groups with different internal ring groups

Hi, I have 4 analogue PSTN lines on my legacy PBX, 2 lines on one number in a rollover group ZAP1 ZAP2and 2 lines on another number ZAP3 ZAP4. Is it possible to have a group of phones ring when lines ZAP1 2 are called and a DIFFERENT set of extensions ring when ZAP3 or ZAP4 receive a

[Asterisk-Users] How to increase delay before incoming call answer with tdm400p

Is there a way of increasing the delay before asterisk picks up the incoming PSTN call? I'm using a tdm400p with fxo card. It seems to pick up the inbound call immediately. I want to delay detecting the call by about 10 secs if poss. Done some searching but couldn't find anything relevant.

Re: [Asterisk-Users] Motherboard and processor recommendations

Thanks Tzafrir and canuck15 for your comments. Yes I don't think the NIC will be saturated, and I'll search the quality of the Onboard RAID. I guess I have to learn more about canuck15's comments though, because I am actually questioning what happens to the board when you're adding onboard

Re: [Asterisk-Users] How to increase delay before incoming call answer with tdm400p

taf taffey wrote: Is there a way of increasing the delay before asterisk picks up the incoming PSTN call? I'm using a tdm400p with fxo card. It seems to pick up the inbound call immediately. I want to delay detecting the call by about 10 secs if poss. Done some searching but couldn't

Re: [Asterisk-Users] Motherboard and processor recommendations

On Thu, 2005-09-08 at 12:01 +0300, Soner Tari wrote: Thanks Tzafrir and canuck15 for your comments. Yes I don't think the NIC will be saturated, and I'll search the quality of the Onboard RAID. I guess I have to learn more about canuck15's comments though, because I am actually questioning

[Asterisk-Users] who use astlinux with booting from DOM?

asterisk-users who use astlinux with booting from DOM? how to do ?thanks oncemore [EMAIL PROTECTED] 2005-09-08 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Contexts are not being created - Asterisk BT100 Password Issue

Hello, I think I might have an inkling as to where the issue may be at. For some reason when I create a new context, a directory is not created in /var/spool/asterisk/voicemail. The default context resides there but new ones are not created. Has anyone ever experienced this or does

Re: [Asterisk-Users] How to increase delay before incoming call answer with tdm400p

If I use the wait command won't this intercept the inbound call? I want the call to stay on the pstn line for 10 seconds before asterisk detects the inbound call. Taff. --- Olle E. Johansson [EMAIL PROTECTED] wrote: taf taffey wrote: Is there a way of increasing the delay before asterisk

[Asterisk-Users] 2 X100P and SIP inbound routing

Sorry about my previous post, outbound routing this was clearly available in AMP. However, How would I go about making all calls from PSTN line 1 (X100P #1) ring call group #1, PSTN line 2 (X100P #2) call group #2, and the SIP incoming calls route to group #1 also. Is this done via dial plans?

[Asterisk-Users] pri gateway

hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What

[Asterisk-Users] Extension a

Hi, I would like to use the * when I am in the asnwer machine, but I received a message asking for the temporary pass code. Where I need to put this pass? I am using asterisk 1.2.0 beta 1 ___ --Bandwidth and Colocation sponsored by Easynews.com --

Re: [Asterisk-Users] How to increase delay before incoming call answer with tdm400p

On Thursday 08 September 2005 05:38, taf taffey wrote: If I use the wait command won't this intercept the inbound call? Why not try it and see? I want the call to stay on the pstn line for 10 seconds before asterisk detects the inbound call. You can't prevent Asterisk from detecting the

Re: [Asterisk-Users] How to increase delay before incoming call answer with tdm400p

Will try it later when i get a chance.. Cheers for the input.. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 08 September 2005 05:38, taf taffey wrote: If I use the wait command won't this intercept the inbound call? Why not try it and see? I want the call to stay on

[Asterisk-Users] sending fax....i'm in trouble !

hi all, i've this problem with app_txfax. Here's the log of the error: Sep 8 13:28:55 VERBOSE[10750]: -- Attempting call on Zap/g1/2430 for application txfax(/var/tmp/ast_fax-1126178934.10240.1804289383.0|caller) (Retry 1) Sep 8 13:28:55 DEBUG[10750]: Using channel 3 Sep 8 13:28:55

Re: [Asterisk-Users] pri gateway

what about a copy of your zapata.conf and zaptel.conf,what color is the leds On Thu, 2005-09-08 at 12:42 +0300, Baris Simsek wrote: hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts

[Asterisk-Users] Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?

Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client.

Re: [Asterisk-Users] pri gateway

hi, my asterisk version is 1.0.9 /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 it is comfortable with Turkish Telecom. i tried before and it works. /etc/asterisk/zapata.conf [channels] switchtype=euroisdn signalling=pri_cpe context=incoming group=1

RE: [Asterisk-Users] Want to use a remotely location POTS phone

Just one line. Do you think you could point me to the SPA3K. A google search doesn't yield any results. Is that a discontinued product? Would I not need something on the other end where the POTS phone line is located? Thanks! -- Patrick Campbell

[Asterisk-Users] Sip clients through proxy

Hi i know that we can use sip clients through nat, like the same way can we use sip clients through a proxy,. is there any sip client that i can specify a proxy address and use or any sip device. regards Kanishka ___ --Bandwidth and Colocation

[Asterisk-Users] Yuxin hardphones feedback

Hello everybody This question has probably already been asked, but I'd like to have feedbacks about Yuxin hardphones Especially series 10, 100 and 200 ( and by the way i didnt found too much technical differences between those models). Is it better than budgettone, or so cheap hardphones ? I

Re: [Asterisk-Users] OT: Differences between test equipment

Given the current discusison regarding ztmonitor, line testing, etc., I've been looking into purchasing a used transmission test set. From my research, it seems that there are two items that might fit the bill: the HP 3551A and the HP 4935A. I know nothing about these specific

[Asterisk-Users] Immediate response dial

Would like to be able to do the following, which is typical of an internal setup: On handset pickup get some kind of internal dial tone (probably user defined in indications.conf) Pressing 9, for dialing an external line, to immediately switch to the PSTN dial tone. continue with dialing

RE: [Asterisk-Users] Want to use a remotely location POTS phone

Just one line. Do you think you could point me to the SPA3K. A google search doesn't yield any results. Is that a discontinued product? Would I not need something on the other end where the POTS phone line is located? Thanks! spa3k is really an spa3000 (k = 000). Try:

Re: [Asterisk-Users] Immediate response dial

On Thu, 2005-09-08 at 13:54 +0300, Dias Badekas wrote: Would like to be able to do the following, which is typical of an internal setup: 1. On handset pickup get some kind of internal dial tone (probably user defined in indications.conf) 2. Pressing 9, for dialing an

[Asterisk-Users] Hangup problem

i have a box running debian sarge with asterisk installed from distribution packages: CLI show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64 running Linux I have managed to configure a simple dialplan (the PBX task is quite simple as this is a small

Re: [Asterisk-Users]

I've been messing with it for a couple weeks with MySQL. It seems pretty good to me though I have had a couple crashes. I cane' say for sure that the crashes were directly related to RealTime though. Also, I'm still using CVS HEAD 2005-09-06 which was right before thebeta release, I think. On

Re: [Asterisk-Users] Motherboard and processor recommendations

Not that it's very widely used, but I thought it worth mentioning, if you intend to use TDMoE with multiple Asterisk servers locally your ethernet will be fairly well saturated and you will want a second NIC connected to a separate isolated network for your TDMoE trunks. MATT---On 9/8/05, Dave

Re: [Asterisk-Users] Hangup problem

On Thu, Sep 08, 2005 at 02:56:28PM +0200, Marek Zachara wrote: i have a box running debian sarge with asterisk installed from distribution packages: CLI show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64 running Linux I have managed to configure a

Re: [Asterisk-Users] Insert Subject Here

Flobi wrote: I've been messing with it for a couple weeks with MySQL. It seems pretty good to me though I have had a couple crashes. I cane' say for sure that the crashes were directly related to RealTime though. Also, I'm still using CVS HEAD 2005-09-06 which was right before the beta

[Asterisk-Users] cvs head and seqno 102 (Critical Response) messages for Cisco 7960

I've being using the same non-nat config for a 7960 for about a year with no issues. I have just upgraded cvs head from the 16th of July to todays. About a minute after I make a call I get the message chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for

Re: [Asterisk-Users] How to increase delay before incoming call answer with tdm400p

If you're talking about asterisk actually doing anything with the call on a logical basis (i.e. processing your dialplan), wait will halt that. Actual detection (as recorded in CDR) and acknowledment (via the lower level SIP/IAX/etc to the requester)begins when the call is received and the wait

Re: [Asterisk-Users] Transfer calls from cellphone

b is possible. See res_features.conf for more information on transferring via DTMF. c is not yet possible. This would require shared call appearances which isn't yet implemented. On 9/8/05, Arnar Birgisson [EMAIL PROTECTED] wrote: Hello,Avaya has a nice feature that allows you toa) ring both a

Re: [Asterisk-Users] Motherboard and processor recommendations

Generally I have used Intel Chipsets on ASUS motherboards. I've always used Kingston RAM. I've used Intel P4 CPU on S478 and LGA775. The Asus boards almost always have NIC and sometimes on board VGA. I've not had any problems with the hardware. Regards, Chris - Original

[Asterisk-Users] Not enough lines available for Asterisk implemetation

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on

[Asterisk-Users] power over ethernet hub/switch

Hi, is there anyone trying a power over ethernet solution to feed IP phones? I'd like to buy a good but cheap hub/switch but I don't know which. Can anybody help me?? TIA Giorgio ___ --Bandwidth and Colocation sponsored by Easynews.com --

[Asterisk-Users] Asterisk Euro-ISDN

I'm trying to setup my Asterisk-PBX (using [EMAIL PROTECTED]) to work as an ISDN-VoiP-Gateway. For this to work, I'm trying to setup a trunk using my Billion PCI-ISDN-Card. And yes: I'm an Asterisk-Newbie. ;-) I can't get it working and was wondering if there is someone out there with the same

RE: [Asterisk-Users] power over ethernet hub/switch

Are you looking for a Mid-span hub or a switch? Basically a Mid-span hub will function as a Powered patch panel where a switch will be just that. A switch that will feed power to each device. What do you consider cheap and how many ports are you looking for?

RE: [Asterisk-Users] IVR Documentation and Samples.

Hi We are using an IVR system based in AGI. The AGI makes querys to our radiator server using radius libraries. Itis based on ASTCC and Net-Radius modules. Look this links: http://www.voip-info.org/tiki-index.php?page=ASTCC http://search.cpan.org/~luismunoz/Net-Radius-1.44/ Regards

[Asterisk-Users] All Circuits are busy

hi all i have a problem, with digit 9. to go outside asterisk give me the error All Circuits are busy what happend?? my extension.conf [globals] VM_PREFIX = * RINGTIMER = 15 REGTIME = 7:55-17:05 REGDAYS = mon-fri RECORDEXTEN = PARKNOTIFY = SIP/200 OUT_1 = ZAP/g0 OPERATOR = NULL = IN_OVERRIDE =

Re: [Asterisk-Users] Motherboard and processor recommendations

On Thursday 08 September 2005 08:33, Chris wrote: Generally I have used Intel Chipsets on ASUS motherboards. I've always used Kingston RAM. I've used Intel P4 CPU on S478 and LGA775. The Asus boards almost always have NIC and sometimes on board VGA. I've not had any problems

[Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)

that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt flood. Is it true ? My boss is just asking me if it is

[Asterisk-Users] 2 X100P and SIP inbound routing

Sorry about my previous post, outbound routing this was clearly available in AMP. However, How would I go about making all calls from PSTN line 1 (X100P #1) ring call group #1, PSTN line 2 (X100P #2) call group #2, and the SIP incoming calls route to group #1 also. Is this done via dial plans?

Re: [Asterisk-Users] All Circuits are busy

On Thu, 2005-09-08 at 11:07 -0300, Pietro U wrote: hi all i have a problem, with digit 9. to go outside asterisk give me the error All Circuits are busy what happend?? [outrt-002-9outside] include = outrt-002-9outside-custom exten = _1XX,1,Macro(dialout-trunk,1,${EXTEN},) exten =

Re: [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)

On Thursday 08 September 2005 10:26, Simone Cittadini wrote: Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from real-life experience ? Don't do

[Asterisk-Users] Multiple Instances of Asterisk (no contexts)

I know I have seen something on the mailing list describing how to run more than one instance of Asterisk. I can't find it anymore. What are the things to look for when running more than one copy. Yes, I know about contexts. thanks, Geoff ___

RE: [Asterisk-Users] power over ethernet hub/switch

gincantalupo wrote: Hi, is there anyone trying a power over ethernet solution to feed IP phones? I'd like to buy a good but cheap hub/switch but I don't know which. Can anybody help me?? We are testing out the 3Com 2226-PWR Plus ($800US roughly). We haven't made it too far but the phones

RE: [Asterisk-Users] power over ethernet hub/switch

We use the old Cisco 3500, and works fine. The issue is which phones are you going to use, in order to have no problems with the polarity and the wiring. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gincantalupo Sent: Thursday,

RE: [Asterisk-Users] MAX PRI for single server (was: Not enough linesavailable for Asterisk implemetation)

On Thursday 08 September 2005 10:26, Simone Cittadini wrote: Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from real-life experience ? Don't

RE: [Asterisk-Users] channels VHF/ HF radio in asterisk

This is an old solution for pbx's, was called autopatch. The issue is that the pbx is full duplex, and most of this radios are half duplex. That is all, so the rpt solution deals with that plus the electrical interface to the system. Carlos Alperin -Original Message- From: [EMAIL

Re: [Asterisk-Users] asterisk frequently dead

This is really not a huge help.. hut I got that exact error message running on a pentium 1.8GIg, 128meg of ram, CentOS 3.0, self compiled CVS-HEAD of 2.0. The crash message is less them useful. When it happens you can not do anything with asterisk and have to kill it. On 9/7/05, stevanus

[Asterisk-Users] Re: MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)

On Thursday 08 September 2005 16:26, Simone Cittadini wrote: My boss is just asking me if it is possible to stuck 4* TE411P in a Doesn't that equal 16 lines, not 480 lines? Or did I miss something? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL

Re: [Asterisk-Users] power over ethernet hub/switch

Hi, I think about 8 ports. It is for big firms so I think about 20-25$ for each port (to tell the truth I don't know how much a POE hub can cost, cannot find any price list on internet...). TIA Giorgio Sean Milheim wrote: Are you looking for a Mid-span hub or a switch? Basically a

Re: [Asterisk-Users] All Circuits are busy

Dave thanks for the reply. in my PBX (digital phones) i dial 9 and get outside line and local numbers have 7 digits. is the conf correct for this numeric plan?On 9/8/05, Dave Cotton [EMAIL PROTECTED] wrote: On Thu, 2005-09-08 at 11:07 -0300, Pietro U wrote: hi all i have a problem, with digit 9.

Re: [Asterisk-Users] Multiple Instances of Asterisk (no contexts)

Geoff Karl wrote: I know I have seen something on the mailing list describing how to run more than one instance of Asterisk. I can't find it anymore. What are the things to look for when running more than one copy. Yes, I know about contexts. thanks, Geoff This begs a repeated question:

[Asterisk-Users] Problem: Got SIP response 481 Call Leg/Transaction Does Not Exist

I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register gets the SIP reponse 481 message Register SIP 4009 at 192.168.200.10 port 2199 expires 120 Unregistered SIP 4009 Register SIP 4009 at

Re: [Asterisk-Users] Motherboard and processor recommendations

We have 14 Asterisk servers with Asus/Intel in production at our four locations. We very much recommend them and also go through zipzoomfly.com to buy the parts. MATT---On 9/8/05, Martin [EMAIL PROTECTED] wrote: On Thursday 08 September 2005 08:33, Chris wrote: Generally I have used Intel

[Asterisk-Users] Slight OT: Multi WAN Router and SIP Calls

We are drafting a plan for a new office setup. The users will be using Cisco 7940 phones registered to a remote Asterisk server. We were thinking of using two ADSL lines coming into a Multi-WAN router to allow for load balancing. As opposed to setting up half the users on one ADSL line, half on

[Asterisk-Users] Re: MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)

In article [EMAIL PROTECTED], Wayne Gemmell [EMAIL PROTECTED] wrote: On Thursday 08 September 2005 16:26, Simone Cittadini wrote: My boss is just asking me if it is possible to stuck 4* TE411P in a Doesn't that equal 16 lines, not 480 lines? Or did I miss something? 16 E1 trunks, each

Re: [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)

On Thu, 2005-09-08 at 16:26 +0200, Simone Cittadini wrote: Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from real-life experience ? You

[Asterisk-Users] Pass through of T.38

Hi, I found some contradicting infos about pass through of T.38 data. Are there any experiences of just passing T.38 via SIP from one T.38 application or gateway trough asterisk to another T.38 application or gateway? Would asterisk maybe even pass T.38 from chan_oh323 to chan_sip (without

RE: [Asterisk-Users] channels VHF/ HF radio in asterisk

Maybe this could be used with the Internet Repeater trunking system I primarily use VHF... But would be interested in setting that up on my asterisk with the Internet 2M Repeater trunking system inter-connect Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. (Desk) 804.422.4401 (Cell)

Re: [Asterisk-Users] Re: MAX PRI for single server

Wayne Gemmell wrote: On Thursday 08 September 2005 16:26, Simone Cittadini wrote: My boss is just asking me if it is possible to stuck 4* TE411P in a Doesn't that equal 16 lines, not 480 lines? Or did I miss something? Yes, you missed something: 4 PRIs = 92 Lines per Card * 4

Re: [Asterisk-Users] Not enough lines available for Asterisk implemetation

Wayne Gemmell wrote: Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS

Re: [Asterisk-Users] MAX PRI for single server

Andrew Kohlsmith wrote: On Thursday 08 September 2005 10:26, Simone Cittadini wrote: Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from

RE: [Asterisk-Users] MAX PRI for single server (was: Not enoughlines available for Asterisk implemetation)

We run many servers with 4 Quad cards and have no problems, SANGOMA works great for this !! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk groups Sent: Thursday, September 08, 2005 6:12 AM To: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] Call goes through, but no audio

Hi, Does someone know what the problem is when the call goes through but one or both parties can't hear the other? What are the common causes? Solutions? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)

On Thursday 08 September 2005 13:12, asterisk groups wrote: You might want to offload some of that PRI termination to an external device such as a Cisco AS53XX, Lucent MAX TNT, Audio Codes or Redfone fonebridge device and then trunk it to your Asterisk servers. But putting more then 2 quad

Re: [Asterisk-Users] All Circuits are busy

On Thu, 2005-09-08 at 11:53 -0300, Pietro U wrote: Dave thanks for the reply. in my PBX (digital phones) i dial 9 and get outside line and local numbers have 7 digits. is the conf correct for this numeric plan? Still I'd like to see the Macro, does it strip the 9 off the front? If not you're

Re: [Asterisk-Users] Motherboard and processor recommendations

I've had lots of luck with the Intel/Asus and I am the part supplier. Chris - Original Message - From: Matt Florell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 08, 2005 9:58 AM Subject: Re:

[Asterisk-Users] Server Brand

Hello, all I apologize for all these questions. But I have so many running through my head for this. What brand of server is a good one to use for running an asterisk box? I did an install the other day on a Compaq Proliant ML370 G2 / Debian Sarge and it is currently working great. But

[Asterisk-Users] Txfax

Is theresome way to know if the fax was received correctly or not? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] Multiple Line Appearances / Why use this?

I apologize for the double post. I am curious as to what the usefullness is of the multiple line appearance feature on Polycom phones. I setup our phones to register one line per extension but I hear the IP501's can do three line appearances. Why and how could this feature be applied? Thanks

[Asterisk-Users] Re: MAX PRI for single server

In article [EMAIL PROTECTED], Matthew Boehm [EMAIL PROTECTED] wrote: Wayne Gemmell wrote: On Thursday 08 September 2005 16:26, Simone Cittadini wrote: My boss is just asking me if it is possible to stuck 4* TE411P in a Doesn't that equal 16 lines, not 480 lines? Or did I miss

Re: [Asterisk-Users] Re: MAX PRI for single server

On Thursday 08 September 2005 11:19, Matthew Boehm wrote: Yes, you missed something: 4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines That is assuming you have 1 D-chan per span. You're also assuming T1. -A. ___ --Bandwidth and Colocation sponsored

[Asterisk-Users] AstriCon Update: Please Register ASAP - Free Phones

_Book Your Hotel Room Today_ We're now a little more than a month away from AstriCon 2005 - The Asterisk Conference and Exhibition. We need everyone who plans on attending to register with the Hyatt ASAP to ensure we have enough hotel rooms. (Last year in Atlanta we over-booked the hotel by over

Re: [Asterisk-Users] unicall and cvs head

Did you use the 1.1.x version of the patch and chan_unicall.c ? Denis. On 05 de set de 2005, at 20:57, Anton Krall wrote: Guys. Anybody gotten unicall to compile under cvs-head? I get a lot of errors while under 1.0.9 everything compiled without a hickup. Any hints?

Re: [Asterisk-Users] g729 test

Michael Welter wrote: My preferred LD vendor requires g729 and SIP. Is there a method to test, prior to initiating a call, whether a g729 codec is available? Will ChanIsAvail test g729 availability? To clarify: I have n g729 licenses for my system. If I have n g729 calls in process then

RE: [Asterisk-Users] Polycom ip301 hangs at Running sip.ld

I'm not using an FTP server :-/ I guess I'll have to setup an FTP server and have it get it's files from there...it appears that something is corrupted on the phone. ... -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent:

RE: [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard

I use them with just the one NIC card. I don't use them as a router so the phones and my gateway are all on the same network. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Tuesday, September 06, 2005 4:33 PM To: Asterisk

[Asterisk-Users] asterisk handling of old voicemail messages

Is it true that asterisk voicemail moves messages from old to inbox after they age a certain number of days? How many days? I just had a case where 60 old messages showed back up in someones inbox and the time interval is unknown because I do not know when they were saved.

Re: [Asterisk-Users] Channalized T1 and PRI with Asterisk

Ben, That is the correct choice for an Asterisk box. good luck. Ben Brown wrote: Thanks for the replys. I'm convinced. PRI it is. Peter Svensson wrote: On Mon, 5 Sep 2005, Ben Brown wrote: So the only difference with PRI is caller ID? What I am trying to

RE: [Asterisk-Users] Call goes through, but no audio

That's almost always due to a NAT problem. Try using a stun server to solve this problem (stun.gist.net for example) --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of Doug -Sent: Thursday, September 08, 2005 11:38 AM -To:

[Asterisk-Users] How to cascade dial status back through IAX

On machine A I have something like the following in extensions.conf: [iax-extensions] exten = _9.,1,Dial(IAX2/machineB/${EXTEN:[EMAIL PROTECTED]) exten = _9.,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten = _9.,3,Hangup On machineB I have something like this: [mycontext] exten =

Re: [Asterisk-Users] All Circuits are busy

ok ok ok. this conf, i copy and paste from internet and i try to use in my asterisk. what i need to get outside line from the pbx? in my pbx i dial 9 and the 7 digit numbers. sorry but im a really newbie. On 9/8/05, Dave Cotton [EMAIL PROTECTED] wrote: On Thu, 2005-09-08 at 11:53 -0300,

Re: [Asterisk-Users] Server Brand

Kenny Kant wrote: Hello, all I apologize for all these questions. But I have so many running through my head for this. What brand of server is a good one to use for running an asterisk box? I did an install the other day on a Compaq Proliant ML370 G2 / Debian Sarge and it is currently

Re: [Asterisk-Users] MAX PRI for single server

Jason Becker wrote: Sage advice, but out of curiousity what happened to Digium's T3 card (the DS3000P)? IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC it will have no on-board EC and no on-board encoding so I can't imagine the machine you would need to process that

[Asterisk-Users] Spandsp

I was only able to find spandsp-0.0.1k, I am trying to follow the AMP install guide, it suggests that I use 0.0.2pre18 but the site (soft-switch) is down. Does anyone have a copy of it or later? Also when attempting to compile the 0.0.1k I get errors that won't let me continue, I assume those are

RE: [Asterisk-Users] channels VHF/ HF radio in asterisk

Yes, but on that case, you still are sending digital signal. On the autopatch you just send analog voice. On the Internet, you 'll going to need to use IP Phones on the other side, on the autopatch, you will be talking with someone on the radio (Two way or half duplex, or simplex) That is the

Re: [Asterisk-Users] Multiple Instances of Asterisk (no contexts)

On 9/8/05, Matthew Boehm [EMAIL PROTECTED] wrote: Geoff Karl wrote: I know I have seen something on the mailing list describing how to run more than one instance of Asterisk. I can't find it anymore. What are the things to look for when running more than one copy. Yes, I know about

Re: [Asterisk-Users] Transfer calls from cellphone

Brilliant, thanks for the response. Arnar [EMAIL PROTECTED] 8.9.2005 13:26:52 b is possible. See res_features.conf for more information on transferring via DTMF. c is not yet possible. This would require shared call appearances which isn't yet implemented. On 9/8/05, Arnar Birgisson

Re: [Asterisk-Users] g729 test

On Thursday 08 September 2005 12:16, Michael Welter wrote: I have n g729 licenses for my system. If I have n g729 calls in process then I don't want to attempt another g729 call. Is there a method to test whether a g729 codec license is available? Currently no. This would be a great starter

Re: [Asterisk-Users] MAX PRI for single server

Matthew Boehm wrote: Jason Becker wrote: Sage advice, but out of curiousity what happened to Digium's T3 card (the DS3000P)? IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC it will have no on-board EC and no on-board encoding so I can't imagine the machine you

[Asterisk-Users] play each person's voicemail

How do I set each extension to play it's own voicemail prompts? I have vm working in that it plays the standard person at extension 1234 is not available. and takes the message. I've recorded seperate .gsm files for each user but can not figure out how to use them. - Gary Edison

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