[asterisk-users] Enough Turkey? Voip Users Conference today at 12:30 EST to help digest it all

Friday at 12:30 PM Eastern, 9:30 AM Pacific, 17:30 GMT join us (or listen to) what is really the Asterisk Users Conference that dare not speak its name, except when called the VOIP Users Conference. http://www.VoipUsersConference.org IRC on freenode.net #voip-users-conference In a nutshell At

[asterisk-users] How to bridge two connected calls

Hi everybody. I am in the following scenario: 1 Customer "A" calls an asterisk box over a Zap channel on a toll free number during night time 2 The incoming call enters an AGI script on the dialplan 3 The AGI script plays back a welcome message, then starts the music-on-hold stream 4 The

Re: [asterisk-users] problem with tdm2400p configuration

On Nov 19, 2007 2:31 PM, Mark Quitoriano <[EMAIL PROTECTED]> wrote: > On Nov 19, 2007 12:10 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > > Mark Quitoriano wrote: > > >>> that's the same question i got(regarding question 1). Is it possible > > >>> for PCI compatibility issue? i need to ch

Re: [asterisk-users] Calling with hidden callerid

At 01:02 PM 11/22/2007, you wrote: I`ve set callerid name to "unknown", that works well, but when I put an empty number it goes out with the name "asterisk". Which is NOT what I want. Is there a standard way to say "hid my number"? I set it to 1234567890 if that would work for you. Ira ___

Re: [asterisk-users] Calling with hidden callerid

The dialplan command 'setcallerpres' is also good. PaulH On Fri, 2007-11-23 at 12:44 +1100, Nick Brown wrote: > You can set callerid within the [general] section of your sip.conf. > This should work for you. > > > On 23/11/07 8:02 AM, "Mike" <[EMAIL PROTECTED]> wrote: > > Hi, >

Re: [asterisk-users] NAT keep-alive

Ugo Bellavance wrote: > Hi, > > On my linksys/sipura phones/ATA, there is a setting called "NAT > Mapping Enable" and another called "NAT Keep Alive Enable" > > These settings must be on in my setup so that my phones/ATA remain > connected to my * server. My setup is: > > Home LAN - Pfse

Re: [asterisk-users] Calling with hidden callerid

You can set callerid within the [general] section of your sip.conf. This should work for you. On 23/11/07 8:02 AM, "Mike" <[EMAIL PROTECTED]> wrote: > Hi, > > I have a wholesale provider that allows me to put any caller id I want when > dialing out. In some cases, I`d like the outgoing caller

[asterisk-users] Work

Hey, we are looking for someone to work to the end of january , and maybe even stay on after that. _Immediate start_. Low to Mid level asterisk work (phone support and onsite install work) You MUST be living in Melbourne, Australia. Email me off list for more details. PaulH __

[asterisk-users] NAT keep-alive

Hi, On my linksys/sipura phones/ATA, there is a setting called "NAT Mapping Enable" and another called "NAT Keep Alive Enable" These settings must be on in my setup so that my phones/ATA remain connected to my * server. My setup is: Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet -

[asterisk-users] Odd bug in Siemens C460IP ?

Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handset

[asterisk-users] Calling with hidden callerid

Hi, I have a wholesale provider that allows me to put any caller id I want when dialing out. In some cases, I`d like the outgoing callerid to be hidden. How do I do this? I`ve set callerid name to "unknown", that works well, but when I put an empty number it goes out with the name "asterisk".

Re: [asterisk-users] Dial problem

Remove callprogress=yes from /etc/asterisk/zapata.conf There is a REASON it is listed as EXPERIMENTAL. It simply does not work well. Rilawich Ango wrote: > HI, > I have 2 TDM400s plugged in a PC. I failed to use same channels to > make a call to PSTN. It shows it can't establish connection

Re: [asterisk-users] Problem installing Asterisk

Matt wrote: > On Nov 21, 2007 11:45 AM, Tilghman Lesher > <[EMAIL PROTECTED] > > wrote: > > On Wednesday 21 November 2007 09:09:13 Matt wrote: > > I have installed Asterisk with FreeTDS many times before (this same > > Asterisk and same TDS version)... but

Re: [asterisk-users] quality after call transfer

Hi, I suspect that you are "transcoding", meaning that the call is comming in a specific codec format, and the second phone uses another codec. So, when you do your tranfert, Asterisk is in the middle and is coding from the original to your phone with two different codecs. If you are passing from

[asterisk-users] Dial problem

HI, I have 2 TDM400s plugged in a PC. I failed to use same channels to make a call to PSTN. It shows it can't establish connection after dial command issued. Below is the log. Actually, the call is established as I can hear voice from the called party but the softphone is still showing ringin

Re: [asterisk-users] Toll fraud detection/password script

Thanks for your contrib On Nov 22, 2007 2:56 PM, J. Oquendo <[EMAIL PROTECTED]> wrote: > > So I was bored yesterday and tried solving a few > problems with one stone: > > 1) Notify me of potential brute forcers (multiple attempts > to register multiple numbers from one address) > 2) Notify me of (

[asterisk-users] Toll fraud detection/password script

So I was bored yesterday and tried solving a few problems with one stone: 1) Notify me of potential brute forcers (multiple attempts to register multiple numbers from one address) 2) Notify me of (l)users who are having password issues So I whipped up a simple script to run in cron and notify me

Re: [asterisk-users] Phones Not Registering

On Thu, 22 Nov 2007, Edwin Kariuki wrote: > > > Brett Crapser <[EMAIL PROTECTED]> wrote: > On Thu, 22 Nov 2007, Edwin Kariuki wrote: >> Hi, >> >> I have a voip platform that has a SIP server where about 450 sipura >> phones & adaptors register. On two occassions some phones (which were >> previous

[asterisk-users] mailbox name length

Hi, is there a way to set the length of the mailbox name - now it is 4 Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailma

Re: [asterisk-users] [asterisk-dev] trunk working under windows!

Drew Gibson wrote: > but ... why? so windows lawyers can sneak a few patents thru the patent office and sue Digium for patent infringement. I am not criticizing Zoa or Luigi here, just reflecting on what ends up happening eventually. Think BSD code into windows, think file & re

Re: [asterisk-users] Queue Drops to Voicemail

I believe you should define the agents in agents.conf as well! :-) l. On Wed, 21 Nov 2007 19:39:46 +0100, Gregory Malsack <[EMAIL PROTECTED]> wrote: > Hello All, > > > I am hoping someone out there can enlighten me on this issue. I am using > asterisk 1.4.11. We have a call queue setup, and o

[asterisk-users] Digium and Asterisk

Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/h

Re: [asterisk-users] spandsp as T.38 termination?

[EMAIL PROTECTED] wrote: > You need a T38 gateway of sorts, sort of like the app_t38gateway of > CallWeaver. > the app_rxfax and app_txfax applications in 0.0.4 of spandsp won't provide it? It seems, functionally, that with spandsp supporting T.38, simple dialplans where you call a SIP exte

Re: [asterisk-users] ACD functionality , Skills for agents

Hi, Örn Arnarson schrieb: > I have often wondered the same thing. > > It seems to me to be random, or it works it out some way I am not > familiar with. I have seen calls with wait time of 30 seconds get > answered before calls with 30 minutes wait time from queues with equal > weight. I can con

Re: [asterisk-users] Phones Not Registering

Brett Crapser <[EMAIL PROTECTED]> wrote: On Thu, 22 Nov 2007, Edwin Kariuki wrote: > Hi, > > I have a voip platform that has a SIP server where about 450 sipura > phones & adaptors register. On two occassions some phones (which were > previously working) have refused to register with certain I

Re: [asterisk-users] ACD functionality , Skills for agents

What about tagging calls with skills and then putting them all in one queue? Skill for agents would be declared in queues.conf just like penalties? member => Agent/1001,Sales,10 where Sales=skilland 10 = weight of skill for agent. Is that feasible? -Original Message- From: [EMAI

Re: [asterisk-users] Phones Not Registering

On Thu, 22 Nov 2007, Edwin Kariuki wrote: > Hi, > > I have a voip platform that has a SIP server where about 450 sipura > phones & adaptors register. On two occassions some phones (which were > previously working) have refused to register with certain IPs but when I > change the IP the phones r

Re: [asterisk-users] Phones Not Registering

The reason could be bad routing, IPs used by multiple devices.. n so on... Edwin Kariuki wrote: > Hi, > > I have a voip platform that has a SIP server where about 450 sipura > phones & adaptors register. On two occassions some phones (which were > previously working) have refused to register

[asterisk-users] Phones Not Registering

Hi, I have a voip platform that has a SIP server where about 450 sipura phones & adaptors register. On two occassions some phones (which were previously working) have refused to register with certain IPs but when I change the IP the phones register. The failing IP can the work after two days.

Re: [asterisk-users] common/shared voicemail box

The only possible way I can think of achieving this would be to mangle the incoming caller ID to include the extension that the call came from. Given that Asterisk's voicemail boxes are separate to extensions, I can't see another solution. Benjamin Jacob wrote: > Hello All, > > I am using ODBC st