RE: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

I'm not familiar with Quintum, but what codec do you mean at the allow= line in sip.conf with h723? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Freddy Setiawan Sent: Sunday, June 25, 2006 8:37 PM To: asterisk-users@lists.digium.com Subject:

[Asterisk-Users] PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM

such cards? Specifications: PCI or MiniPCI up to 120 concurrent transcodings Codecs: G.729/G.729A or G.723.1 or GSM or combinations of them Thank you in advance, Roland Zagler ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

RE: [Asterisk-Users] 8 FXS in Asterisk Server

Subject: Re: [Asterisk-Users] 8 FXS in Asterisk Server Get a 8-port FXS gateway from www.broad-tel.com. That is the single box you need. On 8/16/05, Roland Zagler [EMAIL PROTECTED] wrote: Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones

[Asterisk-Users] Execute script on Answer

Hello, i was wondering if it is possible to execute an AGI or shell script when a channel is answered. Does anyone have suggestions on how to do this? Thanks in advance, Roland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] 8 FXS in Asterisk Server

Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI

RE: [Asterisk-Users] 8 FXS in Asterisk Server

in Asterisk Server Easy and cheap. Get two gateways AG-468 (each have 4 FXS ports) made by Atcom http://www.voip-info.org/tiki-index.php?page=Atcom one is about 88/ea I have two on the way and will let you know how it works. -- #Joseph On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote: Hello

RE: [Asterisk-Users] SOLVED: TE410P card in an HP-Compaq DL380 G4 server

, folks! Best regards, Roland -Original Message- Sent: Wednesday, July 20, 2005 10:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: Roland Zagler Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server What is your dmesg output when you fire up

[Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

Hello list, Did anyone already get the T410P card running in an HP-Compaq DL380 G4 server? If yes, how? I'm using Fedora Core 3 with 2.6.11-1.35_FC3smp Kernel package. Thanks in advance, Roland ___ Asterisk-Users mailing list

RE: [Asterisk-Users] Cisco 7960 on Asterisk?

Hi Michael, i run several 7960 and 7940 on our network, and they run smooth and without any complications. you will have to upgrade them using the SIP firmware from Cisco (i use versions 7.4 and 7.5 at the moment), you can download them from Cisco's Homepage but you will need a CCO account with

RE: [Asterisk-Users] OT: proliant fedora asterisk

Hi Freddy, we use the drivers from RedHat Enterprise Linux 4 and they work great. i think it depends just on the kernel version. e.g. http://h18000.www1.hp.com/support/files/server/us/locate/1116_6011.html for the DL360 regards, roland -Original Message- From: [EMAIL PROTECTED]

AW: [Asterisk-Users] SpanDSP rxfax, no tiff.

is /usr/local/sbin/mailfax flagged to 755? Von: [EMAIL PROTECTED] im Auftrag von Rob Danz Gesendet: Mi 13.07.2005 17:17 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] SpanDSP rxfax, no tiff. Hello, Let me start by saying I have checked the wiki

[Asterisk-Users] Cisco 7940/7960 interdigit timeout

Hello list, does anyone know how to change the interdigit timeout when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

RE: [Asterisk-Users] Any suggestions for an IP phone?

try the cisco 7940 with sip firmware: tons of features and easy to install see http://www.voip-info.org/tiki-index.php?page=cisco%2079xx regards, roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexandre Leclerc Sent: Tuesday, July 12, 2005 6:16

[Asterisk-Users] Cisco SIP Frimware for 7940/7960 v7.5

Hello list, is there anyone out there that could grab the new SIP firmware 7.5 for the 7940/7960 from Cisco's Site and mail it to me ([EMAIL PROTECTED])? i already ordered a support contract but did not get my access data yet! Thanks, Roland ___

RE: [Asterisk-Users] Cisco 7940/7960 interdigit timeout

PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7940/7960 interdigit timeout -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 12 Jul 2005, Roland Zagler wrote: Hello list, does anyone know how to change the interdigit timeout when using

RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

Hi Newbie, I wonder how you could find the mailing list but NOT the wiki at http://www.voip-info.org/tiki-index.php?page=Asterisk, that documents a huge area of how to use Asterisk in many scenarios. First of all, take a week or two to read the wiki and to set up a testing environment and try to

RE: [Asterisk-Users] cisco 7920

Sure! http://www.voip-info.org/tiki-index.php?page=SCCP-HOWTO2 regards, roland From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betül Gözlükoglu Sent: Monday, July 04, 2005 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

RE: [Asterisk-Users] G729 licencing with asterisk, how does it work ??

find it here: http://www.digium.com/index.php?menu=product_detailcategory=extrasprod uct=G729 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis curty Sent: Monday, July 04, 2005 3:45 PM To: Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] Asterisk on Virtual Machine

did you use the zaptel drivers? you need a timer interface for meetme application! use ztdummy! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed Farid Sent: Monday, July 04, 2005 3:59 PM To: Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] Proper way to start * and load modules on a RedHatbox

Hi, after you have done make, make install and maybe make samples in asterisk source-dir just do a make config and all will be done for you. to check if it worked, simply issue chkconfig --list asterisk to see the runlevels asterisk is started or not. to start zaptel drivers do the same after

RE: [Asterisk-Users] play message to callee before connecttoincomingcall

, Roland Zagler wrote: sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should

RE: [Asterisk-Users] Connecting two servers - dial string

) extensions.conf of server2: exten = _1X.,1,Dial(IAX2/server1/${EXTEN:1},30) use deny and permit only with later versions than 1.0.5 of asterisk (best with CVS HEAD) i hope this helps best regards, Roland Zagler -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

RE: [Asterisk-Users] play message to callee beforeconnecttoincomingcall

Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -Original Message- From: [EMAIL PROTECTED]

RE: [Asterisk-Users] Is it possible to setup group voicemail inAsterisk?

Hi Leo, here's a suggestion: in your dialplan (extensions.conf) send multiple users to the same mailbox (e.g. 999) if they do not pick up within 30 seconds: ; SIP Phone 100, Tom exten = 100,1,Dial(SIP/100,30) exten = 100,2,VoiceMail(999) ; SIP Phone 200, Eric exten = 200,1,Dial(SIP/200,30)

[Asterisk-Users] play message to callee before connect to incoming call

be connected to SIP Phone 100 any suggestions on how to implement this in an easy way? Thanks in advance, Roland Zagler ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] play message to callee before connect toincoming call

and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Roland Zagler -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent

RE: [Asterisk-Users] play message to callee before connect toincomingcall

On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote: Thank you, Robert! The announcementfile plays well, but at Dial-option m i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Noted, which is why I offered option two. Background command waits for a user input

[Asterisk-Users] Memory Consumption

Hello, I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21) and i experienced that the memory consumption of the asterisk-process started by the init.d-script raises continously. Now, after 3 hours of operation (on our testing-system we have 30 concurrent connections to another

[Asterisk-Users] OH323 and gatekeeper

Hello! Can i only use one gatekeeper in OH323? Is there any documentation about how to use gatekeeper-ids? Thanks, Roland ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] Asterisk sudo from httpd

Apache httpd 2.0.50 Asterisk 1.0-RC2 Can anyone please help? Thank you in advance! Roland Zagler mailto:[EMAIL PROTECTED] @fog smart partners ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Using AVM Fritz!PCI as zap interface

Hello! Is there a way to use AVM Fritz!PCI as a ZAP interface and have it configured for ZAP channels? Thanx in advance! Roland Zagler mailto:[EMAIL PROTECTED] @fog smart partners ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

[Asterisk-Users] Spandsp - opencall.org offline

Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Than! Roland Zagler mailto:[EMAIL PROTECTED] @fog smart partners ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

[Asterisk-Users] Asterisk as SMS Service Center

Hello! Is it possible to run Asterisk as a SMS Service Center and does it work over a directly connected ISDN (CAPI) interface card? Did anyone already use Asterisk for that? Roland Zagler mailto:[EMAIL PROTECTED] @fog smart partners ___ Asterisk

[Asterisk-Users] Digium TE410P and RedHat Enterprise Server 3.0

Hello! has anyone already successfully installed Digium TE410P card on RedHat Enterprise Server 3.0? Roland Zagler mailto:[EMAIL PROTECTED] @fog smart partners ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] HELP: BYE-request not sent to SIP-peer

..Call-ID: [EMAIL PROTECTED]: [EMAIL PROTECTED] 90.238..Content-type: application/sdp..Max-Forwards: 70..Content-Length: 133v=0..o=none 0 0 IN IP4 [peerIP]..s=-..c=IN IP4 198.31.231.1 7..t=0 0..m=audio 18691 RTP/AVP 8..a=rtpmap:8 PCMA/8000..a=ptime:30.. Than Roland Zagler mailto

RE: [Asterisk-Users] CAPI call transfer

Try specifying your number you want to dial with b in front of, e.g. Dial(CAPI/01824708169:b01824708752,60) in your extensions.conf! Regards, roland Roland Zagler mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Sent

RE: RE: RE: RE: [Asterisk-Users] CAPI call transfer

Can you post your extensions.conf, maybe i can find something! Roland Zagler mailto:[EMAIL PROTECTED] mobile:4369910713694 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Sent: Tuesday, August 10, 2004 1:30 PM To: [EMAIL PROTECTED] Subject: Re

RE: RE: RE: RE: RE: [Asterisk-Users] CAPI call transfer

,CAPI/50:b${EXTEN},60 exten = _.,100,Hangup Roland Zagler mailto:[EMAIL PROTECTED] mobile:4369910713694 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Sent: Tuesday, August 10, 2004 2:03 PM To: [EMAIL PROTECTED] Subject: Re: RE: RE: RE: RE

RE: RE: RE: RE: RE: RE: [Asterisk-Users] CAPI call transfer

You could try to specify incomingmsn *NOT* to * and outgoingmsn in your capi.conf Roland Zagler mailto:[EMAIL PROTECTED] mobile:4369910713694 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Sent: Tuesday, August 10, 2004 2:38 PM To: [EMAIL

[Asterisk-Users] Asterisk and Douglas Telecom

Has anyone experienced in connecting a asterisk pbx to douglas telecom successfully? If yes, could you please post your SIP.CONF and your EXTENSIONS.CONF! Thanx in advance, Roland ___ Asterisk-Users mailing list [EMAIL PROTECTED]