[asterisk-users] b option in Directory

I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b option that let you enter the first name OR last name of a user. I see that to make this work I need a patch. I'm wondering how can I install this patch as it's an option one of my customer would like to have but I never

Re: [asterisk-users] Question about OSLEC or HPEC with AsteriskNow

AsteriskNow use CentOS 5 and it comes preinstalled with dahdi and asterisk with the freepbx GUI interface and it seems to be missing all the dev packages Martin On 2009-11-17, at 02:19, Olivier wrote: 2009/11/17 Martin Roy I was previously using an old computer running Asterisk 1.2

[asterisk-users] Question about OSLEC or HPEC with AsteriskNow

I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my pr

[asterisk-users] Asterisk on Fedora Core 4

I just did a clean install of Fedora Core 4 on a PC with a TDM400 installed. I installed zaptel 1.2.20.1 and asterisk 1.2.24 correctly. I did make config for both to have zaptel and asterisk start when I boot the computer. My main problem right now is that zaptel doesn't load at startup so

Re: [asterisk-users] Asterisk on Fedora Core 4

ilipp Kempgen wrote: > Martin Roy wrote: > >> I just did a clean install of Fedora Core 4 on a PC with a TDM400 >> installed. I installed zaptel 1.2.20.1 and asterisk 1.2.24 correctly. >> I did make config for both to have zaptel and asterisk start when I >> boot the computer

[Asterisk-Users] Re: Caller ID forwarding

To answer your question on how I do the hook flash transfer here it is :in the globals section of extensions.conf put all your cell phone number like this :[globals]MartCell=5141234567Then add this macro in your extensions.conf :[macro-cell_user]exten => s,1,Playback(Call_Transfer)exten => s,2,Flas

[Asterisk-Users] Caller ID forwarding

I doubt it's possible but I'll ask just in case there's a "legal" way to do that. I have an asterisk server setup at work. When someone call from a PSTN line and enter an extension it rings for a few seconds on the SIP phones of that extension and then if there's no answer it transfer the

[Asterisk-Users] *67 with Sipura 3000

How can I dial *67 on a Sipura 3000 if I dial from a SIP phone connected on an asterisk server. I always get a message saying that authentication failed for INVITE for [EMAIL PROTECTED] If I dial a number without doing *67 it's working fine... sip 221 being the extension of my Cisco phone a

[Asterisk-Users] Multiple Sipura 3000

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo

[Asterisk-Users] Echo problem

Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm CG-410 4 FXOs device. Now I just ordered a few Sipura 3000. With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the following : echocancel=yes echocancelwhenbridged=yes echotraining=yes (I tried 800 with TDM04B

[Asterisk-Users] Using zap channels on 2 different servers

Let say I have a server located in Europe and one in North America. The 2 servers are connected together with iax2. Both server are connected to phone lines in there own country. If I want that when a user call a north american phone number from the server in Europe it use a zap channel on t

[Asterisk-Users] X100p cards

I have a customer that has 10 analog lines (he can't get digital where he's located). I'm currently using 3 TDM04B cards but I have the damn echo problem all the time what ever setting I do (I followed all the steps I could find in the Wiki and in this forum but none work). I even tried on

[Asterisk-Users] Tyan Transport GX28 with TDM400

hanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] French SIP or IAX phones

ossible. So I'm wondering if there's any one out there that found a phone that can be change to french. Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSU

[Asterisk-Users] TDM04B in a Mac

. Anyone found a way to make it work? Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[Asterisk-Users] Tired of trying to fix this echo problem

Anyone tried the Clipcomm CG-410? I'm tired of beeing unable to get rid correctly of the echo problem. I have 3 TDM04B installed in one server. It was working fine when I had only one card installed in the server but since I installed 2 more then I can't get rid of it what ever I try... So now

[Asterisk-Users] Re: Upgrading Asterisk

I did the upgrade everything went well. Now I hope it fixed my echo problem... When I installed version 1.0.3 I forgot to do make linux26 so maybe my echo problem was coming from that. I'll know soon enough... Martin Martin Roy wrote: I'm about to upgrade my currently running aster

[Asterisk-Users] Upgrading Asterisk

I'm about to upgrade my currently running asterisk server from 1.0.3 to 1.0.6 is there anything I should do before doing the upgrade? I know since I'm using Fedora Core 3 that I must do for Zaptel : make clean then make linux26 and finally make install. Do I have to remove the version 1.0.3 fir

Re: [Asterisk-Users] IAX Codec

Currently I have one server running Fedora Core 3 AMD 64bits (on a 3mbits DSL with 640kbits upload) and the second server is running on Mac OS X (on a 512kbits SDSL) I'll change it soon for a PC with Fedora Core 3 but I know the G.729 isn't available for Mac OS X. Is there another codec that do t

[Asterisk-Users] IAX Codec

I have 2 Asterisk servers connected with IAX. It's working fine I can call an extension from one phone in an office to another phone in the other office. The only problem I have is lagging. What codec should I use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I configured it

[Asterisk-Users] Application asterisk uses obsolete OSS audio interface

I don't know what happen but my asterisk server was working fine then I rebooted the computer and at next restart I got that message when I tried to start asterisk : Application asterisk uses obsolete OSS audio interface Then I edited zapata.conf and removed the line that was calling musiconhol

Re: [Asterisk-Users] Weird Echo Problem

Yes that's what I have in my current config... : context=incoming signalling=fxs_ks echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=0 txgain=0 immediate=no busydetect=no callprogress-no musiconhold=default usecallerid=yes callerid=asreceived group=1 callgroup=1 pickupgroup=1 ch

Re: [Asterisk-Users] Weird Echo Problem

Well I have 3 Digium TDM04B (4 port FXO) installed in my server. I use 10 channels out of 12. There's 5 PCI slots on my motherboard, currently they fill the first 3 PCI slots. I can try to move them arround leaving one free PCI slot between each of them. The motherboard I use is a Tyan S2875ANR

[Asterisk-Users] Weird Echo Problem

Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from

[Asterisk-Users] Caller ID

How can I change that when there's no Caller ID instead of displaying asterisk it display something like Unknown. Because everyone is confuse when they see a call coming from asterisk. Thanks Martin ___ Asterisk-Users mailing list Asterisk-Users@lists.

[Asterisk-Users] Multiple mailbox on the same SIP extension

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each p

[Asterisk-Users] Different rings

How can I get a different ring tone when I get a call from the PSTN on my Cisco IP Phone 7960? I want one ring tone when it's an internal call (coming from another SIP extension on my network) and another one when it's coming from the PSTN. I'm using TDM04B cards if that make any difference to

[Asterisk-Users] RE: Incoming calls

So if I understand well this should do the trick : (be aware that context first and second include all my extensions that I haven't included in this and in my SIP phones use context firstinternal and secondinternal) zapata.conf : context=firstincoming switchtype=national signalling=fxs_ks echot

[Asterisk-Users] Incoming calls

OK I have 12 phone lines connected to 3 digium TDM04B cards on the same server. I must do the following thing : The first 10 lines will be use by one company and the 2 left by another one. For outgoing calls it's quite easy I just create 2 different group and let them dial on a different one. B

[Asterisk-Users] Re: how to add more TDM04B

Sorry forget my post I forgot to add it in zaptel.conf... now it's working fine... That's what happen when you want to do things a little too fast hehe ;-) Martin Martin Roy wrote: I already have one Digium TDM04B installed in my server working fine. I just received 2 more so I did

[Asterisk-Users] how to add more TDM04B

I already have one Digium TDM04B installed in my server working fine. I just received 2 more so I did added them to my server. When I booted again linux told me that it found new hardware I said ignore. Then I log in as usual. I did ztcfg -vv to see if it sees 12 channels now instead of 4 but i

[Asterisk-Users] Cisco phones config over internet

I saw a previous post about this problem but no one seems to have a real solution for it... I have a few Cisco IP Phones 7960 that I want to install at remote locations. I can easily setup my router to forward the ports of Asterisk BUT I don't want to open my TFTP server over the Internet as al

[Asterisk-Users] RE: Problem with chan_sccp and cisco 7960

I'm wondering why are you using SCCP and not SIP as most of us that use Cisco 7960 phones? Martin From: "Nenad Radosavljevic" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Problem with chan_sccp and cisco 7960 Hi ! On Cisco 7960 (with or without 7914 add-on module)

[Asterisk-Users] RE: TFTP Server Facing the Internet

Well the best solution would be to create a VPN between your network and the one of your customer but that's only possible if you have a VPN router on both side. Otherwise I don't see much solution then the one you already consider doing. Martin From: Michael Welter <[EMAIL PROTECTED]> Since we

[Asterisk-Users] Re: Howto Setup TFTP server on Linux for Cisco 7960

7.1 there) http://ns.goodgrief.com/voice-comm/ (there's 2.x,3.x and 4.x there but the 4 didn't seem to work for me) As weird as it might seem you can also find some on eDonkey and Kazaa if you are willing to wait a day or so to download them... Hope this help Martin Roy Martin Roy wrote: Get a

[Asterisk-Users] RE: Howto Setup TFTP server on Linux for Cisco 7960

n 7.3. But I had a hard time upgrading from 2.1 to 3.3 as the 3.3 image as more then 8 characters and firmware 2.1 support only 8:3 (8 characters plus 3 characters for the extension) If you need any help let me know. Martin Roy ___ Asterisk-Users ma

[Asterisk-Users] Re: Athlon 64 for Asterisk?

SIP for a start. Then I have to install another Asterisk server at another location and connect the 2 together. I'll probably install the same server at the second location with a T1/E1 card instead of the TDM04B cards. Martin Roy From: "Robert Augustyn" <[EMAIL PROTECTED]> S

[Asterisk-Users] Re: Athlon 64 for Asterisk?

I'm using a server with dual AMD opteron processors with a TDM04B without any problem. The server is running Fedora Core 3 AMD 64bits. Hope this answer your question... Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls

will start with channel 4 and not pick a random one between the 3 channels as I'm pretty sure if I put in my dial plan a group having channel 2, 3 and 4 it might do the opposite and start with channel 2 then if it's busy switch to 3 and then 4 instead of 4 then 3 then

[Asterisk-Users] how to manage Digium TDM04B outgoing calls correctly

7;s only one PRI provider where we are currently) Any help would be appreciate :-) Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users