similar to: Re: Re: 8 pstn lines+ on Asterisk supported

Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the "Qs about FXO/FXS cards" thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi

Hi timebandit, I'm realy happy to hear that, and as a matter of fact, all my Asterisk hardware is Intel server products, from chasis to MB. I know I can trust this hardware and I have excellent support from the guys I buy from. Thank you. >Message: 5 >Date: Mon, 3 Jan 2005 15:48:47 -0500 >From: <timebandit001@gmail.com> >Subject: Re: [Asterisk-Users] 8 pstn lines+ on

This is a system for our lab. I have no problem getting rid of X100P clones. But I am just curious why can they work. Even the drivers are not loading correctly. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent: Tuesday, May 02, 2006 10:51 AM To: 'Asterisk Users Mailing List -

Hello, i've googled and can't find a definite answer, so here goes: I have purchased the Digium TE100P, and am setting up the connection, however the telco i'm supposed to work with does not support PRI/ISDN E1 connections. They only support E1/R2 lines. Is there a way i can make the TE100P work with this? I've not seen any zaptel.conf that supports this. Any workarounds?

Hi, I have a problem I will describe. I have PAP2 connected to the internet to an asterisk box with 2 TDM cards, one TE100P E1 with PRI and one TDM400P with 2 FXS an one FXO. When I call to the TDM400 cards from the PAP2 eveything is OK, sound quality is perfect. When I call to terminate the call in PSTN through E100P I hear clicks which aparently are RTP packet looses. This clicks are only heard

Hi, I have a Gateway running in "TE" (terminal equipment mode as "slave" that I need to connect to my asterisk server using a TE100P card. Can anybody give a quick run up of how to run the TE100P's in Network Termination mode to have this working sucessfully? Cheers! Regards, Sahil Gupta VoiceValley

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel

Hmm, I don't have /etc/modprobe.conf, and wctdm is giving me problems. Which device is it talking about? [root@asterix /]# modprobe wctdm /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

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Guys. I have a te100p with unicall and an E1 and Im having problem with DTMF tones but the weird thing is, I only have problems sending the tones to certain phone numbers, anybody seen this behavior? Asterisk shows on the console the dtmf tone been pressed but seems the other side is not getting them, and this just happens with certain phone numbers, not all.. Any clues(tips?

A DID number was dedicated to receive fax, but i have the problem when getting fax call, which call will become a normal phone call and no fax was printed. When fax is detected, the fax extension is executed and dial the extension of the HT486 device (firmware 1.0.5.22). Somehow sending fax out working well. In the mailing lists, i notice some are using HT286 and it work. Could someone share

I would just like to get a refresh of the situation for Australian users. It would seem that the TE400P is currently available and is likely to acheive approval for use in Australia within the next 2 months ? (when is the end of summer?). Once this is done, it will certainly suit larger installations, but it still leaves a number of 'gaps'. Anyone with analog phone lines will need

Hello, I am having some problems with faxing in asterisk. I have a TE100P which is taking my PRI. This seems to be working fine. I also have a TDM400P with 2 FXS. Again card seems to be working fine, I can dial from phones attached to these to ports and everything seems to work fine. I have 2 DID's for my two fax machines that dial the Zap port for the FXS on the TDM400P. The problem

Hi. I have a problem with two asterisk servers with version 1.2.5. In one server there is a Digium TE411P in the second the Digium TE100P. We use E1 and EuroISDN. '/etc/zaptel.conf': ----- begin ----- span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 ----- end ----- /etc/asterisk/zapata.conf': ----- begin ----- [channels] pridialplan = unknown prilocaldialplan = international

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

Hello list, I recently bought a Linksys SPA400 as a PSTN gateway. The gateway is connected to an * server and i have 10 users using this setup. I do have some problems in establishing a call to an outside location (call that goes through the SPA400). The first attempt doesn't get through. I suspect the spa400 being the source of the problem. The Linksys SPA400 has a lot of params on the

There's a project to use asterisk to replace an old analog pbx for a hospital in Mexico. The person in charge wants to do a trial before making a large investment. For this test, he is willing to buy a Wildcard TE410P. Hes got a proposal from his pstn provider to convert his analog lines to digital ones, but I have some doubts. Let's say he converts only 2 or three of the lines. He would

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it