similar to: Please..... need some help

We have an application for Asterisk that will require connecting 144 fax ports into the system. Faxes will route externally over a PRI. The 144 ports are for local fax machines within the building. Not all will be faxing simultaneously. We just need to be able to provide ports in the building to plug in lots of fax machines. The plan is to run an Asterisk server for about 100 phones and these

ðÏÞÅÍÕ ÂÙ É ÎÅÔ? ÷ÏÐÒÏÓ ÔÏÌØËÏ × ÔÏÍ, ËÁË Õ ÔÅÂÑ ÂÕÄÅÔ ÓÏÅÄÉÎÑÔØÓÑ ÐÁÎÁÓÏÎÉË Ó ÁÓÔÅÒÉÓËÏÍ. Date: Wed, 30 Jul 2003 20:06:17 +0400 From: Pavel Zheltouhov <pavlo@comlink.ru> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk,ata186 and Panasonic TD1232 Reply-To: asterisk-users@lists.digium.com I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk.

Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least

hi everyone, Is it possible to serially connect my panasonic KX-TD1232 with a Linux box(Asterisk installed) and have it working? -- P. K. kakarizz@fastmail.fm

Hi, Anybody have an idea how to integrate * with a Panasonic td-1232? We one at the main office, and are installing * in a branch office. We'd like to be able to make calls from * extensions to Panasonic extensions and the other way around. Making outgoing calls from extensions one one side to lines on the other would be nice too. I can put another * machine at the main office, but what is

Hello We have around 50 phones in our company, and I am playing with the thought to gradually go over to using sip services and ip-phones internally. However at first I would liked the Asterisk just to sit between the phone line and the Panaosnic, so I can take out one lin/number at a time to use ip phones. I am new to Asterisk, and haven't done much configuring of the PBX either. So I also

Hi all, There is someone with some experience interconnecting a Panasonic digital PBX (KX-TD1232) with Asterisk? Thank you and best regards, Dan

If Someone did that, How I connect extensions.conf with this type of Hybrid system to work with asterisk inside this schema: PSTN--->PANASONIC KX <------> Asterisk | |--------->send internal call Thanks.

The hardware we are planning to use is: Micronet SP5050 FXO Gateway http://www.micronet.com.tw/Products/VoIP/SP5050.asp Micronet SP5100 IP Phone http://www.micronet.com.tw/Products/VoIP/SP5100.asp We are hoping to use this hardware along with AsteriskPBX to replace our aging PBX system. What I want to acheive is: * Any incoming call from PSTN (via gateway) rings on the receptionists phone for

Hi. I've searched the archives and found nothing regarding collaborating Asterisk with a Panasonic PBX (TD1232 to be exact) Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? On the hardware page for the X100P card is says it's great for

Hello people, I am using Asterisk with a handful of Micronet SP5100 IP Phones and a Micronet SP5052 FXO Gateway. So far I have incoming calls ringing all the phones correctly, outgoing calls working, voicemail working and calls between phones working. The only think I cant get working is Transferring of calls, and the ability to put calls on hold. The phones have both a 'Transfer'

Anyone have any luck with connecting Asterisk to the Panasonic KX-TD500. I have Asterisk connected via crossover to the TE110P. We are able to make internal calls into the Asterisk Box but the PBX vendor (I know nothing about the KX-TD500) tells us it is not possible route DID over the trunk. I find this hard to believe. Anyone have any luck with this? Thanks! -------------- next part

In the mailing list archives, I found a message that indicates that the IAXy has the ulaw, alaw, and g726 codecs, but I cannot find anything official on Digium's site about it. The Installation Manual has an example iax.conf file that indicates the ulaw codec, so I know that one is good. But we are thinking about using the IAXy over a VPN, to replace our MultiVoip. alaw and ulaw are

Does anyone know if it is possible to use the following disconnect tone setting with an x100p card? Disconnect Tone: 350@-19,440@-19;4(.25/.25/1+2) This tone was written for a Sipura SPA-3000 for a Panasonic KX-TD1232. The Panasonic does not support disconnect supervision, so this tone is the only thing that will detect a disconnect. It is not a standard fast busy or offhook tone.

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

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I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

Thank you for response. So a "standard" quad bri card can serve the KX-TD1232, no need for anyt special fxs similar equipment. Would like not to invest much more money into the Panasonic, I wan't it out :P BR Amund -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Peter Svensson Sendt: 14. juni

Hey folks, I have a Digium TE205P working as a man in the middle: PRI line -------- Asterisk/TE205P -------- PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm

I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia