similar to: RJ11 x RJ45

Greetings. I've plugged 3 analog lines on an ethernet cable in an Khomp card to receive it's incoming calls. Without any configuration, when I call those numbers the asterisk server automatically answer the call and play the default music. The problem is: I need to discern the lines and redirect each one to his respective extension. Since they doesn't got any Caller ID Service the

Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens

>what is the wire combination/sequence >for a successful installation? http://www.google.ca/search?hl=en&c2coff=1&q=rj-11+rj-45+site%3Alists.digium .com&meta= hth

See: http://lists.digium.com/pipermail/asterisk-users/2004-September/063348.h tml -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kumara Jayaweera Sent: Tuesday, June 14, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease

<div><BR>Hello,</div> <div>&nbsp;I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same question....how do I go

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the

Hello list , I?m putting to work a new asterisk box. I?m running * 1.0.3 with (one ) Wildcard TDM ( 2FXS* 2FXO) and (one) E-100P. Both boards are working well. ztcfg don?t show me any error. zttool list both cards as "OK" But when i run asterisk on verbose mode i get those errors : [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?

Hello all, I have set my QoS solution and now I am facing a little problem... When I ping to my server it has some lost packages: Estatísticas do Ping para 172.30.0.1: Pacotes: SENDED = 1029, RETURNED = 880, LOST = 149 (14% de perda), Aproximar um número redondo de vezes em milissegundos: Mínimo = 0ms, Máximo = 686ms, Média = 105ms If I disable my QoS ping stats to be ok. I even have

Hola Comunidad, se me presenta el siguiente problema Tengo dos Tablas, la TablaA y la TablaB y en ambas tablas tengo un campo en particular por el cual haría el cruce, algo tipo un Numero de Pasaporte o una Placa de un carro, valores que son únicos por registro. Entonces necesito hacer un cruce de las 2 tablas pero que en vez de que me queden los comunes en ambas tablas, necesito que me queden

Hi list. I have Asterisk installed on a Debian 1.8 6 64-bit. What happens is the following, some channels are not being hangup properly. They run the hangup in dialplan, but the output of the command "core show channels" shows several channels with status "rsrvd." Checking the server's memory, the "top" command shows multiple processes and stopped using the

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

Guys, I saw in the Asterisk documentation (queues.conf) that members can register with technologies such as SIP, Dahdi and Location. But I have a specific need for members to be registered as Khompchannel. Ex: member => Khomp/b0L1/9200 But reloading module app_queue.so when I run the command "queue show", the member registered as Khomp appears as invalid:

Hi guys, just to let u known the pinout for the adapter : Adapter for connect the E1 telco lines on my digium card DB9 RJ45 3 1 8 2 2 4 6 5 Adapter used for connect the Hicom150 traditional PBX on my digium card (cross connection) DB9 RJ45 3 4 8 5 2 1

Hi, I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on it...only a TAE connector. I'd like to create an adapter so I need to know which TAE pins to connect to RJ 11 pins. Is there anybody who knows where I can find a schema of that adapter? Single connector pinout may help too. TIA Giorgio Incantalupo

Hi all, I just got my long awaited FXS modules for a TDM400 card, and was a bit puzzled about how my phones and the FXS interface worked together. Here goes: In Sweden (that's where I am located) phones traditionally are connected in daisy chain configuration. This means that a phone in on-hook condition passes the line through back onto the outer pair of the RJ11, back into the wall

Hello all, I am trying to make a filter into my QoS rules and I founded that when I try to use filters u32 and with fwmark they do not work together. This is the filter I use, just and example, for u32: $TC filter add dev $DL parent 1:0 protocol ip prio 1 u32 match ip sport 22 0xffff flowid 1:10 This is working fine. Now if I try to mark a package that I want it to go to the same

Greetings. I got two asterisk servers, one is connected to another via sip trunk. The incoming calls are routed to the time period an if matches is transfered to the designed extension. If don't, is redirected to a second time period. Then, if the call matches the second time period it need to be transfered to the trunk that directs to the second server. How do I do to configure it this way?

Hi, Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection? thanks Eduardo

Greetings. I've run into a trouble when trying to add a new Win7 machine on a domain. The domain is controlled by a server running Samba + LDAP (samba compiled with ldap support), on a Debian 5 OS at the local network. I've added the machine name to the LDAP three through phpldapadmin using the option "Samba3 Machine" on the related submenu and via terminal on samba. Then I