[asterisk-users] ZapRAS, pppd plugin option

3060 700 pts/1R+ 16:10 0:00 grep asterisk -- Kind Regards Etienne Pretorius ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] E1 Virtual Callcenter

Hello List, I just have a query is it possible to have 2 or more telephone number mapped to the same E1 line and if so will the TE120P card pick up the last 4 digits of each number - as it is currently doing for the one? -- Kind Regards Etienne Pretorius

[Asterisk-Users] [Fwd: Asterisk as an Ascend box]

Sorry not sure the mail was sent to the correct address: -- Kind Regards Etienne ---BeginMessage--- Hello all, I was just wandering if it is possible to make Asterisk become a replacement for an Ascend box and then utilise the unused channels to make outgoing and/or incoming calls? Possibly

[Asterisk-Users] Trying to cut out the paper work...

Hello everyone, Ok. I am at a bit of a loss and would like someone to point me in the right direction...(btw www.google.co.za did not give me ANY solutions). The issue at hand is simple, I get asterisk (1.0.9) to answer the incoming call with no problems... it does the fax detection

[Asterisk-Users] VoiceBlue GSM

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a

Re: [Asterisk-Users] VoiceBlue GSM

just register the voip2GSM devise to Asterisk and then it is ready to receive and send calls just like any other sip phone. Cost of this is around 400 USD / UNIT When you talk about sms capability, dyou want to originate or receive SMSs through the devise? Selon Etienne Pretorius [EMAIL PROTECTED

[Asterisk-Users] UDP Sip Data: GS Grandstream - remote office

Hello all, Tried to get remote office working and found out that the GS Budge Tone 100 takes the ip address inside the UDP packet data (SIP) that asterisk writes to.The Asterisk server is currently setup with a ADSL ZyXEL PRESTIGE 600 series router. My isp does dynamic ip assignment - so I

[Asterisk-Users] Codecs and * pass through...

Hello all, I came a cross a problem yesterday that I don't quite know how to solve. I am trying to use * to connect to net2phone, and have a net2phone MAX IP-10 connect to net2phone. From the settings on http://www.voip-info.org/ it was easy to get asterisk to connect to the network - acting

Re: [Asterisk-Users] Codecs and * pass through...

ing to talk on the proprietry protocol. For G723.1 passthrough, you just allow it, and it should work fine, as long as you do not try playing any voice prompts to the channel. good luck. regards Clive = Phone I.T. http://www.phonehome.co.za On 13 Apr 2005 at 8:52, Etienne Pre

[Asterisk-Users] User Regerstation, allowing non-registered users on *

Hello *users, I would like to know how one would go about to allow every-one that wishes to connect to my * machine to connect without a registration being placed in the conf files. Would this be achieved through a Database that will lookup the UserName and Password and if it does not exist

[Asterisk-Users] User Regerstation, allowing non-registered users on *

Hello *users, I would like to know how one would go about to allow every-one that wishes to connect to my * machine to connect without a registration being placed in the conf files. Would this be achieved through a Database that will lookup the UserName and Password and if it does not exist

[Asterisk-Users] No Voice to POTS.

Hello Every body. I am quite stuck at the moment. I am using X-Lite, and you see when I call say a Cell phone number then the call works fine - voice is on both sides. But when I call through to a Land line then no voice is heard on the other side from me, but I hear the person speaking to me.

[Asterisk-Users] Zaptel group members - dial out on a availible port via trial and error?

Hi ya-all. Little question that has been bothering me somewhot. Say I have only 2 out going analog phone lines. Some1 in the office decides to call their a client... so the Dial command it using a group and it will start at the first Zap channel listed in the group. But now what if I disconnect

[Asterisk-Users] How do you do Line Hunting in Asterisk?

I have come accoross the fact that * can't handle if there is no dialtone So out of interist, can you do Line hunting in * in a sequencial manner and can you also do so in a random fasion? -- Kind Regards Etienne ___ Asterisk-Users mailing list

[Asterisk-Users] Zap - What is going on?

Ok - I was told that you set a group for Zap channels. So I tried to make use of my Zap channels so the 2 I am interisted in is channel 3 and channel 4. I make Channel 3 in use bu calling a line... then I try to call another line so expecting to have Zap channel 4 open and allowing me to

Re: [Asterisk-Users] Zap - What is going on?

see your text as a variable and it tried to call the number $EXTEN on Zap/g2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Etienne Pretorius Sent: 04 April 2005 13:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users

[Asterisk-Users] X-Lite to Zap, no Voice on other phone!

Hello, The problem is - and i was wandering if anyone knows the solution - is that When I dial from my windows machine, to an external phone line through Zap, then the receiving party does not hear my voice - but when the receiving party calls me back, then we have voice on both sides. What

Re: [Asterisk-Users] X-Lite to Zap, no Voice on other phone!

Wilson Pickett wrote: The problem is - and i was wandering if anyone knows the solution - is that When I dial from my windows machine, to an external phone line through Zap, then the receiving party does not hear my voice - but when the receiving party Do you have Transmit Silence=YES on the

Re: [Asterisk-Users] X-Lite to Zap, no Voice on other phone!

Etienne Pretorius wrote: Wilson Pickett wrote: The problem is - and i was wandering if anyone knows the solution - is that When I dial from my windows machine, to an external phone line through Zap, then the receiving party does not hear my voice - but when

Re: [Asterisk-Users] Can I set queue not to hangup?

look under Queues.conf, use a context=something. A small menu can be set up so that the caller can drop out of the queue and go stright to another service. Mind the small menu only supports single digits. Queues.conf [QUEUE-Sales] musiconhold = default strategy = rrmemory timeout = 15 ; How

[Asterisk-Users] {extensions.conf} Dialing plans with queues....

Hi all, This is the situation: I have a call coming in from the POTS line and I pass this through to the [incoming] section by including the [incoming] inside [sip]. Then s,... starts. It picks up the call and then places the call in a "Reception" queue. This is the problem: When the call

Re: [Asterisk-Users] {extensions.conf} Dialing plans with queues....

Oh well, found it with some searching Add the menu as a context in queue.conf and then define that menu in extensions.conf. Mind only single digit numbers are valid. Kind Regards Etienne Etienne Pretorius wrote: Hi all, This is the situation: I have a call coming in from the POTS line

[Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

Hi All * users... Question: In extensions.conf - I am awaire that you can use macro's but what I am wondering about.. is that can you create a macro to do dynamic Zap channel allocation for a out going call? I don't want to reserve a channel/port in the TDM400P card for Out break calls,

Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

Never Mind. oops. I just needed to play around with some syntax. Zap/1,2,3,4/$EXTEN Ps: Is there a better santax because 1-4 doesn't work. Kind Regards Etienne Etienne Pretorius wrote: Hi All * users... Question: In extensions.conf - I am awaire that you can use macro's but what I am

Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

Nope - I jumped to conclusions. It just tries channel 1 the whole time. Any ideas any1 Kind Regards Etienne Etienne Pretorius wrote: Never Mind. oops. I just needed to play around with some syntax. Zap/1,2,3,4/$EXTEN Ps: Is there a better santax because 1-4 doesn't work. Kind Regards Etienne

Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

Thank you very much, that sorted out the problem. Kind Regards Etienne Steven Critchfield wrote: On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote: Never Mind. oops. I just needed to play around with some syntax. Zap/1,2,3,4/$EXTEN Ps: Is there a better santax

Re: [Asterisk-Users] Dynamic Zap/{channel} allocation for out going possible?

ind Regards Etienne Technical Support Kingsley Technologies Etienne Pretorius wrote: Thank you very much, that sorted out the problem. Kind Regards Etienne Steven Critchfield wrote: On Sat, 2005-04-02 at 18:39 +0200, Etienne Pretorius wrote: Never Mind. oops.