similar to: Cellsocket with @home

I need help from someone who has a working cellsocket, I have received couple email of people who wanted to help, but they just think they know how it supposed to work, but they don't have a working units, and they confused more...I need someone with a working solution to get my cellsocket going.. Thanks!!! Write offlits @ mawise (AT) mail.com -------------- next part --------------

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

Nik, Just curious - what is your telco setup? Do you have PRI with the specified D channels? You need to make sure that your telco is set up to have the D channels on 16 and 47. When you first start Asterisk, or when you log on to the CLI, do you ever see messages stating the B channels are successfully started? Let us know. -MC -----Original Message----- From:

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

I recently purchased a Cellsocket, which is a cradle that holds some older Nokia GSM/PCS phones and converts them to an FXS interface. My test phone is a Nokia 5190 on the T-Mobile GSM network. After going through the ordeal of unlocking the phone (T-Mobile provided unlock codes that didn't work), I finally got it up and running. The good: - Disconnect supervision works - Outstanding sound

When I posted my last message about the cellsocket, I hadn't thought to try incoming calls. (I bought the cellsocket to use only for outgoing calls.) Unfortunately, I cannot get the cellsocket to work for inbound calls. CallerID does not work even though I have a GSM phone. The cellsocket answers the phone and then starts ringing any attached phones after answering. Most importantly,

Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is

Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet.

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone

I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is

Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, it?s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : <vendor> ----SIP--? <Asterisk> ?----IAX2---? <Asterisk with Dhadi channels> Here: -- Executing [966505103150 at from-internal:1]

I am planning on dabbling with some VOIP providers. I was thinking of Teliax first. My thinking is that the first LD call would go to teliax and the second (etc.) calls would go out to the PSTN. I could then verify bandwidth and quality to decide when to add more trunks and to Internet connections. I have been doing some concept testing with FWD for toll free calls, but I am using 393 as a