similar to: Can't hear the caller

<div><FONT size=2>Hello,</FONT></div> <div><FONT size=2>two questions: </FONT></div> <div><FONT size=2></FONT>&nbsp;</div> <div><STRONG><FONT size=2>1: How can I open/enable network connection to B?</FONT></STRONG></div> <div><FONT

Hello, I just started using asterisk, and have a question. I have setup two asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1 FSX modules) and is connected to the PSTN. B has same, but is NOT connected to PSTN. I want to configure B to call A via iaxtel, and connect to the PSTN using A's line. How can I configure iaxtel dial plan for B in extensions.conf? I want to be

<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

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

Hey, I am a new to asterisk and softphones. Ihave recently installed and configured linux and 2 xlite clients all in linux fedora core 6. I have also made a dial plan for the two users. But when i dial from one xlite client to another i can hear the ring tone but when i answer the call i can not hear any sound. I have checked my microphone and its working fine. Please could anyone help me on

Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 "No proxies are up - can not send message to anyone" Xlite is not able to register to the

I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice. They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice. The microphone and speakers are

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

Hi all, I am attempting to call from softphone to softphone, I am using X-lite to call another X-lite. I get the phones to call each other and finnaly connecting, but cannot hear the voice at all. Is there any ideas as to why this is happening. (I don't have sound card in my linux server. I need one in my linux server ??) PS: callonhold is working but cannot hear the music too. look at

I've been working on the local side of asterisk for several days, and I have the in-house dial plan pretty well corn fingered to my satisfaction. Today I began working on the other side to make asterisk do things like place an outgoing call to PSTN and route an incoming call from PSTN. I'm using a TDM11B with a single fxs and a single fxo. My analog handset is plugged into the port

We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either.

I have a question for those of you out there using VoicePulseConnect for incoming did I have in my Realtime extensions Database (the x's are replaced with my phone number) context = voicepulse-in-01 exten = xxxxxxxxxx/ Priority=1 app=NoOp appdata = Incoming call with no callerid on xxxxxxxxxx However it never triggers I also tried using one of my other providers (voipjet for outbound) and

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

I've heard good things about VoipJet here, so I was going to set up an account. Then I noticed their Terms of Service here: https://www.voipjet.com/tos.php Several things there are very concerning to me, and I'm interested in what other people here think of them. * The ToS specifically forbids use for any call relating to medical, financial, or government matters -- as well as any

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Somebody know if they are down? Let me know, Julius C. Barber ventas@gringotel.com www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060509/924605b6/attachment.htm

No Dice so far, anyone now how to add anIAX trunk? What are the settings exactly? I have added everything but I do not know what are the registration strings? Jonathan

Hi all, I'm using VOIPJET to make international calls with an IAX2 connection between my local asterisk server and their server(s). At times I seem to have a problem if 5 or more international calls are made at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the asterisk server uses this DSL line). Today I switched the codec from ulaw to ilbc in an attempt to lower