similar to: re: putting 2 SIP channels together - hangup issues

Hello all, I'm having a bit for a problem with the dial command limit option. I have the following dial command (executed from inside the a2billing agi) AGI Script Executing Application: (Dial) Options: ( IAX2/username@voipjet/18005551212|30|HL(60000:20000:00000)0) Now, from what i read in the wiki, this is supposed to limit me to one minute (60000 ms), and warn me when there are 20

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.

Hi all, i'm trying to integrate cepstral and asterisk, and i have a problem i'd appreciate any help with (i know it's a bit tangential, but i figure this is the place with the most knowledge of app_swift and asterisk). I've installed swift from cepstral.com with alison's voice, and it works fine, from the command line i can do swift "hello there" -o test.wav and then

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of

I've been trying to resolve some quality issues and I was hoping someone might be able to provide some insight. To give you an idea the calls are coming in via a SIP DID and sent out via an IAX2 connection. Latency to both the SIP equipment and IAX equipment are around 80ms with 0 packet loss accoridng to ping tests. The server is located in a data centre so bandwidth is not an issue. Most

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

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

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

I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>

One would assume they have better things to do as they are quite busy. I think this is just a proactive measure meaning they say you cannot do it upfront so that in the event of a problem, it was predeclared. As to the rest of the TOS, I could be wrong but I got the impression that the owner of VoipJet speaks English as a second language due to some email exchanges. If that is the case, the TOS

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

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

So far my experience with TOS has been that most of them are pretty odd. No one wants the liability of a stock trade gone foul or a call to the doctor that gets disconnected. Essentially, those things in the TOS are just a CYA. They are un-enforced but should someone decide to attempt to sue based upon a financial loss, the ITSP is covered. So, yep. That is weird but not unexpected. Heaven

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

Hi, I just setup a VoipJet test account (one with 25c credit) to test, they seem to offer good rates to 02 Uk mobiles :) Anyway, everything went ok, iax.conf amended and extensions.conf too, however when I try to make a call I see:- rt*CLI> -- Executing SetCallerID("SIP/2008-d747", "4153574000") in new stack -- Executing Dial("SIP/2008-d747",

I try to connect to voipjet, but I get always busy, ... it worked yesterday, ... no changes on my side.... -- Executing SetGroup("SIP/615-829b", "iax-voipjet") in new stack -- Executing Dial("SIP/615-829b", "IAX2/17567@voipjet/011886228357765") in new stack May 20 18:16:26 NOTICE[9733]: app_dial.c:973 dial_exec_full: Unable to create channel

How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet Example in my extension.conf I have: [iaxtel] exten => _1700NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1888NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten =>

Hi List, I have set up the following in my extensions.conf ; local numbers look like 0262XXXXXX ; but must be dialed 262 262XXXXXX exten => _0262XXXXXX,1,Dial,IAX2/543@voipjet/011262262${EXTEN:4} exten => _0262XXXXXX,2,Dial,IAX2/jhiver@NuFone/011262262${EXTEN:4} exten => _0262XXXXXX,3,Congestion It did work for a while, however when dialing I get: stargate*CLI> -- Executing