similar to: voipjet anyone?

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

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 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

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

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

Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks! G. ----- Original Message ----- From: <asterisk-users-request@lists.digium.com>

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, Is there anyway to eliminate the color coding (for example [1;36;40m) to be stored in asterisk log file? Regards. __________________________________ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/

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, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the

I've been searching the forums and on the list to see if this has been addressed. If it has, could someone point me to the thread to fix or at least acknowledge it is an issue and what is causing it. Posting to the list was last resort as I couldn't find a solution anywhere else. Setup: Asterisk@Home 1.3 (this is my first system, so path of least resistance) Digium TDM400P (2 FXS on ports

I ran into a problem where systems without DNS entries could not connect to the rsync server with the IPV6 patch applied. Here is a fix to the problem. Basically they were checking the list of IP addresses returned by getaddr even if getaddr failed. I just changed it so they only check the list of IP addresses if getaddr succeeds. Any comments on this please email me directly because I do not

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw) exten =>

On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote: > > You specify the transport in the SIP URI. For example: > > sip:test at example.com;transport=tcp Hrm. This is probably going to be pretty basic, but some googling didn't seem to come up with anything. How do you do this when you are assigning to a variable in extensions.conf such as: FRED=PJSIP/outgoing/sip:fred at

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

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

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

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson