similar to: Cisco Call manager

Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem

Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying "trabajo mensjes" would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton

For outbound only, I have traditionally recommended VoipJet. They just recently has a spat of issues that seem to have resolved though. I am still using them via their east coast server and it seems to work quite well again. Cost is around 1.3 cents minute I believe. Use IAX and g711 for best quality to VoipJet. Thanks, Wiley -----Original Message----- From:

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :)

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

Guys.. which free softphone is the best,grandest,most recommended one out there? based on your own experiences..

Guys. How do those callcenter apps work with Asterisk where a call comes in and * send a URL and some screen popup up based on callerid or something or username or id and shows all the customers info? Anybody done that? What do you need to do that? If you are using ATAs or IP Phones, how do those integrate with the PC so the screen would popup?

Guys, Im looking for a good text file editor for asterisk config files that can be embedded on a web page for online editing (on an interface), any recommendations? ________________________________________________________________ Anton Krall Direccion General Intruder Consulting A Division of IntruderEnterprises S.A. de C.V. www.Intruder.com.mx www.IntruderStore.com.mx Tel. 3872-2200 ext. 201

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

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 was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

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

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

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

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 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 =>

I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? Regards AK

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, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?