similar to: 1.6.2.0-rc1 intermittent voicemail problem ?

Hi all, I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

Is it possible to receive a call via IPKall through IAX connectivity without registration? If so how to set it up. I've run-into and old link; http://forum.voxilla.com/ipkall-support-forum/ipkall-beta-testing-iax-connectivity-without-registration-26728.html -- Joseph

hello- i was messing around with a did from ipkall.com, and asterisk seems to be ignoring the context specified in the sip config. in sip.conf, i've added: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = "ipkall incoming" <7508> nat = no in extensions,conf, i have: [remote] exten => 7508,1,DISA(1111|internal) [internal] exten =>

Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing

HI I am having a problem with the new IPKall number I just got. Other sip numbers work that cost money. The problem I am having to one way audio. I can not hear the outside party when they call in. Is there something special about IPkall I'm missing?

nmbd is already running unfortnately... i can say that when running samba version 3.11 on Solaris it is working also with 3.23... ???? thanks for your suggestion tough ! /J > Date: Wed, 14 Mar 2007 15:59:17 +0700> From: garasi9@gmail.com> To: jflory@aeiconsultants.com> Subject: Re: [Samba] Re: can?t access share by name, but on ip> CC: samba@lists.samba.org> > Hi...>

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part --------------

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

I use IP Kall to forward my missed cell phone calls to. This way, if my phone is off, or out of a service area, calls will go to my * box. Concurrently, all incoming calls to my * box cause it to dial my local extensions at home, my extension at work, and my cell phone via NuFone. Problem: A loop can be created if my cell phone is not on. Say a call comes into my * box, it uses NuFone to call my

Trying to use gtalk: -- Executing [andy at ipkall:2] Dial("SIP/ipkall-00000000", "gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr =

Hey guys, extension to extension calling seems to work but when I setup my ipkall number, I keep getting this error: pbx.c:1317 pbx_extension_helper: Cannot find extension context 'INVALID' I set the extension to 100 (a valid extension) in ipkall control panel. Anyone have any ideas -------------- next part -------------- An HTML attachment was scrubbed... URL:

Calling 7777 from a local extension on my local network, I get good voice quality from asterisk, and asterisk reliably recognizes my dtmf input. I set up a sipphone trunk (free) and called in to it via a separate sipphone account on another computer, and got slightly lower, but still good, audio quality. I set up a FWD trunk (free) and called in from the other computer, and got somewhat lower

Hi all, I've searched high and low on this and found nothing of help. I'm using v2.6.2 and trying to write a function that will count how many people from a dataset fall under a poverty line of 50% of the mean income. a9 is my 100-element vector of incomes. I want pa9 to be my vector that counts how many of these are classed as in poverty. My problem is that my command if(a9[i] <

Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which work very well but I still had the 30 day problem. Is there a way that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm

Hi All ! i?m running Clearcase (IBM Rational) and have some strange problems ... when i use Samba ver 3.21b i can?t access the samba share by name (\\servername\sambashare) but i can access it on ip (\\192.168.1.100\sambashare\) i?m running debug level 10 and it seems like it can?t authenticate when access on netbios/dns/host name but on ip it can ? when running samba ver 3.23b from

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the