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This Friday at 12:30 PM EDT We hope to hear more about Astricon and the 1.6 version. A UK legal professional, John Halton of Cripps Harries Hall LLP, joins us to discuss how the law is coming to terms with VOIP. We also expect a visit from Arick of IPKall about what's cooking with them. Most of all, we expect you, the community to share in this experience! For more info see:

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

1.6.2.0-rc1. I am having trouble with voice mail intermittently not working correctly on CHANUNAVAIL. (it may happen for other statuses too, haven't checked). Basically here's what happens: -- Executing [1651xxxxxx at mydids:1] Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack -- Executing [s at macro-phone:1]

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 =>

Fridays @ 12 Noon EST - 9 AM PST - 17:00 GMT etc ad infinitum THIS WEEK Among other topics, we'll be hearing about TringMe (http://www.tringme.com) a Phone Me solution that works for just about anyone with audio and a headset. Anything else about VOIP is permitted, but the following words are permanently banned from the conference: "Intermittent", "Unable to reproduce"

Hi, I'm not sure at this point who will be with us, but there's always something to talk about on the conference, Fridays at 12 Noon Eastern Time, 9 AM Pacific, 4PM GMT. I have invited Garrett Smith whose blog you should consider following. PSTN: (724) 444-7444 Call ID: 22622 SIP: exten => 1234, 1,Dial(SIP/123 at 66.212.134.192, 60, D(22622# ${MY_PIN} #) ) If you have no PIN use 1#

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?

The poll is still open here: http://food4wine.ning.com/poll Here is a CSV file of the 99 answers. http://voipusersconference.org/poll/ There is also an XML version, but it was created by Excel so I don't know if it's worth dealing with: http://voipusersconference.org/poll/results.xml Because I screwed up (mea culpa, we're all human, or almost) there were less answers now than

Hi, Tomorrow, Friday Nov 16th, 2007 at 12:30 PM, we'll be exploring a simple, well-commented example of an AGI script for asterisk. I have absolutely nothing against GUI, but if you want to unleash the real power of asterisk, you'll need to get into AGI (or pay someone else to do it). Because asterisk solutions are mostly limited to your imagination (and a hired hand may or may not have

I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian

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: