search for: ipkal

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'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 i...

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 something wrong? I've tried switching dtmfmode to all the options, but still nothing. Thanks for your help! - Josh...

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...

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

...chine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ; [sip] type=peer username=fiducia_ag fromuser=fiducia_ag authuser=fiducia_ag secret=password host=64.56.64.64 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 [ipkall.com] host=voiper.ipkall.com context=from-ipkall dtmfmode=rfc2833 insecure=invite type=friend canreinvite=no disallow=all allow=ulaw Extension.conf: _________________ [from-ipkall] exten => 901835,1,NoOp(from-ipkall) exten => 901835,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM}) exten => 901835,...

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,DIS...

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?

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] Dial("SIP/ipkall-trunk-14838bc8", "SIP/1651xxxxxx,25") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 [Sep 15 14:19:34] WARNING[26239]: app_dial.c:...

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

...file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895 ,4,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----2062036895-----Closer---------------999-----1) exten => 2062036895 ,5,Hangup() 2062036895 is with IPKall. What could be the wrong. ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090114/fc6fbf13/attachment.htm

...reinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip[1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com nat=no[1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com nat=no I pasted the log here -> http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info) and did not have the same problems. Any suggestions would be great. Thank...

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,...

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

...that if it takes less than 10 seconds from the time I forward a call to my cell, and a forwarded cell call comes into my * box, then it must be the beginnings of a loop), and if less than 10 to send the call on to be hungup, or else process it normally. Thus, I submit the beginning of my [inbound-ipkall] context from my extensions.conf: [inbound-ipkall] Exten => _X.,1,GotoIf($[${ZAPCALLEDTIME} = ""]?3) ; This is to see check that ZAPCALLEDTIME exists at all - I am not sure I do it correctly here. Exten => _X.,2,GotoIf($[${EPOCH} - ${ZAPCALLEDTIME} < 10]?1000) ; This determines...

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 PTSN, get the dialtone, and enter the password. In the CLI I'm getting duplicate/extra/incorrect digits. I've tried dtmfmode=auto, dtmfm...

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...

hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. I am experimenting with asterisk, for home use. thanks,

We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This als...

Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode:...