search for: 18924

Displaying 20 results from an estimated 32 matches for "18924".

Did you mean: 1892
2003 Sep 28
1
Forwarding SIP over IAX problem: No One Available
...p") in new stack -- Called madsen:password@nattedbox/55555@from-sip -- Hungup 'IAX2[207.61.247.201:4569]/1' == No one is available to answer at this time And here are the config files: *** NAT'd Box *** extensions.conf --------------- [globals] PHONE1=Zap/1 PHONE1VM=18924 CALLFILENAME=foo FOO=foo [intern] include => outbound-fwd include => from-sip [outbound-fwd] exten => _7.,1,Dial(IAX2/madsen:password@liveipbox/${EXTEN}@intern) exten => _7.,2,DISA,no-password|intern [from-sip] exten => 18924,1,Dial(${PHONE1},30,t) exten => 18924,2,Voicemail(u...
2003 Oct 06
5
Help with questions for initial Asterisk wizard (GUI)
...to setup registration for FWD, IAXtel, SIPPhone or iptel? + -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +------------------------------------------+ -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com...
2003 Sep 10
9
Free World Dialup (FWD).
Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers
2003 Oct 06
2
Asterisk, X-Lite and iLBC..still..
Hi, Has anyone managed to get X-Lite to work with Asterisk using the iLBC codec.. I have just tried updating the the latest version 1079 (BTW this new version supports up to 10 proxy configurations, Not that I can see a reason to have 10 proxies setup, I would rather have the ability to transfer calls).. I can make a call using iLBC but the sound that I hear is just a lot of pop's and
2003 Oct 06
1
MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash
...If you have any idea's on things I can test for, it'd be greatly appreciated. Thanks in advance, +-------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com | +-------------------------------------------+ | @| leif at hacklocalhost dot com | | FWD| 18924 | | IAX| 1700-363-0761 | | SMS| sms at hacklocalhost dot com | | ICQ| 3445119 | |iptel| 8972-1969 | |sipph| 1-747-386-1618 | +------------------------------------...
2003 Sep 11
1
How much to charge for Asterisk installations?
...mail, call parking, call queueing? I'm trying to get an idea of how a standard Asterisk installation might be charged to a business. Thanks in advance, - -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus @: leif.madsen@sheridanc.on.ca ICQ: 3445119 FWD: 18924 IAX: 1-700-674-5480 -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.8 for non-commercial use <http://www.pgp.com> iQA/AwUBP2DBheoKt3kNIKTVEQJfaQCfe1wqiKxU7zgNTrV80w+2DwYUwYIAn3HZ PdhQp7NI7zCw+FIHohMbQgOt =BYhb -----END PGP SIGNATURE-----
2003 Nov 27
13
Asterisk behind NAT << How to do it.
...le in pwd /usr/src/asterisk/channels/ patch -p0 < /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen <leif@hacklocalhost.com> http://www.hacklocalhost.com
2003 Nov 04
2
Does externalip= do anything to help with SIP behind a Linux based NAT router?
...g if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +------------------------------------------+
2003 Nov 11
4
OT: Document Control System?
...ingle user license :) (or any money at all actually) Thanks, -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +------------------------------------------+
2003 Sep 26
2
Creating a SIP gateway for use behind NAT
...his is my work around until then. I have built the IAX registrations between the boxes, so they register with each other. What I am trying to figure out is how to use my existing dial plan which worked on the gateway. This is going to work with FWD, so I should be able to receive calls on my 18924 number, and the 55555 welcome line as well. Couple of questions: 1) Should the GW or the second * box register with FWD? 2) Am I basically going to be using a blank configuration on the GW box, and having the second * box with all the fancy dial plan stuff, or the other way around? 3) Am I loo...
2003 Sep 06
7
OT: Creating documentation using a web interface
...as well as maybe a history timeline or something like that? Just some thoughts. If you know of something like this, please let me know. In the meantime, I'll be googling some more. Thanks, -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus ICQ: 3445119 FWD: 18924
2006 Apr 17
2
Email Multipart message
Hi, I use the multipart functionality from the action mailer. But when i send an html message, my outlook opens the mail and shows this: --mimepart_4443b76d12268_6f34..fdacc616831e6 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: Quoted-printable Content-Disposition: inline <!!Message> = = --mimepart_4443b76d12268_6f34..fdacc616831e6-- No html will appear. When
2003 Sep 08
0
Is this use of DISA secure?
...include => local include => iaxtel-out [local] exten => 6500,1,Wait,2 exten => 6500,2,VoicemailMain2 exten => 6500,3,DISA,no-password|intern - -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus @: leif.madsen@sheridanc.on.ca ICQ: 3445119 FWD: 18924 -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.8 for non-commercial use <http://www.pgp.com> iQA/AwUBP11haOoKt3kNIKTVEQLSgwCZAWWAcBaur03DpMDCZ2FEZqjH2IQAn2fI TTMU0TvyhciYBjZQrtSpvdD0 =FIAx -----END PGP SIGNATURE-----
2003 Sep 09
2
Has the "allow=all" function changed in sip.conf?
I had posted earlier asking about a Snom200 communicating with a C7960 and lots of noise in one direction. Turned out the problem was created by me removing the allow=all statement in sip.conf. Someone had suggested that statement is no longer needed, and using allow=ulaw, etc, had an issue where one or more deny's had to be used as well. By adding allow=ulaw in the sip.conf file, the Snom
2003 Sep 10
1
MOH - White noise, static
...Has anyone had this problem and resolved it? I am calling from analog device to analog device, using call parking. Suggestions? Using the newest CVS also. - -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus @: leif.madsen@sheridanc.on.ca ICQ: 3445119 FWD: 18924 IAX: 1-700-674-5480 -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.8 for non-commercial use <http://www.pgp.com> iQA/AwUBP1/CJuoKt3kNIKTVEQIDZwCfRxNIu0YofAhMie0x4+S5PChvYfMAoNBq LT0EPMh2fW8rXQgfXA0pzZIH =svR7 -----END PGP SIGNATURE-----
2003 Sep 11
1
newbie - sip, pxb, ata, nat
hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for
2003 Oct 22
2
new codec for grandstreams
Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines
2003 Nov 20
2
Change the all announcement
Hello all I would like change the all announcemennt(Voicemailmain,Voicemail etc.). But I don't know how to change the these each prompt. Do we have any guide book for this? Please teach me about changing the voicemail or other prompt. Thanks
2004 Jul 28
1
Access voicemail from Cisco 7960
Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? Regards, Evert
2003 Nov 20
1
Linux Voice Mail Application??
Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV