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...om-sip mailbox=201 callerid="101" <2125> nat=yes My extensions.conf has: exten => 101,1,Dial(SIP/101,20,tr) exten => 101,2,VoiceMail,u101 exten => 101,102,VoiceMail,b101 My voicemail.conf has: 101 => 2348,Emma, kidjue@yahoo.co.uk Any ideas are most welcome. -- Rgds, Juki _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

...om-sip mailbox=201 callerid="101" <2125> nat=yes My extensions.conf has: exten => 101,1,Dial(SIP/101,20,tr) exten => 101,2,VoiceMail,u101 exten => 101,102,VoiceMail,b101 My voicemail.conf has: 101 => 2348,Emma, kidjue@yahoo.co.uk Any ideas are most welcome. -- Rgds, Juki

Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, "Juki" wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to } sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones } on two different PCs. My problem is that whe...

...edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Searchable Mailing Lists & NooB Question (Rich Adamson) 2. RE: TelIAX troubles (Scott Bussinger) 3. Re: Asterisk not acceptingmultiple SIP phone logins (Juki) 4. Re: Asterisk not accepting multiple SIP phone logins (Juki) 5. RE: dtmfmode and IAX protocol (Rich Adamson) 6. RE: dtmfmode and IAX protocol (Michael Giagnocavo) 7. Re: Why echo occurs (Rich Adamson) 8. RE: dtmfmode and IAX protocol (Rich Adamson) 9. Re: Why echo occurs (Steve...

I am currently working on a small database project to track company assets. Right now my Database consists of 3 tables. 1 for the equipment, 1 for the users info, and 1 for the type of equipment. I have all 3 databases working on a model test site using RoR. Where i am having trouble is getting the databases to talk together. I can''t find this answer anywhere else. My main table

I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius.

Dear all, before installing asterisk would like to know if it is possible to config the software to forward an incoming call from a sip server1 to a sip server2. I need to route the call to anoter number using another sip server. Thx a lot. Juki ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory:

Hi David > I think the trouble with branches is the lockdown of the root repository > directory. Surely not (at the server) > git svn init --stdlayout https://<user>@llvm.org/svn/llvm-project/llvm \ >

...ion 0.2.Rca to last RC7k and when try to compile zaptel > (after patched it) i've this error: > > make: *** [zaptel.o] Error 1 > > > ------------------------------ > > Message: 23 > Date: Fri, 11 Mar 2005 16:23:56 +0300 (EAT) > From: "Julius Kidubuka" <juki@one2net.co.ug> > Subject: [Asterisk-Users] Voicemail - No Audio Output! > To: asterisk-users@lists.digium.com > Message-ID: <3203.81.199.88.27.1110547436.squirrel@mail.one2net.co.ug> > Content-Type: text/plain;charset=iso-8859-1 > > Hi all, > > I am able to receive...

Hi, I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. Hence phpconfig would be great, however I am having difficulty getting it to work. I have searched the message boards and the wiki, and

Hi all, I have setup two X-Lite phones and an Asterisk box. They are all on the same LAN and have private IP addresses assigned to them. I am able to place a call from either phone but the moment it is picked up (trying to be answered), it goes dead - as in no sound! I get two errors, "Unknown RTP codec 72 received" and "RFC3389 support incomplete." How can I go about this?

Hi, I have installed two X-Lite phones and they're able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX.

Hi all, I am able to receive voicemail in my mail box but when I try to play the audio file attachment, I hear nothing at all (yet the caller on the other end does leave a voicemail message)! Anyone had a similar problem before? Ideas are welcome! Note: I am using Asterisk@Home 0.6 Thanks in advance, -- Rgds, Julius Kidubuka. "My advice to you is get married: if you find a good wife

Hi, I believe there is a script that reads the contents of voicemail.conf and goes on to send the voice e-mail messages to whatever e-mail address specified in voicemail.conf. What's the name of this script and where is it located? Thanks, Julius.

Hello, I have come up with a php script that I believe should be able to send sms alerts to cell phones as specified. I have added an option under the extension account settings (am using Asterisk@home 0.6) successfully. I would like to embed this script into the various specific configuration files so that it performs it function. I have looked at the conf files under /var/www/html/admin, which

Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory:

Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; <?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham <dave.packham@utah.edu> *

Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; <?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham <dave.packham@utah.edu> *

Hi, I have installed two X-Lite phones and they're able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX.