Re: [asterisk-users] Asterisk from Debian Packages
On Sun, 10 Dec 2006 20:54:10 -0500 Paul <[EMAIL PROTECTED]> wrote: > If you run etch before it is released as stable, you might run into > problems that are over your head. I have run into a few that weren't > over my head but they were very inconvenient. Yes Paul, I'm running 2 etch with asterisk, but it is my own risk. In Debian I trust. Charlie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with Cisco ATA 186/MGCP
On Mon, 18 Apr 2005 19:32:20 -0300 (GMT+3) navarrocarlos <[EMAIL PROTECTED]> wrote: > CA0orCM0: 192.168.1.254:2727__ I changed it for: CA0orCM0: 192.168.1.254:2427 And it worked. Now I'm trying with eyeP Phone in windows. CHarlie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+Oracle
On Fri, 21 Jan 2005 09:42:15 -0800 (PST) R A <[EMAIL PROTECTED]> wrote: > Have some bady working asterisk with oracle? I will use odbc. I did use odbc with MySQL without problem. regards Charlie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2100 and Asterisk one-way audio
Hello People, I have a Sipura SPA-2100 with default configuration and the last software upgrade, and a * from Debian Sarge with the simple configuration: [general] port = 5060 bindaddr = 0.0.0.0 [103] username=103 type=friend secret=qaz123wsx qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="X" <103> allow=all I have the normal codecs (Debian Installation). The Sipura was registered OK, I make a call, I listen but the other site cannot. I can call, with the Sipura, direct to other SIP Phone without problem. I change the Sipura SPA-2100 with a Sipura SPA-1000 in the same peer and it work fine. It is the same problem in other Sipura SPA-841. Do you have any clue about it? Thank in advance Charlie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with CISCO, SIP and Asterisk
Hello People, I'm newbie in * 1.0.1, running a Linux 2.6.7 in a Debian Sarge, and this is my situation: ++ +-+ | Sip Server |-|CISCO PSTN GW| ++ +-+ \ || \ || \ +--+|| | Asterisk |= +--+ The * and CISCO are authorized by the sip. The call is coming from PTSN via CISCO to *. I'm seen the sip debug in the * console, I see that * send "demo-enterkeywords" but in this moment the CISCO hangup the connexion. In the SER server the CISCO send BYE to SER but nothing to *. If the call is coming from the x-lite authorized by the same SIP server, there are no problems. The peer configuration in the CISCO 5XXX is the same that: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cisco%20FXO My sip.conf is: [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 context=from-sip autocreatepeer=yes register => ::[EMAIL PROTECTED] [666] context=local-phones type=friend user=666 secret=666 auth=md5 host=dynamic defaultip=192.168.10.167 reinvite=no canreinvite=no qualify=1000 callerid="diavolo" <666> disallow=all allow=ulaw My extensions.conf is: [default] include => mainmenu include => lan-phones [mainmenu] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,15 exten => s,4,ResponseTimeout,35 exten => s,5,Background(demo-enterkeywords) exten => 1,1,VoicemailMain() exten => 1,2,Hangup exten => 2,1,Playback(demo-echotest) exten => 2,2,Echo exten => 2,3,Playback(demo-echodone) exten => 2,4,Goto(mainmenu,s,6) exten => 3,1,MusicOnHold(default) exten => 3,2,Goto(mainmenu,s,6) exten => 4,1,Playback(demo-thanks) exten => 4,2,Hangup exten => t,1,Goto(4,1) exten => i,1,Playback(invalid) [lan-phones] exten => 666,1,Dial(SIP/666,20) exten => 666,2,Voicemail(u666) [from-sip] include => mainmenu include => lan-phones Asterisk show while running: linux*CLI> sip show registry HostUsername Refresh State sip.example.com:5060 105 Registered linux*CLI> could you give me some clue about it? Thanks in advance Charlie pgphhTuwRvNix.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
