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Hey, does anyone know why i'd be receiving: Aug 28 19:40:04 DEBUG[1875]: ##### Testing 66.27.233.241 with 10.0.10.0 Aug 28 19:40:04 DEBUG[1875]: Target address 66.27.233.241 is not local, substituting externip I get tons of them, usually when the phone is registering/calling/receiving calls. Thanks! Chris

In article <1251.165.146.69.140.1144596935.squirrel@www.ecntelecoms.com>, yusuf@ecntelecoms.com says... > Hi, > > I have had the exact same problem last week. I have not yet solved it. > So instead I am using ooh323, but would prefer to use oh323. Can anyone > help? I'm glad that I'm not the only one :)) Hopefully we'll find sol...

Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, I just want out find out how to do bill recon's when you send calls to a provider. They send me their CDR's, and when I compare it to my * CDR's, some calls are 1 second off, either way. How in general is it done by others? -- thanks, Yusuf

I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I

...mailing list >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > > > >------------------------------ > >Message: 2 >Date: Thu, 19 Jul 2007 16:14:46 +0200 >From: Yusuf <yusuf at ecntelecoms.com> >Subject: Re: [asterisk-users] Upgrade Procedure >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> >Message-ID: <469F71D6.2080302 at ecntelecoms.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed &g...

Hi all, I am using Asterisk CVS, and I am getting a huge delay in dialing SIP. This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP users. So, a user '0251' dials from his phone, the PABX sends it the my Asterisk box, no delay, then I get a 15 sec delay, before it actually dials the end SIP user. 1 -- Accepting call from '0251' to '0834541083' on

Hey all, I have read on voip-info.org that to configure MoH asterisk requires the use of mpg123. I have installed mpg123 and restarted asterisk. But, when i put a call on hold i get this error: May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865 local_ast_moh_start: No class: default Can you help, Thanks yusuf

X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: yusuf at ecntelecoms.com X-Spam-Status: No My third try, humph! Yusuf wrote: > Hi, > > I am trying to understand the RTCP stats in Asterisk. > > 1. I am using the 'h' exten to store the RTCP records in CDRS. > However, only if the > caller hangups does the RTCP values have anything...

Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw

Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 <- 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]:

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

; ; Music on hold class definitions ; [classes] default => /var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces) ;manual =>

Hi all, I have a sirrix bri card, connected to 2 ISDN lines.I used to get a lot of slips on it, so i put the 'master' setting to 'yes'. But every couple of hours the machine completly hangs, and i have to reboot it. I only get 1 or 2 slips,I dont know whats wrong? yusuf

I am using 0h323 on Asterisk CVS HEAD 19/07/2005. I am dialling a h323 gatekeeper. He can hear me, but I cannot hear him. I have a suspicion that it could be the rtp traffic, since he said that they need rtp traffic from ports 4500 - 65000. So in 0h323.conf i set updstart and udpend, and in rtp.conf i set the ports here. a tcpdump confirms there is two way traffic. unfortunately, a 0h323

Hi guys, I am busy planning to implement SE Linux on my asterisk box. Either that or I will use AppArmor from Suse. I just want to know what are others experiences/incidents with SE Linux or AppArmor thanks, yusuf

Hi all, I have a couple of asterisk servers running. When one asterisk server dials another asterisk server over IAX, i want to match that call in both of the cdr's. How do i make both asterisk servers use the same uniqueid for that call, if this is possible. Or is this a dumb question since it is 'unique'. I just want to match a call on different asterisk servers from the

Hi all i have been succesfully using OpenH323 (oh323) for a few months. the versions are: asterisk CVS HEAD 19-07-2005, OpenH323 v1.13.5, PWlib v1.6.6, asterisk-oh323-0.7.2-pre1 I now have moved to Asterisk 1.2.4, so as per the directions i am using: Asterisk 1.2.4 pwlib_Mimas_patch2 openh323_Mimas_rc2 asterisk-oh323-0.7.3 The problem is that when asterisk starts it fails on loading the module