similar to: Asterisk: SIP Gateway or Proxy

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

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

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

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 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

Postfix <http://www.postfix.org> does a chattr +S on its spool directory when it is installed on Linux. This is what is written in the postfix startup script -- # # LINUX by default does not synchronously update directories - # that's dangerous for mail. # if [ -f /usr/bin/chattr ] then CHATTR="/usr/bin/chattr +S" else CHATTR=echo fi -- Is this still required on ext3 ?

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

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, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'.

Hello All, What would be the lists recommendation for an 8 port controller with native support for Centos 5.0? Currently have a 3ware 9650SE which does not... Cheers Steve

On Mon, Nov 26, 2001 at 10:00:39PM +0800, Yusuf Goolamabbas wrote: > Hi Andrea, The following is a thread on ext3-users in which sct mentions > that this is due to a core VFS bug introduced in 2.4.10 which applies > ulimits to block devices. Maybe this could be due to some interaction > with your blockdevice in pagecache > > I don't know if you already have a fix in your

Hi, my name is Yusuf, I just join with this groups. I have using samba PDC with LDAP as backend. I have a problem to change user password from web. I tried using sudo smbldap-passwd, change permission every file so apache can read / execute that file, but I'm still can't change the user password. Is there any way to change the password only with change the LDAP password (using

Hi all, I've installed samba-3.0.21rc2 on a solaris 10 machine (latest patches applied) by compiling from source: #uname -a SunOS newton 5.10 Generic_118844-20 i86pc i386 i86pc User's home is mounted using automount without any problem. But when I try to mount using samba, it can not access the users home directory when it 's share is not mounted and gives following error:

A playback of a mono vorbis track on a system using the i810 audio driver via ogg123 results in a double speed playback (chipmunk sound). It seems that this chipset doesn't support mono and while xmms seems to compensate for it, ogg123 doesn't I have a bugzilla report at redhat.com filed on this with more details https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=90433 Regards, Yusuf

I had sent email to Andrea asking him if his blkdev-in-pagecache might cause the recent reports about ulimits being applied to block devices. This is his response which he asked me to forward to ext3-users since he can't post being a non-subscriber BTW, Redhat should whitelist various kernel hackers email address on their mailserver so that they can post freely to various lists Regards,

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

Hello! I'm sorry in advance if I'm asking stupid questions, this is my first time dealing with a development list, so please excuse me if something is wrong with this message... I'm pretty interested in the OpenSSH codebase, and a couple of questions arose while I was investigating it, and I guess this is the place where I can find answers. 1. There are a lot of allocations, even for

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf

Thanks a million for replying. Your answer helped me. I inspected the errors more closely - with the help of my supervisor - and managed to solve some problems. Finally the making was successful, by just following steps 1,2 and 3. I found that one of the problems was that I modify in a wrong Passes.h, as there are 2. My question now is: How I can apply this pass? (call, use ... etc) >From what