similar to: Asterisk Realtime and SIP Registration

Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten => _X.,1,Absolutetimeout(20) exten =>

Hi guys, When I try to get ring tones when dialing out with the command Dial(SIP/sipout/${PHONE},15,r), I get the error message indicated in the subject. I've checked my indications.conf file using the sample file provided with asterisk 1.4.10 (the version I'm using) and it's not better. Any idea ? Regards. -- Cyril SCETBON

To be clear I meant using Chan)_h323 with Call Manager where CM is configured with * as a H.323 gateway, not client. CM supports H.323 to direct calls through gateways, and in fact until recently that is all they supported. They now also have MGCP, but only to their IOS platforms, and SIP is coming soon. There are NO sccp-based gateways, from Cisco anyways. Dan -----Original Message-----

Hi all, with an (attended) transfer i see the following happening: 1) A calls B1 2) B2 calls C 3) B2 transfers call to A 4) A talks to C At step 3, the channel A is connected to channel C and B1 and B2 are hung up. In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and i see that the channel variables of A have been merged into B2<ZOMBIE>. If there were

Hi there, I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with my asterisk server. I already changed the name of the user to "anonymous" since it looks like the phone sends that name. The WiFi Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200 What is it that I am missing? Any help very much appreciated!!! The error message I get is: Jan

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten =>

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone PROBLEM : I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing. INFORMATION : This is the output on the CLI of the local Asterisk-server : [Oct 3 17:40:33]

Hi, I'm trying to use a mobile phone (ericsson W300i) with asterisk through bluetooth. After some sutrggling, I foun chan_mobile. As some one already used this mobile with what result? I'm considering a simple asterisk system (for home use/test purpose) with : - one SIP service provider (with link to PSTN) - one SIP phone (a softphone on the asterisk box) - one mobilephone (ericsson

Hi, we have a 2 node / 3 node RAC installation with OCFS. We have the following observations. 1. TIME STAMP Issue We have noticed that the time stamp which is shown on the datafiles doesnt remain the same even after a shutdown normal /shutdown immediate i.e If I shutdown all RAC instances ( A , B , C) using shutdown normal / immediate, the timestamp on the datafiles are not the same. Even

I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?" AT&T Local did not, But XO communications said they did. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national

I've asked this question several times before and was always told it wasn't possible. However, after reading a thread posted to the list today, I'm not so sure my question was understood. So, here I go again... Is it possible to have multiple Asterisk systems share a common realtime database for the purposes of accepting registrations from phones, and maintaining SIP location (ie

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

Dear all, I am studying a dataset using the aov() function. The independant variable 'cds' is a factor() with 8 levels and here is the result in studying the dependant variable 'rta' with aov() : > summary(aov(rta ~ cds)) Df Sum Sq Mean Sq F value Pr(>F) cds 7 0.34713 0.04959 2.3807 0.02777 Residuals 92 1.91635 0.02083 The dependant variable

Hey all, I''m looking for more detailed info on the netlink api. I''ve read Andi Kleen''s man pages (which describe themselves as lacking and incomplete), rfc3549 -- Linux Netlink as an IP Services Protocol (which was a nice general overview, but lacking in implementation detail), both the iproute2/ip/tc source and the kernel source (which have no usable

Okay, here is a very simple QOS scheduler that delays packets for 2.6. It is good for testing, and might be useful for people who want to put some class of traffic into a "penalty box". Alexey''s tc command is really easy to extend to new disciplines. There is a version which knows about this scheduler at: http://developer.osdl.org/shemminger/tcp/iproute2-delay.tar.bz2 The

Hi all Iam trying to implement "route add " using netlink. The changes are not reflected in the routing table. I have given my code and screen shots of the routing tables. Can anybody tell me is there any mistake iam making in defining the fields . or any other mistake iam commiting thanxs viji //////////////////////////////////// CODE

Hi, I use Centos 7.? I think the samba-dsdb-modules package is debian based. Best, Robert 2020. 07. 20. 13:33 keltez?ssel, L.P.H. van Belle via samba ?rta: > Is : samba-dsdb-modules > Installed ? > > dpkg-query -s samba-dsdb-modules |grep Status > > Greetz, > > Louis > > > >> -----Oorspronkelijk bericht----- >> Van: samba [mailto:samba-bounces at

Good afternoon, I have a problem with rsync when I sync my virtual machines files between a Linux box and a NAS. This is the output: rsync: failed to set times on "/mnt/backup/Windows Server 2003/.Windows Server 2003.nvram.GSfBdw": No such file or directory(2) rsync: failed to set times on "/mnt/backup/Windows Server 2003/.Windows Server 2003.vmdk.0ykiw7": No such file or