similar to: SIP REGISTER script

Kevin wrote: > There is a reboot script posted on the wiki to reboot Polycom > telephones. When I execute this script, I get the following messages. > I am concerned as this is causing issues with asterisk and the PRI. > Does anyone have any ideas why this would be happening? > > > > asterisk console: > > -- Remote UNIX connection > -- Remote UNIX

With my thanks to Brian West and his offering in the thread, "Subject: Re: [Asterisk-Users] Remote reload Cisco 7960" I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones PolyReboot.pl takes an IP address as a single argument and reboots the phone. You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all lower case. Further, you need to use ftp for

any way the you can get * to send a NOTIFY SIP message to all SIP phones? to have the SIP sets recheck thier configs etc?? Like this? NOTIFY sip:sip@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.0/UDP ipaddress From: <sip:webadim@192.168.0.1> To: <sip:sip@192.168.0.3> Event: check-sync Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: test@192.168.0.1

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

Hi, yesterday I updated my second server from Debian Squeeze to Debian Wheezy. Since todaay I get the followinig errors in my logs: Error-Log: ... Jun 26 09:08:28 mailstore dovecot: imap(user at domain.net pid:28898 session:<iuMX3Lf8fACXLrFC>): Fatal: master: service(imap): child 28898 killed with signal 11 (core dumped) ... Mail-log ... Jun 26 09:08:28 mailstore dovecot: imap-login: ID

Hello. I have eth1 for WAN(0.0.0.0) and eth0 for LAN (192.168.10.0/24), need to setup that local user get access to $LOCAL_IP network and ip 192.168.10.2, 192.168.10.3 (will be more in future) to internet, but bandwidth to $LOCAL_IP is 128kbps and for internet is 8kbps. i wrote rc.firewall #!/bin/bash #env IPTABLES="/usr/sbin/iptables" LOCAL_IP="62.64.80.0/21 62.221.38.0/24

Hello, I tried to code a R socket server but I did not succeed. The problem is that once the R socket server is created, I call the readLines function and then R gets blocked. The client seems to work fine since I tested it with a PERL server. I tried many combination of params in the socketConnection but none of them worked. I have seen some examples where the server sends something to a client

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Since I've been using this for maybe a year now, maybe someone else is interested in restricting IMAP and POP logins via libwrap. In addition to the attached patch (against 1.0.5) to src/login-common/main.c, src/{imap,pop3}-login/Makefile.in have to be modified to link against libwrap. Of course, the option needs to be integrated into configure in the long run. -------------- next part

Hello, I'm trying to setup the revised Polycom remote reboot script as found on: http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script I'm not sure how to use this script, it's just a perl script, so I tried creating an executable perl script and running it, but I get the following: [root@asterisk1 agi-bin]# ./polycom_reboot.pl 192.168.3.205 Checking ARP

plain text document attachment (lguest64-loader.patch) I noticed that the lguest loader code for i386 was in Documentation/lguest. Well, that's fine (I guess) but it can't just be for i386. So I made a separate directory to put the loader code in. So now we have: Documentation/lguest/i386/... for the lguest i386 loader. and Documentation/lguest/x86_64/... for the lguest x86_64

plain text document attachment (lguest64-loader.patch) I noticed that the lguest loader code for i386 was in Documentation/lguest. Well, that's fine (I guess) but it can't just be for i386. So I made a separate directory to put the loader code in. So now we have: Documentation/lguest/i386/... for the lguest i386 loader. and Documentation/lguest/x86_64/... for the lguest x86_64

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

hi all, how to establish a call between two asterisk servers for the sip users registered for the servers. ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Sunday, February 10, 2008 11:30 PM Subject: asterisk-users Digest, Vol 43, Issue 30 > Send asterisk-users mailing list submissions to >

found response from asterisk. =20 =20 On asterisk's log I see messages like: "Looking for conference on conference-context (domain serverIP)" =20 And: "Call from 'conference' to extension 'conference' rejected because extension not found." =20 =20 Does anyone have an ideia of why I'm getting that message? =20 Why does asterisk seem to be using domain

Hi, I'd like to call my friends through their SIP URLs. I've found two approaches for doing this in Asterisk: - one is to prepend some numbers at start and catch them - the rest of called string is used for SIP URL - another approach (that I like better) is to use catchall pattern at the end of context _. and then parse string with help of SIPDOMAIN variable. But there is a catch into

Hi I'm struggling with setting up SIP domains. If I specify a domain and a context in [general], that context overrides any set in type=peer blocks elsewhere. This results in incoming calls from PSTN gateways I use arriving in the wrong context. If I don't specify a context (which the docs I've found suggest is valid), then I get: 2006-05-12 07:36:16 WARNING[95290]:

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

I'm using Dovecot (1.0.10) locally to test SugarCRM. When I tried to set up a mail account in Sugar, it complains with -- SECURITY PROBLEM: insecure server advertised AUTH=PLAIN Please check your settings and try again. -- don't know if that behaviour is a bug or a feature of php-imap. The case is that I'm unable to set up the mail account in Sugar. Timo answered to me on IRC about

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem