similar to: Finding the position of a character in a string

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm getting a strange problem with a Mediatrix 3631 Gateway connected to the PSTN via an E1 PRI link configured for Euro ISDN signaling. The Mediatrix sends incoming calls from the PSTN to an Asterisk server via SIP: this works fine. But when the caller hangs up, the Mediatrix doesn't send "Bye" to Asterisk, so the call is

I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118',

2014-02-06 11:09 GMT+01:00 giovanni.v <iax at keybits.org>: > Il 05/02/2014 8.42, Olivier ha scritto: > > channel then it depends upon what you have the priindication option >> set to. With >> priindication=outofband then a busy cause code is sent to the >> network and the call >> is hung up. With priindication=inband then a busy tone

Hi all, I have a need to strip some characters from a variable to get the right data but have only found how to strip all but the last or middle stuff, need to keep the beginning. EG: With $(SIPURI) I want to keep just the sip number and delete the remainder '@server.com'. Ideally I'd like to use 'SayDigits($(sipuri[-@server.com])' All replies greatfully accepted. Phil

Hi all I'd like to log the client IP addr and port used for SIP and RTP *during* in a call. The IPs must be the real source IPs (internet accessible). How are these parameters available from dialplan? For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. I need the external IP:port Regards Ethy

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I''ve written a minimal sort of Perl module that dynamically creates and destroys traffic control rules for specific IPs. I''m currently using it for a user bandwidth control application at a client site. The module essentially gets Ethernet device(s), IP address and in/out speeds as input and dynamically creates classes, queues

Hi, I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5 digits). ChanSpy is working fine for listening in to conversations initiated by these channels, and I can use '*' to randomly switch channels. However, is there any way in this scenario to be able to switch ChanSpy to a specific channel by typing in a ...# key sequence during a spy session?

#!/bin/bash # checksetexternip.sh # Author: John Cahill email at johncahill.net # Licence: GPL v3 # Description: script that queries checkip.dyndns.com to find the server's external IP address. Updates asterisk's externip value and does a sip reload if necessary. # Last modified 06/02/2012 is_ip(){ input=$1 octet1=$(echo $input | cut -d "." -f1) octet2=$(echo $input

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

Hi, I have a client connected to the ''net through 3 ISP''s. Have set up a Linux box to do routing and load sharing for the 3 connections. A fourth interface is connected to the LAN with private IP addresses. Am using iptables to SNAT traffic to the appropriate IP depending on the interface the packet gets routed onto. The setup looks something like this: Interface IP

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

hits=-2.5 tests=BAYES_00,FORGED_RCVD_HELO X-USF-Spam-Flag: NO Hi, I have a requirement to scan Apache logs and discover ``exceptions''. Exceptions can be of two types: 1. A single IP generating a large amount of traffic within a given time frame (for definable values of ``large'' and ``time frame''). 2. A single IP hitting a wide set of URLs on the server (indicates

I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably earlier versions too) remove the group write permissions from /etc/asterisk/. which is different than 1.4. And 1.6. Is this expected behavior? If so, what's the rationale? If not, I'll submit a bug report if someone hasn't beaten me to it. -K -------------- next part -------------- An HTML attachment was

Hi everybody! I''m trying to implement the load balancing for a LAN with two ISPs. I''ve installed a Suse Linux Enterpise Server 9 with iproute2 for that porpouse. The server have two NICs, one of them is for both the LAN and ISP 1. I''ve setup both NICs with YAST (if I use ip for this, then the whole thing doesn''t work!) and execute the following commands to

Hi All, i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that. is it possible to run system like that ? is it good idea , can

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body bgcolor="#ffffff" text="#000000"> <font face="sans-serif">Hello all,<br> <br> I've googled

Hi all, I have a problem with voicemail. My boss has asked me to send via email, the message that a user leaves on the voicemail. This is very easy. :) After, he asked me to check before sending the email, if the receiver's mailbox is full. If the mailbox is full, Asterisk should call the receveir intern (example 2001) and using a Playback tell him that his mailbox is full. How can I do?

Hi All, I'm trying to achieve the following: - If <sip/iax user> logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If <sip/iax user> logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

Hi. I've just bought SIP telephony service from a Swedish telco. I've managed to make and receive calls with kphone. Now I want to set up asterisk to be able to add fancy features like voice mail and recording conversations. But first I have to get the basic setup right. I'm running asterisk and kphone on the same machine, behind at NAT-router. When I make a call (from my regular