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Hi, I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4 I am looking for the following functionality: `````````````````````````````````````````````````````````````````````````````````````````````` I receive a call from Mr. A. I put Mr. A on hold. I dial Mr. B I connect Mr. A's call (which was on hold) to Mr. B and I get out of the call. Mr. A & Mr. B are in

Hi, I seem to be facing an intrusion issue, inspite of firewall (script attached). What am I missing ?? Any suggestions / recommendation are welcome pls. Best regards, Sans -------------- next part -------------- #!/bin/bash echo 0 > /proc/sys/net/ipv4/ip_forward # Clear any existing firewall stuff before we start /sbin/iptables --flush # As the default policies, drop all incoming

Hi, I am taking my first steps towards virtualization. I have 3 Gateway Servers; all on CentOS 5.3 (non-graphical mode) + Squid. We have 3 separate teams each of 5-7 users who use 1 ISP each via their own Gateway server. I am keen to use virtualization and consolidate all 3 servers on a single server. I plan to install ZEN on a Pentium Core2Duo with 2GB RAM & 1 nic. Then create 3 virtual

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai

Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/816aecdd/attachment.htm

Hello, I have Elastix ISO install (FreePBX 2.7.0.3) My current Setup is as follows: Inbound Route > Queue > (Dynamic Agents) The queue distributes calls based on rrMemory. I have been asked to redesign the call distribution as follows: Calls will be delievered to Level-1 Agents (say 4 dynamic agents) in rrMemory format. When Level-1 Agents are busy, distribute calls to Level-2 Agents

Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : "sip show channels" [trixbox ~]# /usr/sbin/asterisk -rx "sip show channels" Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No 192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0

Hi, I am using Trixbox 2.6.2.3, ISO install I am getting the below error in `/var/log/asterisk/full` Unable to create channel of type 'SIP' (cause 3 - No route to destination) Is there anyway to figure out which extension is causing this error ? Thank you. Best regards, Sanjay

can i use MP3 files as an IVR prompts directly without converting to .gsm format? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091010/67205e07/attachment.htm

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

Hi, Can someone help me formulate MySQL Query(s) which will help me extract the following details for a given DID (date range can be excluded for simplicity). Date-Time DNID (I am recording this is `userfield`) CLID time-1 (when call was received) time-2 (when call was answered by agent) time-3 (when call was hung-up) My Call flow is as follows: - Caller dials a DNID - Call enters queue - Call

Hi, I am using Asterisk 1.4 along with FreePBX. My call flow is as follows: Inbound DID > Inbound Route > Time Condition > Queue. My welcome greeting MP3 is setup under "System Recordings" & its called under Queue > "Join Announcement". Not sure why, the MP3 audio file starts to play after a 5 sec pause. Any thoughts or pointers are appreciated. Thanks in