similar to: Firewall Issue

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 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 all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

Incident Information:- Originator: "G Lalithambika"<lalithambika@mumbai.tcs.co.in> Recipients: samba@samba.org Subject: FOCHAP06 WARNING: The file FOCHAP06.DOC.pif you received was infected with the W32/SirCam@MM virus. The file attachment was not successfully cleaned.

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

Hi. We run Samba 2.2.1a in production and it seems to be working fine. We run it on RedHat 7.2, kernel 2.4.7-10smp. All of a sudden (I'm sure it's for a reason, but I can't tell why), this shows up in the logs: Oct 15 10:36:03 mumbai kernel: smb_errno: class ERRDOS, code 112 from command 0xb Oct 15 10:36:03 mumbai kernel: smb_errno: class ERRDOS, code 112 from command 0xb Oct

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 small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

Hi, THIS IS IN DUBAI. I am having PRI line with 100 DID's (00-99) and when we call to any landline or mobile number then it shows us our board number or pilot number (i.e 4663000 means 00).. As i give all the extensions a particular DID, so people from outside world can call them. The problem is the CALLERID ... When we call from any of other extension PSTN line carries out our pilot number

Hi , In vicidial dialer I need small Dialplan require. when i call from hardphone , in that has 1to9 no.s i want define the dipositions like when i press the 1 it will goes NotIntrest, press 2 for NotAvailable. How can i configure for this. -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75

Hi all. This is my first message at R-help...so I'm hoping I have some beginner's luck and get some good help for my problem! FYI I have just started using R recently so my knowledge of R is pretty preliminary. Okay here is what I need help with - I need to know how to use agrep in a for loop. I need to compare elements of a vector of names with other elements of the same vector.

Sir, i am using goautodial server , bri card is showing ok but when i try to call that showing below , This configuration is in doing in dubai , so kindly help me how can connet the call from this , what is my mistake is in this :::chan-dahdi.conf [channels] #include dahdi-channels.conf language=en context=default usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes