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...put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring and how about their stability and reliability? Or should I stay with 1.4? Sincerely, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100824/e384ddf4/attachment.htm

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi, I'm trying to use Asterisk to place Automated Voice Calls. A verbose log from Asterisk CLI taken when I place a call in the spool directory looks like this: -- Attempting call on SIP/MTN-NEW/my-number for application MP3Player(/myfile) (Retry 1) == Using SIP RTP CoS mark 5 > Channel SIP/MTN-NEW-00000005 was never answered. [Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

...the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101023/5c56d155/attachment.htm

...r is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101030/7af3ae2e/attachment.htm

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve

Hi, I use asterisk with sip3000 device with "sip-aho" connected to PSTN and "sip-ahi" connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten => 99,n,Dial(SIP/sip-ahi,30,g) exten => 99,n,Hangup() The asterisk properly detects hangup of the caller as I

...ave any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101215/23e46bcd/attachment.htm>

...201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only from eth0, and if this port fails, sends registration coming in from eth1? Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101018/1c5459f8/attachment.htm

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter t...

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

...ocumentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If anybody has experience with Primus, it'll be more helpful. Thanks Zeeshan A Zakaria -- www.visionvoip.com www.ilovetovoip.com www.pbxforall.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110121/bc7e691a/attachment.htm>

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at mary at accessgate.net cell 407-267-1463 or Jonathan at jsnyder at accessgate.net cell 407-267-0056 or call our main number 888-227-9337.

Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch => Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries from the database? Thanks Dan -------------- next part -------------- An HTML attachment was

...to look for, where to start from. My experience with Cisco is limited to their networking equipment, IOS, their 7960 series phones and making them work with asterisk, and also using Cisco press's wonderful book 'Taking charge of Your VoIP Project'. Sincerely, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/ac9cd562/attachment.htm

Hi, Does anybody have experience working with Aculab groomer II, to convert between ISDN E1 and non-ISDN T1, or anything similar. I am looking for sample config files. We have asterisk as ISDN E1, but for testing we set it up as regular T1 if we get sample config files. Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100705/06e1b910/attachment.htm

Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101&SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks

Hi List, When doing 'ael reload' on two servers, which are setup with asterisk 1.4.22 and 1.4.35 respectively, I am getting multiple lines of this strange error: ERROR[15483]: ael.flex:647 ael_yylex: Unhandled char(s): On three other servers with same versions of asterisk, i.e. 1.4.22, I don't see this error. Number of lines of the error are the same as the number of lines of the