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...n't want to 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

...ylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/59296095/attachment.htm

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

...n noticing that 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

...t; _NXXNXXXXXX,s,1,Dial(SIP/XXXXXX/${EXTEN},30) exten => h,1,Macro(hangupcall,${EXTEN},${CDR(accountcode)}) [macro-hangupcall] NoOp(${CDR(dst)}) Set(dialout_num=${ARG1}) Set(user_id=${ARG2}) ResetCDR(vw); NoCDR(); Hangup(); -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090713/0dfcfea4/attachment.htm

...n asterisk server 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

...va based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas, suggestions whom to go with? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090710/1396790d/attachment.htm

...uctions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523" line1_displayname: "Zeeshan A Zakaria" line1_authname: "523" line1_password: "523" line1_shortname: "x523" line2_name: "523" line2_displayname: "Zeeshan" line2_authname: "523" line2_password: &...

...ning*: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg0000.txt): failed to open stream: Permission denied in * /var/www/html/recordings/modules/voicemail.module* on line *525* It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061031/29626458/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 international calls anyways, I wonder how would it benefit any hacker in a...

...time/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for different users, groups or departments. Can this be accomplised at all in asterisk? Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081017/5314a36a/attachment.htm

...means extension 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

I've got a 778 DID for vancouver, but don't know if it will be a local call if dialed 604 and vice versa. What are the different area codes in Vancouver and why its easier to get 778 DID than 604? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070718/cecacac2/attachment.htm

...what price. I may need it for upto 100 phones. What else should I consider so that phones work without problem along with the computers on the same network? Phones will use their bridged ethernet connections, so that both computer and phones can work on the same connections. Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061210/fdde9964/attachment.htm

...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 code. But the AEL code works fine otherwise. Any idea what is the reason of this error? I couldn't find any useful info on google regarding this error. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100825/8212a68d/attachment.htm

...roISDN and National. Although the documentation 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>

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip

...otocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080223/d8ce7457/attachment.htm

...ur help as what 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

Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some