search for: nabeel

> Many thanks Julian. Are you looking for the pinout for a single plug 2.5mm (cellphone) headset or a dual plug 3.5mm (computer) headset? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net

...This is not just a problem with LiveVOIP - for some other countries where VoipJet is primary I've had similar problems). Are there any ways to get around this problem? Is there a way to timeout if "ringing" doesn't happen in 5 secs (for example) and go to the backup provider? -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeel<at>jafferali.net

I just downloaded the new astcc and it includes now a new field in the list of the cards: Brand Great! How can I use it in the dialplan? bye Ronald

...oice packets? Does the remote server just use the "received from" IP address and port to respond? Finally, would an IAXy work seamlessly in a configuration where it is plugged into a NAT router which is plugged into another NAT router - double NATted? The * server is on a public IP. -- Nabeel Jafferali tel: 647.722.8457 x201 718.606.4190 x201 fwd: 46990 x201 email/msn: nabeel<at>jafferali.net

...I write it, I was wondering if anyone had an extensive dialplan or an AGI script that already did something like this. I know it'll only take a couple of hours to write and test this, but I thought if someone has something already written, I could just "borrow" it from you. Thanks, Nabeel

...providers (all IAX). I need something that can handle monthly fees and per call charges (depending on destination, obviously), and should provide a web interface for customers and administrators. Something that can tie in to one of the existing management GUIs would be a big plus. Any ideas? -- Nabeel Jafferali tel: 647.722.8457 x201 718.606.4190 x201 fwd: 46990 x201 email/msn: nabeel<at>jafferali.net

Hello. I have my * server set up and working perfectly. I wanted to allows calls to sip:nabeel@sip.myserver.net. In sip.conf, I have: [general] context=default Also, in extensions.conf, I have: [default] exten => myname,1,Goto(internal,nabeel,1) However, when I make a call using a "Direct Dial IP" account in X-Lite, I get the following error in *: Failed to authenticat...

...box' prompt and 'password' prompt when a user tries to access their voicemail. I can remove the 'password' prompt by not setting a password for the user, but the 'mailbox' prompt is always heard. Please let me know how Asterisk can be configured to remove these prompts. Nabeel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160730/e9f46f69/attachment.html>

On Thu, 4 Aug 2016 14:03:39 +0100 Nabeel <nabeelshikder at gmail.com> wrote: > I should add, a password is *always* asked if a password has been set. > There isn't a way to bypass that. Then something is wrong. http://darcy.vex.net/star98.mp3 -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:...

If I remove the password, how can anyone access the mailbox if the 'mailbox' prompt is not played? Nabeel On 30 Jul 2016 3:19 p.m., "D'Arcy J.M. Cain" <darcy at vex.net> wrote: > On Sat, 30 Jul 2016 06:43:47 +0100 > Nabeel <nabeelshikder at gmail.com> wrote: > > I am using Asterisk voicemail on a CentOS 7 server. I would like to > > be able to remove the ...

I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @<proxy-addr>... So the callerID on the phone looks like: 2145551212@10.10.10.10 which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? 7.5

I am trying to get a SPA-3000 to work behind NAT - for the sake of the exercice. The SPA is on the local network at the address 192.168.0.125 behind a NATted linux router. The machine I am trying to work with is a friend's (let's call it lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it. I can see the SPA register but when I try to make an outbound call I get the message:

...h) to the routes table, so "011416..." would pick up "0114" instead of "1416", but it didn't work. Reordering the mySQL table so these 8 non-NANPA catch-alls appeared at the top of the table (before the "1416" and other NANPA entries) fixed it though. -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net

...ry to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have 1-2 international destinations that don't go through and eventually time out, but ASTCC just says "the party is not answering" since the IAX channel hung up. -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeel<at>jafferali.net

...quot; for the specific device. I can't seem to figure out what this refers to. Anyone? Also, if I ever changed the IP of my * server, would leaving an * server on the old IP for nothing but provisioning for a day or two work to migrate these IAXy's? Finally, how do I upgrade firmware? -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeel<at>jafferali.net

...e, and you can see why > replying to a message, erasing its contents and starting an > entirely new email about a different topic is frowned upon > (yours is the highlighted message). I know this is OT, but can you recommend an email program for Windows that does something like that? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net

...to the provider's gateway (hopefully reducing my bandwidth consumption). I have it working with VoicePulse Connect but SetCIDNum doesn't appear to work. Is this something with VoicePulse Connect only or is it generally difficult to set the CallerID info on a per-call basis when using SIP? Nabeel

I saw these adapters on eBay. 2.5mm stereo jack to modular RJ-9 jack. I think original site is http://www.ciscoheadsetadapter.com Mike >Nabeel, > I am very interested in what you came up with for a 2.5mm to RJ-10 > adapter. > >I played with every combination I could think of but the best I was able to >come up with had a echo of the far end voice back to the far end. >Could you post your schematics for me? > >th...

Hi all I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page. I need to now the name os de file or a specific category link where i can download it. If you can send me the file is beter ;-) Thanks in advance Regards Wert --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.

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