search for: jafferali

...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

> 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

...ckets? 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

...rs (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

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:

...he 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

...iew 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

...or 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

...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

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.

...eir home router has a "real" IP, not a private IP. So, for some reason, STUN (or something else) is seeing their IP as their ISP's router. It almost seems like their connection is double-NATted? I am unsure what steps to take next, so any help would be greatly appreciated. -- Nabeel Jafferali tel: 647.722.8457 x201 718.606.4190 x201 fwd: 46990 x201 email/msn: nabeel<at>jafferali.net

Hello All, I am looking for a sip phone that is capable of automatic nat. The Cisco ata186 for example works fine for natting with iconnecthere, but as for asterisk, both my 7960 and polycom ip600 require you to set the nat ip on the tftp. Does anyone know a good phone (or ata) that can do this automatically? For example, I want to give a phone to my brother, who is going to europe. His ICH

...39;s Topics: 1. Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore) (Ronald Wiplinger) 2. Re: Dial Plan Agents (1 of 2) agent-dialplan.conf (Michael Loftis) 3. Number of Calls per Proxy on Cisco 7960G? (Glenn Powers) 4. RE: Is anybody using an IAXy? (Nabeel Jafferali) 5. RE: Number of Calls per Proxy on Cisco 7960G? (Nabeel Jafferali) 6. Re: Auto Protocol (depending upon registration.... (Freddi Hansen) 7. Re: RE: Issue compiling zaptel on FC 3 kernel 2.6.10-1.737 (Eric Bishop) 8. Re: Out of 5 Grandstream BudgeTone 101 THREE are defect !!!...

...around just in case.. .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nabeel Jafferali Sent: Friday, December 17, 2004 5:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Total newbie here looking to do a VoIPconference call? > My packet8 "dta310" adapter has the SIP server hardcoded into > it. If I could change that, I c...

...dAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2}) > Once the call is finished I would like to have the balance > shown in the display by sending a sip message to the phone(if > possible otherwise not important). This would require adding code to the AGI, if it's even possible. -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (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

...gt;come up with had a echo of the far end voice back to the far end. >Could you post your schematics for me? > >thanks >mike > >-- >----=> Mike Dewey >All Technologies Unlimited >mdewey-at-alltechunlimited-dot-com > >On Tuesday 25 January 2005 07:16 am, Nabeel Jafferali wrote: >> Mike Dent wrote: >> > Neither, the one I am looking for is the tiny (similar to RJ11) plug. >> > Which are used on telephony headsets. >> >> The RJ10. Well, >> http://www.mml.uni-hannover.de/einhorn/headset/index_e.html has the >> Cisco 7960...

Just upgraded to the current stable ver. when I start asterisk with -vvvvvcg I get the following error [pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! Asterisk

Hello, Ever since I started using Asterisk I always get this error: Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463 monmp3thread: Request to schedule in the past?!?! I have a dedicated system system that really runs only Asterisk: - Pentium III 500Mhz - 128MB of RAM - 10GB of Disk Space - SuSE v9.2 - MySQL - Apache (only for use with Asterisk) - NTP client for clock synch There is no X

Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Thanks, Ben Blakely -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/a7e575cc/attachment.htm