[asterisk-users] Registering Asterisk to a SIP Provider

? -- Regards, Sahil Gupta Director Tigercom Pte. Limited 998 Toa Payoh North #07-22/23 Singapore 318993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] Dedicated Servers

Hi,I am looking for a reliable provider that can provide 3 dedicated linux servers asap. Unfortunately, the provider I have used for YEARS has become way too slack in recent times and we have to move on. Cheers, Sahil ___ -- Bandwidth and Colocation

[asterisk-users] Off Topic: 8x FXO Gateway

Hi, I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist -- Regards, Sahil Gupta Corporate Advisor TigerCom Pte. Limited 296 River Valley Road Singapore 238337 ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Hylafax

Hi, We seem to be having some teething issues with a new Hylafax - happy to pay someone to complete the installation. Please contact offlist. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499

Re: [asterisk-users] Alcatel - Asterisk setup

Hi, You need to enable overlapdial. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 On Tue, 29 May 2007, Carlos Hernandez wrote: Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk

Re: [asterisk-users] Could two Asterisk servers connect through VPN

Hi, Yes they can - relatively straight forward. Regards, Sahil Gupta VoiceValley On Mon, 7 May 2007, Tielin Xu wrote: Hi list: Has anyone done to set up two servers in different remote offices through VPN in order to get the VoIP communication? Thanks for your information. Tielin Xu

[asterisk-users] SpanDSP (RxFax)

Hi, We had an install working quite well of SpanDSP on our machine until recently where it has began spitting out an error stating unable to translate from unknown to unknown. Any ideas ? Regards, Sahil Gupta VoiceValley ___ --Bandwidth

[asterisk-users] Asterisk Hangups on PRI Interface

hangup request -- Hungup 'IAX2/voicevalley-7' == Spawn extension (context, 099092428, 1) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' I have upgraded to the latest versions and have also ensured that busydetect and callprogress are turned off. Any ideas? Regards, Sahil Gupta

[asterisk-users] Asterisk Hangups on PRI Interface

hangup request -- Hungup 'IAX2/voicevalley-7' == Spawn extension (context, 099092428, 1) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' I have upgraded to the latest versions and have also ensured that busydetect and callprogress are turned off. Any ideas? Regards, Sahil Gupta

[asterisk-users] Off Topic: Hardware Required

Hi, Apologies for the off-topic post, is there anybody in NYC with a bunch of video cards lying around that I might be able to get picked up this evening or early tomorrow ? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided

[Asterisk-Users] PABX Setup

is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

RE: [Asterisk-Users] PABX Setup

Thanks mate. All going well. Regards, Sahil Gupta VoiceValley On Tue, 6 Jun 2006, Boris Bakchiev wrote: Samsung PABX? Its TEPRI probably configured in overlap mode so you need to configure asterisk span that is connected to PABX to overlap mode as well. When user selects the outside line

Re: Fwd: [Asterisk-Users] Prices of g729 codec

Gupta VoiceValley On Mon, 5 Jun 2006, Jon Lewis wrote: On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - Sahil Gupta [EMAIL PROTECTED] wrote: We recently had around 60-80 licenses become useless because Digium refused to renew the keys on that. That was a bit of money kissed goodbye

[Asterisk-Users] ASTCC Developer

Hi, I need a few things modified on the current version of astcc. If there is someone competent, please contact me off-list. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] Prices of g729 codec

We recently had around 60-80 licenses become useless because Digium refused to renew the keys on that. That was a bit of money kissed goodbye. Regards, Sahil Gupta VoiceValley On Sat, 3 Jun 2006, Chris Mason (Lists) wrote: I have no problem with paying Digium the $10 for G729 licenses

[Asterisk-Users] Contract Work : On-site NYC

Hi, We require a technical person to do some on-site installation work for us in New York, must be proficient with Linux and Cisco. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

[Asterisk-Users] Canada Termination

Hi, We will have a DS3 of capacity available directly into Canada with a Tier-1 Carrier there - we will almost certainly be able to thrash your existing rates for high volume traffic. If anybody is keen on routes into Canada, please contact me off-list. Regards, Sahil Gupta VoiceValley

[Asterisk-Users] Colocation Denmark

Hi there, Is there anybody on the list that offers or can put me in touch with somebody that offers quality colocation services in Denmark? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Work available - India

Hi there, If there is anybody on-list looking for VoIP related work in India, please contact me off=list with your details. Positions are of a full-time nature. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided

[Asterisk-Users] Part-Time work available

contact me off-list. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Cisco AS5350

on this, it would be appreciated. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] VoIP in India

http://www.dov.gov.in Regards, Sahil Gupta VoiceValley On Wed, 25 Jan 2006, Code Lover wrote: Hi all, I would like to set an VoIP Gateway in India. Could any one tell me, is VoIP is legal in India? How I can obtain the license to start my VoIP gateway? -- Thank You, Code Lover

Re: [Asterisk-Users] VoIP in India

Network, you need not purchase off an ILD operator but are still required to purchase off a local ITSP license holder (there are a dozen of them in each city). Hope that clarifies the fog. Regards, Sahil Gupta VoiceValley On Thu, 26 Jan 2006, Vamsi Pottangi wrote: Nope, convergence with public

[Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there

Hi, Not very reliable for commercial setups, they do have issues hanging up ports etc. Quintum over Antek any day. Regards, Sahil Gupta VoiceValley On Mon, 2 Jan 2006, Rehan AllahWala wrote: www.antek.com.tw Had 4 port fxo, for around 200 to 250$ They are OEM, and can change things

Re: [Asterisk-Users] How to check how many G729 codec license installed

Right :) Regards, Sahil Gupta VoiceValley On Sun, 13 Nov 2005, Angelito Manansala wrote: *CLI show g729 No such command 'show g729' (type 'help' for help) this means i have no g729 codec installed, right? On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote: That's easy... Just go

Re: RE : [Asterisk-Users] codecs

You simply need to have g729/g723 codecs. Asterisk comes with gsm by default. Regards, Sahil Gupta VoiceValley On Wed, 9 Nov 2005, Olivier Taylor wrote: Right, I must suppose I need gsm codec to hear gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto

Re: [Asterisk-Users] didgium card in india

Such hardware I believe incurs a stock standard duty of 35% plus some other charges. All up, AFAIK it will cost you $2300USD to import the card (based on the $1495 price for a 4 E1 card). You can try guys like Drishti in Delhi, they can help out. Regards, Sahil Gupta VoiceValley On Sat

[Asterisk-Users] Asterisk - Dying Signal 11

Hi, Asterisk keeps dying reporting error signal 11. There is no segmentation fault etc and full logging reports nothing with respect to reasons of why it restarts. Any ideas? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation

Re: [Asterisk-Users] Asterisk PBX

Hi Kapil, AFAIK, there are no such PDF's that exist unless someone has really spent time compiling such information, which will be great to see. However, if you check out www.voip-info.org, its a complete mine of useful information regarding doing what you wish to. Regards, Sahil Gupta

Re: [Asterisk-Users] internet connection between Africa and Europe

FlagTel offer dedicated circuits between Egypt and Europe, if that helps... Regards, Sahil Gupta VoiceValley On Thu, 15 Sep 2005, [ISO-8859-1] Stéphane LAVRI wrote: Hi I'm looking for a company who can provide me an Internet connection between africa and Europe. Plesa If someone can give

[Asterisk-Users] Oh323 and Asterisk with MERA

whilst Asterisk is still running happily. We have to then kill asterisk and start it again. This is a problem that crops up randomly and goes away randomly as well. A permanent solution would make life easy Regards, Sahil Gupta VoiceValley

[Asterisk-Users] Call drops

Hi, We are facing an issue with ALL calls simply dropping during peak times (this is happening upto 10-13x an hour) on certain gear: We have a setup like this: Client --- SIP --- Asterisk --- IAX --- Asterisk --- ISDN --- Provider Any ideas? Regards, Sahil Gupta VoiceValley

[Asterisk-Users] Balancing traffic between two routes

(IAX2/pop2/${EXTEN}) But the above.. would hammer pop1 any tips ? Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

RE: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations

Hi, Whilst the talks are on regarding DS3's, what is the maximum number of simultaneous channels Asterisk should be able to push through in pure pass-through mode? Regards, Sahil Gupta VoiceValley On Wed, 13 Jul 2005, Brian C. Fertig wrote: Trust me dude.. You don't want a lucent TNT

Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

Why not look at getting a provider that can port your numbers to their network and buying the DID's off them over VoIP? Regards, Sahil Gupta VoiceValley On Wed, 13 Jul 2005, Ed Pastore wrote: Thanks for all the great replies. I guess I over-asked my question (since so many kept popping up

Re: [Asterisk-Users] Using G729 in pass through mode

Hi, If you are terminating the call from/to a T1/E1 card or modifying the call in anyway e.g. playing IVR prompts not just voice in - voice out, you will require the codec. Regards, Sahil Gupta VoiceValley On Thu, 7 Jul 2005, Obelix wrote: Is it possible to use G729 on asterisk without

Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?

Hi, I spent quite a few days with this and in the end I find that the 1.07 release is by far the most stable. I had a lot of trouble with the CVS release. Ofcourse, thats just in my case, what do the others feel on this? Regards, Sahil Gupta VoiceValley On Thu, 7 Jul 2005, Christoph wrote

Re: [Asterisk-Users] g.729 codec -- open source?

Check out http://www.readytechnology.co.uk/open/g729/ Regards, Sahil Gupta VoiceValley On Wed, 6 Jul 2005, Juraj Bednar wrote: Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents

[Asterisk-Users] cdr_mysql

A search on google says to use an older release, done that, no help.. any ideas guys? Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Colocation/Telehousing

Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users

[Asterisk-Users] Error with app_addon_sql_mysql.c

app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 This is a installation of Slackware 10.1 with Mysql 4.1.12 (source). Any ideas? Regards, Sahil Gupta VoiceValley On Fri, 1 Jul 2005, Brian West wrote: You could have just done ln -s asterisk-1.0.9

Re: [Asterisk-Users] Can't build cdr_addon_mysql.

to get it going but have failed (incl. reinstalling zlib)... any ideas? Regards, Sahil Gupta VoiceValley On Fri, 1 Jul 2005, Brian West wrote: You could have just done ln -s asterisk-1.0.9 asterisk and it would have fixed that. It should by default do -I../asterisk /b --- Anakin: “You’re

[Asterisk-Users] TE100P

Hi, I have a Gateway running in TE (terminal equipment mode as slave that I need to connect to my asterisk server using a TE100P card. Can anybody give a quick run up of how to run the TE100P's in Network Termination mode to have this working sucessfully? Cheers! Regards, Sahil Gupta

Re: [Asterisk-Users] LiveVoip is Bankrupt

E-mail me off-list, we'll help out :-) Regards, Sahil Gupta VoiceValley On Sun, 26 Jun 2005, trixter http://www.0xdecafbad.com wrote: On Sun, 2005-06-26 at 23:29 +0200, Matt Riddell wrote: Andres wrote: So it looks like Livevoip went Bankrupt Sh1t. Looks like the Daily Asterisk News

[Asterisk-Users] H323 with Asterisk

at default,12126599878,1 failed so falling back to exten 's' == Starting H323/ip$1.2.3.4:12914/16313 at default,s,1 still failed so falling back to context 'default' Any help would be appreciated... Regards, Sahil Gupta VoiceValley ___ Asterisk-Users

Re: [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition

Take this off list please.. Regards, Sahil Gupta VoiceValley On Sun, 12 Jun 2005, Bob Goddard wrote: On Sunday 12 Jun 2005 16:10, trixter http://www.0xdecafbad.com wrote: On Sun, 2005-06-12 at 15:06 +0100, Bob Goddard wrote: On Sunday 12 Jun 2005 08:56, trixter http://www.0xdecafbad.com

Re: [Asterisk-Users] G711 ( alaw or ulaw ) pass-thru

Hi, Both of those are fully uncompressed codecs and free to use. Regards, Sahil Gupta VoiceValley On Fri, 10 Jun 2005, Edgardo Bermejo wrote: Hi, Its possible to make a pass-trhu conection with alaw or ulaw? Thanks -- Este mensaje ha sido analizado por C4I S.A. Mail Server en busca de

Re: [Asterisk-Users] English vs American voice files

Like to share who can record NZ / Australian voices? Regards, Sahil Gupta VoiceValley On Wed, 8 Jun 2005, Mark Phillips wrote: I think you miss the point Andrew. She's not from NZ but from England. She speaks English. Says six and not sex etc. Mark Andrew Thrift wrote: I also have

[Asterisk-Users] Message Playback

Hi, I'd like to know how I can playback a pre-recorded message to a user using our system without answering the call. I want to do the above in the scenario where the user dials a number and the number has been dialled incorrectly. Regards, Sahil Gupta VoiceValley

Re: [Asterisk-Users] SIP--h323 conversion

This is relatively straight forward, you can either use Nufones Implementation or the OH323 package. Both work relatively well. However, I've had issues presenting a GateKeeper ID from Asterisk to carriers that authenticate based on that in the past. Regards, Sahil Gupta VoiceValley On Mon

Re: [Asterisk-Users] voipjet anyone?

VoipJet are not too bad, little pricey though.. theres better around.. a matter of looking :-) Regards, Sahil Gupta VoiceValley On Fri, 13 May 2005, Andrew Latham wrote: Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do

Re: [Asterisk-Users] voipjet anyone?

International calls must be prefixed as 011 to voipjet. Regards, Sahil Gupta VoiceValley On Thu, 12 May 2005, JD wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work

Re: [Asterisk-Users] Re: Who's happy with their voip service?

We've recently broken off with RNK, major service issues.. For some weird reason, during test time all carriers are great. When they get the money of you all of a sudden, the quality goes bad, the account manager is on holiday, the NOC is down and the list goes on... Regards, Sahil Gupta

[Asterisk-Users] VoIP A-Z Carriers

Hi, If you recommend any good carriers, please let me know :-) Volume is no problem, prepayment is no issue. We require good quality routes with high ASR. Preferably on ulaw. Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk

Re: [Asterisk-Users] Music on Hold

! Regards, Sahil Gupta VoiceValley On Sat, 7 May 2005, Matt Riddell wrote: Sahil Gupta wrote: Assuming you have 0.59r (which you should), which codec is the call using? ulaw to the box. And do you definitely have 0.59r? Paste us the output you get when you type mpg123 -v -- Cheers, Matt Riddell

Re: [Asterisk-Users] Music on Hold

ulaw to the box. Regards, Sahil Gupta VoiceValley On Sat, 7 May 2005, Matt Riddell wrote: Sahil Gupta wrote: Hi, I've been trying to get music on hold going on one of our servers: Upon dialling extension 005, it plays: -- Executing WaitMusicOnHold(SIP/parssyd1-4dbe, 30) in new stack

[Asterisk-Users] Music on Hold

-04-21 10:25 fpm-calm-river.mp3 -rw-r--r-- 1 root root 2582496 2005-04-21 10:25 fpm-sunshine.mp3 -rw-r--r-- 1 root root 2217563 2005-04-21 10:25 fpm-world-mix.mp3 Any clues ? Seems like it actions things but isn't playing the mp3 files.. Regards, Sahil Gupta VoiceValley

[Asterisk-Users] SPANDSP

Hi, I'm having troubles getting SPANDSP working with Asterisk (for faxes), on a search of google.. I came up with a few links but the rxfax and txfax modules wouldn't patch or compile into asterisk Any hints? Regards, Sahil Gupta VoiceValley

[Asterisk-Users] Cisco 7940G

the correct provider. Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Digium Card Issues

dchan = 16 defaultzone = au loadzone = au Any ideas? Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Cisco ATA Help

Hi, I have a Cisco ATA 186 that I bought on my recent overseas trip and its the I2 series which has higher impedance than the New Zealand standard 600ohm. Is there something I can do to make it listen to my DTMF tones? Regards, Sahil Gupta VoiceValley

Re: [Asterisk-Users] wcfxo problem

I'm having similar issues using an X100P Ambient Chipset Clone Card any ideas? Regards, Sahil Gupta VoiceValley On Mon, 11 Apr 2005, Dave Weis wrote: I've got a X100P in a compaq proliant 3000. My system stops taking calls and making calls. I had been getting the FXO PCI Master abort before

Re: [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323

Hi, Try the OH323 implementation, we found it works better. Everyone has different experiences oviously.. Cheers, Sahil On Sat, 9 Apr 2005, Adam Rybak wrote: Hello, have successfully installed Asterisk 1.o with H.323 driver and made configuration: GW (Hardware)- GnuGK - Asterisk and i call