similar to: RE: [asterisk-biz] Colocation Denmark

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

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

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

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 VoiceValley

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it

Hi, We seem to be having an interesting issue with Asterisk whereby, it keeps routing calls coming in to the 'default' context.... regardless of what changes occur to h323.conf. <SNIP> [POP-A] type=user host=1.2.3.4 context=international </SNIP> == Starting H323/ip$1.2.3.4:12914/16313 at default,12126599878,1 failed so falling back to exten 's' == Starting

Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley

Hi, I'm looking for someone to do time-to-time mantainence on some of our machines going up in New York. The person *MUST* be stationed in New York. Areas of expertise required: - Proficiency in Linux: Slackware, Fedora - Proficiency on Cisco Routers If anybody is interested, please contact me off-list. Regards, Sahil Gupta VoiceValley

We've got a white route w/ VSNL. $0.09 / min, billing is 1/1 prepaid only. SIP or H323 w/ G729 Codec. E-mail me off-list for testing. Thanks, Jon ----- Original Message ----- From: "Jerry Romney" <Jerry@uspcom.com> To: <daniel.silaro@gmail.com>; <asterisk-biz@lists.digium.com>; <asterisk-dev@lists.digium.com>; <asterisk-users@lists.digium.com>

I believe there were some new prompts added for 1.4 for Directory Info. These have now been added to http://www.tel.net Have a good 2007. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

Why is what he is doing different than having the fax machine on a Sipura ATA? Just because both those ports are on the pci card that doesn't make them not Voice in between....if I'm wrong....eh...oh well.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Wednesday, September 28, 2005 9:45

Hi sahil: I think the reason why you cannot get page shared is due to the gref you got. Gref is responsible for a page allocated from domU, in my understanding it should not be 0, that is a gref 0 can not be shared, that''s why I skip gref 0 to be nominated. The gref is nominated to Xen and later used to find a corrspond MFN, so it shall not always be the same.

Thanks For support. One more doubt is that , As Centos 7 major release is launched , now support for centos 6.x where x is latest mirror point release is supported or not ? ? and further 6.y where y>x will be released or not . You are requested to calrify . On Fri, Aug 1, 2014 at 5:30 PM, <centos-docs-request at centos.org> wrote: > Send CentOS-docs mailing list submissions to >

Hello all, My name is Sahil Yerawar, Btech 3rd Year Undergraduate student at IIT-Hyderabad. I am currently working on improving vectorization diagnostic remarks in LoopVectorize.cpp. I have some familiarity with LLVM as we studied it in Compiler Engineering Course at IITH. Here is our first effort to improve one of the remarks. https://reviews.llvm.org/D44067 You might want to have a look at

In the next couple of weeks we will be starting the beta phase of our Guatemala POP. If you could wait, welcome. LTenorio -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Phil Astin Sent: Sunday, January 16, 2005 6:23 PM To: asterisk-biz@lists.digium.com; asterisk-users@lists.digium.com Subject: [Asterisk-biz]

1. Search the archives 2. Search again 3. Now search the internet 4. The fact is that faxing over VoIP without T.38 doesnt really work. Because it works 60% of the time it doesn't mean that it works, until it works 99% of the time. Since Asterisk does NOT support T.38 it doens't support faxing over VoIP other solutions are available, search the wiki. On 9/26/05, Mark Armstrong

I am running version 1.1.1 with mail_location: maildir:~/Maildir. This is working great as all our users have UNIX accounts with nologin shells. New domains (and their users) are about to come online and we would like to migrate to a setup with virtual mailboxes/users. From the wiki and comments within dovecot.conf, I see it is possible to do this piecemeal so both local and virtual users

Can they be used from any unsecured access point (eg they have a browser to enter in a password etc) or can you only use them from home AP's etc. Cheers, Dean > -----Original Message----- > From: asterisk-biz-bounces at lists.digium.com [mailto:asterisk-biz- > bounces at lists.digium.com] On Behalf Of Mike White > Sent: Wednesday, 27 October 2010 8:37 PM > To: Commercial