HA Asterisk

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Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf,
but its not yet production ready. Can someone please pitch in about HA
feature in Asterisk ?
(HA -> High Availability.) Also, What would be the pros and cons of using
AsteriskNow over Asterisk ? Are the versions same in Asterisk and
AsteriskNow ? We have been evaluating Asterisk for our Voice Application and
it seems it would fit the requirement. Is Asterisk a CPU Intensive or a
Memory Intensive application. Please suggest/guide. Regards, Kaushal

Asterisk Users 4.4 years ago 11 Answers

SIP bad request

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Try to look in 'sip set debug peer user4444'. On 29.04.2011 18:10, Mike wrote:
>
> Hi,
>
> I have been getting reports phones ringing only a tiny moment and then
> going to voicemail. CLI output shows:
>
> -- SIP/user4444-0006fcdd is ringing
>
> -- Got SIP response 400 "Bad Request" back from 23.23.23.23
>
> -- SIP/user4444-0006fcdd is circuit-busy
>
> == Everyone is busy/congested at this time (1:0/1/0)
>
> Which does explain it. How can I find the root…

Asterisk Users 4.4 years ago 1 Answer

odbc error - server is gone

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Yes I have it there, here the content of the file: i think the code is buggy, here is a comment from the function which generated the error
(ast_odbc_smart_execute in res_odbc.c line 155 ) /* This is a really bad method of trying to correct a dead connection. It
* only ever really worked with MySQL. It will not work with any other
* database, since most databases prepare their statements on the server,
* and if you disconnect, you invalidate the statement handle. Hence, if
* you disconnect, you're going to fail anyway, whether…

Asterisk Users 4.4 years ago 4 Answers

Jabber/XMPP

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Hi all, Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org
(formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz)
about Jabber, something the Asterisk community is becoming more
interested in by the day. Join us to learn more about Jabber and SIP
or to share your knowledge and experience. As always, the VUC
discussion includes people from very diverse backgrounds, so it should
be a unique approach to the subject. All the info to connect is on this page: http://vuc.me - SIP:200901@login.zipdx.com (g722, g711)
- Skype:vuc.me…

Asterisk Users 4.4 years ago 0 Answers

Asterisk 1.6.2.18 Now Available

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Hi Matt,
On 29/04/11 11:26, Matt Riddell wrote:
> On 29/04/11 11:19 AM, Jan Bakuwel wrote:
>> Hi,
>>
>> I'm about to deliver a production system based on Debian Squeeze and
>> Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
>> packages for Debian& Ubuntu are available from packages.asterisk.org.
>> Observing some recent discussions on this list, it seems that 1.8 might
>> not yet be ready for production use. Would whoever kindly makes the
>> Asterisk 1.8 packages available also consider doing that for 1.6
>>…

Asterisk Users 4.4 years ago 1 Answer

anybody out there sucessfully using gnugk?

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Hi List,
I have a client that wants me to replace their existing H323
gateway. I am able to get ooh323 and h323 to work fine in a native
environment, but the whole thing goes to heck when I have to cross networks.
Gnugk seems to be the answer to this, but I can't seem to get it to work
right. Any ideas? Thanks
Danny Nicholas

Asterisk Users 4.4 years ago 0 Answers

Best modem for chan_datacard

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I used succesfully huawei E1550 On 24 April 2011 16:45, Dovid Bender wrote: > Hi List,
>
> I am looking to "play around" with chan_datacard. Any advice on the "best"
> device to test with (that I can find on eBay) ?
>
> Regards,
>
> Dovid
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello

Asterisk Users 4.4 years ago 1 Answer

How to create distortion, echo, and chopping sound in a SIP trunk?

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In article ,
Bruce B wrote:
>
> How can I introduce some distortion, echo, chopping sound and all other bad
> quality things that can happen to a SIP trunk? I have plenty of bandwidth
> and crisp clear lines so the only thing that I can think of is to limit
> bandwidth but even that requires quite some scripting work.
>
> Is there any easy way to simulate a distorted SIP line temporarily for
> testing?
>
> I am appreciate experienced…

Asterisk Users 4.4 years ago 3 Answers