On 30/10/2011 05:53 a.m., Raj Mathur (राज माथुर) wrote:
On Sunday 30 Oct 2011, Sammy Govind wrote:
hmmm so IAX channel is playing with you guys.
1- Cant you guys use SIP, does this happen with SIP trunk as well !?
2- Which version of asterisk are there on both servers.
3- See the output of the command "core show file versions" in your
both asterisk servers. Mainly lookout for IAX channel version.
Also try enabling IAX debug and paste the output on console.
1.6.2.9-2+squeeze3 on the SIP server, 1.6.2.9-2+squeeze1 on the Dial
server.
I doubt if we'll be able to change the architecture of an infrastructure
handling up to 450 simultaneous calls for the past 6 months at this
stage, so SIP is out. IAX2 has been working beautifully for our needs
up to this point, and we need to find a solution that we can integrate
into this architecture itself!
Incidentally, if anyone's interested, the installation itself is
detailed at:
http://www.mail-archive.com/[email protected]/msg28166.html
Regards,
-- Raj
Sorry if i missed it, but is IAX2 trunked? IF so, perhaps you are
running out of bandwidth in your IAX2 trunk. The setting 'trunkmaxsize'
defaults to 128000 bytes.
From the readme file:
"...Once this limit is
; reached, calls may be dropped or begin to lose audio. Depending on the codec
in use and
; number of channels to be supported this value may need to be raised, but in
most cases the
; default value is large enough."
--
Alex
--
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