[asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2

[Alex Kauffmann]

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/ilugd@lists.linux-delhi.org/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