[asterisk-dev] Please consider adding new sip peer/Google Voice retry options

[Kinsey Moore]

On Fri, Nov 9, 2012 at 9:37 AM, Paul Belanger
<paul.belanger at polybeacon.com>wrote:

> On 12-11-09 10:20 AM, MichiganTelephone wrote:
>
>> I'd like to see the following settings added for SIP configurations for
>> peers and (in particular) in the settings for whatever is used in a Google
>> Voice connection in Asterisk 11, unless there is already some mechanism to
>> handle it (I'm not on 11 yet so don't know):
>>
>> errorretries= (default 0)
>> errorretrydelay= (in ms, suggested default 500)
>> retryonerrorcodes= (comma separated list, suggested default 503)
>>
>> The idea is that when you are creating a trunk to Google Voice, it is far
>> too often the case that Google rejects the call with a 503 error.  But if
>> you immediately retry the call, it will go through on the second attempt.
>>  Rather than force the user to hang up redial, why not simply provide a
>> mechanism that will retry the call one or more times on the same trunk or
>> connection prior to giving up, with an optional short delay between
>> attempts, and the ability to specify that these retries should occur only
>> if particular errors (in particular the dreaded 503 error) are the cause of
>> the rejection.
>>
>> Sane values (for Google Voice) might be something like  errorretries=2,
>> errorretrydelay=500, and retryonerrorcodes=503.  If the errorretries value
>> is not set it should default to 0, meaning the call fails without any
>> retries if an error is received, as happens now.
>>
>> If possible it would be great if this could be backported as far as
>> Asterisk 1.8.  Thank you for your consideration.
>>
>> By the way, FreePBX users could easily work around the lack of such a
>> setting by adding a trunk more than once the the Outbound Route trunk
>> selection list, however FreePBX doesn't allow that.  It's possible to
>> circumvent FreePBX's disapproval by directly manipulating the MySQL
>> database that FreePBX uses but that's a pain, and in any case, such a
>> workaround would not help uses that don't use FreePBX.
>>
> You should be able to do what you are asking using 'Who Hung Up?'[1],
> however I have not tested it.


Looking at the code, it doesn't appear that chan_motif exports this
information.  This is probably because the work for "Who Hung Up?" went in
before chan_motif was committed.

Kinsey
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