[asterisk-dev] Please consider adding new sip peer/Google Voice retry options
Paul Belanger
paul.belanger at polybeacon.com
Fri Nov 9 09:50:04 CST 2012
On 12-11-09 10:42 AM, Kinsey Moore wrote:
> 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.
>
Ya, I noticed that too. Not sure what would be needed into chan_motif to
support this.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
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