Branch: refs/heads/2.1
Home: https://github.com/OpenSIPS/opensips
Commit: de7bcfc5f8b635f3d098ee1ddb7ff46d22281397
https://github.com/OpenSIPS/opensips/commit/de7bcfc5f8b635f3d098ee1ddb7ff46d22281397
Author: David Sanders <dsander...@ucsbalum.com>
Date: 2015-09-24 (Thu,
Currently the dispatcher module won't start if only `db_default_url` is set and
the `db_url` modparam for dispatcher is not. This fixes that.
```
Sep 23 13:43:55 [46257] DBG:dispatcher:mod_init: initializing ...
Sep 23 13:43:55 [46257] DBG:dispatcher:init_ds_bls: Initialising ds blacklists
Sep
Same change as #648.
You can view, comment on, or merge this pull request online at:
https://github.com/OpenSIPS/opensips/pull/649
-- Commit Summary --
* Dialplan module should honor db_default_url
-- File Changes --
M modules/dialplan/dialplan.c (2)
-- Patch Links --
I'm currently trying to use the B2B refer scenario (from the tutorial) for
inbound calls. After advice from @bogdan-iancu on the user mailing list I fixed
the natted SDP issue I was having, but now I've broken the ability to send
calls to voicemail.
Previously I was using the same method as in
On further investigation, it is possible to set the `t_on_failure` route in
`local_route` (this B2B stuff hurts the head a little) and get the same
voicemail behavior as before.
It works, it's just a bit awkward because you lose all context in `local_route`
about the leg (all the logic leading
If `avp_db_query` is used in the script but `db_url` was not set as a modparam
for avpops then OpenSIPS will segfault on startup.
```
* thread #1: tid = 0x, 0x000104325a8f avpops.so`avpops_db_bind + 159 at
avpops_db.c:157, stop reason = signal SIGSTOP
* frame #0: 0x000104325a8f
@saghul, fair points.
This problem is known, but it never seemed worth fixing because it would
involve keeping track of the registered AoR for NAT_Contact.
It was worth fixing for us, hence this fix which worked for our needs. It would
seem that the multiple accounts from the same IP and
@bogdan-iancu, yea, it's good for older versions. We ran it on 1.8 for a long
time.
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Branch: refs/heads/1.11
Home: https://github.com/OpenSIPS/opensips
Commit: 36ad430f0bf50bb20fcdfd1180c9262348e41f79
https://github.com/OpenSIPS/opensips/commit/36ad430f0bf50bb20fcdfd1180c9262348e41f79
Author: David Sanders dsander...@ucsbalum.com
Date: 2014-11-05 (Wed, 05 Nov
Branch: refs/heads/1.8
Home: https://github.com/OpenSIPS/opensips
Commit: 346029318e9c9aab4019e29f4c9204e7c1a69bce
https://github.com/OpenSIPS/opensips/commit/346029318e9c9aab4019e29f4c9204e7c1a69bce
Author: David Sanders dsander...@ucsbalum.com
Date: 2014-11-05 (Wed, 05 Nov
@bogdan-iancu, long delay here, but have been testing out the patch (applied to
a 1.11 source tree) and all is working well. The filename makes debugging a lot
easier, thanks!
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You can merge this Pull Request by running:
git pull https://github.com/dsanders11/opensips 1.11
Or you can view, comment on it, or merge it online at:
https://github.com/OpenSIPS/opensips/pull/372
-- Commit Summary --
* Backporting filename in runtime error messages to 1.11
-- File
@bogdan-iancu, pull request opened:
https://github.com/OpenSIPS/opensips/pull/372
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https://github.com/OpenSIPS/opensips/issues/299#issuecomment-60318755___
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Observed cases where keepalives from `nat_traversal` continued after unregister
were traced back to a few lines of code which prevented stopping keepalives on
unregister events.
There are three main changes here:
1. Allow `SIP_Registration_update` to decrease a keepalive time
2. If a successful
It's a custom solution for our project, but it isn't anything special. Just an
extra param tacked onto the contact when sent for registration.
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Yes, this only makes sense with CONTACT_ONLY matching mode. That mode, along
with a UAC which changes call IDs when re-registering, will provoke this case.
I observed it in a situation where we added our own custom matching mode which
matched off of a token sent by UACs to better handle
RFC 3261 does not force UACs to use the same call ID for all registrations,
only strongly suggests it. Some UAC implementations may not use the same call
ID, causing the cseq check in `urecord.c` after a contact match to not occur in
situations where an out of order request may be received.
New PV (`$rT`) which holds the current route type as a string. Useful for
determining the original route type when inside another route in script, such
as a route called from an `onreply_route`. Allows for more generic and reusable
routes such as a logging route which includes the route type in
Allows for assertions which can be enabled or disabled for easier script
debugging.
Usage:
assert($var(foo) == 42, quot;Whoops, var(foo) wasn#39;t equal to
42quot;);
You can merge this Pull Request by running:
git pull https://github.com/dsanders11/opensips master
Or you can
Unfortunately the PVs don't get expanded. I'm not enough of an OpenSIPS expert
to know what needs to be done to allow that without a good amount of
investigating, but you're right, that would be very useful.
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@bogdan-iancu, thanks for the quick turn around. I'll try it out ASAP, but it
may be a week or two due to a release coming up.
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This is based on OpenSIPS 1.8.0, so this may already be fixed, but glancing at
the source code I don't think it has been.
With the `include_file` functionality, parsing errors at startup include the
file name and line number of the bad syntax.
However, at runtime, the error messages don't
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