Hi All,
I need to change the number in the Message-Account: part in the body
of the NOTIFY message coming from Exchange 2010. My thought was to
use the textopts module. Is this the right way of doing it? Is
there a better way?
Thanks!
Kyle
___
Use
Thanks Bogdan, I will check that out.
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Thanks Anca and Bogdan. Both of you are right in a way. SIPP is, after a
while, leaving a lot of dialogs in "state 3". It could be because I am
using the same server as the SIPP UAS and UAC. I will need to split the
SIPP processes up to see if that changes things.
I am not sure if all the Dial
take care that the BYEs which are generated by opensips do not go
through the opensips main route, you need to configure a local_route{}
to get them...and make there a sip_trace()
Regards,
Bogdan
rad bogdan wrote:
Bogdan,
The lines related to siptrace are these:
modparam("siptrace", "db_url
Bogdan,
The lines related to siptrace are these:
modparam("siptrace", "db_url", "mysql://user:password@localhost/opensips")
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_flag",22)
modparam("siptrace", "trace_local_ip", "localhost")
modparam("siptrace", "traced_user_avp", "$avp(s:
Hi Bogdan,
What kind of tracing do you do? dialog based? with flags ?
Regards,
Bogdan
rad bogdan wrote:
Hi Bogdan,
I've seen that when CallControl notifies OpenSIPS (1.6.4) that a call
must be interrupted because the balance is 0, OpenSIPS sends BYE to
both the caller and the callee but the
Dan,
Brett and I communicated privately. It turns out the majority of both our
headaches with this issue is from the same, rather large carrier. Does
anything come to mind on how we might handle it?
I'd love an option I could include in engage_media_proxy to tell it to
immediately set up the
Hi Bogdan,
I've seen that when CallControl notifies OpenSIPS (1.6.4) that a call must be
interrupted because the balance is 0, OpenSIPS sends BYE to both the caller and
the callee but the messages are not being written into sip_trace.
Is this a normal behavior or it is a bug ?
Thanks,
Bogdan
On Fri, Jan 21, 2011 at 8:42 AM, Jeff Pyle chagrin I've seen a few more
carriers exhibit this type of behavior. One
>
> way or another I think we're going to have to find a way to accommodate
> it.
>
>
I'd just like to add my $0.02 here. I've had this exact same problem. It was
a while ago, so I d
Dan,
As I re-read this, I should clarify one point:
"...I think we're going to have to find a way to accommodate it."
I was referring to my team and I, not you. You don't "have" to do
anything. :)
- Jeff
On 1/21/11 9:42 AM, "Jeff Pyle" wrote:
>Dan,
>
>Excellent info.
>
>Here's the call
Dan,
Excellent info.
Here's the call flow I'm having trouble with:
- INVITE from Opensips to carrier (Sonus GSX), mediaproxy previously
engaged.
- GSX fires back 100.
- GSX tries a route on SS7 to terminate the call.
- Some media comes from the GSX on port, say, 16400 as a function of this
first
Ha! I looked everywhere for it but inside the binary itself. Yes, this will
likely take care of it. Thanks.
- Jeff
From: Duane Larson mailto:duane.lar...@gmail.com>>
Reply-To: OpenSIPS users mailling list
mailto:users@lists.opensips.org>>
Date: Thu, 20 Jan 2011 23:14:50 -0500
To: OpenSIPS
Hi Alan,
If still in memory, it means the dialog is considered as alive - please
also print the "state" field (both from fifo and DB).
Even so, the record is strange:
start_time is Thu, 20 Jan 2011 20:00:56 GMT
timeout is Sat, 01 Jan 2011 17:23:06 GMT
So timeout is before the start_tim
Hi Duane,
actually, checking the dump you sent, I see no trace of a leak - the pkg
dump shows only script and DB conn mem, while the shm is empty.
So, in your case, you may have a memory overload because because of
runtime issues (but not a leak). If dialogs are not removed by BYEs, I
guess
Hi Anton,
On 01/19/2011 06:15 PM, Anton Zagorskiy wrote:
Hi.
Please help me with advice.
I want to do a call limitation based on prefixes for each user in each
domain. In other words, I need to deny or allow call depend on username and
domain and value on a field in a sql table.
Which module s
Hi Duane,
On 01/20/2011 11:22 PM, Duane Larson wrote:
After looking a little more do you think this would cause the memory
issue...
Looks like my SIPP test starts the call, does whatever I want and
sends the BYE, but the dialog is still in the database after the call
has ended between the UA
On Thu, Jan 20, 2011 at 3:08 PM, Adrià Vidal wrote:
>
>
> On Thu, Jan 20, 2011 at 2:33 PM, Tijmen de Mes wrote:
>
>> Hi,
>>
>> This one line is not the only thing changed in cdr_opensips.php. You
>> should make sure you got the all the new files that are in the release.
>>
>> Did you also upgrad
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