I was reading Flavio's "Building Telephony Systems with OpenSER" chapter
about AVPOPs and he mentions that AVP's can be used for a whole domain. I
was thinking that I might be able to configure a area code for Company A's
domain and then route calls that way. If not that then I can set the AVP o
The following new features have been added to OpenSIPS:
1. Generic AAA API
The AAA API represents a set of generic callbacks and structures needed
for AAA operations.
The purpose of the API is to move all the AAA specific (Radius)
implementations
in a single module. The AAA API will hide the imp
Hey, asking is free. :)
My solution is to increase my min_se value to the least common denominator.
That reduces its usefulness for me but it's better than not having it at
all.
Thanks for the reality check.
- Jeff
On 8/18/09 9:44 AM, "Alex Balashov" wrote:
> I see what you mean, yeah.
I see what you mean, yeah. Unfortunately, session timers are for the
UAs to negotiate from a design and SIP specification standpoint. The
only thing the SST module provides is a thin layer of SE value
enforcement by the proxy.
In keeping with the sort of thing that a proxy is, it is not an ul
G'day mate,
is there an uninstall command for opensips? I am trying to uninstall the old
version and install a new version?
if there is no "make uninstall" available, do you know this feature will be
available in future release?
Cheers mate,
opensipser
Right. That was my fear. In my case the UAC knows nothing of session
timers. Its UAS (Opensips) adds the SST headers and relays the request. If
the far end replies with a 422, by default Opensips will relay the 422 to
the UAC who, well, won't know what to do with it.
It just doesn't seem fair
The SST module is designed for a scenario in which the proxy serves as
the endpoint of the SST negotiation. Otherwise, SST is up to the UA
endpoints to negotiate amongst themselves. So, SST does not deal with a
situation in which the proxy *receives* a 422; it only equips the proxy
to *send*
It seems very strange to me to have to manually manipulate headers that an
Opensips module added in the first place. Seems like bad things could
happen if the modules expects them to be there with certain values and they
have different values or gone altogether. If these headers are added in the
If I'm understanding the documentation correctly, you'd probably have to
do this with manual header manipulation.
Jeff Pyle wrote:
> On 8/18/09 8:51 AM, "Alex Balashov" wrote:
>
>> Sure, use a failure route and append_branch().
>
> Ok, but how do I adjust the timer value so it doesn't get 422
On 8/18/09 8:51 AM, "Alex Balashov" wrote:
> Sure, use a failure route and append_branch().
Ok, but how do I adjust the timer value so it doesn't get 422'd again? Or
is this handled automatically? The SST module documentation doesn't appear
to cover this.
- Jeff
__
Jeff Pyle wrote:
> For example, my SST module min_se value is set to 300. Let's say a far-end
> device responds with a 422 that contains "Min-SE: 1800". Is there a way
> within Opensips to handle this and re-relay the call with an adjusted
> Min-SE/Session Expires header?
Sure, use a failure ro
Hello,
Is there a way with the SST module to handle the receipt of a 422?
For example, my SST module min_se value is set to 300. Let's say a far-end
device responds with a 422 that contains "Min-SE: 1800". Is there a way
within Opensips to handle this and re-relay the call with an adjusted
Min-
Hi!
Now we have an old CUCM 4.1 Installation but in a few weeks we make an
upgrade to CUCM 7.
CUCM 7 understands SIP but i dont want to have the CUCM reachable from
outside.
Does anyone know if it works to setup OpenSips as an proxy for CUCM 7?
Has anyone done this before, are there tu
smadhoo6 wrote:
> How to configure Opensips (version 1.5.0) to use a particular CODEC say..
> Speex.?
This is like asking how to put the milk back in the cow with JSON.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+
smadhoo6 wrote:
> How to configure Opensips (version 1.5.0) to use a particular CODEC say..
> Speex.?
OpenSIPS does not "use codecs", OpenSIPS is a SIP Proxy.
Regards,
Thomas Gelf
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How to configure Opensips (version 1.5.0) to use a particular CODEC say..
Speex.?
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View this message in context: http://n2.nabble.com/CODEC-tp3465202p3465202.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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Hi Dmitri,
You can do this very simple with OpenSIPS/OpenSIP:
use the "registered" function ( see
http://www.opensips.org/html/docs/modules/1.5.x/registrar.html#id271315)
to check if the callee is registered with OpenSIPS or not.
Looking at the default OpenSIPS config file (
http://opensips.s
Hi Ghaith,
when you use RTPproxy in bridging mode, you need to explicitly indicate
(when calling force_rtp_proxy() ) what interface to use - see the "i"
and "e" flags :
http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362
You need to pass these flags when you do force
Hi,
Requirements on the format of CONTACT and TO headers are nonsense as
they are not used for routing at all. Only FROM (which provides info on
the caller) and RURI (request URI) (which provide info on callee).
So, bottom line, only the normalization of the RURI should be required
on the sys
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