Anyone has any clue where to look?
Van: MT bern...@buitenhuis.numailto:bern...@buitenhuis.nu
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Datum: maandag 14 april 2014 12:35
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Hi
I have a problem with accc modules with the below scenario :
OPENSIPS receive the 200 ok from the trunk gateway ,and forward it to UA,
UA replies by ACK and opensips forward this ACK to Gateway ,..but for some
reason the Gateways resends a lot of 200 OK with SDP and opensips relay
Hello,
I was able to install OpenSIPS and RTPProxy ( taken from here ) on
Amazon EC2, and onn the mobile side I am using PJSIP library.
I am able to make VOIP call from ios simulator - iphone5 and it
works well, if the iphone5 is connected to a WiFi, but if I turn off
the WiFi on the device and
Hi Liviu,
I think this would be a very useful feature as well. Specifically I'd like
to be able to say something like
:dlg_end_profile
profile_name
and it'd hang up all calls (dlg_end_dlg method) all calls in that profile.
Given that you already know what calls are in a profile, this should be a
Very useful indeed. +1
- Jeff
On Tue, Apr 15, 2014 at 10:28 AM, Brett Nemeroff br...@nemeroff.com wrote:
Hi Liviu,
I think this would be a very useful feature as well. Specifically I'd like
to be able to say something like
:dlg_end_profile
profile_name
and it'd hang up all calls
Hello,
It works, but:
1) get_timestamp doesnt work inside has_totag section
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
get_timestamp($avp(secbye),$avp(usecbye));
.
.
but works if called before that section
2)
You don't really need Options for TCP. You use Options to keep UDP
ports/connection open. TCP is a stateful connection. You only need to make
sure that opensips keeps the connection open for the period of the time it
is needed.
is_from_gw should work since it works based on the src_ip.
Regards,
Have a look at your system network configuration and aliases in opensips?
Where do these messages come from and where do they go? Have you checked if
they are set properly before they reach opensips.
Regards,
Ali Pey
On Tue, Apr 15, 2014 at 7:14 AM, B. Buitenhuis bern...@buitenhuis.nuwrote:
I second this as well. +1
Thanks,
Ali Pey
On Tue, Apr 15, 2014 at 10:48 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Very useful indeed. +1
- Jeff
On Tue, Apr 15, 2014 at 10:28 AM, Brett Nemeroff br...@nemeroff.comwrote:
Hi Liviu,
I think this would be a very useful feature as
Hi ,
I'm not using any aliases in my config and I've tested the config with
different machines in different networks.
About the messages
They are set properly before they reach open sips. Weird thing is that after
running the open sips deamon (about a hour) the problem looks solved.
Van: Ali
Are you running on a VM environment? If so, try your script on a bare metal
hardware. VMs mess with OS quite a bit.
Regards,
Ali Pey
On Tue, Apr 15, 2014 at 1:33 PM, B. Buitenhuis bern...@buitenhuis.nuwrote:
Hi ,
I'm not using any aliases in my config and I've tested the config with
I'll try it on a bare metal.
Good point!
Van: Ali Pey ali...@gmail.commailto:ali...@gmail.com
Beantwoorden - Aan: OpenSIPS users mailling list
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Datum: dinsdag 15 april 2014 19:36
Aan: OpenSIPS users mailling list
Thanks Razvan.
I'm still trying to troubleshoot this problem, i've seen that in the opensips
logs, when i place a call, there are the following error messeges:
Apr 15 21:21:31 sip-lab /usr/local/opensips110/sbin/opensips[14226]:
INFO:core:probe_max_sock_buff: using snd buffer of 255 kb
Apr 15
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