I learned the hard way that sipp doesn't support record routing in their
default scenarios, at first i thought the example for load balancing was wrong
or something had changed from 1.5 to 1.7, sipp to asterisk directly worked
fine, but sipp -> opensips -> 2 asterisk didn't, the acks were reachi
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Thank you Bogdan for replying,
UAC send the ACK to the correct port of opensips but after it reaches
opensips, opensips doesn't know where to send this ACK
I tried to perform record routing in my cfg as in the performance test
tutorial( http://www.opensips.org/Resources/StressTests#toc6 test_E.c
Hello Guys,
Someone know this error ?
I'm using opensips 1.6.1 with rtpproxy 1.2.1 and Centos 6.0.
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27117]:
ERROR:sl:sl_send_reply_helper: response building failed
Dec 28 14:01:09 veoscol /usr/local/sbin/opensips[27116]:
ERROR:core:do_action: memor
The bottom idea is that you need to send a custom NOTIFY to the phone,
in order to restart (the ctl function has a builtin functionality for
the cisco-like phone).
So, you can directly use the MI function from TM to fire a NOTIFY
request (via t_uac_dlg command) :
http://www.opensips.org/
Thanks for the help Bogdan. I will do some tests and see if I can figure it.
Regards
Lee
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 28 December 2011 13:08
To: OpenSIPS users mailling list
Cc: Lee Archer
Subject: Re: [OpenSIPS-Users] B2BUA + RTPproxy + Asterisk direct media
He
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Hi guys,
I just opened a ticket, to allow these functions in STARTUP route.
Regards,
Bogdan
On 12/20/2011 11:35 PM, ddgiants wrote:
Hmmm thats what I am doing using startup and timer routes except I don't use
avp query to get data I use exec_avp.
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View this message in context:
http://open
Hi Schneur,
That depends on the phone you are using - so you need to check the docs
for that.
Regards,
Bogdan
On 12/28/2011 03:09 PM, Schneur Rosenberg wrote:
Is there a way to reboot phones, similar to asterisk "sip notify" command?
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Hi Rodrigo,
To be honest never tried the makeann tool, but to be sure the conversion
is correct, use "file ann_.wav" and "file ann_.wav.8" to
check if the the properties (stereo, sampling, etc) are the same.
Regards,
Bogdan
On 12/20/2011 09:03 PM, Rodrigo Ferreira wrote:
Hi
Ok, thank you, I will try
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Wednesday, December 28, 2011 5:05 PM
To: OpenSIPS users mailling list
Cc: dpa
Subject: Re: [OpenSIPS-Users] drouting and free memory
Hello,
DR module use private memory (pkg mem) in order to load info f
Is there a way to reboot phones, similar to asterisk "sip notify" command?
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Hello Lee,
Asterisk is doing the "direct media" by firing some re-INVITEs after the
call is up in order to exchange the media IPs of the the end points.
So, if this does not work, most probably you do not correctly handle the
re-INVITEs in opensips, like you are no forcing again rtpproxy for
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Hello,
DR module use private memory (pkg mem) in order to load info from DB.
Usually OpenSIPS has 2M of pkg memory (which is also used by other
module aside DR), so maybe that available amount of mem is not enough
for loading a 40K rows at a time.
Try to decrease the "fetch_rows" param to 10
Hi Cindy,
What you describe as bug #2 is not really a bug - First of all opensips
works at codec level - you are able to remove /change order of codecs,
but you cannot remove streams or sessions from SDP. So whatever info is
associated to the sessions/streams will stay there. Secondly, opensi
Hello!
There is such problem
Opensips 1.6.4-2
I am using drouting module to relay any calls.
modparam("drouting", "fetch_rows", 4)
When I am try using fifo dr_reload I see such error
“Dec 28 16:38:24 opensips /usr/local/opensips1.6.4-2/sbin/opensips[27360]:
ERROR:core:db_all
Hello,
Your opensips script does not do Record-Routing, so the sequential
requests (in-dialog requests, like ACK) will not go through opensips,
but rather directly between UAC and UAS, based on the Contact IPs.
Check on the UAC side, where the ACK request is send to.
Regards,
Bogdan
On 12/2
Hi Dmitriy,
The docs on the MI functions (names and what they are doing) can be
found as part of the module online docs (like
http://www.opensips.org/html/docs/modules/1.7.x/tm.html#id294662).
MI commands can be triggered via FIFO files, via XMLRPC or via simple
UDP package.
For same examp
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Hi all, I wonder if someone can help me. I have a system where I use the B2B
module and RTPproxy for inbound calls but once answered the call might jump
between Asterisk servers depending on what service is required. I would like
to use the Asterisk direct media option for SIP calls but when e
Hello, Bogdan.
Thank you for answer
Where i can read more about MI and how to use it. maybe examples.
Regards,
Dmitriy Abramov.
On Dec 27, 2011, at 2:15 PM, Bogdan-Andrei Iancu wrote:
> Hello Dmitriy,
>
> A call pickup scenario is related to call setup, so it is more appropriate to
> handle it
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