Hello everybody!
There was a news in web site of Opensips about ACC module and CDR generation
http://lists.opensips.org/pipermail/news/2010-August/96.html
I use 1.6.3. Can I already install ACC for the Opensips with new feature? Or it
will be accessed only in next release of
On 2 Dec 2010, at 20:04, Richard Revels wrote:
Here is the m line from an INVITE/200 after the messages were
modified by use_media_proxy in each direction. The call happened to
be mine and although it did not stay up long enough for me to be
100% sure, I think the person on the other end
Hi, using Opensips 1.6.2. I have observed that soemtimes Opensips changes the
RTP port in the SDP description. This is when using Bria Counterpath client
with another softphone. It seems to be because the Bria client has an RTP port
of 4000 which gets remapped to something higher eg 4100. This
Hi Nauman,
OpenSIPS by itself does not change the SDP content, only if instructed
by the config file, by using function from nathelper/mediaproxy
modules. Do you use any of them ?
Regards,
Bogdan
Nauman Sulaiman wrote:
Hi, using Opensips 1.6.2. I have observed that soemtimes Opensips
Hi Denis.
The feature to directly generate CDRs in available in OpenSIPS trunk
(devel) and it will be available in opensips 1.6.4 stable (in mid December)
Regards,
Bogdan
Denis Putyato wrote:
Hello everybody!
There was a news in web site of Opensips about ACC module and CDR
Hi Gavin,
Do you think it will be helpful for you to see (from OpenSIPS) which
are the these ghost calls ? (I can describe a procedure to get such a
listing).
Now, about what to do to prevent...More or less is about detecting so
called ghost callsand you have several options:
1)
Bogdan, i understand, thank you
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, December 06, 2010 1:49 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ACC module
Hi
Hello list,
I've been played with b2bua module using the top hiding scenario
using 1.6.2 opensips version.
It seems to be working fine, but I need to call force_rtp_proxy() for
the INVITE requests to force the uac send rtp to rtpproxy IP.
I called it in the script at this point:
Hello.
I'm using a dialog module and store values in dialogs. Is it possible to
fetch values from dialog while ACC logging?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
Hello Anton
And a reason of this? And what do you mean while ACC logging?
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
Sent: Monday, December 06, 2010 5:27 PM
To: 'OpenSIPS users mailling list'
Subject:
When INVITE is sent and OK isn't received $DLG_status equals 2. But in the
documentation $DLG_status can be NULL, 3, 4, 5.
When does $DLG_status is 2? Can it has other values?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
If I were to use freeradius for accounting, where should I put user list?
and also I'd like to check credit as well before calling.
Thanks
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Hi,
I was wondering if anyone had any experience getting a Cisco 7960
phone to register to opensips when the phone is behind a PIX firewall.
I'm having a hell of a time getting it to register.
I see these messages:
U nat.ip:2260 - opensips.ip:5060
REGISTER sip:opensips.ip SIP/2.0..Via:
On 6 December 2010 11:04, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
Hi Gavin,
Do you think it will be helpful for you to see (from OpenSIPS) which are
the these ghost calls ? (I can describe a procedure to get such a listing).
Yes, that would be great. We have tracked them down and
I use the command 'opensipsctl fifo dlg_list' to check the active dialogs in
our system (with a interface to our CRM).
I want to match the dialog on username, this is not always possible (not in
to_uri, callee_contact etc).
Question: Is it possible to add one or more custom fields to the
James,
On your TFTP server, add this line to either SIPDefault.cnf or
SIPmacaddress.cnf:
nat_enable : 1
Then reboot your phone.
Mario
http://advantia.ca
On Mon, Dec 6, 2010 at 10:17 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I was wondering if anyone had any experience getting a Cisco
Hi Mario,
On Mon, Dec 6, 2010 at 11:08 AM, Advantia VoIP Systems i...@advantia.ca wrote:
James,
On your TFTP server, add this line to either SIPDefault.cnf or
SIPmacaddress.cnf:
nat_enable : 1
Then reboot your phone.
Adding that results in a problem where OpenSIPS does not send the
reply
Yes, you are right there is no information about status 2 in doc. But as I
understand status 2 exists during time after create dialog and until final
reply.
Then status can be 3,4 (if final reply received) or 5(if there is no final
reply).
-Original Message-
From:
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