Hello,
I try use OpenSips to route VoIP between SipProxy-A and SipProxy-B.
SipProxy-A -> OpenSips --> SipProxy-B
OpenSips send INVITE with To: section contens OpenSips_IP.
"To: "
How to setup OpenSips when send INVITE to SipProxy-B, section To: contens
SipProxy-B_IP like this?
"To: "
Thanks for the suggestion -- unfortunately I saw this previously in a thread
and thought that'd be the key, but the results from cranking debug up the
same show it to be a different transaction on receiving the ACK and thus the
value is null. :(
I've played around with this just a little bit more
See if setting this param on helps.
http://www.opensips.org/html/docs/modules/devel/tm.html#id293118
-Brett
On Aug 4, 2011, at 7:48 PM, Bobby Smith wrote:
All,
Here's why I'm trying to accomplish:
route [subsequent_request] {
if (has_totag && is_method("ACK") {
if (MY_VAR = "cisco") {
Sorry, there were a couple of typos in that:
if (MY_VAR = "cisco") { >>>
<<< if (MY_VAR =~/== "cisco")
and assume onreply_route[foo] has a trigger armed (or is the default onreply
route).
I will post the context of my routing script if necessary but I really think
this is just a fundamental und
There is an opensips admin panel; Its php based; It is probably much
better supported.
I am not so sure the sermyadmin is well supported anymore;
On 8/4/11 9:12 PM, aelix systems wrote:
Hello,
I am first time user on this list.
When I try to create a "User Register" using OpenSIPS (ver: 1.6.4
Hello,
I am first time user on this list.
When I try to create a "User Register" using OpenSIPS (ver: 1.6.4-2) and
SerMyAdmin (2.0.1a ), I get Grails Runtime Exception -- see below for
details. I have seen this problem documented previous in the OpenSIPS' list,
but unfortunately, I didn't find a s
All,
Here's why I'm trying to accomplish:
route [subsequent_request] {
if (has_totag && is_method("ACK") {
if (MY_VAR = "cisco") {
lookup("location");
... relay and exit
}
else {
... relay and exit lr
}
}
}
onreply_route [foo] {
Hi all -
I was wondering if anyone ever saw this in their OpenSips/Postgres
installation: I am unable to add users using opensipsdbctl, but it is able
to read them fine.
On the command line, I'll give it a opensipsctl add aaron aaron
and the error I get back is
INFO: user 'aaron' already exists
Hi,
it is somehow that username from sip uri to be non case sensitive when
we talk about presence and xcap storage? I mean, if userA add userB, in
his contact list, i need userA to be able to add userB even he add
him(type) as USERB.
Dani
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Thanks,
I already did that.
Dani
On 08/04/11 18:19, Razvan Crainea wrote:
Hi Dani,
You can try by deleting the most common video codecs (like H261, H263,
H264).
You can do that using the codec_delete[1] functions from the textops
module.
I think you should also replace the video port with
Ok,
thanks for quick response.
Dani
On 08/04/11 18:26, Vlad Paiu wrote:
Hello,
Is it possible that you upgrade to 1.7 ? It is possible that this
issue was fixed in the latest OpenSIPS version.
If not, go to Makefile.defs, uncomment the line with
-DDBG_QM_MALLOC \
and comment the line w
bump :)
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bump :)
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Hello,
Is it possible that you upgrade to 1.7 ? It is possible that this issue
was fixed in the latest OpenSIPS version.
If not, go to Makefile.defs, uncomment the line with
-DDBG_QM_MALLOC \
and comment the line with
-DF_MALLOC \
and then recompile OpenSIPS.
Also set memlog=1 in you
Hi Dani,
You can try by deleting the most common video codecs (like H261, H263,
H264).
You can do that using the codec_delete[1] functions from the textops module.
I think you should also replace the video port with 0.
[1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910
Hi,
I know that, but how can i check why :)
root@test:~# opensips -V
version: opensips 1.6.4-2-tls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, USE_SCTP, DISABLE_NAGLE,
USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV
Hi,
In fact, i have some problems with one of my pstn gw's that send "400
Incorrect content length", i think, because of too long sip packet. So,
because it is pstn, i want to remove video capability(many lines in
first invite packet).
Dani
On 08/04/11 17:02, Razvan Crainea wrote:
Hi Dani,
Hi Dani,
It seems you are out of memory. What version of OpenSIPS are you using?
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 16:07, Dani Popa wrote:
Hi,
How can i solve this kind of problems ? Opensips doesn't crash, but it
not respond to any sip requests.
Aug 3 07:36:48 t
Hi Dani,
Why would you do that? If you don't want to allow video, you can simply
replace the video port in the "m=" line with 0.
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 16:58, Dani Popa wrote:
Hi all,
How can i remove all sip video body headers regardin video. Should i
r
Hi all,
How can i remove all sip video body headers regardin video. Should i
remove any line from body after "m=video", or how. Please give me a
hint, if you have.
Thanks,
Dani
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Hi,
How can i solve this kind of problems ? Opensips doesn't crash, but it
not respond to any sip requests.
Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]:
WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation
Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]:
Hi,
I have a requirement where I must set up an RCS compliant server to support
group chat and IM conference.
Now I have OpenSIPS set up for the VoIP calls.
How easy would it be to integrate RCS and OpenSIPS server ?
Is it possible to extend OpenSIPS to make it RCS Compliant?
Thank you,
Nethra Ch
Check out the uac_registrant module avaliable in 1.7:
http://www.opensips.org/html/docs/modules/1.7.x/uac_registrant.html
Please note that future private e-mails will be ignored. Please keep
the mailing list in cc.
Regards,
Ovidiu Sas
On Thu, Aug 4, 2011 at 1:56 AM, wrote:
>
> Hi
> What shal
Hello Alan,
Are you using other functions in your script that might alter the
Contact header ?
Because of OpenSIPS internals, you can alter the same message part only
once, and if you use the topology hiding from the dialog module along
with other Contact header alterations from the script,
Hello,
I try function.pl from PERL samples of OpenSips v 1.7.0.
I get follow messages:
ERROR:core:moduleFunc: Module function 'sl_send_reply' is unsafe. Call is
refused.
ERROR:core:XS_OpenSER__Message_moduleFunction: calling module function
'sl_send_reply' failed. Missing loadmodule?
Where is p
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