Hi,
I'm using Kamilio 3.3.0 as registrar server. I`m using an outbound proxy
so 'use_path' parameter or 'registrar' module is enabled.
According to RFC 5626, a re-registration from a specific combination of
AoR, instance_id and reg_id must update the binding.
If the registrar receives a
I just installed a Siremis 3.2.1 on a server with an already working Kamailio
3.2.1. bellow is a typical error message I keep getting.
Can you point me to a solution
-Thanks
[http://192.168.105.196/siremis/themes/default/images/icon_forbidden.gif]
System Internal Error
The detailed error
Hi, full agree with this bug report. Adding the devel maillist. More
comments at the end of the mail:
2012/7/26 José Luis Millán jmil...@aliax.net:
Hi,
I'm using Kamilio 3.3.0 as registrar server. I`m using an outbound proxy so
'use_path' parameter or 'registrar' module is enabled.
Antanas Masevicius writes:
it looks like column order a bit changed in my system after using alter table
during migrations.
That is prefix col is not it its place. Thus this data:
(1424,90,'OP1','10.10.10.10','',5060,NULL,1,0,NULL,'11#',1,NULL,NULL);
should be fixed as:
Hello Juha,
i have my lcr_count set as modparam(lcr, lcr_count, 102).
This is really messy then. Is there something else which could help to
understand the cause of this?
I see you use sip-proxy as flavorful, maybe there something is a bit
different?
Antanas
On 2012.07.27 12:41, Juha Heinanen
Antanas Masevicius writes:
i have my lcr_count set as modparam(lcr, lcr_count, 102).
This is really messy then. Is there something else which could help to
understand the cause of this?
i did my tests on amd64 architecture, but it should not matter because
all ints in your data fit into
I want retransmit some me ssages to other sip server as is.
f.e server sip1.com receives invite message
INVITE u...@somewhere.org
i want to send it to server sip2.com as is.
Unfortunately, if i use rewritehostport() function, or modify $rp $rd
pseudovariables
fe $rp='sip2.com', kamailio changes
Hello,
just to be sure before going to any further investigation (as I
remember, such case I tested a bit with some command line tools due to
lack of a sip phone with good ob/gruu support), do you have in the config:
modparam(registrar, gruu_enabled, 1)
The default config file in 3.3, has
Hi Users,
This is my ultimate try to ask for some help to debug the two cores
generated by fm_realloc() function. If somebody can give me a feedback
about my questions or doubts I will be very grateful, this user list of
kamailio always help me to solve my problems.
Best Regards
2012/7/23
I'm not yet using Kamailio. I am looking for something to do a job which
sounds really simple but may not be. Our PBX connects to a SIP service
(voicetrading.com) but does not supply our CallerID, this means that
voicetrading.com provide some very weird callerids for the person at the
other
Thanks Charles for that info, I will look at it (but I may not
understand it...)
Our PBX is Voispeed v3 (runs on Windows!). Current Voispeed is v4.6
and probably has SIP CallerID functionality, but the older version
is otherwise fine and we have built some stuff
Hi Gary,
It was fixed already by Richard in 3.3 branch:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=28be16549831df46dd1b8312da223b02359d8a9c
(and master)
Thank you for the report.
On 07/24/2012 09:44 PM, Gary Chen wrote:
Sorry, it should be Kamailio 3.3.0 not 3.2.0.
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