Hi,
I also had similar problems to log into Siremis some weeks ago. It
was caused by a combination of circumstances.
First I had an error with the mysql password configured for Siremis.
No error messages pointed to that direction. After correcting the
password, I couldn't connect.
El Thu, 27 May 2010 04:44:53 +0100
Ricardo Coelho escribió:
> Hi,
>
> Right now I have 2 machines (one on domainA and the other on domainB). I
> want to allow a phone registered in the openser from domainA to call a phone
> registered in the openser from domainB (and vice-versa). I have alread
In fact I want to send 1941 from Mitel IPBX to kamailio and kamailio
return only 941 in the cisco call manager, that's why I say that
kamailio absorbs the number one.
Cisco Call Manager<->kamailio<--->IPBX Mitel
kamailio send 941<--receives 1941 and absorbs the 1<---send 1941
I think the database entries are setup ok, I used the example in the
module doc, but I can share that if needed. Any guidance will be
appriciated.
Thanks.
JR
Hello JR
Can you provide more info regarding the problem. Any special logging
info in the debug log?
You can increase the deb
Thank you very much. It works with strip(1);
best regards
Henning Westerholt wrote:
On Wednesday 26 May 2010, dotnetdub wrote:
I want to absorb a number with kamailio just to make out only three but I
can not. In fact I send 1941 kamailio and i spring 941 with kamailio so I
absorb a number.
Hello,
On 5/25/10 8:32 PM, Daniel-Constantin Mierla wrote:
Hello,
I am planning to release kamailio 3.0.2 this Thursday. There were some
fixes since 3.0.1 that worth to be packaged. If you have major reports
for current stable version, please write to sr-...@lists.sip-router.org
if anyone ha
On May 27, 2010 at 10:56, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 5/25/10 8:32 PM, Daniel-Constantin Mierla wrote:
> >Hello,
> >
> >I am planning to release kamailio 3.0.2 this Thursday. There were
> >some fixes since 3.0.1 that worth to be packaged. If you have
> >major reports for curre
On May 26, 2010 at 21:36, Alex Balashov wrote:
> On 05/26/2010 09:28 PM, Zahid Mehmood wrote:
>
> >Can it be set to a value> 30 now? I remember 30
> >being the max at some point.
>
> It seems to me, from my inspection of the code, Jiri Kuthan's
> answer, and a priori reasoning, that it can
On Thursday 27 May 2010, JR Richardson wrote:
> I am lab testing carrierroute modue on kamailio 1.5.4-notls
> (i386/linux) and have a question on how to continue processing a call
> if kamailio sends a call with t_relay() but does not get a response
> from the gateway.
>
> I read about the timers
On 5/27/10 11:12 AM, Andrei Pelinescu-Onciul wrote:
On May 27, 2010 at 10:56, Daniel-Constantin Mierla wrote:
Hello,
On 5/25/10 8:32 PM, Daniel-Constantin Mierla wrote:
Hello,
I am planning to release kamailio 3.0.2 this Thursday. There were
some fixes since 3.0.1 that worth to b
Hello,
Kamailio v 1.4.2
I have problems with canceling calls. When CANCEL arrive to kamailio
function t_check_trans always return false. Cancel message arrive after
75s. I was try with "t_newtran", but without success. The
only thing which I found that could be a problem is different branch on
On May 27, 2010 at 11:50, Ernest Mavrel wrote:
>
Please try to send mails as plain text and not as html.
> Hello,
>
> Kamailio v 1.4.2
> I have problems with canceling calls. When CANCEL arrive to kamailio
> function t_check_trans always return false. Cancel message arrive after
> 75s. I was
On 05/27/2010 03:16 AM, Vicente wrote:
> Hi,
>
> I also had similar problems to log into Siremis some weeks ago. It
> was caused by a combination of circumstances.
>
> First I had an error with the mysql password configured for Siremis.
> No error messages pointed to that directio
I wasn't complaining :) I just wanted to confirm if there was an upper bound.
So far I have not had the need for more than 30 branches.
Thanks for the clarification.
--
Zahid
On May 27, 2010, at 5:29 AM, Andrei Pelinescu-Onciul wrote:
> On May 26, 2010 at 21:36, Alex Balashov wrote:
>> O
On Thu, May 27, 2010 at 1:10 PM, Joe Micciche wrote:
> On 05/27/2010 03:16 AM, Vicente wrote:
>> Hi,
>>
>> I also had similar problems to log into Siremis some weeks ago. It
>> was caused by a combination of circumstances.
>>
>> First I had an error with the mysql password configured f
Insert both domains in both mysql databases of each openser
On May 27, 2010, at 8:25 AM, Jon Bonilla (Manwe) wrote:
> El Thu, 27 May 2010 04:44:53 +0100
> Ricardo Coelho escribió:
>
>> Hi,
>>
>> Right now I have 2 machines (one on domainA and the other on domainB). I
>> want to allow a phone
Hi,
regarding the missing Contact header in the REFER message, it can be solved
by including some lines in /modules_k/dialog/dlg_ transfer.c. In this file,
the Contact of the initial INVITE generated with dlg_bridge command can be
easily changed as well.
The problem now is only in the processing
On 05/27/2010 01:10 PM, Joe Micciche wrote:
On 05/27/2010 03:16 AM, Vicente wrote:
Hi,
I also had similar problems to log into Siremis some weeks ago. It
was caused by a combination of circumstances.
First I had an error with the mysql password configured for Siremis.
No err
I just tested with Asterisk trunk version and it does not have this bug.
Maybe you should update your Asterisk server.
regards
klaus
Am 27.05.2010 11:50, schrieb Ernest Mavrel:
Internet Protocol, Src: 80.81.82.83 (80.81.82.83), Dst: 123.124.125.126
(123.124.125.126)
User Datagram Protocol, Sr
On Thu, May 27, 2010 at 2:39 AM, marius zbihlei wrote:
>
>>
>> I think the database entries are setup ok, I used the example in the
>> module doc, but I can share that if needed. Any guidance will be
>> appriciated.
>>
>> Thanks.
>>
>> JR
>>
>
> Hello JR
>
> Can you provide more info regarding th
On Thu, May 27, 2010 at 4:38 AM, Henning Westerholt
wrote:
> On Thursday 27 May 2010, JR Richardson wrote:
>> I am lab testing carrierroute modue on kamailio 1.5.4-notls
>> (i386/linux) and have a question on how to continue processing a call
>> if kamailio sends a call with t_relay() but does not
Hi,
I couldn't find the module that has the functions for modifying pAsserted
field. Your help would be appreciated.
Thanks,
AR
_
Hotmail has tools for the New Busy. Search, chat and e-mai
El Thu, 27 May 2010 14:01:03 +0100
Ricardo Coelho escribió:
> Insert both domains in both mysql databases of each openser
>
Do you mean that for calling a u...@company.com account I should add
company.com to my database?
I am sorry but I still don't understand what you mean. In which tables a
Alex rsm writes:
> I couldn't find the module that has the functions for modifying
> pAsserted field. Your help would be appreciated.
you don't modify it. instead you remove it in incoming request and add
a new one to outgoing request.
-- juha
___
SI
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 05/27/2010 10:15 AM, Elena-Ramona Modroiu wrote:
> Note as well that in some OSes, connecting to openser database from
> 127.0.0.1 is not same as localhost (which is used by kamctl to create
> openser user and database), so try with localhost instea
JR Richardson wrote:
On Thu, May 27, 2010 at 2:39 AM, marius zbihlei wrote:
I think the database entries are setup ok, I used the example in the
module doc, but I can share that if needed. Any guidance will be
appriciated.
Thanks.
JR
Hello JR
Can you provide more info regarding
On Thursday 27 May 2010, JR Richardson wrote:
> > i assume according your description that the GW in question don't send a
> > provisional response. Then the tm module should generate a internal 408
> > after fr_timer interval which you could then catch in a failure_route. Do
> > you armed the appr
Hi,
Forget the domains and tables. What I want is: I have 2 machines running
openser and some phones are registered on openserA and some on openserB. I want
to allow a phone in openserA to call a phone in openserB.
Sorry for the mess and thanks
On May 27, 2010, at 3:47 PM, Jon Bonilla (Manwe)
alternatively you could use substitution expressions in
textops module's replace function.
-jiri
Juha Heinanen wrote:
Alex rsm writes:
I couldn't find the module that has the functions for modifying
pAsserted field. Your help would be appreciated.
you don't modify it. instead you remove it
El Thu, 27 May 2010 16:09:35 +0100
Ricardo Coelho escribió:
> Hi,
>
> Forget the domains and tables. What I want is: I have 2 machines running
> openser and some phones are registered on openserA and some on openserB. I
> want to allow a phone in openserA to call a phone in openserB.
>
> Sorry
Am 27.05.2010 17:09, schrieb Ricardo Coelho:
I have 2 machines running openser and some phones are registered on openserA
and some on openserB.
Why are some phones registered at openserA and others at openserB?
- Do they have different SIP domains where one SIP domain point to A and
other
On Thu, May 27, 2010 at 10:09 AM, Henning Westerholt
wrote:
> On Thursday 27 May 2010, JR Richardson wrote:
>> > i assume according your description that the GW in question don't send a
>> > provisional response. Then the tm module should generate a internal 408
>> > after fr_timer interval which
On Thu, May 27, 2010 at 9:57 AM, marius zbihlei wrote:
> JR Richardson wrote:
>>
>> On Thu, May 27, 2010 at 2:39 AM, marius zbihlei
>> wrote:
>>
I think the database entries are setup ok, I used the example in the
module doc, but I can share that if needed. Any guidance will be
>>
JR,
Maybe for subsequent routes you'd like to use $oU (original URI Username)
that is the original number you intend to route.
Rgds,
Uriel
On Thu, May 27, 2010 at 1:23 PM, JR Richardson wrote:
> On Thu, May 27, 2010 at 10:09 AM, Henning Westerholt
> wrote:
> > On Thursday 27 May 2010, JR Richa
Hello,
Kamailio v3.0.2 was just released. It is a patch release that includes
the fixes done since release of v3.0.1. You should consider to upgrade
if you run v3.0.0 or or v3.0.2, there is no change that you should do to
configuration file or database, just reinstall the binaries.
Source ta
Hi All,
Kamailio 3
I currently store a default RPID in our database. We allow some asterisk
customers to send us the RPID.
I want to store the RPID sent by the customer on the radius record.
We test:
if!(is_present_hf("Remote-Party-ID")) {
#RPID Not Set by Asterisk - Will set
On 05/27/2010 03:56 PM, dotnetdub wrote:
Not really sure how to extract the RPID from the SIP message sent from
the asterisk.
You can get the full header value of any header using the $hdr(...)
container[1], e.g.
$hdr(Remote-Party-ID)
There is also a dedicated pseudovariable to expose t
On Thu, May 27, 2010 at 1:24 PM, Uriel Rozenbaum
wrote:
> JR,
>
> Maybe for subsequent routes you'd like to use $oU (original URI Username)
> that is the original number you intend to route.
>
> Rgds,
> Uriel
>
> On Thu, May 27, 2010 at 1:23 PM, JR Richardson
> wrote:
>>
>> On Thu, May 27, 2010 a
I am working on a project that aims to improve scalability of a system and it
is required that only some phones are attached to openserA and all the others
are attached to openserB. I am new to openser so forgive me for the lack of
knowledge.
Thanks
On May 27, 2010, at 5:05 PM, Klaus Darilio
Hi, Ricardo...
Just help us understand your needs so that we can better help you...
When you talk on scalability, on how many registered users are we
talking about?
Are all this users just registered, or are you planning to use RTPProxy?
If so, on the same machine?
Just let say that just for
Am 27.05.2010 23:09, schrieb Ricardo Coelho:
I am working on a project that aims to improve scalability of a
system and it is required that only some phones are attached to
openserA and all the others are attached to openserB. I am new to
openser so forgive me for the lack of knowledge.
Ok. B
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