Re: [SR-Users] siptrace + flatstore

2013-04-29 Thread Daniel-Constantin Mierla

Hello,

set debug=3 in you configuration file and look at the logs to see if you 
spot any hint in the messages.


Cheers,
Daniel

On 4/25/13 4:31 PM, Victor V. Kustov wrote:

Hi all

loadmodule siptrace.so
modparam(siptrace, db_url, flatstore:/var/log/siptrace)

call sip_trace();

nothing in /var/log/siptrace, no errors in logs. whats wrong?



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Re: [SR-Users] kamaillio as SBC

2013-04-29 Thread Daniel-Constantin Mierla

Hello,

On 4/23/13 8:35 AM, Vu Ngo thanh wrote:
I am interested in testing Kamaillio acting as an SBC. The SBC as 
Outbound proxy server, the IPPhone can register to SIP Server via SBC 
and make a call via SBC. the topology is bellow


The IP Phone A, The IP Phone B - register ---  FreeSwitch 
(SBC-Outbound Proxy)  SIP Server


Could you please help me how to install and configure FreeSwitch as 
SBC-Outbound Proxy ?
probably you get more help about installing and configuring FreeSwitch 
from its mailing lists.


Here is a tutorial showing kamailio - freeswitch integration, in 
different topology, but it may help to some extent:


  - http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc

Cheers,
Daniel

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Re: [SR-Users] kamaillio as SBC

2013-04-29 Thread Alex Balashov
And if you're interested in learning more about the idea of Kamailio's 
suitability for an SBC role, see my blog post at Likewise.am:


http://www.likewise.am/2013/03/kamailio-as-an-sbc-session-border-controller/

-- Alex

On 04/29/2013 03:54 AM, Daniel-Constantin Mierla wrote:

Hello,

On 4/23/13 8:35 AM, Vu Ngo thanh wrote:

I am interested in testing Kamaillio acting as an SBC. The SBC as
Outbound proxy server, the IPPhone can register to SIP Server via SBC
and make a call via SBC. the topology is bellow

The IP Phone A, The IP Phone B - register ---  FreeSwitch
(SBC-Outbound Proxy)  SIP Server

Could you please help me how to install and configure FreeSwitch as
SBC-Outbound Proxy ?

probably you get more help about installing and configuring FreeSwitch
from its mailing lists.

Here is a tutorial showing kamailio - freeswitch integration, in
different topology, but it may help to some extent:

   - http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc

Cheers,
Daniel

--
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   *http://asipto.com/u/katu  *



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Re: [SR-Users] siptrace source ip causes martians

2013-04-29 Thread Thilo Bangert
On Monday, April 22, 2013 10:14:39 AM you wrote:
 Hi,
 
 i have a kamailio instance, which is connected to two subnets (A and B) and
 exchanges sip messages between them. I have setup the siptrace module to
 send hep tracing messages of all sip messages to a host in subnet B.
 
 This works really well for sip messages arriving on the IP of subnet B - for
 those arriving on subnet A however it doesnt work. The hep messages have
 the interface address of subnet A set as source address.
 Since the machine receiving the HEP messages also is connected to both
 subnet A and B, these messages are (correctly) rejected as martians and
 never reach the sipcapture module on that host.
 
 this is on kamailio 3.3.4.
 
 Is there a way to force the source IP of outgoing HEP messages in the
 siptrace module?

setting 

mhomed=1

fixed this for me. although i didnt need it for my sip routing before.

thanks
kind regards
Thilo

 
 thanks
 kind regards
 Thilo

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[SR-Users] Fwd:

2013-04-29 Thread Charles Chance
http://www.toyosoken.co.jp/ivtduw.php


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[SR-Users] Rewrite RURI

2013-04-29 Thread Grant Bagdasarian
Hello,

Is it possible to have Kamailio rewrite the Request URI of the INVITE message 
but sent the INVITE to another address first?
For example

-  Kamailio (10.0.0.1) receives an INVITE

-  Kamailio rewrites the RURI to 10.0.0.3

-  Kamailio sends the INVITE to 10.0.0.2

-  The application at 10.0.0.2 sends the INVITE to 10.0.0.3

Regards,

Grant
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Re: [SR-Users] Rewrite RURI

2013-04-29 Thread Alex Balashov
Yes, this is achieved by changing the destination set ($du), or direct 
arguments to t_relay():


   http://kamailio.org/docs/modules/4.0.x/modules/tm.html#t_relay

This has the effect of saying, set the Request URI to X, but on the 
network (IP) and transport (UDP, TCP, etc.) level, send the request 
physically to Y.


-- Alex

On 04/29/2013 04:32 AM, Grant Bagdasarian wrote:


Hello,

Is it possible to have Kamailio rewrite the Request URI of the INVITE
message but sent the INVITE to another address first?

For example

-Kamailio (10.0.0.1) receives an INVITE

-Kamailio rewrites the RURI to 10.0.0.3

-Kamailio sends the INVITE to 10.0.0.2

-The application at 10.0.0.2 sends the INVITE to 10.0.0.3

Regards,

Grant



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235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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[SR-Users] !!! about private emails

2013-04-29 Thread Daniel-Constantin Mierla

Hello,

I would like to remind that, at least in my case, writing emails 
directly without cc-ing the mailing list is not recommended. The rule is 
also suggested in the mailing lists presentation page:


  - http://www.kamailio.org/w/mailing-lists/

Unless I asked explicitly for a private email with some specific 
details, the message will not be noticed and therefore not replied. I 
use this email address for many public mailing lists and it is intended 
only for that usage, I am reacting based on the filters I created, 
unmatched messages getting the lowest priority.


There are several common sense as well as technical reasons. Here are 
some along with other suggestions to improve the likeliness of getting 
an answer:
- we are not answering questions on mailing lists to help only one 
person, but also other people that may have similar issues in the future 
-- they can find the answers in the archive with a search engine
- the amount of unexpected messages is very high -- it is practically a 
zero chance to get to the folder with messages that didn't match any 
filter (thus your message is not going to be replied)
- whenever I find time, the first emails I answer are those coming on 
mailing lists. There are many there as well, people are traveling or 
having other personal or business projects, so sometime is good to send 
a reminder if a question does not get an answer
- starting with negative approach or no technical content, like you are 
going to use something else or this application is missing what so ever 
tutorial giving the solution you need or simply asking for full config 
of complex requirements, is not helping, making people ignore your 
messages. Start with what you tried and where you got stuck. Everything 
you get is for free, nobody here owe you anything.


Therefore, if anyone sent me a private message asking about kamailio or 
continuing a discussion started on mailing lists, then the messages have 
to be resent on mailing list if one wants an answer. A quick look in my 
unfiltered folder revealed hundreds of messages during the past year, 
soon many will be deleted anyhow. Note that you can always mask 
sensitive data (e.g., replace passwords, IP addresses) before sending 
logs or config file.


Cheers,
Daniel

--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24016, 2013
  * http://asipto.com/u/katu *


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Re: [SR-Users] Rewrite RURI

2013-04-29 Thread Carsten Bock
Hi Grant,

that's easy:

$ru = sip:$rU@10.0.0.3;
$du = sip:10.0.0.2;

Done.

Carsten

2013/4/29 Grant Bagdasarian g...@cm.nl:
 Hello,



 Is it possible to have Kamailio rewrite the Request URI of the INVITE
 message but sent the INVITE to another address first?

 For example

 -  Kamailio (10.0.0.1) receives an INVITE

 -  Kamailio rewrites the RURI to 10.0.0.3

 -  Kamailio sends the INVITE to 10.0.0.2

 -  The application at 10.0.0.2 sends the INVITE to 10.0.0.3



 Regards,



 Grant


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[SR-Users] Rewrite RURI

2013-04-29 Thread Juha Heinanen
Grant Bagdasarian writes:

 Is it possible to have Kamailio rewrite the Request URI of the INVITE
 message but sent the INVITE to another address first?

yes, you can set $du and t_relay() sends the request accordingly no
matter what r-uri is.

-- juha

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Re: [SR-Users] Fwd:

2013-04-29 Thread Charles Chance
Hi,

Sincere apologies, not sure how this happened - please ignore, it was not
me who sent it :/

Regards,

Charles


On 2 April 2013 08:21, Charles Chance charles.cha...@sipcentric.com wrote:

 http://www.toyosoken.co.jp/ivtduw.php


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Re: [SR-Users] Rewrite RURI

2013-04-29 Thread Grant Bagdasarian
Cool. Thanks!

-Original Message-
From: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov
Sent: Monday, April 29, 2013 10:37 AM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Rewrite RURI

Yes, this is achieved by changing the destination set ($du), or direct 
arguments to t_relay():

http://kamailio.org/docs/modules/4.0.x/modules/tm.html#t_relay

This has the effect of saying, set the Request URI to X, but on the network 
(IP) and transport (UDP, TCP, etc.) level, send the request physically to Y.

-- Alex

On 04/29/2013 04:32 AM, Grant Bagdasarian wrote:

 Hello,

 Is it possible to have Kamailio rewrite the Request URI of the INVITE 
 message but sent the INVITE to another address first?

 For example

 -Kamailio (10.0.0.1) receives an INVITE

 -Kamailio rewrites the RURI to 10.0.0.3

 -Kamailio sends the INVITE to 10.0.0.2

 -The application at 10.0.0.2 sends the INVITE to 10.0.0.3

 Regards,

 Grant



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United States
Tel: +1-678-954-0670
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Re: [SR-Users] Rewrite RURI

2013-04-29 Thread Olle E. Johansson

29 apr 2013 kl. 10:32 skrev Grant Bagdasarian g...@cm.nl:

 Hello,
  
 Is it possible to have Kamailio rewrite the Request URI of the INVITE message 
 but sent the INVITE to another address first?
 For example
 -  Kamailio (10.0.0.1) receives an INVITE
 -  Kamailio rewrites the RURI to 10.0.0.3
 -  Kamailio sends the INVITE to 10.0.0.2
 -  The application at 10.0.0.2 sends the INVITE to 10.0.0.3

Check the $du pseudo-variable that sets the destination of the message 
(outbound proxy) for a request. You can also
use the functions that relay/forward to specific address instead of the RURI.

So yes, it's possible.

/O

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Re: [SR-Users] !!! about private emails

2013-04-29 Thread Olle E. Johansson

29 apr 2013 kl. 10:37 skrev Daniel-Constantin Mierla mico...@gmail.com:

 Hello,
 
 I would like to remind that, at least in my case, writing emails directly 
 without cc-ing the mailing list is not recommended. The rule is also 
 suggested in the mailing lists presentation page:
 
  - http://www.kamailio.org/w/mailing-lists/
 
 Unless I asked explicitly for a private email with some specific details, the 
 message will not be noticed and therefore not replied. I use this email 
 address for many public mailing lists and it is intended only for that usage, 
 I am reacting based on the filters I created, unmatched messages getting the 
 lowest priority.
 
 There are several common sense as well as technical reasons. Here are some 
 along with other suggestions to improve the likeliness of getting an answer:
 - we are not answering questions on mailing lists to help only one person, 
 but also other people that may have similar issues in the future -- they can 
 find the answers in the archive with a search engine
 - the amount of unexpected messages is very high -- it is practically a zero 
 chance to get to the folder with messages that didn't match any filter (thus 
 your message is not going to be replied)
 - whenever I find time, the first emails I answer are those coming on mailing 
 lists. There are many there as well, people are traveling or having other 
 personal or business projects, so sometime is good to send a reminder if a 
 question does not get an answer
 - starting with negative approach or no technical content, like you are going 
 to use something else or this application is missing what so ever tutorial 
 giving the solution you need or simply asking for full config of complex 
 requirements, is not helping, making people ignore your messages. Start with 
 what you tried and where you got stuck. Everything you get is for free, 
 nobody here owe you anything.
 
 Therefore, if anyone sent me a private message asking about kamailio or 
 continuing a discussion started on mailing lists, then the messages have to 
 be resent on mailing list if one wants an answer. A quick look in my 
 unfiltered folder revealed hundreds of messages during the past year, soon 
 many will be deleted anyhow. Note that you can always mask sensitive data 
 (e.g., replace passwords, IP addresses) before sending logs or config file.

Just to chime in: These rules apply to many of us active in the Open Source 
projects. We answer general questions when we have time and help when we 
have time - but on the mailing list. It's better for you, since many can answer 
and reuse the discussion later when googling for answers.
It's also better for the developer who gets to decide when he/she has time  
available to assist you. If you have urgent needs for help, then 
you need to pay for personal assistance.

Cheers,
/O
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[SR-Users] rfc: accounting records time details

2013-04-29 Thread Daniel-Constantin Mierla

Hello,

there were old and recent discussions about the representation of time 
in accounting records. At this moment the acc module stores the unix 
timestamp as datetime value.


In some countries is required to store more accuracy, beyond the 
seconds. Also, the datetime gives some troubles with converting back to 
unix timestamp, specifically with time zones and daylight saving times.


I want to open a discussion that is visible for all users and devs, 
being something affecting probably everyone, input from anyone that is 
interested being relevant to select the right approach.


Here are some suggestions presented so far.

1) store seconds.miliseconds as double - there is a patch (which 
probably needs some tunings itself) on tracker. The precision is shifted 
from seconds to milliseconds. Internally computed from microseconds from 
timeval structure. The other modules might need updates (iirc dbtext has 
only 2 decimals precisions for double, while miliseconds will require at 
least three -- could be the case for flatstore or other modules).


Advantage is the computing of duration directly by subtraction of two 
values.


2) store seconds and microseconds as two separate values. Should be no 
issues with existing modules. Even more accuracy than above, no issues 
with other modules, but will require to use two columns (thus four 
values to compute the duration)


Any other suggestions?

Personally, at this moment, I will go with 2), but that might not meet 
everyone's needs. So a solution can be to make it configurable, as a 
bitmask of what time representation to store.


Say, there will be a new parameter  timestamp_mode for acc module:
- bit one set - store seconds timestamp as for now (default)
- bit two set - store seconds and microseconds in separate integer columns
- bit three set - store seconds.miliseconds as double value in one column

I could add the parameter and tune acc for bit one and two. For bit 
three a deeper review is needed for other modules, probably the 
developer that submitted to the tracker can do it.


More suggestions? Pro or cons opinions?

Cheers,
Daniel

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Re: [SR-Users] BUG? - Dialog module timeout BYE is not ok.

2013-04-29 Thread Julia
Thank you,

 

The patch solves the problem with dlgtimeout BYE (also for chain of proxy
servers), but causes a new one.

 

Now, OPTIONs from dispatcher module sent with angle brackets, but 200 OK
(received with same format) are ignored by kamailio.

 

ERROR: dispatcher [dispatch.c:2373]: Setting the state failed
(sip:) .

 

BR,

 

Julia

  _  

From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel-Constantin Mierla
Sent: Thursday, April 25, 2013 4:24 PM
To: Uri Shacked
Cc: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] BUG? - Dialog module timeout BYE is not ok.

 

Here is the link to the patch:

http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modul
es/tm/t_msgbuilder.c;h=b5ff3094af2594e62a050ccc56768ad9237eb633;hp=aa0144c84
43760c9e81ee121ef907be812fa9da8;hb=4e4b1339bfd3a832f5feeb1d2a2380c7455ec82b;
hpb=744a8d317b894a1360e3441a9e69ac9190a1745b

You have to add the lines starting with '+' in t_msgbuilder.c -- there are
six.

Not sure it really works replacing the entire file, it is safe to apply the
patch.

Cheers,
Daniel

On 4/25/13 3:17 PM, Uri Shacked wrote:

I have no internet access from my kamailio servers.

where can i download it and install?

or just download the t_msgbuilder.c and reinstall 3.3.2 ?

 

On Thu, Apr 25, 2013 at 3:55 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:

Hello,

I committed a patch to add angle brackets around From/To URI for local
generated requests. Here is the link to commit:

-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=4e4b1339
bfd3a832f5feeb1d2a2380c7455ec82b

No time to test it, so give it a try. If you are using 4.0 (even 3.3) should
work by just cherry-picking. Let me know if works fine.

Depending on feedback will be part of 4.0.1 (out in few hours), or let to be
backported for 4.0.2.

Cheers,
Daniel 

 

On 4/25/13 2:23 PM, Uri Shacked wrote:

Hi,

As i wrote, when the BYE is normal not generated form dlg_timeout. srv2
that forwards it, does it OK.

The TO/From headers in my scenario are the same as the initial request in
both servers (there are 2 legs for each call (incoming and outgoing), on
each leg i use the same initial headers.Ii hope i was more clear now.

Thanks,

Uri

 

On Thu, Apr 25, 2013 at 2:50 PM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:

IMO the timeout-triggered BYE should have identical From/To headers as the
initial request - otherwise it may cause issues with clients that are
strict, or rely on identical headers (like in your setup).

regards
Klaus 



On 25.04.2013 13 tel:25.04.2013%2013 :30, Uri Shacked wrote:

Hi,

Following this issue, and the issue BYE dialog timeout bad syntax from
Julia.

I have 2 kamailio servers. srv1 creates a dlg_timeout BYE and sends it
to srv2 that forwards the BYE to the next sip server (some other server...).

I can definitely see that the BYE that is being forward is malformed.
The TO header info is not complete (the port section for example).

When a normal bye is received from srv1 (not dlg_timeout) all works fine.

One thing that i do in this scenario, on srv2 i use uac_replace() on the
TO header. So, it is OK that the final destination is different (i do
translations).

I attached the BYE from both servers.

Any ideas?

Thanks,

Uri

You probably meant the To and From headers, I'm guessing. Yeah, as long
as there is no display name component, URIs in headers like that don't
have to go in s. The way Kamailio sends it is grammatically valid, per
RFC 3261. The user agents are at fault for not understanding it
correctly. Alex Balashov abalashov at evaristesys.com

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users wrote:
 /BYE headers? Which headers? Uri Shacked ushacked at
gmail.com
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users wrote:
Hi, I think there is a bug with the BYE that is sent to
the caller and //callee //when dialog timeout happends. The
BYE headers are sent with no  or . //So, some sip singaling
points decline the BYE. BR, //Uri
//

___ //SIP Express
Router (SER) and Kamailio (OpenSER) - sr-users mailing //list
//sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
//http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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might //expect from a fully-fledged keyboard. Alex Balashov -
Principal //Evariste Systems LLC //235 E Ponce de Leon Ave //Suite
106 //Decatur, GA 30030 //United States //Tel: +1-678-954-0670
tel:%2B1-678-954-0670 
//Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
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Router (SER) and Kamailio (OpenSER) - sr-users mailing list //sr-users
at 

Re: [SR-Users] rfc: accounting records time details

2013-04-29 Thread Thilo Bangert

 
 More suggestions? Pro or cons opinions?

I'd just save one timestamp, ie TAI64 or as java does miliseconds since epoch, 
in a single, new field. saving the timestamp in two fields seems messy.

miliseconds since epoch is probably preferable, since it can be converted to 
human readable dates by the database server.

then have a flag that may be enabled, which additionally saves the time in the 
current format (with less precision) -- for backward compatability.

but a change like this i'd save for a 5.0 release.

kind regards
Thilo

 
 Cheers,
 Daniel

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Re: [SR-Users] rfc: accounting records time details

2013-04-29 Thread Alex Hermann
On Monday 29 April 2013 11:05:36 Daniel-Constantin Mierla wrote:
 Here are some suggestions presented so far.
 
 1) store seconds.miliseconds as double - there is a patch (which

Please do not use floating point respresentations for values that will be used 
in accounting. Floating point is imprecise. As the time related columns will 
most probably be used for billing, the values should be exact. In SQL this 
means using the DECIMAL or NUMERIC column type.


 2) store seconds and microseconds as two separate values. Should be no
 issues with existing modules. Even more accuracy than above, no issues
 with other modules, but will require to use two columns (thus four
 values to compute the duration)

Difficult to use in calculations.


 Any other suggestions?

3) Use native mili/microseconds support for DATETIME or TIMESTAMP in the 
database. At least MariaBD and PostgreSQL support this.

4) Store mili/microseconds since epoch in a BIGINT column.


 Say, there will be a new parameter  timestamp_mode for acc module:
 - bit one set - store seconds timestamp as for now (default)
 - bit two set - store seconds and microseconds in separate integer columns
 - bit three set - store seconds.miliseconds as double value in one column
 - bit three set - store seconds.miliseconds as DECIMAL value in one column
 - bit four set - add mili/microseconds to DATETIME (only valid when bit 1 is 
set too)
 - bit five set - store mili/microseconds since epoch as BIGINT value in one 
column


Alternatively, 2 settings can be used, one for storage format and one to 
choose the precision/resolution. This provides the most flexibility for the 
user.

timestamp_format:
  datetime (TIMESTAMP or DATETIME)
  epoch ((BIG)INT or DECIMAL, depending on resolution)
  split_epoch (2x INT)

timestamp_resolution: seconds, miliseconds, microseconds

Default would be the current situation: datetime + seconds


-- 
Greetings,

Alex Hermann


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Re: [SR-Users] rfc: accounting records time details

2013-04-29 Thread Andreas Granig

Hi,

On 04/29/2013 01:42 PM, Alex Hermann wrote:

On Monday 29 April 2013 11:05:36 Daniel-Constantin Mierla wrote:

1) store seconds.miliseconds as double - there is a patch (which


Please do not use floating point respresentations for values that will be used
in accounting. Floating point is imprecise. As the time related columns will
most probably be used for billing, the values should be exact. In SQL this
means using the DECIMAL or NUMERIC column type.


Just for clarification, we're using DECIMAL(13,3) with the patch on the 
tracker, not DOUBLE. What's missing in this patch is the table 
definition, so it needs to be updated in any case.


Andreas

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Re: [SR-Users] BUG? - Dialog module timeout BYE is not ok.

2013-04-29 Thread Daniel-Constantin Mierla

Hello,

there are few follow up patches that you need to get, here are the 
commits id:


57ed79b9d45c29d37c405c3fa582c1d1011a2315
f22dcd559c739dd99275cd2444cf481d458d2fab
ff890a4eee1888ed3e1e080a18bd72124ab99690
aefea5477dc7878d5e818628e04ddcb088fd2858

Cheers,
Daniel


On 4/29/13 11:31 AM, Julia wrote:


Thank you,

The patch solves the problem with dlgtimeout BYE (also for chain of 
proxy servers), but causes a new one.


Now, OPTIONs from dispatcher module sent with angle brackets, but 200 
OK (received with same format) are ignored by kamailio.


*ERROR: dispatcher [dispatch.c:2373]: Setting the state failed 
(sip:) .*


BR,

Julia



*From:*sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] *On Behalf Of 
*Daniel-Constantin Mierla

*Sent:* Thursday, April 25, 2013 4:24 PM
*To:* Uri Shacked
*Cc:* Kamailio (SER) - Users Mailing List
*Subject:* Re: [SR-Users] BUG? - Dialog module timeout BYE is not ok.

Here is the link to the patch:

http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modules/tm/t_msgbuilder.c;h=b5ff3094af2594e62a050ccc56768ad9237eb633;hp=aa0144c8443760c9e81ee121ef907be812fa9da8;hb=4e4b1339bfd3a832f5feeb1d2a2380c7455ec82b;hpb=744a8d317b894a1360e3441a9e69ac9190a1745b

You have to add the lines starting with '+' in t_msgbuilder.c -- there 
are six.


Not sure it really works replacing the entire file, it is safe to 
apply the patch.


Cheers,
Daniel

On 4/25/13 3:17 PM, Uri Shacked wrote:


I have no internet access from my kamailio servers.

where can i download it and install?

or just download the t_msgbuilder.c and reinstall 3.3.2 ?

On Thu, Apr 25, 2013 at 3:55 PM, Daniel-Constantin Mierla 
mico...@gmail.com mailto:mico...@gmail.com wrote:


Hello,

I committed a patch to add angle brackets around From/To URI for 
local generated requests. Here is the link to commit:


- 
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=4e4b1339bfd3a832f5feeb1d2a2380c7455ec82b


No time to test it, so give it a try. If you are using 4.0 (even 3.3) 
should work by just cherry-picking. Let me know if works fine.


Depending on feedback will be part of 4.0.1 (out in few hours), or 
let to be backported for 4.0.2.


Cheers,
Daniel

On 4/25/13 2:23 PM, Uri Shacked wrote:


Hi,

As i wrote, when the BYE is normal not generated form dlg_timeout. 
srv2 that forwards it, does it OK.


The TO/From headers in my scenario are the same as the initial 
request in both servers (there are 2 legs for each call (incoming 
and outgoing), on each leg i use the same initial headers.Ii hope i 
was more clear now.


Thanks,

Uri

On Thu, Apr 25, 2013 at 2:50 PM, Klaus Darilion 
klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at 
wrote:


IMO the timeout-triggered BYE should have identical From/To headers 
as the initial request - otherwise it may cause issues with clients 
that are strict, or rely on identical headers (like in your setup).


regards
Klaus



On 25.04.2013 13 tel:25.04.2013%2013:30, Uri Shacked wrote:

Hi,

Following this issue, and the issue BYE dialog timeout bad
syntax from
Julia.

I have 2 kamailio servers. srv1 creates a dlg_timeout BYE and
sends it
to srv2 that forwards the BYE to the next sip server (some other
server...).

I can definitely see that the BYE that is being forward is
malformed.
The TO header info is not complete (the port section for example).

When a normal bye is received from srv1 (not dlg_timeout) all
works fine.

One thing that i do in this scenario, on srv2 i use
uac_replace() on the
TO header. So, it is OK that the final destination is different
(i do
translations).

I attached the BYE from both servers.

Any ideas?

Thanks,

Uri

You probably meant the To and From headers, I'm guessing. Yeah,
as long
as there is no display name component, URIs in headers like that
don't
have to go in s. The way Kamailio sends it is grammatically
valid, per
RFC 3261. The user agents are at fault for not understanding it
correctly. Alex Balashov abalashov at evaristesys.com
http://evaristesys.com

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
wrote:
 /BYE headers? Which headers? Uri Shacked ushacked at
gmail.com http://gmail.com
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
wrote:
Hi, I think there is a bug with the BYE that is
sent to
the caller and //callee //when dialog timeout happends.
The
BYE headers are sent with no  or . //So, some sip singaling
points decline the BYE. BR, //Uri

//
___ //SIP
Express
Router (SER) and Kamailio (OpenSER) - sr-users mailing 

Re: [SR-Users] rfc: accounting records time details

2013-04-29 Thread Andreas Granig

Hi,

On 04/29/2013 11:47 AM, Thilo Bangert wrote:

I'd just save one timestamp, ie TAI64 or as java does miliseconds since epoch,
in a single, new field. saving the timestamp in two fields seems messy.


Agreed.


miliseconds since epoch is probably preferable, since it can be converted to
human readable dates by the database server.


What I like about the DECIMAL approach is that it's (at least in MySQL) 
usable with from_unixtime functions, in case you need quick access to 
human readable format. Up until 5.1 it only shows seconds precision in 
that case, not sure about high resolution precision in 5.6 where 
timestamp seems to support microseconds.


Also not sure about compatibility with other DB engines.

Andreas

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Re: [SR-Users] var vs avp...

2013-04-29 Thread Victor V. Kustov
В Fri, 19 Apr 2013 14:46:00 +0200
Olle E. Johansson o...@edvina.net пишет:

 We will certainly look into this. There are trainings, but no books
 and very few guides on the web. Please feel free to contribute,
 after all it is Open Source and we depend on community
 contributions in order to get documentation written.

OK, i think any function need examples. Not for call only, but for use
result.

radius_load_callee_avps(callee)

I see in doc:

radius_load_callee_avps($rU@$rd); # take callee from Request-URI

I call this function, but where are results? How can i check errors, where i 
must look results? In which avps?

-- 
 SY,
Victor
  JID: coy...@bks.tv
  JID: coy...@bryansktel.ru
  I use FREE operation system: 3.8.4-calculate GNU/Linux

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Re: [SR-Users] var vs avp...

2013-04-29 Thread Juha Heinanen
Victor V. Kustov writes:

 radius_load_callee_avps(callee)
 
 I see in doc:
 
 radius_load_callee_avps($rU@$rd);   # take callee from Request-URI
 
 I call this function, but where are results? How can i check errors,
 where i must look results? In which avps?

as the name of the function indicates, the result is in the avps.  if
the function call fails, result value will be -1 as usual and you get
error message to syslog.

-- juha

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Re: [SR-Users] var vs avp...

2013-04-29 Thread Olle E. Johansson

29 apr 2013 kl. 14:29 skrev Victor V. Kustov coy...@bks.tv:

 В Fri, 19 Apr 2013 14:46:00 +0200
 Olle E. Johansson o...@edvina.net пишет:
 
 We will certainly look into this. There are trainings, but no books
 and very few guides on the web. Please feel free to contribute,
 after all it is Open Source and we depend on community
 contributions in order to get documentation written.
 
 OK, i think any function need examples. Not for call only, but for use
 result.
 
 radius_load_callee_avps(callee)
 
 I see in doc:
 
 radius_load_callee_avps($rU@$rd);   # take callee from Request-URI
 
 I call this function, but where are results? How can i check errors, where i 
 must look results? In which avps?

It is explained in section 1. Overview.

Now, I realize you missed that so there propably needs to be a pointer to that 
section in the function descriptions.

/O
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Re: [SR-Users] var vs avp...

2013-04-29 Thread Victor V. Kustov
В Mon, 29 Apr 2013 15:34:11 +0300
Juha Heinanen j...@tutpro.com пишет:


 as the name of the function indicates, the result is in the avps.  if
 the function call fails, result value will be -1 as usual and you get
 error message to syslog.

in which avps? 

kk, radius answer:

Session-Timeout  = '4284860'
next-hop-ip  = 'SIP/1@cisco-out'
session-protocol = 'SIP'

how i can access to reply?

-- 
 SY,
Victor
  JID: coy...@bks.tv
  JID: coy...@bryansktel.ru
  I use FREE operation system: 3.8.4-calculate GNU/Linux

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Re: [SR-Users] var vs avp...

2013-04-29 Thread Olle E. Johansson

29 apr 2013 kl. 14:44 skrev Victor V. Kustov coy...@bks.tv:

 В Mon, 29 Apr 2013 15:34:11 +0300
 Juha Heinanen j...@tutpro.com пишет:
 
 
 as the name of the function indicates, the result is in the avps.  if
 the function call fails, result value will be -1 as usual and you get
 error message to syslog.
 
 in which avps? 
 
 kk, radius answer:
 
 Session-Timeout  = '4284860'
 next-hop-ip  = 'SIP/1@cisco-out'
 session-protocol = 'SIP'
 
 how i can access to reply?
Documentation says:

All functions of this module load AVPs from SIP-AVP reply items received from 
RADIUS upon a successful request. Value of the SIP-AVP reply item must be a 
string of form:

• value = SIP_AVP_NAME SIP_AVP_VALUE

• SIP_AVP_NAME = STRING_NAME | '#'ID_NUMBER

• SIP_AVP_VALUE = ':'STRING_VALUE | '#'NUMBER_VALUE


/O
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Re: [SR-Users] var vs avp...

2013-04-29 Thread Victor V. Kustov
В Mon, 29 Apr 2013 14:46:34 +0200
Olle E. Johansson o...@edvina.net пишет:

 All functions of this module load AVPs from SIP-AVP reply items
 received from RADIUS upon a successful request. Value of the SIP-AVP
 reply item must be a string of form:
 
   • value = SIP_AVP_NAME SIP_AVP_VALUE
 
   • SIP_AVP_NAME = STRING_NAME | '#'ID_NUMBER
 
   • SIP_AVP_VALUE = ':'STRING_VALUE | '#'NUMBER_VALUE
 
 

sorry its not clear for me. 

$avp('next-hop-ip') is ok in my case? Or i need _special_ radius answer?



-- 
 SY,
Victor
  JID: coy...@bks.tv
  JID: coy...@bryansktel.ru
  I use FREE operation system: 3.8.4-calculate GNU/Linux

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Re: [SR-Users] BUG? - Dialog module timeout BYE is not ok.

2013-04-29 Thread Julia
Now it works,

 

Thank you!

 

Julia.

  _  

From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] 
Sent: Monday, April 29, 2013 3:11 PM
To: Julia; 'Kamailio (SER) - Users Mailing List'
Cc: 'Uri Shacked'
Subject: Re: [SR-Users] BUG? - Dialog module timeout BYE is not ok.

 

Hello,

there are few follow up patches that you need to get, here are the commits
id:

57ed79b9d45c29d37c405c3fa582c1d1011a2315
f22dcd559c739dd99275cd2444cf481d458d2fab
ff890a4eee1888ed3e1e080a18bd72124ab99690
aefea5477dc7878d5e818628e04ddcb088fd2858

Cheers,
Daniel



On 4/29/13 11:31 AM, Julia wrote:

Thank you,

 

The patch solves the problem with dlgtimeout BYE (also for chain of proxy
servers), but causes a new one.

 

Now, OPTIONs from dispatcher module sent with angle brackets, but 200 OK
(received with same format) are ignored by kamailio.

 

ERROR: dispatcher [dispatch.c:2373]: Setting the state failed
(sip:) .

 

BR,

 

Julia


  _  


From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel-Constantin Mierla
Sent: Thursday, April 25, 2013 4:24 PM
To: Uri Shacked
Cc: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] BUG? - Dialog module timeout BYE is not ok.

 

Here is the link to the patch:

http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modul
es/tm/t_msgbuilder.c;h=b5ff3094af2594e62a050ccc56768ad9237eb633;hp=aa0144c84
43760c9e81ee121ef907be812fa9da8;hb=4e4b1339bfd3a832f5feeb1d2a2380c7455ec82b;
hpb=744a8d317b894a1360e3441a9e69ac9190a1745b

You have to add the lines starting with '+' in t_msgbuilder.c -- there are
six.

Not sure it really works replacing the entire file, it is safe to apply the
patch.

Cheers,
Daniel

On 4/25/13 3:17 PM, Uri Shacked wrote:

I have no internet access from my kamailio servers.

where can i download it and install?

or just download the t_msgbuilder.c and reinstall 3.3.2 ?

 

On Thu, Apr 25, 2013 at 3:55 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:

Hello,

I committed a patch to add angle brackets around From/To URI for local
generated requests. Here is the link to commit:

-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=4e4b1339
bfd3a832f5feeb1d2a2380c7455ec82b

No time to test it, so give it a try. If you are using 4.0 (even 3.3) should
work by just cherry-picking. Let me know if works fine.

Depending on feedback will be part of 4.0.1 (out in few hours), or let to be
backported for 4.0.2.

Cheers,
Daniel 

 

On 4/25/13 2:23 PM, Uri Shacked wrote:

Hi,

As i wrote, when the BYE is normal not generated form dlg_timeout. srv2
that forwards it, does it OK.

The TO/From headers in my scenario are the same as the initial request in
both servers (there are 2 legs for each call (incoming and outgoing), on
each leg i use the same initial headers.Ii hope i was more clear now.

Thanks,

Uri

 

On Thu, Apr 25, 2013 at 2:50 PM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:

IMO the timeout-triggered BYE should have identical From/To headers as the
initial request - otherwise it may cause issues with clients that are
strict, or rely on identical headers (like in your setup).

regards
Klaus 



On 25.04.2013 13 tel:25.04.2013%2013 :30, Uri Shacked wrote:

Hi,

Following this issue, and the issue BYE dialog timeout bad syntax from
Julia.

I have 2 kamailio servers. srv1 creates a dlg_timeout BYE and sends it
to srv2 that forwards the BYE to the next sip server (some other server...).

I can definitely see that the BYE that is being forward is malformed.
The TO header info is not complete (the port section for example).

When a normal bye is received from srv1 (not dlg_timeout) all works fine.

One thing that i do in this scenario, on srv2 i use uac_replace() on the
TO header. So, it is OK that the final destination is different (i do
translations).

I attached the BYE from both servers.

Any ideas?

Thanks,

Uri

You probably meant the To and From headers, I'm guessing. Yeah, as long
as there is no display name component, URIs in headers like that don't
have to go in s. The way Kamailio sends it is grammatically valid, per
RFC 3261. The user agents are at fault for not understanding it
correctly. Alex Balashov abalashov at evaristesys.com

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users wrote:
 /BYE headers? Which headers? Uri Shacked ushacked at
gmail.com
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users wrote:
Hi, I think there is a bug with the BYE that is sent to
the caller and //callee //when dialog timeout happends. The
BYE headers are sent with no  or . //So, some sip singaling
points decline the BYE. BR, //Uri
//

___ //SIP Express
Router (SER) and Kamailio (OpenSER) - sr-users mailing //list
//sr-users at lists.sip-router.org

[SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus Dispatcher module

2013-04-29 Thread Aldo Antignano
I have read and applied the excellent guide found on:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb

Now I have added to Kamailio the HA/Load Balancer support, with the 
dispatcher module.
This way I have 1 Kamailio and 2 Asterisk machines.

How can I change the routing logic of the sections route[REGFWD] | 
route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the 
guide above the asterisk binded ip address is cabled in the kamailio config 
code)

Thanks,
Anty






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[SR-Users] riud access from route block

2013-04-29 Thread Juha Heinanen
when i call t_relay() on a tcp contact, branch route is executed.  in
that branch route i call t_on_branch_failure.  now if the tcp connection
to the contact is broken, for example, due to uas crash, the branch
failure route is never executed and thus i cannot get access to
$T_reply_ruid.

there is $branch(name) pv that gives access to other branched, but not
the main branch.  is there any way to get access to ruid of the main
branch when t_relay() fails when called from route block?  for example,
if there is only one contact (the $ru one), which results in t_relay()
failure, is there a way to find out the ruid of this contact?

-- juha

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