Re: [SR-Users] Regd : SIP Configuration in Kamailio

[Henning Westerholt]

Hello,

please keep the list in CC. I can see Kamailio in your log:

Oct 14 11:21:59 nickar /usr/sbin/kamailio[24567]: INFO:  [main.c:842]: 
sig_usr(): signal 15 received
Oct 14 11:21:59 nickar systemd[1]: Stopping Kamailio (OpenSER) - the Open 
Source SIP Server...
Oct 14 11:21:59 nickar /usr/sbin/kamailio[24568]: INFO:  [main.c:842]: 
sig_usr(): signal 15 received

About tracing/debugging, one good option is sngrep.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: Santanu B 
Sent: Wednesday, October 14, 2020 1:41 PM
To: Henning Westerholt 
Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio

Hi,
My syslog file is enclosed for your reference.

Thanks,
Santanu

On Wed, Oct 14, 2020 at 1:07 PM Henning Westerholt 
mailto:h...@skalatan.de>> wrote:
Hello,

have a look to /var/log/messages or /var/log/syslog, depending on the 
distribution. Kamailio might be logged there.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Santanu B
Sent: Wednesday, October 14, 2020 7:09 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio

Dear David,
In /var/log folder I am unable to see any kamailio log. Only syslog file is 
available. Please help me to find the Kamailio log.
Also please guide me to find out the trace also as I am very new to SIP Server.

Thanks a lot.

Regards,
Santanu

On Tue, Oct 13, 2020 at 4:05 PM David Villasmil 
mailto:david.villasmil.w...@gmail.com>> wrote:
Hello,

You will need to provide more information. What happens? What errors do you see 
in kamailio’s log? A trace would also be nice.



On Tue, 13 Oct 2020 at 06:24, Santanu B 
mailto:santanu...@gmail.com>> wrote:
Hi,
We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint 
to join in the conference room in Jitsi. For that i am trying to configure 
Kamailio SIP Server in our environment. I have installed the Server and my 
endpoints are registered in my SIP server. Both endpoints are able to connect 
to each other. But I am unable to configure the SIP endpoints to join in the 
Meet Conference Room. Please guide me to configure.

Thanks and regards,
Santanu


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phone: +34669448337
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[SR-Users] To Header Manipulation

[Sam Ware]

Wondering if this is intended behavior.  We have a customer doing some
craziness with the TO header.  They are including their tech prefix in the
TO header as well as the RURI.  It doesn't cause any issue in our systems
but apparently it does with one of our vendors.  In an effect to prevent
the issue, we thought we would check if the tech prefix was added on the TO
header and remove it.

Initially, I tried this by changing the $tU variable.

$avp(tp_len) = $(avp(techprefix){s.len}) + 1; # Get the Length of the Tech
Prefix and add 1 for the # or *
$tU = $(tU{s.strip, $avp(tp_len)});


The result ended up with the duplication of the DNIS: sut 
Where 911999# was the tech prefix and 1812555 is the DNIS which was
still an issue for the vendor.

Through additional reading, I found the suggestion to use
the uac_replace_to function in the UAC module which I implemented as
follows:

$avp(new_to_hdr_uri) = $(tu{re.subst,/sip:(.*[\*#])(.*)/sip:\2/});
uac_replace_to("$avp(new_to_hdr_uri)");


This resulted in the original TO header username to be appended at the end
of the uri: sut 

Wondering if I am doing something wrong or this is the way these functions
are designed to protect the TO header contents for Dialog matching.


-- 
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Re: [SR-Users] Error in RTP engine setup

[David Villasmil]

I believe you need to add the table yourself like:

https://github.com/sipwise/rtpengine


# this only needs to be one once after system (re-) boot
modprobe xt_RTPENGINE
iptables -I INPUT -p udp -j RTPENGINE --id 0
ip6tables -I INPUT -p udp -j RTPENGINE --id 0

# ensure that the table we want to use doesn't exist - usually needed
after a daemon
# restart, otherwise will error
echo 'del 0' > /proc/rtpengine/control

# start daemon
/usr/sbin/rtpengine --table=0 --interface=10.64.73.31
--interface=2001:db8::4f3:3d \
--listen-ng=127.0.0.1:2223 --tos=184 --pidfile=/run/rtpengine.pid --no-fallback


Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Wed, Oct 14, 2020 at 5:06 PM Pavithra Mohanraja 
wrote:

> Hi,
> I tried to setup rtpengine to make my sip clients communicate through
> proxy following https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/
>
> I started sip communication since the service was running and saw that the
> RTP packets go to exposed rtp interface instead of other sip client. But i
> don't receive packets from the other end.
>
> When analyzed I noticed the rtp-engine daemon gives a few errors.
> *modprobe: FATAL: Module xt_RTPENGINE not found in directory
> /lib/modules/4.15.0-118-generic*
>
>
> *iptables: No chain/target/match by that name.ip6tables: No
> chain/target/match by that name. * Starting RTP/media proxy
> ngcp-rtpengine-daemon
> [1602690253.130392] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such
> file or directory), KERNEL FORWARDING DISABLED*
> *(Note: *uname -r gives 4.15.0-118-generic and  /lib/modules/ has only
> 4.15.0-115-generic*)*
>
> Please help me with this issue. Thanks in Advance.
>
> Regards,
> Pavithra
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[SR-Users] Error in RTP engine setup

[Pavithra Mohanraja]

Hi,
I tried to setup rtpengine to make my sip clients communicate through proxy
following https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/

I started sip communication since the service was running and saw that the
RTP packets go to exposed rtp interface instead of other sip client. But i
don't receive packets from the other end.

When analyzed I noticed the rtp-engine daemon gives a few errors.
*modprobe: FATAL: Module xt_RTPENGINE not found in directory
/lib/modules/4.15.0-118-generic*


*iptables: No chain/target/match by that name.ip6tables: No
chain/target/match by that name. * Starting RTP/media proxy
ngcp-rtpengine-daemon
[1602690253.130392] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such
file or directory), KERNEL FORWARDING DISABLED*
*(Note: *uname -r gives 4.15.0-118-generic and  /lib/modules/ has only
4.15.0-115-generic*)*

Please help me with this issue. Thanks in Advance.

Regards,
Pavithra
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Re: [SR-Users] how to get a value from SIP INVITE Header

[sai sudheer]

hii,,
I tried {uri.user} from the URI transformations. but i just get 0 or Null in 
the log.

Now i tried the string transformation as per your reply, and 
{s.select,index,separator} is thing i was expecting .. 
It worked now, and i am getting +34xxx alone, which is what i need. 

Thanks a lot. 





On Wednesday, October 14, 2020, 08:36:37 AM GMT+1, Henning Westerholt 
 wrote: 








Hello,

 

can you share the error you have got? If you just want to get the substring of 
the +34x@domain URI, have a look to the string transformations on the same 
link.

 

Cheers,

 

Henning

 

-- 

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com 

 


From: sai sudheer  
Sent: Tuesday, October 13, 2020 5:04 PM
To: Henning Westerholt ; Henning Westerholt 
; Kamailio (SER) - Users Mailing List 

Subject: RE: [SR-Users] how to get a value from SIP INVITE Header


 

Hi.


Thanks for your response. I went through the uri transformation page, tried it 
but no luck.



 



Is there any other method to slice the string from +34xx...@sub.domain.com 
to +34??



 



Regards


Sent from Yahoo Mail on Android


 

>  
>  
>  
> On Tue, Oct 13, 2020 at 17:30, Henning Westerholt
> 
> 
>  
>  wrote:
> 
> 
> 
>  
>  
>  
>  
> Hello,
> 
>  
> 
> have a look to the transformations for URIs:  
> https://www.kamailio.org/wiki/cookbooks/devel/transformations#uri_transformations
> 
> There is one to get parameters as well.
> 
>  
> 
> $shm variables are shared in shared memory, so you can share them in other 
> calls. The $avp/$xavp are shared between processes, but belongs to the SIP 
> transaction.
> 
>  
> 
> About routing calls by prefix – have a look to the lcr, carrierroute or 
> drouting modules, for example.
> 
>  
> 
> Cheers,
> 
>  
> 
> Henning
> 
>  
> 
> -- 
> 
> Henning Westerholt – https://skalatan.de/blog/
> 
> Kamailio services – https://gilawa.com 
> 
>  
> 
>  
>  
> From: sr-users  On Behalf Of sai sudheer
> Sent: Tuesday, October 13, 2020 12:44 PM
> To: sr-users@lists.kamailio.org; sr-...@lists.kamailio.org
> Subject: [SR-Users] how to get a value from SIP INVITE Header
> 
> 
>  
> 
>>  
>>  
>>  
>> Hii
>> 
>> 
>>  
>>  
>> 
>> 
>>  
>> Please help me to get the a value from SIP INVITE header reached to kamailio 
>> like INVITE sip:+341930203...@sub.domain.com;myid=+3...@sub.domain.com 
>> SIP/2.0 and i want to save the myid value +34 into a variable, without the 
>> domain name.
>> 
>> 
>>  
>>  
>> 
>> 
>>  
>>  
>> 
>> 
>>  
>>      $var(uri) = $sel(ruri);
>> 
>> 
>>  
>>      xavp_params_explode("$(var(uri){s.unbracket})", "uri");
>> 
>> 
>>  
>>      xlog("L_INFO", "$var(uri) Received converted to $xavp(uri=>myid[0])\n");
>> 
>> 
>>  
>>  
>> 
>> 
>>  
>> I tried above and it prints +3...@sub.domain.com But i want to just save +34 
>> into a variable to further check the prefix based routing from the database.
>> 
>> 
>>  
>>  
>> 
>> 
>>  
>> 1) Could you please help how to get it or If there is any alternate/single 
>> line approach to get this value?
>> 
>> 
>>  
>> 2) $var & $xavp are process-local variables, and they cant be shared with 
>> other calls? Right
>> 
>> 
>>  
>> 3) What is the best module to route calls based their prefix ? pstn_route or 
>> dynamic routing or any suggestions?
>> 
>> 
>>  
>>  
>> 
>> 
>>  
>> Please help, thanks in advance.
>> 
>> 
>> 
> 
> 
> 
> 
> 
> 




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Re: [SR-Users] SRTP

[Olle E. Johansson]

> On 14 Oct 2020, at 16:48, Andy Kama  wrote:
> 
> Hi Fred
> 
> So i am running rtpengine but prob i am having is i receive the call as rtp 
> and need it to become srtp before i send it to the carrier
> then the carrier sends me a call as srtp and need that to become normal rtp 
> before i can go to the uac

THat is exactly what rtpengine does for you. Go explore their documentation!

/O :-)
> 
> On Wed, Oct 14, 2020 at 3:27 PM Fred Posner  > wrote:
> On Wed, 2020-10-14 at 11:51 +0100, Andy Kama wrote:
> > Hi All,
> > 
> > I have setup a tls connection to a sip provider and it works
> > perfectly
> > But they are requesting i send srtp
> > i can find any documentation how i do it with rtpengine etc?
> > 
> > any guidance would be appreciated
> > 
> > 
> 
> Generally this would be initiated by the endpoint making/receiving the
> call and can be as simple as making sure it's offered in the phone
> settings.
> 
> If you need to bridge rtp with srtp, you can look at rtpengine (
> https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html 
> )
> for information on flagging the offer/manage.
> 
> Of course, this would also require you to install and run rtpengine.
> 
> -- 
> Fred Posner
> f...@palner.com 
> https://www.palner.com 
> 
> Need Fred? Call Fred. 336-HEY-FRED
> Matrix: @fred:matrix.lod.com 
> 
> 
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Re: [SR-Users] SRTP

[Andy Kama]

Hi Fred

So i am running rtpengine but prob i am having is i receive the call as rtp
and need it to become srtp before i send it to the carrier
then the carrier sends me a call as srtp and need that to become normal rtp
before i can go to the uac

On Wed, Oct 14, 2020 at 3:27 PM Fred Posner  wrote:

> On Wed, 2020-10-14 at 11:51 +0100, Andy Kama wrote:
> > Hi All,
> >
> > I have setup a tls connection to a sip provider and it works
> > perfectly
> > But they are requesting i send srtp
> > i can find any documentation how i do it with rtpengine etc?
> >
> > any guidance would be appreciated
> >
> >
>
> Generally this would be initiated by the endpoint making/receiving the
> call and can be as simple as making sure it's offered in the phone
> settings.
>
> If you need to bridge rtp with srtp, you can look at rtpengine (
> https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html)
> for information on flagging the offer/manage.
>
> Of course, this would also require you to install and run rtpengine.
>
> --
> Fred Posner
> f...@palner.com
> https://www.palner.com
>
> Need Fred? Call Fred. 336-HEY-FRED
> Matrix: @fred:matrix.lod.com
>
>
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Re: [SR-Users] SRTP

[Fred Posner]

On Wed, 2020-10-14 at 11:51 +0100, Andy Kama wrote:
> Hi All,
> 
> I have setup a tls connection to a sip provider and it works
> perfectly
> But they are requesting i send srtp
> i can find any documentation how i do it with rtpengine etc?
> 
> any guidance would be appreciated
> 
> 

Generally this would be initiated by the endpoint making/receiving the
call and can be as simple as making sure it's offered in the phone
settings.

If you need to bridge rtp with srtp, you can look at rtpengine (
https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html)
for information on flagging the offer/manage.

Of course, this would also require you to install and run rtpengine.

-- 
Fred Posner
f...@palner.com
https://www.palner.com

Need Fred? Call Fred. 336-HEY-FRED
Matrix: @fred:matrix.lod.com


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Re: [SR-Users] [sr-dev] pstn_route add_route(): route name not defined error

[Daniel-Constantin Mierla]

Hello,

do not send emails to both sr-dev and sr-users mailing lists when you
have questions about using Kamailio, sr-dev is only for discussing about
code development for project or what is only in development branch (not
part of a stable release). I removed sr-dev from recipients.

I think you haven't understood what prefix_route module does: it execute
route blocks from configuration file, it does not set the destination
address based on a prefix. For that look at pdt module for a simple
solution. Other modules such as mtree or lcr could be alternatives, but
from simplicity point of view, pdt is better fr this case imo.

Cheers,
Daniel


On 14.10.20 14:27, sai Sudheer wrote:
> Hii,
> I am trying to use pstn_route module for the first time, and i want to
> route the invite to specific pstn server based on its prefix.
> I have loaded the module & setup the DB, and it's giving error saying
> route name (from DB) is not defined after adding the a prefix into DB
> table., and there were no errors when DB table is empty.
> Not sure how to defined it, or i dont know what did i miss. Please help.
>
>  CONFIG ##
> loadmodule "prefix_route.so"
> modparam("prefix_route", "db_url",
> "mysql://x:x@localhost/kamailioDB")
> modparam("prefix_route", "db_table", "new_prefix_route")
>
> #PSTN Routing table contains
>          if (!prefix_route("+44")) {
>               xlog("L_INFO", "+44 prefix didnt match with prefix_route
> DB\n");
>          }
>
> ## DB
> MariaDB [kamailio]> select * from new_prefix_route;
> Empty set (0.00 sec)
>
> MariaDB [kamailio]> exit;
> ##
>
>
> Kamailio looks good untill here, no errors.. 
>
> # Now i have added a prefix into sql DB
> INSERT INTO new_prefix_route VALUES ("+44", "10.10.1.1:5060", "route
> for this prefix");
> ###
> Error log after adding a prefix into new_prefix_route table & restart
> of kamailio
>
>  0(5802) CRITICAL: prefix_route [prefix_route.c:72]: add_route():
> route name '10.10.1.1:5060' is not defined
>  0(5802) NOTICE: prefix_route [prefix_route.c:170]: pr_db_load():
> Total prefix routes loaded: 1
>  0(5802) ERROR: prefix_route [prefix_route.c:174]: pr_db_load(): error
> flushing tree
>  0(5802) CRITICAL: prefix_route [prefix_route.c:212]: mod_init(): db
> load failed
>
> I have tried a "name" instead of IP (though i dont know how to convert
> that "name" into a IP:Port), But i am still getting same error using
> "name" instead of IP:port @ route field in DB.
>
>
> Please help.
>
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Re: [SR-Users] how to get a value from SIP INVITE Header

[Santanu B]

Please help to configure  uri transformation page.

Thanks
Santanu

On Wed, Oct 14, 2020 at 1:08 PM Henning Westerholt  wrote:

> Hello,
>
>
>
> can you share the error you have got? If you just want to get the
> substring of the +34x@domain URI, have a look to the string
> transformations on the same link.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* sai sudheer 
> *Sent:* Tuesday, October 13, 2020 5:04 PM
> *To:* Henning Westerholt ; Henning Westerholt <
> h...@skalatan.de>; Kamailio (SER) - Users Mailing List <
> sr-users@lists.kamailio.org>
> *Subject:* RE: [SR-Users] how to get a value from SIP INVITE Header
>
>
>
> Hi.
>
> Thanks for your response. I went through the uri transformation page,
> tried it but no luck.
>
>
>
> Is there any other method to slice the string from
> +34xx...@sub.domain.com to +34??
>
>
>
> Regards
>
> Sent from Yahoo Mail on Android
> 
>
>
>
> On Tue, Oct 13, 2020 at 17:30, Henning Westerholt
>
>  wrote:
>
> Hello,
>
>
>
> have a look to the transformations for URIs:
> https://www.kamailio.org/wiki/cookbooks/devel/transformations#uri_transformations
>
> There is one to get parameters as well.
>
>
>
> $shm variables are shared in shared memory, so you can share them in other
> calls. The $avp/$xavp are shared between processes, but belongs to the SIP
> transaction.
>
>
>
> About routing calls by prefix – have a look to the lcr, carrierroute or
> drouting modules, for example.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* sr-users  *On Behalf Of *sai
> sudheer
> *Sent:* Tuesday, October 13, 2020 12:44 PM
> *To:* sr-users@lists.kamailio.org; sr-...@lists.kamailio.org
> *Subject:* [SR-Users] how to get a value from SIP INVITE Header
>
>
>
> Hii
>
>
>
> Please help me to get the a value from SIP INVITE header reached to
> kamailio like INVITE sip:+341930203...@sub.domain.com;myid=+
> 3...@sub.domain.com SIP/2.0 and i want to save the myid value +34 into a
> variable, without the domain name.
>
>
>
>
>
>  $var(uri) = $sel(ruri);
>
>  xavp_params_explode("$(var(uri){s.unbracket})", "uri");
>
>  xlog("L_INFO", "$var(uri) Received converted to
> $xavp(uri=>myid[0])\n");
>
>
>
> I tried above and it prints +3...@sub.domain.com But i want to just save
> +34 into a variable to further check the prefix based routing from the
> database.
>
>
>
> 1) Could you please help how to get it or If there is any alternate/single
> line approach to get this value?
>
> 2) $var & $xavp are process-local variables, and they cant be shared with
> other calls? Right
>
> 3) What is the best module to route calls based their prefix ? pstn_route
> or dynamic routing or any suggestions?
>
>
>
> Please help, thanks in advance.
>
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[SR-Users] SRTP

[Andy Kama]

Hi All,

I have setup a tls connection to a sip provider and it works perfectly
But they are requesting i send srtp
i can find any documentation how i do it with rtpengine etc?

any guidance would be appreciated
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Re: [SR-Users] '302 Moved temporarily' contact parsing error

[David VILLAUME]

Hi,

Sorry for the noise and thanks for your help, i forgot to exclude the 302 from 
the relaying in my reply_route.

Regards,
David

From: Daniel-Constantin Mierla 
Sent: Wednesday, October 14, 2020 9:54 AM
To: David VILLAUME ; Kamailio (SER) - Users Mailing 
List 
Subject: Re: [SR-Users] '302 Moved temporarily' contact parsing error


Hello,

but do you have reply_route or onreply_route[x] where you access the contact 
with a transformation? Can you give here those lines from your config accessing 
the contact header?

parse_to_param() is not used by core Contact header parser, thus that can 
happen only on some config operations...

Cheers,
Daniel
On 14.10.20 09:41, David VILLAUME wrote:
Hi Daniel,

I already use the select for getting the contact, but I still got the error as 
it happens before the fist line of my failure_route :

Oct 14 09:32:53 fr-kam-misc-lab-vma-1 kamailio[21633]: ERROR: { 2  udp 
172.18.0.15 17 REGISTER # 
8_466109112@10.244.3.114 * }  
[core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid character ',' in 
status 27: [;q=0.5]
Oct 14 09:32:53 fr-kam-misc-lab-vma-1 kamailio[21633]: INFO: { 1  tcp 
65.39.65.250 17 REGISTER # 
8_466109112@10.244.3.114 * } 

Re: [SR-Users] '302 Moved temporarily' contact parsing error

[Daniel-Constantin Mierla]

Hello,

but do you have reply_route or onreply_route[x] where you access the
contact with a transformation? Can you give here those lines from your
config accessing the contact header?

parse_to_param() is not used by core Contact header parser, thus that
can happen only on some config operations...

Cheers,
Daniel

On 14.10.20 09:41, David VILLAUME wrote:
>
> Hi Daniel,
>
>  
>
> I already use the select for getting the contact, but I still got the
> error as it happens before the fist line of my failure_route :
>
>  
>
> Oct 14 09:32:53 fr-kam-misc-lab-vma-1 kamailio[21633]: ERROR: { 2  udp
> 172.18.0.15 17 REGISTER # 8_466109112@10.244.3.114
>  * } 
> [core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid
> character ',' in status 27: [;q=0.5]
>
> Oct 14 09:32:53 fr-kam-misc-lab-vma-1 kamailio[21633]: INFO: { 1  tcp
> 65.39.65.250 17 REGISTER # 8_466109112@10.244.3.114
>  * } 

Re: [SR-Users] '302 Moved temporarily' contact parsing error

[David VILLAUME]

Hi Daniel,

I already use the select for getting the contact, but I still got the error as 
it happens before the fist line of my failure_route :

Oct 14 09:32:53 fr-kam-misc-lab-vma-1 kamailio[21633]: ERROR: { 2  udp 
172.18.0.15 17 REGISTER # 
8_466109112@10.244.3.114 * }  
[core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid character ',' in 
status 27: [;q=0.5]
Oct 14 09:32:53 fr-kam-misc-lab-vma-1 kamailio[21633]: INFO: { 1  tcp 
65.39.65.250 17 REGISTER # 
8_466109112@10.244.3.114 * } 

Re: [SR-Users] Regd : SIP Configuration in Kamailio

[Henning Westerholt]

Hello,

have a look to /var/log/messages or /var/log/syslog, depending on the 
distribution. Kamailio might be logged there.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sr-users  On Behalf Of Santanu B
Sent: Wednesday, October 14, 2020 7:09 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio

Dear David,
In /var/log folder I am unable to see any kamailio log. Only syslog file is 
available. Please help me to find the Kamailio log.
Also please guide me to find out the trace also as I am very new to SIP Server.

Thanks a lot.

Regards,
Santanu

On Tue, Oct 13, 2020 at 4:05 PM David Villasmil 
mailto:david.villasmil.w...@gmail.com>> wrote:
Hello,

You will need to provide more information. What happens? What errors do you see 
in kamailio’s log? A trace would also be nice.



On Tue, 13 Oct 2020 at 06:24, Santanu B 
mailto:santanu...@gmail.com>> wrote:
Hi,
We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint 
to join in the conference room in Jitsi. For that i am trying to configure 
Kamailio SIP Server in our environment. I have installed the Server and my 
endpoints are registered in my SIP server. Both endpoints are able to connect 
to each other. But I am unable to configure the SIP endpoints to join in the 
Meet Conference Room. Please guide me to configure.

Thanks and regards,
Santanu


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Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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Re: [SR-Users] how to get a value from SIP INVITE Header

[Henning Westerholt]

Hello,

can you share the error you have got? If you just want to get the substring of 
the +34x@domain URI, have a look to the string transformations on the same 
link.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sai sudheer 
Sent: Tuesday, October 13, 2020 5:04 PM
To: Henning Westerholt ; Henning Westerholt 
; Kamailio (SER) - Users Mailing List 

Subject: RE: [SR-Users] how to get a value from SIP INVITE Header

Hi.
Thanks for your response. I went through the uri transformation page, tried it 
but no luck.

Is there any other method to slice the string from 
+34xx...@sub.domain.com to +34??

Regards
Sent from Yahoo Mail on 
Android

On Tue, Oct 13, 2020 at 17:30, Henning Westerholt
mailto:h...@skalatan.de>> wrote:

Hello,



have a look to the transformations for URIs: 
https://www.kamailio.org/wiki/cookbooks/devel/transformations#uri_transformations

There is one to get parameters as well.



$shm variables are shared in shared memory, so you can share them in other 
calls. The $avp/$xavp are shared between processes, but belongs to the SIP 
transaction.



About routing calls by prefix – have a look to the lcr, carrierroute or 
drouting modules, for example.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of sai sudheer
Sent: Tuesday, October 13, 2020 12:44 PM
To: sr-users@lists.kamailio.org; 
sr-...@lists.kamailio.org
Subject: [SR-Users] how to get a value from SIP INVITE Header



Hii



Please help me to get the a value from SIP INVITE header reached to kamailio 
like INVITE 
sip:+341930203...@sub.domain.com;myid=+3...@sub.domain.com
 SIP/2.0 and i want to save the myid value +34 into a variable, without the 
domain name.





 $var(uri) = $sel(ruri);

 xavp_params_explode("$(var(uri){s.unbracket})", "uri");

 xlog("L_INFO", "$var(uri) Received converted to $xavp(uri=>myid[0])\n");



I tried above and it prints +3...@sub.domain.com 
But i want to just save +34 into a variable to further check the prefix based 
routing from the database.



1) Could you please help how to get it or If there is any alternate/single line 
approach to get this value?

2) $var & $xavp are process-local variables, and they cant be shared with other 
calls? Right

3) What is the best module to route calls based their prefix ? pstn_route or 
dynamic routing or any suggestions?



Please help, thanks in advance.
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Re: [SR-Users] '302 Moved temporarily' contact parsing error

[Daniel-Constantin Mierla]

Hello,

the Contact header body looks ok, using comma to separate many contact
values is ok.

My guess is that {to.body} transformation is used in this case, it is
not coming from core or a module. The To header cannot appear many
times, thus the transformation is throwing error in this case. If I am
right with my guess, a solution is to plug {s.select,0,,} transformation
before use of {to.body}. The index of s.select can be a variable, in
case one needs to get all contact values, a while loop can be used.

Cheers,
Daniel

On 13.10.20 22:07, Sergiu Pojoga wrote:
> Hmm, it's a tricky one, not sure actually.
>
> https://tools.ietf.org/html/rfc3261#page-167
> 
>
> The answer might be in:
> Even if the "display-name" is empty, the "name-addr" form MUST be used if the 
> "addr-spec" contains a comma, semicolon, or question mark.
>
> On Tue, Oct 13, 2020 at 3:39 PM Sergiu Pojoga  > wrote:
>
> Aren't the Contact params supposed to be delimited by semicolon?
>
> On Tue, Oct 13, 2020 at 1:06 PM David VILLAUME
> mailto:david.villa...@sewan.fr>> wrote:
>
> Hello,
>
>  
>
> I see parsing errors on 302 contact header that seems
> compliant to RFC :
>
>  
>
> Contact:;q=0.5,;q=0.25
>
>  
>
>  [core/parser/parse_addr_spec.c:479]: parse_to_param():
> invalid character ',' in status 27: [;q=0.5]
>
>  
>
> Is there an error in my contact or is it truly a parsing error ?
>
>    
>
> Regards,
>
> David
>
>  
>
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www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

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