[SR-Users] Internet RTP filters

2011-06-30 Thread Dominguez Jover, Ricardo
Hi everybody,

First to say sorry if this is not the right list to send this post, anyway we 
hope somebody can help us.

We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT 
issues in the server side and STUN for NAT issues in the internet client side. 
Calls in Spain are working fine.

This month one of our users has gone to Rwanda (our University has created a 
Hospital and a School in a region called Nemba). They are using an internet 
connection by satellite, with a low bandwith (128kb down, 16 up).

In some countries we have trouble with internet calls. SIP signaling works 
fine, the call rings and is stablished, but there is a problem with RTP streams:

- RTP streams sent by RTP proxy to client in internet never arrives to the 
client in the internet (although they RTP is sent to the right IP and port)
- RTP streams sent by client in the internet never arrives to RTP proxy 
(although RTP is sent to the right IP and port)

As I said this happens sometimes, and it's happening in Rwanda. We usually 
solve this problem making a VPN connection to the University, then the RTP 
stream works fine.

However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps 
down and 8kbps up), so they can hear as perfectly, but we can't hear them.

Does anyone know if T companies are filtering RTP traffic? I've tried to change 
ports in RPT Proxy with no result.

How could we solve this?

Thanks in advance and best regards,

Ricardo Dominguez
Universidad Miguel Hernández de Elche




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Re: [SR-Users] Internet RTP filters

2011-06-30 Thread Dominguez Jover, Ricardo
Hi Andreas, I've tried with TLS (using TCP port 5061) with no RTP encryption 
and with SRTP, but I don't know what minimum encryption means.

Regards,
Ricardo


-Mensaje original-
De: sr-users-boun...@lists.sip-router.org en nombre de andreas kaschner
Enviado el: jue 30/06/2011 22:04
Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
UsersMailingList
Asunto: Re: [SR-Users] Internet RTP filters
 
hi, have you tried TLS with Minimum encryption?
regards Andreas



On 30. juni 2011, at 19:43, Dominguez Jover, Ricardo djo...@umh.es wrote:

 Hi everybody,
 
 First to say sorry if this is not the right list to send this post, anyway we 
 hope somebody can help us.
 
 We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for 
 NAT issues in the server side and STUN for NAT issues in the internet client 
 side. Calls in Spain are working fine.
 
 This month one of our users has gone to Rwanda (our University has created a 
 Hospital and a School in a region called Nemba). They are using an internet 
 connection by satellite, with a low bandwith (128kb down, 16 up).
 
 In some countries we have trouble with internet calls. SIP signaling works 
 fine, the call rings and is stablished, but there is a problem with RTP 
 streams:
 
 - RTP streams sent by RTP proxy to client in internet never arrives to the 
 client in the internet (although they RTP is sent to the right IP and port)
 - RTP streams sent by client in the internet never arrives to RTP proxy 
 (although RTP is sent to the right IP and port)
 
 As I said this happens sometimes, and it's happening in Rwanda. We usually 
 solve this problem making a VPN connection to the University, then the RTP 
 stream works fine.
 
 However when our Rwanda's users connect to VPN, the Bandwith goes down (45 
 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear 
 them.
 
 Does anyone know if T companies are filtering RTP traffic? I've tried to 
 change ports in RPT Proxy with no result.
 
 How could we solve this?
 
 Thanks in advance and best regards,
 
 Ricardo Dominguez
 Universidad Miguel Hernández de Elche
 
 
 
 
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Re: [SR-Users] Kamailio 3.1.3 uncontrolled core dump

2011-05-17 Thread Dominguez Jover, Ricardo
Hi everybody,

In the config file I had set debug level to 9 (debug=9). If I change it to its 
default debug=2, then Kamailio doesn´t crash.

For example, if I call a non existent domain it crashes.

Is there any reason for this behaviour?

Regards,
Ricardo

-Mensaje original-
De: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Klaus Darilion
Enviado el: viernes, 13 de mayo de 2011 14:14
Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - 
UsersMailing List
Asunto: Re: [SR-Users] Kamailio 3.1.3 uncontrolled core dump

1. seems like your config is broken and request uri is set to
  sip:@192.168.64.36

2. Anyways, Kamailio shouldn't crash. I tried to reproduce the failure,
but my Kamailio does not crash.

regards
klaus

Am 13.05.2011 10:19, schrieb Dominguez Jover, Ricardo:
 06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]:
 ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable
 destination (478/SL) 
 May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]:
 ERROR: core [resolve.c:1540]: ERROR: sip_hostport2su: could not
 resolve hostname: @192.168.64.36 

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[SR-Users] Kamailio 3.1.3 uncontrolled core dump

2011-05-13 Thread Dominguez Jover, Ricardo
Hi everybody.

As posted in
http://lists.sip-router.org/pipermail/sr-users/2011-May/068514.html we
are having an uncontrolled core dump. It seems that the crash occurs
when some users registers to Kamailio. After upgrading to Kamilio
release 3.1.3 we are able to generate the following GDB:

GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-32.el5_6.2)
Copyright (C) 2009 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later
http://gnu.org/licenses/gpl.html
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type show
copying
and show warranty for details.
This GDB was configured as i386-redhat-linux-gnu.
For bug reporting instructions, please see:
http://www.gnu.org/software/gdb/bugs/...
Reading symbols from /usr/local/kamailio-3.1.3/sbin/kamailio...done.
Reading symbols from /lib/libdl.so.2...(no debugging symbols
found)...done.
Loaded symbols for /lib/libdl.so.2
Reading symbols from /lib/libresolv.so.2...(no debugging symbols
found)...done.
Loaded symbols for /lib/libresolv.so.2
Reading symbols from /lib/libc.so.6...(no debugging symbols
found)...done.
Loaded symbols for /lib/libc.so.6
Reading symbols from /lib/ld-linux.so.2...(no debugging symbols
found)...done.
Loaded symbols for /lib/ld-linux.so.2
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so...done.
Loaded symbols for
/usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so
Reading symbols from /usr/lib/mysql/libmysqlclient.so.15...(no debugging
symbols found)...done.
Loaded symbols for /usr/lib/mysql/libmysqlclient.so.15
Reading symbols from /usr/lib/libz.so.1...(no debugging symbols
found)...done.
Loaded symbols for /usr/lib/libz.so.1
Reading symbols from /lib/libcrypt.so.1...(no debugging symbols
found)...done.
Loaded symbols for /lib/libcrypt.so.1
Reading symbols from /lib/libnsl.so.1...(no debugging symbols
found)...done.
Loaded symbols for /lib/libnsl.so.1
Reading symbols from /lib/libm.so.6...(no debugging symbols
found)...done.
Loaded symbols for /lib/libm.so.6
Reading symbols from /lib/libssl.so.6...(no debugging symbols
found)...done.
Loaded symbols for /lib/libssl.so.6
Reading symbols from /lib/libcrypto.so.6...(no debugging symbols
found)...done.
Loaded symbols for /lib/libcrypto.so.6
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1...done.
Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1...done.
Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1
Reading symbols from /usr/lib/libgssapi_krb5.so.2...(no debugging
symbols found)...done.
Loaded symbols for /usr/lib/libgssapi_krb5.so.2
Reading symbols from /usr/lib/libkrb5.so.3...(no debugging symbols
found)...done.
Loaded symbols for /usr/lib/libkrb5.so.3
Reading symbols from /lib/libcom_err.so.2...(no debugging symbols
found)...done.
Loaded symbols for /lib/libcom_err.so.2
Reading symbols from /usr/lib/libk5crypto.so.3...(no debugging symbols
found)...done.
Loaded symbols for /usr/lib/libk5crypto.so.3
Reading symbols from /usr/lib/libkrb5support.so.0...(no debugging
symbols found)...done.
Loaded symbols for /usr/lib/libkrb5support.so.0
Reading symbols from /lib/libkeyutils.so.1...(no debugging symbols
found)...done.
Loaded symbols for /lib/libkeyutils.so.1
Reading symbols from /lib/libselinux.so.1...(no debugging symbols
found)...done.
Loaded symbols for /lib/libselinux.so.1
Reading symbols from /lib/libsepol.so.1...(no debugging symbols
found)...done.
Loaded symbols for /lib/libsepol.so.1
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so...done.
Loaded symbols for
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1...done.
Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so...done.
Loaded symbols for
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1...done.
Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so...done.
Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so...done.
Loaded symbols for
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so...done.
Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so...done.
Loaded symbols for
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so
Reading symbols from
/usr/local/kamailio-3.1.3/lib/kamailio/modules_k/pv.so...done.
Loaded symbols for

Re: [SR-Users] Kamailio 3.1.3 uncontrolled core dump

2011-05-13 Thread Dominguez Jover, Ricardo
Hi Klaus,

I'm investigating why my config tries to solve @192.168.64.36, I know it is 
related to presence subscriptions. Many @192.168.64.36 petitions are being 
processed, however Kamailio only crashes when it comes from one of our users. 
¡! I'm working on it

Anyway I think that Kamailio shouldn't crash because of this sip uri, if you 
are trying to reproduce it and need any more logs or debug info please tell me. 

Regards,

Ricardo


-Mensaje original-
De: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Klaus Darilion
Enviado el: viernes, 13 de mayo de 2011 14:14
Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - 
UsersMailing List
Asunto: Re: [SR-Users] Kamailio 3.1.3 uncontrolled core dump

1. seems like your config is broken and request uri is set to
  sip:@192.168.64.36

2. Anyways, Kamailio shouldn't crash. I tried to reproduce the failure,
but my Kamailio does not crash.

regards
klaus

Am 13.05.2011 10:19, schrieb Dominguez Jover, Ricardo:
 06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]:
 ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable
 destination (478/SL) 
 May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]:
 ERROR: core [resolve.c:1540]: ERROR: sip_hostport2su: could not
 resolve hostname: @192.168.64.36 

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Re: [SR-Users] core dump

2011-05-12 Thread Dominguez Jover, Ricardo
Thanks Klaus, I found the commit which is included in release 3.1.3. We've 
upgraded some hours ago and no crash at the moment (so no core dump is neither 
generated).

I'll let you know about this issue.

Regards,
Ricardo

-Mensaje original-
De: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Klaus Darilion
Enviado el: miércoles, 11 de mayo de 2011 11:40
Para: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] core dump

IIRC there were some changes in recent kernels which needs some more
tweaking (there was a commit from Daniel, but I can't remember the details)

regards
Klaus

Am 11.05.2011 10:28, schrieb Dominguez Jover, Ricardo:
 Hi everybody,
 
 We are having an uncontrolled crash in Kamailio 3.1:
 
 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23366]: : core 
 [pass_fd.c:293]: ERROR: receive_fd: EOF on 39
 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: 
 core [main.c:741]: child process 23364 exited by a signal 11
 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: 
 core [main.c:744]: core was not generated
 
 
 I've read in several posts I must generate the Core Dump to know what is 
 happening. I'm trying to generate it in this way.
 
 /etc/init.d/Kamailio file:
 if test $DUMP_CORE = yes ; then
 1. set proper ulimit
 ulimit -c unlimited
 directory for the core dump files
 COREDIR=/dumps/
 [ -d $COREDIR ] || mkdir $COREDIR
 chmod 777 $COREDIR
 echo $COREDIR/core.%e.sig%s.%p  /proc/sys/kernel/core_pattern
 fi 
 
 
 /etc/default/Kamailio file:
 DUMP_CORE=yes
 
 
 But the core is not yet generated.
 
 Then I've added -w option:
 
 OPTIONS=-P $PID_FILE -m $MEMORY -u $USER -g $GROUP -w /dumps/
 
 But core still not generated
 
 In kamailio.cfg I´ve also added:
 disable_core_dump=no 
 
 
 User running Kamailio is kamailio who has 777 permissions in /dumps/ 
 directory. No way.
 
 
 Anyhelp would be appreciated.
 
 Cheers,
 
 Ricardo Domínguez
 
 
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Re: [SR-Users] core dump

2011-05-11 Thread Dominguez Jover, Ricardo
Thanks Hervé. I'm using Kamailio 3.1.0 in a RHEL 5. Do you know if this issue 
is solved in Kamailio release 3.1.3?

Cheers,
Ricardo Domínguez

De: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Hervé Cochet
Enviado el: miércoles, 11 de mayo de 2011 11:20
Para: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] core dump

Hi,

I also have the same problem with kamailio 3.1

I made a modification to the file daemonize.c because if lim.rlim_cur is set to 
-1 the test with size parameter at line 491 do not work because rlimit 
parameters are unsigned int.

--- kamailio-3.1.0/daemonize.c.ori    2011-04-12 12:24:14.0 +0200
+++ kamailio-3.1.0/daemonize.c    2011-04-12 12:24:57.0 +0200
@@ -488,7 +488,7 @@
                 strerror(errno));
         goto error;
     }
-        if (lim.rlim_cursize){
+        if ((int)lim.rlim_cursize){
         /* first try max limits */
         newlim.rlim_max=RLIM_INFINITY;
         newlim.rlim_cur=newlim.rlim_max;

Thanks to this patch the core dump should be generated.

This work for me on my testing servers, BUT with my production servers (debian 
5.0.1) where the core is not generated and I cannot understand why...


Hervé

On 11/05/2011 10:28, Dominguez Jover, Ricardo wrote: 
Hi everybody,

We are having an uncontrolled crash in Kamailio 3.1:

May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23366]: : core 
[pass_fd.c:293]: ERROR: receive_fd: EOF on 39
May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: 
core [main.c:741]: child process 23364 exited by a signal 11
May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: 
core [main.c:744]: core was not generated


I've read in several posts I must generate the Core Dump to know what is 
happening. I'm trying to generate it in this way.

/etc/init.d/Kamailio file:
if test $DUMP_CORE = yes ; then
1. set proper ulimit
ulimit -c unlimited
directory for the core dump files
COREDIR=/dumps/
[ -d $COREDIR ] || mkdir $COREDIR
chmod 777 $COREDIR
echo $COREDIR/core.%e.sig%s.%p  /proc/sys/kernel/core_pattern
fi 


/etc/default/Kamailio file:
DUMP_CORE=yes


But the core is not yet generated.

Then I've added -w option:

OPTIONS=-P $PID_FILE -m $MEMORY -u $USER -g $GROUP -w /dumps/

But core still not generated

In kamailio.cfg I´ve also added:
disable_core_dump=no 


User running Kamailio is kamailio who has 777 permissions in /dumps/ 
directory. No way.


Anyhelp would be appreciated.

Cheers,

Ricardo Domínguez


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-- 


Hervé COCHET.

Ingénieur en développement logiciel.
Tel Direct:
+33(0)482 531 303
TECHNOSENS SAS
Donnons du sens à la Technologie
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+33(0)476 230 240
www.technosens.fr 

Ce message et les documents l'accompagnant sont confidentiels. Ils contiennent 
des informations qui sont destinées uniquement à la personne ou l'entité dont 
le nom est indiqué ci-dessus. Toute reproduction, divulgation ou autre 
utilisation de ces informations, même partiellement, par un autre destinataire 
est strictement interdite. Si ce message vous est parvenu par erreur, veuillez 
le détruire immédiatement et nous le faire savoir par téléphone, Fax ou e-mail.

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Re: [SR-Users] ACK not sent and rr-enforced

2011-03-16 Thread Dominguez Jover, Ricardo
Hi Alex,

As I said I was doubting about this inference, but as calls are working
with other providers and I read the post I linked, I don't really know
in what side the solution is.


The scenario is as follows:

Softphone A - providerProxy - myProxy - Softphone B

Softphone A sends the invite to Softphone B through providerProxy and
myProxy
Softphone B sends the 200OK with CONTACT: user@softphone_B_contact_URI
to myProxy
myProxy sends the 200OK with CONTACT: user@softphone_B_contact_URI to
providerProxy
providerProxy sends the 200OK with CONTACT: user@myproxy_IP_address to
Softphone A
Softphone A sends ACK sip:user@myproxy_IP_address and when it arrives
to myproxy it is never sent to Softphone B and there is a timeout 30
seconds later.

Is there anything I can do to solve this?

Thank you,
Ricardo



-Mensaje original-
De: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Alex
Balashov
Enviado el: martes, 15 de marzo de 2011 17:25
Para: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] ACK not sent and rr-enforced

On 03/15/2011 08:28 AM, Dominguez Jover, Ricardo wrote:

 Should I infer IPTEL.org is not implementing SIP RFC 3261 in the
 right way? It seems odd to me...

No, Ricardo, that is not the correct inference.  First, if the ACK is an

end-to-end ACK (as for a 200 OK), it is generated by the sending 
endpoint, and the SER proxy is not responsible for constructing it. 
Secondly, there are various reasons why an ACK may have a request line 
not equal to the Contact URI established as the dialog target, having to

do with backward compatibility with RFC 2543.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [SR-Users] ACK not sent and rr-enforced

2011-03-15 Thread Dominguez Jover, Ricardo
Hi,

I've found this post where it says:

 ...the Contact header in the 200 OK and the request URI in the ACK. They MUST 
be the same!!!...

http://www.mail-archive.com/users@lists.kamailio.org/msg00606.html

Should I infer IPTEL.org is not implementing SIP RFC 3261 in the right way? It 
seems odd to me... 

Cheers,
Ricardo


De: Daniel-Constantin Mierla [mailto:mico...@gmail.com] 
Enviado el: lunes, 14 de marzo de 2011 10:58
Para: Dominguez Jover, Ricardo
CC: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] ACK not sent and rr-enforced

Hello,

I will look over it very soon. As a hint for the future, if you catch me 
traveling, rar files won't work for me, use tgz or zip as they are easy to 
expand very easy even on web mail clients. If the trace is not big, plain text 
is faster or eventually use some pastebin sites out there.

Cheers,
Daniel

On 3/10/11 1:49 PM, Dominguez Jover, Ricardo wrote: 
Hello Daniel, here it is.

Thanks

Ricardo

De: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Enviado el: jueves, 10 
de marzo de 2011 12:49
Para: Dominguez Jover, Ricardo
CC: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] ACK not sent and rr-enforced

Hello,

can you post the ngrep trace of such call (fron incoming invite, to the bye, 
taken on your server)? That will help to see what could be mismatching there.

Cheers,
Daniel
On Thu, Mar 10, 2011 at 11:06 AM, Dominguez Jover, Ricardo djo...@umh.es 
wrote:
Hi again,

I'm still working in this issue. I've noticed that iptel proxy is writing in 
the ACK message the following:

ACK sip:username@myproxyIP:5060;.   - ACK is not sent to the client. 
tcheck_trans fails. If a force the transfer - t_relay do nothing

while sip2sip and VoIP-Talk are writing:

ACK sip:username@userprivateIP:5060;  - ACK is sent to the client

In both cases, contact URI sent in the 200 OK message by my proxy is the 
private IP address of the client  sending the 200 OK, so I don't know why IPtel 
doesn't use it in the ACK. I find a lot of information about lost ACKs in 
posts, but not this particular issue.

Could anyone give me some related information that can help me to solve this 
issue?

Best regards,

Ricardo Dominguez




De: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Dominguez Jover, 
Ricardo Enviado el: lunes, 07 de marzo de 2011 20:03
Para: sr-users@lists.sip-router.org
Asunto: [SR-Users] ACK not sent and rr-enforced

Hi everybody.

I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external 
test accounts to check if the calls are established and the media flow is ok.

When I use a sip2sip.info or  VoIP Talk accounts, then all is working fine 
between my internal and these external accounts.

But when I use a iptel.org account and this account calls to an internal 
account (registered with kamailio), then callee sends the 200 OK to the SIP 
proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy 
with this lines at the end of the packet:

P-hint:  rr-enforced\r\n
P-hint:  rr-enforced\r\n

And my SIP proxy never resends the ACK to the callee, so the callee resends OK 
200 periodically and after 32 seconds sends a BYE message and the call is 
finished.

I've been reading posts about missing ACKs but I can't find the answer to my 
problem, that it seems like t_check_trans doesn´t recognize the ACK as 
related to a transaction. But this is only with IPTEL accounts, my proxy SIP is 
working with other SIP providers, so I don't know if forcing relay of every ACK 
packet is a good idea.

Any help would be appreciated.

Thanks,

Ricardo
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  http://www.asipto.com


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-- 
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http://www.asipto.com

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Re: [SR-Users] ACK not sent and rr-enforced

2011-03-10 Thread Dominguez Jover, Ricardo
Hello Daniel, here it is.

Thanks

Ricardo

De: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Enviado el: jueves, 10 
de marzo de 2011 12:49
Para: Dominguez Jover, Ricardo
CC: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] ACK not sent and rr-enforced

Hello,

can you post the ngrep trace of such call (fron incoming invite, to the bye, 
taken on your server)? That will help to see what could be mismatching there.

Cheers,
Daniel
On Thu, Mar 10, 2011 at 11:06 AM, Dominguez Jover, Ricardo djo...@umh.es 
wrote:
Hi again,

I'm still working in this issue. I've noticed that iptel proxy is writing in 
the ACK message the following:

ACK sip:username@myproxyIP:5060;.   - ACK is not sent to the client. 
tcheck_trans fails. If a force the transfer - t_relay do nothing

while sip2sip and VoIP-Talk are writing:

ACK sip:username@userprivateIP:5060;  - ACK is sent to the client

In both cases, contact URI sent in the 200 OK message by my proxy is the 
private IP address of the client  sending the 200 OK, so I don't know why IPtel 
doesn't use it in the ACK. I find a lot of information about lost ACKs in 
posts, but not this particular issue.

Could anyone give me some related information that can help me to solve this 
issue?

Best regards,

Ricardo Dominguez




De: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Dominguez Jover, 
Ricardo Enviado el: lunes, 07 de marzo de 2011 20:03
Para: sr-users@lists.sip-router.org
Asunto: [SR-Users] ACK not sent and rr-enforced

Hi everybody.

I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external 
test accounts to check if the calls are established and the media flow is ok.

When I use a sip2sip.info or  VoIP Talk accounts, then all is working fine 
between my internal and these external accounts.

But when I use a iptel.org account and this account calls to an internal 
account (registered with kamailio), then callee sends the 200 OK to the SIP 
proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy 
with this lines at the end of the packet:

P-hint:  rr-enforced\r\n
P-hint:  rr-enforced\r\n

And my SIP proxy never resends the ACK to the callee, so the callee resends OK 
200 periodically and after 32 seconds sends a BYE message and the call is 
finished.

I've been reading posts about missing ACKs but I can't find the answer to my 
problem, that it seems like t_check_trans doesn´t recognize the ACK as 
related to a transaction. But this is only with IPTEL accounts, my proxy SIP is 
working with other SIP providers, so I don't know if forcing relay of every ACK 
packet is a good idea.

Any help would be appreciated.

Thanks,

Ricardo
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--
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  http://www.asipto.com


ack_trace.rar
Description: ack_trace.rar
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[SR-Users] ACK not sent and rr-enforced

2011-03-07 Thread Dominguez Jover, Ricardo
Hi everybody.

 

I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external 
test accounts to check if the calls are established and the media flow is ok.

 

When I use a sip2sip.info or  VoIP Talk accounts, then all is working fine 
between my internal and these external accounts.

 

But when I use a iptel.org account and this account calls to an internal 
account (registered with kamailio), then callee sends the 200 OK to the SIP 
proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy 
with this lines at the end of the packet:

 

P-hint:  rr-enforced\r\n

P-hint:  rr-enforced\r\n

 

And my SIP proxy never resends the ACK to the callee, so the callee resends OK 
200 periodically and after 32 seconds sends a BYE message and the call is 
finished.

 

I've been reading posts about missing ACKs but I can't find the answer to my 
problem, that it seems like t_check_trans doesn´t recognize the ACK as 
related to a transaction. But this is only with IPTEL accounts, my proxy SIP is 
working with other SIP providers, so I don't know if forcing relay of every ACK 
packet is a good idea.

 

Any help would be appreciated.

 

Thanks,

 

Ricardo

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[SR-Users] rtpproxy and connection information field

2011-03-01 Thread Dominguez Jover, Ricardo

Hi all,

I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in almost every 
case. However I have a problem with some call.

When softphone A using sip2sip.info account calls softphone B using my Kamailio 
server account, the Kamilio receives SIP packets from IP1 (81.23.228.129). The 
invite packet has IP2 (81.23.228.150) in the conecction information field (c=IN 
IP4 81.23.228.150):


INVITE sip:12...@.xxx;transport=udp SIP/2.0
Record-Route: sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0
Via: SIP/2.0/UDP 
192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e37bb89-1---d8754z-;rport=7964
Max-Forwards: 69
Contact: sip:54...@88.xxx.xxx.xxx:7964;transport=udp
To: 2205sip:12...@.xxx
From: R sip2sipsip:54...@sip2sip.info;tag=d9e99adf
Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Content-Type: application/sdp
Supported: replaces
User-Agent:x
Content-Length: 279

v=0
o=- 12943454020854250 1 IN IP4 192.168.xxx.xx
s=
c=IN IP4 81.23.228.150
t=0 0
m=audio 52854 RTP/AVP 107 0 8 18 101
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


Then Kamilio translates c= IP address to my RTPproxy IPaddress and sends the 
INVITE to softphone B. After the OK softphone B  sends RTP packets to the 
RTPproxy as specified in the c= field, however the RTPproxy sends RTP packets 
to IP1 instead of send packets to IP2. How can I tell the RTPproxy to send 
packets to IP2 (the one in the c= field?)


Thanks,
Ricardo Dominguez
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Re: [SR-Users] rtpproxy and connection information field

2011-03-01 Thread Dominguez Jover, Ricardo
I have to use r flag. Sorry for my quick posting...

 

 

De: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Dominguez
Jover, Ricardo
Enviado el: martes, 01 de marzo de 2011 13:16
Para: sr-users@lists.sip-router.org
Asunto: [SR-Users] rtpproxy and connection information field

 

 

Hi all,

I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in
almost every case. However I have a problem with some call.

When softphone A using sip2sip.info account calls softphone B using my
Kamailio server account, the Kamilio receives SIP packets from IP1
(81.23.228.129). The invite packet has IP2 (81.23.228.150) in the
conecction information field (c=IN IP4 81.23.228.150):


INVITE sip:12...@.xxx;transport=udp SIP/2.0
Record-Route: sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0
Via: SIP/2.0/UDP
192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e
37bb89-1---d8754z-;rport=7964
Max-Forwards: 69
Contact: sip:54...@88.xxx.xxx.xxx:7964;transport=udp
To: 2205sip:12...@.xxx
From: R sip2sipsip:54...@sip2sip.info;tag=d9e99adf
Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent:x
Content-Length: 279

v=0
o=- 12943454020854250 1 IN IP4 192.168.xxx.xx
s=
c=IN IP4 81.23.228.150
t=0 0
m=audio 52854 RTP/AVP 107 0 8 18 101
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


Then Kamilio translates c= IP address to my RTPproxy IPaddress and
sends the INVITE to softphone B. After the OK softphone B  sends RTP
packets to the RTPproxy as specified in the c= field, however the
RTPproxy sends RTP packets to IP1 instead of send packets to IP2. How
can I tell the RTPproxy to send packets to IP2 (the one in the c=
field?)


Thanks,
Ricardo Dominguez 

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[SR-Users] Radius and kamailio: avp unknown attribute

2011-02-01 Thread Dominguez Jover, Ricardo
Hi everybody,

 

I have a problem with Kamalio and Radius, when a try to authenticate a
user I get the following:

 

Feb  1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]:
rc_avpair_new: unknown attribute 5

Feb  1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]:
ERROR: auth_radius [sterman.c:412]: authorization failed

 

In fact I can't see any packet sended to the Radius Server from
Kamailio. I've read several posts and I've chmod of /var/run/radius.seq,
include dictionary.sip in dictionary.radius and so on.

 

Changing if (!radius_www_authorize($td)) with 
if (!radius_www_authorize()) I don't get any error but and can't
neither see any packet sended to the Radius Server. I've checked the
connection between radius client and radius server and it's OK, I get:

 

Received response ID 91, code 2, length = 211

 

Any help would be appreciated.

 

Kind regards,

Ricardo Dominguez

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Re: [SR-Users] Radius and kamailio: avp unknown attribute

2011-02-01 Thread Dominguez Jover, Ricardo
Solved, bad dictionary

 

Thanks

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[SR-Users] RTP to SRTP bridge

2010-12-22 Thread Dominguez Jover, Ricardo
Hi everybody,

 

I'm trying to deploy an scenario where all calls in the LAN are encrypted with 
SRTP, so I've forced my sip softphones to mandatory SRTP. As in the Internet 
not all providers and clients have SRTP enabled, I couldn´t communicate them in 
this way, so I would need some kind of border element to bridge RTP and SRTP 
calls. Is there any bridge/proxy who can receive rtp packets and convert them 
into srtp and viceversa?

 

I've been reading RTPproxy and Media proxy docs and also some forums, but I 
didn´t find anything useful.

 

Regards,

 

Ricardo Dominguez

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Re: [SR-Users] kamailio restart and TLS ( relay_to_tls() )

2010-12-21 Thread Dominguez Jover, Ricardo
TCP ASYNC=YES fixed the problem (set_forward_no_connect() didn´t.)

I don't know if you notice that TCP ASYNC in Core Cook Book v3.1 is not 
updated with this feature with TLS:
http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#tcp_async

although in Kamailio 3.1 New Features DOC it is:
http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.1.x#asynchronous_tls

Thanks a lot for your help.

Best regards,

Ricardo Domínguez


-Mensaje original-
De: Klaus Darilion [mailto:klaus.mailingli...@pernau.at] 
Enviado el: martes, 21 de diciembre de 2010 20:03
Para: Dominguez Jover, Ricardo
CC: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] kamailio restart and TLS ( relay_to_tls() )



Am 21.12.2010 18:46, schrieb Dominguez Jover, Ricardo:
 Hi again Klaus,
 I understand (now better) what you mean with timing parameters, I was
 just testing to close the first connection. The reason is because when I
 restart kamailio the clients I use reopen a second connection, as you
 said to me. So the solution to this issue could be not to open newer
 connection. I tested, as you said, set_forward_no_connect(); but may
 be not well enough. I imagine the solution goes by using it.

The TCP connection should be kept alive as long as possible. If for some 
reason the TCP connection is lost (client crash, network failure, 
kamailio restart) there are two things the should be done:
- the client has to create a new registration on a new TCP connection
- the proxy should ignore contacts without existing TCP connections 
(thus use set_forward_no_connect())

 About the question on making TLS connection to the clients, I'm only
 relaying TLS connections to the gateway, who has a certificate. I set
 TCP ASYNC=NO, because I had an error running TLS, as documentation says
 if I use TLS I have to disable asynch TCP.

That was valid with 3.0 release. Since 3.1 release this is fixed and 
asynch mode can be used also with TLS.

regards
Klaus


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[SR-Users] kamailio restart and TLS ( relay_to_tls() )

2010-12-20 Thread Dominguez Jover, Ricardo
Hi everybody,

 

Since I implemented Kamailio 3.1 with TLS I've found a strange behavior. That 
is, with some clients (Bria and Blink) registered, if I restart Kamailio, then 
when the clients re-register the strange behaivour happens. This behavior 
consist on receiving calls, it took about 15 seconds to receive the first tone 
since the call was made.

 

I made the following modification in the route[Relay] config. The reason is I 
wanted my gateway and Kamailio to make signaling by TLS. Without this 
modification the signaling was unencrypted (SIP):

 

route[RELAY] {

#!ifdef WITH_NAT

if (check_route_param(nat=yes)) {

setbflag(FLB_NATB);

}

if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {

route(RTPPROXY);

}

#!endif

 

/* example how to enable some additional event routes */

if (is_method(INVITE)) {

#t_on_branch(BRANCH_ONE);

t_on_reply(REPLY_ONE);

t_on_failure(FAIL_ONE);

}

 

# Se comunica con el GWa traves de TLS 

if(!( ($od=~mydomain.com)  ( ($rU=~[a-z]{3,20}$) || 
($rU=~^xx[0-9][0-9]$) ) ) ) {### If I'm calling a PBX extension do the 
signaling by TLS with the gateway (Cisco 2811)

 

if (!t_relay_to_tls()) {

sl_reply_error();

}

} else if {

 

if (!t_relay()) {

sl_reply_error();

}

}

exit;

}

 

The rest of functionalities are working really fine. Any idea about what is 
happening?

 

Cheers!

 

Ricardo Domínguez

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[SR-Users] append_rpid_hf() and append_hf() issues

2010-11-10 Thread Dominguez Jover, Ricardo
Hi everybody,

 

I've been testing and setting up kamalio 3.0.3 with MySQL since a few
weeks ago. All the basic functionalities are working properly, but now I
have an issue which I'm not been able to solve by myself. I've read
documentation and posts, but no idea about what is happening. This is
the issue.

 

I'm trying to append Remote Party ID field to the SIP message header.
I've been trying with both append_rpid_hf() and append_hf() and I
can't see any Remote Party Header at the SIP messages sniffed with
wireshark. Here is the configuration I've tried:

 

#para cargar los Remote party ID de la tabla suscriber

modparam(auth_db, load_credentials, $avp(s:rpid)=rpid)

modparam(auth, rpid_avp, $avp(s:rpid))

modparam(auth, rpid_prefix, sip:)

modparam(auth, rpid_suffix,
;user=phone;party=calling;screen=yes;id-type=subscriber;privacy=off)

 

#Modificacion para no pedir autenticacion si viene desde un trusted

if (!allow_trusted($si, $(proto{s.toupper}))) {

 

if (!proxy_authorize(umh.es, subscriber)) {

proxy_challenge(, 0);

exit;

}

if (is_method(PUBLISH))

{

if ($au!=$tU) {

sl_send_reply(403,Forbidden
auth ID);

exit;

}

} else {

if ($au!=$fU) {

sl_send_reply(403,Forbidden
auth ID);

exit;

}

}

 

append_rpid_hf($fU,
;party=calling;id-type=subscriber;privacy=off;screen=yes);

consume_credentials();

 

 

### I also tested the following

# append_hf(Remote-Party-ID:
sip:$avp(rpid)@$avp(domain);user=phone;privacy=$avp(privacy);party=cal
ling\r\n);

# append_hf(P-Asserted-Identity:
sip:$avp(rpid)@$avp(domain)\r\n, Call-ID);

# consume_credentials();

 

}

}

 

 

In a similar post I've read that the user had a problem because of the
domain table. I have both the domain name and the IP of the host
defined, and I also get response with $avp(s:rpid) , when RPID field is
defined to a subscriber.

 

As I said I can't find the mistake, so any help would be appreciated.

 

Thanks in advance and best regards,

Ricardo Dominguez

 

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Re: [SR-Users] append_rpid_hf() and append_hf() issues

2010-11-10 Thread Dominguez Jover, Ricardo
Hi Alex, I think I'm doing what you say, look at the configuration my config 
file:

 

append_rpid_hf($fU, 
;party=calling;id-type=subscriber;privacy=off;screen=yes);

consume_credentials();

 

(the following lines are commented)

 

### I also tested the following

# 
append_hf(Remote-Party-ID:sip:$avp(rpid)@$avp(domain);user=phone;privacy=$avp(privacy);party=calling\r\n);

# 
append_hf(P-Asserted-Identity:sip:$avp(rpid)@$avp(domain)\r\n, Call-ID);

# consume_credentials();

 

 

Is that right or you mean something like this:

 

append_rpid_hf($avp(rpid), 
;party=calling;id-type=subscriber;privacy=off;screen=yes);

consume_credentials();

?

 

Anyway I've just tested this last configuration and I can't see Remote Party 
ID headers in SIP message.

 

Kind regards,

Ricardo

 

-Mensaje original-
De: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Alex Balashov
Enviado el: miércoles, 10 de noviembre de 2010 11:24
Para: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] append_rpid_hf() and append_hf() issues

 

Ricardo,

 

The 'rpid' value is loaded from the 'subscriber' table as part of the 

authentication functions;  if you consume_credentials() before it, it 

will be gone.  You should append_rpid_hf() the RPID header first, and 

consume_credentials() afterward.

 

Cheers,

 

-- Alex

 

 

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