13 Feb
2009
13 Feb
'09
3:26 p.m.
Hello,
On 02/13/2009 12:08 PM, Abdul Hakeem wrote:
Hi,
Are you able to put me in touch with the guys using the SDP parser ?
direct your emails to mailing list all the time. Those people are there,
the developer of SDP parser and others using it. You will get much
faster response and maybe better than writing private emails.
Thanks,
Daniel
Cheers,
Abdul Hakeem
-----Original Message-----
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Sent: 11 February 2009 15:20
To: alhakeem(a)ipextelecom.net; kamailio
Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
Hello,
On 02/11/2009 03:20 PM, Abdul Hakeem wrote:
--
Daniel-Constantin Mierla
http://www.asipto.com
Hi,
Just wanted to touch base on the parser to find out if you are still
pursuing this.
it is still on my todo list, however didn't make it in the upcoming
1.5.0.
Ovidiu Sas did some commits to the SDP parser and I know couple of others
using it with success.
Please open a feature request at:
http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=br…
It is easier to track and not forgot about it for the next release.
Thanks,
Daniel
I am in the mind of purchasing some Cisco
equipment to do the Sip-MAP
gateway from a GSM network.
I have someone else from the OpenBTS with me on this.
The object is to provide a full fledged sip based routing of voice,
sms and multimedia calls from gsm handsets to BTS with a Sip server
backend.
Cheers,
Abdul Hakeem
-----Original Message-----
From: users-bounces(a)lists.kamailio.org
[mailto:users-bounces@lists.kamailio.org] On Behalf Of
Daniel-Constantin Mierla
Sent: 03 September 2008 17:45
To: Kristian Kielhofner
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
Hello,
On 09/03/08 19:10, Kristian Kielhofner wrote:
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello everyone,
Feel free to set $OSSPROXY to whatever you like =
Kamailio/OpenSIPS/OpenSER (for those of you who can't let go).
$OSSPROXY is my favorite SIP toolkit. In addition to being an
excellent proxy and registrar it serves as a platform to do SIP
mangling and wrangling. With that in mind let me propose a crazy
idea...
How hard would it be to get $OSSPROXY to be able to parse what is
commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has
no business getting too involved in the SDP but I feel it makes sense
in some instances (like this one). What if I want to make a routing
decision based on an encapsulated ISUP parameter of some sort? Who
knows what it could be!?!? Maybe there are some other uses but that
makes the most sense and seems to be the most practical. It should
be possible to develop an $OSSPROXY module to:
- Use the SDP parser to find the ISUP SDP part:
http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing
- Pass that part into a "real" ISUP parser (if necessary). I
couldn't seem to find the perfect library but perhaps something like
this could
work:
http://www.openss7.org/isup.html
- Make the ISUP parameters available via AVPs, pseudo variables, etc
for use in the $OSSPROXY script.
I don't really have a need for this but it seems like it could be
cool. Am I nuts?
I would expose the body as a dynamic structure, so components will be
accessible in the script, e.g. (naming schema just for example),
if($body(0=>content-type)=="application/sdp")
{
if($body(0=>m=>type) == "audio")
{
}
}
Based on the available content type parsers (sdp, isup), the structure
has different attributes.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
13 Feb
13 Feb
9:07 p.m.
New subject: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
The multipart body content parsing is embedded in the sdp parser.
The sdp parser is available in the source tree under the ./parser/sdp directory.
Hope this helps,
Ovidiu Sas
On Fri, Feb 13, 2009 at 8:26 AM, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
Hello,
On 02/13/2009 12:08 PM, Abdul Hakeem wrote:
Hi,
Are you able to put me in touch with the guys using the SDP parser ?
direct your emails to mailing list all the time. Those people are there,
the developer of SDP parser and others using it. You will get much
faster response and maybe better than writing private emails.
Thanks,
Daniel
Cheers,
Abdul Hakeem
-----Original Message-----
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Sent: 11 February 2009 15:20
To: alhakeem(a)ipextelecom.net; kamailio
Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
Hello,
On 02/11/2009 03:20 PM, Abdul Hakeem wrote:
--
Daniel-Constantin Mierla
http://www.asipto.com
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Hi,
Just wanted to touch base on the parser to find out if you are still
pursuing this.
it is still on my todo list, however didn't make it in the upcoming 1.5.0.
Ovidiu Sas did some commits to the SDP parser and I know couple of others
using it with success.
Please open a feature request at:
http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=br…
It is easier to track and not forgot about it for the next release.
Thanks,
Daniel
I am in the mind of purchasing some Cisco
equipment to do the Sip-MAP
gateway from a GSM network.
I have someone else from the OpenBTS with me on this.
The object is to provide a full fledged sip based routing of voice,
sms and multimedia calls from gsm handsets to BTS with a Sip server
backend.
Cheers,
Abdul Hakeem
-----Original Message-----
From: users-bounces(a)lists.kamailio.org
[mailto:users-bounces@lists.kamailio.org] On Behalf Of
Daniel-Constantin Mierla
Sent: 03 September 2008 17:45
To: Kristian Kielhofner
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
Hello,
On 09/03/08 19:10, Kristian Kielhofner wrote:
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello everyone,
Feel free to set $OSSPROXY to whatever you like =
Kamailio/OpenSIPS/OpenSER (for those of you who can't let go).
$OSSPROXY is my favorite SIP toolkit. In addition to being an
excellent proxy and registrar it serves as a platform to do SIP
mangling and wrangling. With that in mind let me propose a crazy
idea...
How hard would it be to get $OSSPROXY to be able to parse what is
commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has
no business getting too involved in the SDP but I feel it makes sense
in some instances (like this one). What if I want to make a routing
decision based on an encapsulated ISUP parameter of some sort? Who
knows what it could be!?!? Maybe there are some other uses but that
makes the most sense and seems to be the most practical. It should
be possible to develop an $OSSPROXY module to:
- Use the SDP parser to find the ISUP SDP part:
http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing
- Pass that part into a "real" ISUP parser (if necessary). I
couldn't seem to find the perfect library but perhaps something like
this could
work:
http://www.openss7.org/isup.html
- Make the ISUP parameters available via AVPs, pseudo variables, etc
for use in the $OSSPROXY script.
I don't really have a need for this but it seems like it could be
cool. Am I nuts?
I would expose the body as a dynamic structure, so components will be
accessible in the script, e.g. (naming schema just for example),
if($body(0=>content-type)=="application/sdp")
{
if($body(0=>m=>type) == "audio")
{
}
}
Based on the available content type parsers (sdp, isup), the structure
has different attributes.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
16 Feb
16 Feb
2:10 p.m.
New subject: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
On Friday 13 February 2009, Ovidiu Sas wrote:
The multipart body content parsing is embedded in the
sdp parser.
The sdp parser is available in the source tree under the ./parser/sdp
directory.
There is also some documentation about the parser on our wiki:
http://www.kamailio.org/dokuwiki/doku.php/development:sdp-parsing
Cheers,
Henning
2:54 p.m.
New subject: [Kamailio-Users] Call Stateful Server
Hi list,
I need some explanation about using Kamailio as a Call Stateful Proxy.
I understand that Kamailio is a transaction stateful proxy. And I would
like to learn how I can achieve a Call Stateful Proxy using Kamailio.
I'm going to use Asterisk for transcoding to receive and send calls to the
users and Kamailio should do the routing process using carrierroute,
deciding to which carrier the calls should be sent to.
Customer Carrier1
Group1 \ /
\ /
\ /
Customer ---->GW ---> Kamailio ---> GW ---> Carrier2
Group2 Asterisk Asterisk
/ \
/ \
Customer / Carrier3
Group3
Now it works with x-lite, if customer1 calls to certain number the call is
send to the correspondent carrier base on the prefix number.
But when I add the GWs, how I can send the call to the GW IP address
(Asterisk) and then forward to the IP address of the carrier that I
assigned for that prefix number in my carrierroute table.
Can any of you could help me to clarify this or guide me about what to
read, so I could implement this.
Thanks in advance,
Juan.-
3:24 p.m.
New subject: [Kamailio-Users] Call Stateful Server
Kamailio is not, in fact, a call-stateful proxy, although for certain
applications the 'dialog' module can help confer the illusion that it
is.
--
Sent from mobile device
On Feb 16, 2009, at 7:54 AM, "Juan Asencio" <juan(a)asencio.dk> wrote:
Hi list,
I need some explanation about using Kamailio as a Call Stateful Proxy.
I understand that Kamailio is a transaction stateful proxy. And I
would
like to learn how I can achieve a Call Stateful Proxy using Kamailio.
I'm going to use Asterisk for transcoding to receive and send calls
to the
users and Kamailio should do the routing process using carrierroute,
deciding to which carrier the calls should be sent to.
Customer Carrier1
Group1 \ /
\ /
\ /
Customer ---->GW ---> Kamailio ---> GW ---> Carrier2
Group2 Asterisk Asterisk
/ \
/ \
Customer / Carrier3
Group3
Now it works with x-lite, if customer1 calls to certain number the
call is
send to the correspondent carrier base on the prefix number.
But when I add the GWs, how I can send the call to the GW IP address
(Asterisk) and then forward to the IP address of the carrier that I
assigned for that prefix number in my carrierroute table.
Can any of you could help me to clarify this or guide me about what to
read, so I could implement this.
Thanks in advance,
Juan.-
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
4:40 p.m.
New subject: [Kamailio-Users] Call Stateful Server
Hello,
your issue has nothing to do with a call stateful server.
You simply do record route in your config so for within dialog requests
the routing is don properly.
For initial invite, I assume (as I never used it) carrierroute sets the
R-URI so you just use:
t_relay("udp:asteriskip:asteriskport");
and in asterisk you dial the address in the r-uri.
If carrierroute is setting the dst_uri, then add it as special header,
use same to forward to asterisk, but there use the special header to
dial out.
Cheers,
Daniel
On 02/16/2009 02:54 PM, Juan Asencio wrote:
Hi list,
I need some explanation about using Kamailio as a Call Stateful Proxy.
I understand that Kamailio is a transaction stateful proxy. And I would
like to learn how I can achieve a Call Stateful Proxy using Kamailio.
I'm going to use Asterisk for transcoding to receive and send calls to the
users and Kamailio should do the routing process using carrierroute,
deciding to which carrier the calls should be sent to.
Customer Carrier1
Group1 \ /
\ /
\ /
Customer ---->GW ---> Kamailio ---> GW ---> Carrier2
Group2 Asterisk Asterisk
/ \
/ \
Customer / Carrier3
Group3
Now it works with x-lite, if customer1 calls to certain number the call is
send to the correspondent carrier base on the prefix number.
But when I add the GWs, how I can send the call to the GW IP address
(Asterisk) and then forward to the IP address of the carrier that I
assigned for that prefix number in my carrierroute table.
Can any of you could help me to clarify this or guide me about what to
read, so I could implement this.
Thanks in advance,
Juan.-
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
--
Daniel-Constantin Mierla
http://www.asipto.com
4:45 p.m.
New subject: [Kamailio-Users] Call Stateful Server
On Monday 16 February 2009, Daniel-Constantin Mierla wrote:
Hello,
your issue has nothing to do with a call stateful server.
You simply do record route in your config so for within dialog requests
the routing is don properly.
For initial invite, I assume (as I never used it) carrierroute sets the
R-URI so you just use:
Hi Daniel,
cr rewrites the request URI.
Cheers,
Henning
18 Feb
18 Feb
1:31 p.m.
New subject: [Kamailio-Users] Call Stateful Server
Hello Daniel and Henning, thank you for your help.
I add the t_relay(...) and now the call is sent to the gateway. But I'm
getting some errors.
I will appreciate your advice:
I'm using it to route calls base on the prefix number. This seems to work
some how (with some errors) and I'm not satisfy with the results that I'm
getting.
I would expect that on the second invite, when it sends from Kamailio to
the Gateway it would change to the host that the carrierroute assigned,
something like 456789(a)host5.dk-ws.
You can see some more info in the following lines.
I'm calling from a X-lite softphone to the number 456789, carrierroute
reads the prefix and base on the carrierroute db table assigns the host
that the call should be route to. In this case is host5.dk-ws
When I check # cat /var/log/messages I get the following errors and I
can't figure it out what they are:
Feb 18 12:38:59 localhost /sbin/kamailio[14687]:
INFO:carrierroute:cr_do_route: uri 456789 was rewritten to
sip:456789@host5.dk-ws
Feb 18 12:38:59 localhost /sbin/kamailio[14687]:
ERROR:tm:t_forward_nonack: no branch for forwarding
Feb 18 12:38:59 localhost /sbin/kamailio[14687]: ERROR:tm:w_t_relay:
t_forward_nonack failed
Feb 18 12:39:24 localhost kernel: device eth0 left promiscuous mode
Feb 18 12:39:28 localhost /sbin/kamailio[14695]:
CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for
code 408
[root@localhost kamailio]#
Also, if I do a # ngrep 456789 -qt
interface: eth0 (192.168.50.0/255.255.255.0)
match: 456789
U 2009/02/18 12:40:51.606688 192.168.50.118:8236 -> 192.168.50.93:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:51.611241 192.168.50.93:5060 -> 192.168.50.118:8236
SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.50.118:8236;
U 2009/02/18 12:40:51.612321 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
U 2009/02/18 12:40:51.617089 192.168.50.93:5060 -> 192.168.50.118:8236
SIP/2.0 500 Server error occurred (19/SL)..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:51.623317 192.168.50.118:8236 -> 192.168.50.93:5060
ACK sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP 192.168.50.118:8236;
U 2009/02/18 12:40:52.031349 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
U 2009/02/18 12:40:53.031163 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
And keeps sending invites...
I would expect that on the second invite, when it sends from 192.168.50.93
to 192.168.50.114 it would change to the host that the carrierroute
assigned, something like 456789(a)host5.dk-ws.
My question is if it should change it or is doing what it suppose to do?
All the ip addresses where I'm sending the traffic are fake, so it is not
a gateway to respond the call. Is that what is causing the errors and why
is not passing the call with the host address that cr assigned?
Cheers,
Juan.-
Hello,
your issue has nothing to do with a call stateful server.
You simply do record route in your config so for within dialog requests
the routing is don properly.
For initial invite, I assume (as I never used it) carrierroute sets the
R-URI so you just use:
t_relay("udp:asteriskip:asteriskport");
and in asterisk you dial the address in the r-uri.
If carrierroute is setting the dst_uri, then add it as special header,
use same to forward to asterisk, but there use the special header to
dial out.
Cheers,
Daniel
On 02/16/2009 02:54 PM, Juan Asencio wrote:
Hi list,
I need some explanation about using Kamailio as a Call Stateful Proxy.
I understand that Kamailio is a transaction stateful proxy. And I would
like to learn how I can achieve a Call Stateful Proxy using Kamailio.
I'm going to use Asterisk for transcoding to receive and send calls to
the
users and Kamailio should do the routing process using carrierroute,
deciding to which carrier the calls should be sent to.
Customer Carrier1
Group1 \ /
\ /
\ /
Customer ---->GW ---> Kamailio ---> GW ---> Carrier2
Group2 Asterisk Asterisk
/ \
/ \
Customer / Carrier3
Group3
Now it works with x-lite, if customer1 calls to certain number the call
is
send to the correspondent carrier base on the prefix number.
But when I add the GWs, how I can send the call to the GW IP address
(Asterisk) and then forward to the IP address of the carrier that I
assigned for that prefix number in my carrierroute table.
Can any of you could help me to clarify this or guide me about what to
read, so I could implement this.
Thanks in advance,
Juan.-
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
--
Daniel-Constantin Mierla
http://www.asipto.com
2:02 p.m.
New subject: [Kamailio-Users] Call Stateful Server
On Wednesday 18 February 2009, Juan Asencio wrote:
I add the t_relay(...) and now the call is sent to the
gateway. But I'm
getting some errors.
I will appreciate your advice:
I'm using it to route calls base on the prefix number. This seems to work
some how (with some errors) and I'm not satisfy with the results that I'm
getting.
I would expect that on the second invite, when it sends from Kamailio to
the Gateway it would change to the host that the carrierroute assigned,
something like 456789(a)host5.dk-ws.
You can see some more info in the following lines.
I'm calling from a X-lite softphone to the number 456789, carrierroute
reads the prefix and base on the carrierroute db table assigns the host
that the call should be route to. In this case is host5.dk-ws
When I check # cat /var/log/messages I get the following errors and I
can't figure it out what they are:
Feb 18 12:38:59 localhost /sbin/kamailio[14687]:
INFO:carrierroute:cr_do_route: uri 456789 was rewritten to
sip:456789@host5.dk-ws
Feb 18 12:38:59 localhost /sbin/kamailio[14687]:
ERROR:tm:t_forward_nonack: no branch for forwarding
Feb 18 12:38:59 localhost /sbin/kamailio[14687]: ERROR:tm:w_t_relay:
t_forward_nonack failed
Feb 18 12:39:24 localhost kernel: device eth0 left promiscuous mode
Feb 18 12:39:28 localhost /sbin/kamailio[14695]:
CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for
code 408
[root@localhost kamailio]#
Hi Juan,
it seems that it can't forward the message, and then it gets a 408.
Also, if I do a # ngrep 456789 -qt
interface: eth0 (192.168.50.0/255.255.255.0)
match: 456789
U 2009/02/18 12:40:51.606688 192.168.50.118:8236 -> 192.168.50.93:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:51.611241 192.168.50.93:5060 -> 192.168.50.118:8236
SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.50.118:8236;
U 2009/02/18 12:40:51.612321 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
U 2009/02/18 12:40:51.617089 192.168.50.93:5060 -> 192.168.50.118:8236
SIP/2.0 500 Server error occurred (19/SL)..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:51.623317 192.168.50.118:8236 -> 192.168.50.93:5060
ACK sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:52.031349 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
U 2009/02/18 12:40:53.031163 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
And keeps sending invites...
I would expect that on the second invite, when it sends from 192.168.50.93
to 192.168.50.114 it would change to the host that the carrierroute
assigned, something like 456789(a)host5.dk-ws.
My question is if it should change it or is doing what it suppose to do?
All the ip addresses where I'm sending the traffic are fake, so it is not
a gateway to respond the call. Is that what is causing the errors and why
is not passing the call with the host address that cr assigned?
yes, this is probably the problem. Kamailio needs to lookup the DNS names for
relaying the message to the appropriate target. If this fails, it can't
forward the message. You can use IP addresses in the cr table for testing,
though.
Cheers,
Henning
2:13 p.m.
New subject: [Kamailio-Users] Call Stateful Server
Thanks Henning,
Carrierroute is assigning the correct host address. So I guess that works,
but I'm worried about the invite, that is send with ip address of the
kamailio, instead of the one assigned by carrierroute.
Do you think the address would change if it could find host address
assigned by cr?
Cheers,
Juan.-
On Wednesday 18 February 2009, Juan Asencio wrote:
I add the t_relay(...) and now the call is sent
to the gateway. But I'm
getting some errors.
I will appreciate your advice:
I'm using it to route calls base on the prefix number. This seems to
work
some how (with some errors) and I'm not satisfy with the results that
I'm
getting.
I would expect that on the second invite, when it sends from Kamailio to
the Gateway it would change to the host that the carrierroute assigned,
something like 456789(a)host5.dk-ws.
You can see some more info in the following lines.
I'm calling from a X-lite softphone to the number 456789, carrierroute
reads the prefix and base on the carrierroute db table assigns the host
that the call should be route to. In this case is host5.dk-ws
When I check # cat /var/log/messages I get the following errors and I
can't figure it out what they are:
Feb 18 12:38:59 localhost /sbin/kamailio[14687]:
INFO:carrierroute:cr_do_route: uri 456789 was rewritten to
sip:456789@host5.dk-ws
Feb 18 12:38:59 localhost /sbin/kamailio[14687]:
ERROR:tm:t_forward_nonack: no branch for forwarding
Feb 18 12:38:59 localhost /sbin/kamailio[14687]: ERROR:tm:w_t_relay:
t_forward_nonack failed
Feb 18 12:39:24 localhost kernel: device eth0 left promiscuous mode
Feb 18 12:39:28 localhost /sbin/kamailio[14695]:
CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for
code 408
[root@localhost kamailio]#
Hi Juan,
it seems that it can't forward the message, and then it gets a 408.
Also, if I do a # ngrep 456789 -qt
interface: eth0 (192.168.50.0/255.255.255.0)
match: 456789
U 2009/02/18 12:40:51.606688 192.168.50.118:8236 -> 192.168.50.93:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:51.611241 192.168.50.93:5060 -> 192.168.50.118:8236
SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.50.118:8236;
U 2009/02/18 12:40:51.612321 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
U 2009/02/18 12:40:51.617089 192.168.50.93:5060 -> 192.168.50.118:8236
SIP/2.0 500 Server error occurred (19/SL)..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:51.623317 192.168.50.118:8236 -> 192.168.50.93:5060
ACK sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP
192.168.50.118:8236;
U 2009/02/18 12:40:52.031349 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
U 2009/02/18 12:40:53.031163 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
<sip:192.168.50.93;lr=on>..Via: SIP/2.0/UDP 192.168.50.93;
And keeps sending invites...
I would expect that on the second invite, when it sends from
192.168.50.93
to 192.168.50.114 it would change to the host that the carrierroute
assigned, something like 456789(a)host5.dk-ws.
My question is if it should change it or is doing what it suppose to do?
All the ip addresses where I'm sending the traffic are fake, so it is
not
a gateway to respond the call. Is that what is causing the errors and
why
is not passing the call with the host address that cr assigned?
yes, this is probably the problem. Kamailio needs to lookup the DNS names
for
relaying the message to the appropriate target. If this fails, it can't
forward the message. You can use IP addresses in the cr table for testing,
though.
Cheers,
Henning
2:27 p.m.
New subject: [Kamailio-Users] Call Stateful Server
On Wednesday 18 February 2009, Juan Asencio wrote:
Carrierroute is assigning the correct host address. So
I guess that works,
but I'm worried about the invite, that is send with ip address of the
kamailio, instead of the one assigned by carrierroute.
Do you think the address would change if it could find host address
assigned by cr?
Hi Juan,
> U 2009/02/18 12:40:51.606688 192.168.50.118:8236
-> 192.168.50.93:5060
> INVITE sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP
> 192.168.50.118:8236;
>
> U 2009/02/18 12:40:51.611241 192.168.50.93:5060 -> 192.168.50.118:8236
> SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.50.118:8236;
>
> U 2009/02/18 12:40:51.612321 192.168.50.93:5060 -> 192.158.50.114:5060
> INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
.118 is the address of your test box, .114 the kamailio and .93 the gateway?
Yes, cr should rewrite the request uri to the one in its table. I just tested
with a non existing host name on my system (test.box):
Feb 18 13:26:30 ca ../kamailio[2430]: INFO:carrierroute:cr_do_route: uri
49721123456787 was rewritten to sip:49721123456787@test.box:7000, carrier 0,
domain 10
Feb 18 13:26:30 ca ../kamailio[2430]: CRITICAL:core:mk_proxy: could not
resolve hostname: "test.box"
Feb 18 13:26:30 ca ../kamailio[2430]: ERROR:tm:uri2proxy: bad host name in URI
<sip:49721123456787@test.box:7000>
Feb 18 13:26:30 ca ../kamailio[2430]: ERROR:tm:t_forward_nonack: failure to
add branches
Cheers,
Henning
2:51 p.m.
New subject: [Kamailio-Users] Call Stateful Server
Hi Henning, you are right those are the right addresses.
If I compare with your message, so cr rewrites the uri.
It is when I do a ngrep that it keeps using the kamailio ip address.
I will try to send it to a working gateway and see what happens.
:) thanks for your help.
Juan.-
Feb 18 12:40:51 localhost /sbin/kamailio[14765]:
INFO:carrierroute:cr_do_route: uri 456789 was rewritten to
sip:456789@host5.dk-ws
Feb 18 12:40:51 localhost /sbin/kamailio[14765]:
ERROR:tm:t_forward_nonack: no branch for forwarding
Feb 18 12:40:51 localhost /sbin/kamailio[14765]: ERROR:tm:w_t_relay:
t_forward_nonack failed
Feb 18 12:41:20 localhost /sbin/kamailio[14769]:
CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for
code 408
On Wednesday 18 February 2009, Juan Asencio wrote:
Carrierroute is assigning the correct host
address. So I guess that
works,
but I'm worried about the invite, that is send with ip address of the
kamailio, instead of the one assigned by carrierroute.
Do you think the address would change if it could find host address
assigned by cr?
Hi Juan,
> U 2009/02/18 12:40:51.606688
192.168.50.118:8236 -> 192.168.50.93:5060
> INVITE sip:456789@192.168.50.93 SIP/2.0..Via: SIP/2.0/UDP
> 192.168.50.118:8236;
>
> U 2009/02/18 12:40:51.611241 192.168.50.93:5060 -> 192.168.50.118:8236
> SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.50.118:8236;
>
> U 2009/02/18 12:40:51.612321 192.168.50.93:5060 -> 192.158.50.114:5060
> INVITE sip:456789@192.168.50.93 SIP/2.0..Record-Route:
.118 is the address of your test box, .114 the kamailio and .93 the
gateway?
Yes, cr should rewrite the request uri to the one in its table. I just
tested
with a non existing host name on my system (test.box):
Feb 18 13:26:30 ca ../kamailio[2430]: INFO:carrierroute:cr_do_route: uri
49721123456787 was rewritten to sip:49721123456787@test.box:7000, carrier
0,
domain 10
Feb 18 13:26:30 ca ../kamailio[2430]: CRITICAL:core:mk_proxy: could not
resolve hostname: "test.box"
Feb 18 13:26:30 ca ../kamailio[2430]: ERROR:tm:uri2proxy: bad host name in
URI
<sip:49721123456787@test.box:7000>
Feb 18 13:26:30 ca ../kamailio[2430]: ERROR:tm:t_forward_nonack: failure
to
add branches
Cheers,
Henning
20 Feb
20 Feb
3:18 p.m.
New subject: [Kamailio-Users] add uri as a special header
Hi Daniel,
I tried to add the new uri from carrierroute as a special header.
But I can't figure it out how to do it. Do you have an example or can you
point me where I could find the information about it?
Now I forward the invite to the gateway IP address, but I'm not sure if
the new assigned uri by cr is on the header. How can I see that?
The address 50.118 is my x-lite, 50.93 is Kamailio and 50.114 is my gateway
U 192.168.50.118:60560 -> 192.168.50.93:5060
INVITE sip:456789@192.168.50.93 SIP/2.0.
Via: SIP/2.0/UDP
192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport.
Max-Forwards: 70.
Contact: <sip:1002@192.168.50.118:60560>.
To: "456789"<sip:456789@192.168.50.93>.
From: "1002"<sip:1002@192.168.50.93>;tag=2a692168.
U 192.168.50.93:5060 -> 192.168.50.118:60560
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport=60560.
To: "456789"<sip:456789@192.168.50.93>.
From: "1002"<sip:1002@192.168.50.93>;tag=2a692168.
U 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0.
Via: SIP/2.0/UDP 192.168.50.93;branch=z9hG4bK81cd.0e7f7944.0.
Via: SIP/2.0/UDP
192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport=60560.
Max-Forwards: 69.
Contact: <sip:1002@192.168.50.118:60560>.
To: "456789"<sip:456789@192.168.50.93>.
From: "1002"<sip:1002@192.168.50.93>;tag=2a692168.
This is my log:
Feb 20 14:47:57 localhost /sbin/kamailio[18371]: time [Fri Feb 20 14:47:57
2009] method (INVITE) r-uri (sip:456789@192.168.50.93) 2nd via (<null>)
Feb 20 14:47:57 localhost /sbin/kamailio[18371]:
INFO:carrierroute:cr_do_route: uri 456789 was rewritten to
sip:456789@host5.dk-ws
Feb 20 14:47:57 localhost /sbin/kamailio[18371]:
ERROR:tm:t_forward_nonack: no branch for forwarding
Feb 20 14:47:57 localhost /sbin/kamailio[18371]: ERROR:tm:w_t_relay:
t_forward_nonack failed
Feb 20 14:47:57 localhost /sbin/kamailio[18371]: time [Fri Feb 20 14:47:57
2009] method (INVITE) r-uri (sip:456789@host5.dk-ws) 2nd via (<null>)
Feb 20 14:48:27 localhost /sbin/kamailio[18377]:
CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for
code 408
Cheers,
Juan.-
Hello,
your issue has nothing to do with a call stateful server.
You simply do record route in your config so for within dialog requests
the routing is don properly.
For initial invite, I assume (as I never used it) carrierroute sets the
R-URI so you just use:
t_relay("udp:asteriskip:asteriskport");
and in asterisk you dial the address in the r-uri.
If carrierroute is setting the dst_uri, then add it as special header,
use same to forward to asterisk, but there use the special header to
dial out.
Cheers,
Daniel
5:34 p.m.
New subject: [Kamailio-Users] add uri as a special header
Hello,
On 02/20/2009 03:18 PM, Juan Asencio wrote:
Hi Daniel,
I tried to add the new uri from carrierroute as a special header.
But I can't figure it out how to do it. Do you have an example or can you
point me where I could find the information about it?
append_hf("My-Hdr: $ru\n");
Cheers,
Daniel
Now I forward the invite to the gateway IP address,
but I'm not sure if
the new assigned uri by cr is on the header. How can I see that?
The address 50.118 is my x-lite, 50.93 is Kamailio and 50.114 is my gateway
U 192.168.50.118:60560 -> 192.168.50.93:5060
INVITE sip:456789@192.168.50.93 SIP/2.0.
Via: SIP/2.0/UDP
192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport.
Max-Forwards: 70.
Contact: <sip:1002@192.168.50.118:60560>.
To: "456789"<sip:456789@192.168.50.93>.
From: "1002"<sip:1002@192.168.50.93>;tag=2a692168.
U 192.168.50.93:5060 -> 192.168.50.118:60560
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport=60560.
To: "456789"<sip:456789@192.168.50.93>.
From: "1002"<sip:1002@192.168.50.93>;tag=2a692168.
U 192.168.50.93:5060 -> 192.158.50.114:5060
INVITE sip:456789@192.168.50.93 SIP/2.0.
Via: SIP/2.0/UDP 192.168.50.93;branch=z9hG4bK81cd.0e7f7944.0.
Via: SIP/2.0/UDP
192.168.50.118:60560;branch=z9hG4bK-d8754z-0924d953ee50f428-1---d8754z-;rport=60560.
Max-Forwards: 69.
Contact: <sip:1002@192.168.50.118:60560>.
To: "456789"<sip:456789@192.168.50.93>.
From: "1002"<sip:1002@192.168.50.93>;tag=2a692168.
This is my log:
Feb 20 14:47:57 localhost /sbin/kamailio[18371]: time [Fri Feb 20 14:47:57
2009] method (INVITE) r-uri (sip:456789@192.168.50.93) 2nd via (<null>)
Feb 20 14:47:57 localhost /sbin/kamailio[18371]:
INFO:carrierroute:cr_do_route: uri 456789 was rewritten to
sip:456789@host5.dk-ws
Feb 20 14:47:57 localhost /sbin/kamailio[18371]:
ERROR:tm:t_forward_nonack: no branch for forwarding
Feb 20 14:47:57 localhost /sbin/kamailio[18371]: ERROR:tm:w_t_relay:
t_forward_nonack failed
Feb 20 14:47:57 localhost /sbin/kamailio[18371]: time [Fri Feb 20 14:47:57
2009] method (INVITE) r-uri (sip:456789@host5.dk-ws) 2nd via (<null>)
Feb 20 14:48:27 localhost /sbin/kamailio[18377]:
CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for
code 408
Cheers,
Juan.-
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello,
your issue has nothing to do with a call stateful server.
You simply do record route in your config so for within dialog requests
the routing is don properly.
For initial invite, I assume (as I never used it) carrierroute sets the
R-URI so you just use:
t_relay("udp:asteriskip:asteriskport");
and in asterisk you dial the address in the r-uri.
If carrierroute is setting the dst_uri, then add it as special header,
use same to forward to asterisk, but there use the special header to
dial out.
Cheers,
Daniel
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13 comments
5 participants
participants (5)
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Alex Balashov -
Daniel-Constantin Mierla -
Henning Westerholt -
Juan Asencio -
Ovidiu Sas