I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD
I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc.
After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch.
I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other.
Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.
Hi,
You need to handle in dialog routing - check one of the configs that ships with kamailio. Right now Kamailio forwards all SIP packets to freeswitch, even the ones that freeswitch sends to Kamailio.
/Morten
2011/10/5 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD
I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc.
After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch.
I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other.
Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Morten.
Do you mean anything specific in the standard config: http://pastebin.com/Aj4mHAJq
Because that handles registrations, subscriber list etc. etc... I'm only interested in Kamailio as a dispatcher.
And I've already tried adding the PATH module with the use_received parameter and add_path() and add_path_received() functions. That didn't help.
On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk wrote:
Hi,
You need to handle in dialog routing - check one of the configs that ships with kamailio. Right now Kamailio forwards all SIP packets to freeswitch, even the ones that freeswitch sends to Kamailio.
/Morten
2011/10/5 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
I have a setup with Kamailio as dispatcher in front of a FreeSwitch
server.
This is my kamailio.cfg: http://pastebin.com/8PR2GFBD
I'm currently getting "Too many hops" when calling between SIP clients. I
am
able to call to FreeSwitch and listen to voicemail, hold music etc.
After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch.
I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other.
Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any
luck.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
This part
# handle requests within SIP dialogs route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
Do you mean anything specific in the standard config: http://pastebin.com/Aj4mHAJq
Because that handles registrations, subscriber list etc. etc... I'm only interested in Kamailio as a dispatcher.
And I've already tried adding the PATH module with the use_received parameter and add_path() and add_path_received() functions. That didn't help.
On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk wrote:
Hi,
You need to handle in dialog routing - check one of the configs that ships with kamailio. Right now Kamailio forwards all SIP packets to freeswitch, even the ones that freeswitch sends to Kamailio.
/Morten
2011/10/5 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD
I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc.
After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch.
I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other.
Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Morten.
I've tried to add that part: http://pastebin.com/MmKnbKLz
But now it won't even register. Do you know any config-example for a working dispatcher for Kamailio?
On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk wrote:
This part
# handle requests within SIP dialogs route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
Do you mean anything specific in the standard config: http://pastebin.com/Aj4mHAJq
Because that handles registrations, subscriber list etc. etc... I'm only interested in Kamailio as a dispatcher.
And I've already tried adding the PATH module with the use_received parameter and add_path() and add_path_received() functions. That didn't help.
On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk wrote:
Hi,
You need to handle in dialog routing - check one of the configs that ships with kamailio. Right now Kamailio forwards all SIP packets to freeswitch, even the ones that freeswitch sends to Kamailio.
/Morten
2011/10/5 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD
I'm currently getting "Too many hops" when calling between SIP
clients.
I am able to call to FreeSwitch and listen to voicemail, hold music etc.
After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch.
I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other.
Can anyone help me out? I've tried to Google a lot for this problem
and
asked in several IRC channels, mailing lists and forums. Without any luck.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
I've tried to add that part: http://pastebin.com/MmKnbKLz
But now it won't even register. Do you know any config-example for a working dispatcher for Kamailio?
On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk wrote:
This part
# handle requests within SIP dialogs route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
Do you mean anything specific in the standard config: http://pastebin.com/Aj4mHAJq
Because that handles registrations, subscriber list etc. etc... I'm only interested in Kamailio as a dispatcher.
And I've already tried adding the PATH module with the use_received parameter and add_path() and add_path_received() functions. That didn't help.
On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk wrote:
Hi,
You need to handle in dialog routing - check one of the configs that ships with kamailio. Right now Kamailio forwards all SIP packets to freeswitch, even the ones that freeswitch sends to Kamailio.
/Morten
2011/10/5 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD
I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc.
After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch.
I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other.
Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk wrote:
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
I've tried to add that part: http://pastebin.com/MmKnbKLz
But now it won't even register. Do you know any config-example for a
working
dispatcher for Kamailio?
On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk wrote:
This part
# handle requests within SIP dialogs route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
Do you mean anything specific in the standard config: http://pastebin.com/Aj4mHAJq
Because that handles registrations, subscriber list etc. etc... I'm
only
interested in Kamailio as a dispatcher.
And I've already tried adding the PATH module with the use_received parameter and add_path() and add_path_received() functions. That
didn't
help.
On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk
wrote:
Hi,
You need to handle in dialog routing - check one of the configs that ships with kamailio. Right now Kamailio forwards all SIP packets to freeswitch, even the ones that freeswitch sends to Kamailio.
/Morten
2011/10/5 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD
I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc.
After a long conversation with a FreeSwitch expert, and some tests,
I
was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch.
I've also run tshark on both FreeSwitch and Kamailio and when
calling
between clients they just send the packets between each other.
Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without
any
luck.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Morten.
I've tested it a lot know, your latest config-example. At it actually works when I connect 2 devices, 1 iPhone and 1 Android. But when connecting 1 phone and my laptop with SFLPhone or Linphone I cannot call the laptop. Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk wrote:
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
I've tried to add that part: http://pastebin.com/MmKnbKLz
But now it won't even register. Do you know any config-example for a
working
dispatcher for Kamailio?
On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk wrote:
This part
# handle requests within SIP dialogs route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
Do you mean anything specific in the standard config: http://pastebin.com/Aj4mHAJq
Because that handles registrations, subscriber list etc. etc... I'm
only
interested in Kamailio as a dispatcher.
And I've already tried adding the PATH module with the use_received parameter and add_path() and add_path_received() functions. That
didn't
help.
On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk
wrote:
Hi,
You need to handle in dialog routing - check one of the configs that ships with kamailio. Right now Kamailio forwards all SIP packets to freeswitch, even the ones that freeswitch sends to Kamailio.
/Morten
2011/10/5 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
:
> I have a setup with Kamailio as dispatcher in front of a
FreeSwitch
> server. > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD > > I'm currently getting "Too many hops" when calling between SIP > clients. > I am > able to call to FreeSwitch and listen to voicemail, hold music
etc.
> > After a long conversation with a FreeSwitch expert, and some
tests, I
> was > told that Kamailio delivers the wrong IP (NAT problems) to > FreeSwitch. > > I've also run tshark on both FreeSwitch and Kamailio and when
calling
> between clients they just send the packets between each other. > > Can anyone help me out? I've tried to Google a lot for this
problem
> and > asked in several IRC channels, mailing lists and forums. Without
any
> luck. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening to voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is wrong with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've tested it a lot know, your latest config-example. At it actually works when I connect 2 devices, 1 iPhone and 1 Android. But when connecting 1 phone and my laptop with SFLPhone or Linphone I cannot call the laptop. Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk wrote:
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
I've tried to add that part: http://pastebin.com/MmKnbKLz
But now it won't even register. Do you know any config-example for a
working
dispatcher for Kamailio?
On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk
wrote:
This part
# handle requests within SIP dialogs route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
Do you mean anything specific in the standard config: http://pastebin.com/Aj4mHAJq
Because that handles registrations, subscriber list etc. etc... I'm
only
interested in Kamailio as a dispatcher.
And I've already tried adding the PATH module with the use_received parameter and add_path() and add_path_received() functions. That
didn't
help.
On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk
wrote:
> > Hi, > > You need to handle in dialog routing - check one of the configs
that
> ships with kamailio. Right now Kamailio forwards all SIP packets to > freeswitch, even the ones that freeswitch sends to Kamailio. > > /Morten > > 2011/10/5 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
:
> > I have a setup with Kamailio as dispatcher in front of a
FreeSwitch
> > server. > > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD > > > > I'm currently getting "Too many hops" when calling between SIP > > clients. > > I am > > able to call to FreeSwitch and listen to voicemail, hold music
etc.
> > > > After a long conversation with a FreeSwitch expert, and some
tests, I
> > was > > told that Kamailio delivers the wrong IP (NAT problems) to > > FreeSwitch. > > > > I've also run tshark on both FreeSwitch and Kamailio and when
calling
> > between clients they just send the packets between each other. > > > > Can anyone help me out? I've tried to Google a lot for this
problem
> > and > > asked in several IRC channels, mailing lists and forums. Without
any
> > luck. > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
> > list > > sr-users@lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > Morten Isaksen > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
> sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening to voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is wrong with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've tested it a lot know, your latest config-example. At it actually works when I connect 2 devices, 1 iPhone and 1 Android. But when connecting 1 phone and my laptop with SFLPhone or Linphone I cannot call the laptop. Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk wrote:
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
I've tried to add that part: http://pastebin.com/MmKnbKLz
But now it won't even register. Do you know any config-example for a working dispatcher for Kamailio?
On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk wrote:
This part
# handle requests within SIP dialogs route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com: > Hi Morten. > > Do you mean anything specific in the standard config: > http://pastebin.com/Aj4mHAJq > > Because that handles registrations, subscriber list etc. etc... I'm > only > interested in Kamailio as a dispatcher. > > And I've already tried adding the PATH module with the use_received > parameter and add_path() and add_path_received() functions. That > didn't > help. > > On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk > wrote: >> >> Hi, >> >> You need to handle in dialog routing - check one of the configs >> that >> ships with kamailio. Right now Kamailio forwards all SIP packets >> to >> freeswitch, even the ones that freeswitch sends to Kamailio. >> >> /Morten >> >> 2011/10/5 Henrik Aagaard Sørensen >> henrikaagaardsorensen@gmail.com: >> > I have a setup with Kamailio as dispatcher in front of a >> > FreeSwitch >> > server. >> > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >> > >> > I'm currently getting "Too many hops" when calling between SIP >> > clients. >> > I am >> > able to call to FreeSwitch and listen to voicemail, hold music >> > etc. >> > >> > After a long conversation with a FreeSwitch expert, and some >> > tests, I >> > was >> > told that Kamailio delivers the wrong IP (NAT problems) to >> > FreeSwitch. >> > >> > I've also run tshark on both FreeSwitch and Kamailio and when >> > calling >> > between clients they just send the packets between each other. >> > >> > Can anyone help me out? I've tried to Google a lot for this >> > problem >> > and >> > asked in several IRC channels, mailing lists and forums. Without >> > any >> > luck. >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> > mailing >> > list >> > sr-users@lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Morten Isaksen >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them
produce
the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening to voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is wrong with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without
Kamailio,
makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've tested it a lot know, your latest config-example. At it actually works when I connect 2 devices, 1 iPhone and 1 Android. But when
connecting
1 phone and my laptop with SFLPhone or Linphone I cannot call the
laptop.
Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk
wrote:
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Hi Morten.
I've tried to add that part: http://pastebin.com/MmKnbKLz
But now it won't even register. Do you know any config-example for a working dispatcher for Kamailio?
On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk wrote: > > This part > > # handle requests within SIP dialogs > route(WITHINDLG); > > 2011/10/6 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
:
> > Hi Morten. > > > > Do you mean anything specific in the standard config: > > http://pastebin.com/Aj4mHAJq > > > > Because that handles registrations, subscriber list etc. etc...
I'm
> > only > > interested in Kamailio as a dispatcher. > > > > And I've already tried adding the PATH module with the
use_received
> > parameter and add_path() and add_path_received() functions. That > > didn't > > help. > > > > On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk > > wrote: > >> > >> Hi, > >> > >> You need to handle in dialog routing - check one of the configs > >> that > >> ships with kamailio. Right now Kamailio forwards all SIP packets > >> to > >> freeswitch, even the ones that freeswitch sends to Kamailio. > >> > >> /Morten > >> > >> 2011/10/5 Henrik Aagaard Sørensen > >> henrikaagaardsorensen@gmail.com: > >> > I have a setup with Kamailio as dispatcher in front of a > >> > FreeSwitch > >> > server. > >> > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD > >> > > >> > I'm currently getting "Too many hops" when calling between SIP > >> > clients. > >> > I am > >> > able to call to FreeSwitch and listen to voicemail, hold music > >> > etc. > >> > > >> > After a long conversation with a FreeSwitch expert, and some > >> > tests, I > >> > was > >> > told that Kamailio delivers the wrong IP (NAT problems) to > >> > FreeSwitch. > >> > > >> > I've also run tshark on both FreeSwitch and Kamailio and when > >> > calling > >> > between clients they just send the packets between each other. > >> > > >> > Can anyone help me out? I've tried to Google a lot for this > >> > problem > >> > and > >> > asked in several IRC channels, mailing lists and forums.
Without
> >> > any > >> > luck. > >> > > >> > _______________________________________________ > >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > >> > mailing > >> > list > >> > sr-users@lists.sip-router.org > >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > > >> > > >> > >> > >> > >> -- > >> Morten Isaksen > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
> >> list > >> sr-users@lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
> > list > > sr-users@lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > Morten Isaksen > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening to voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is wrong with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've tested it a lot know, your latest config-example. At it actually works when I connect 2 devices, 1 iPhone and 1 Android. But when connecting 1 phone and my laptop with SFLPhone or Linphone I cannot call the laptop. Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk wrote:
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com: > Hi Morten. > > I've tried to add that part: http://pastebin.com/MmKnbKLz > > But now it won't even register. Do you know any config-example for > a > working > dispatcher for Kamailio? > > On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk > wrote: >> >> This part >> >> # handle requests within SIP dialogs >> route(WITHINDLG); >> >> 2011/10/6 Henrik Aagaard Sørensen >> henrikaagaardsorensen@gmail.com: >> > Hi Morten. >> > >> > Do you mean anything specific in the standard config: >> > http://pastebin.com/Aj4mHAJq >> > >> > Because that handles registrations, subscriber list etc. etc... >> > I'm >> > only >> > interested in Kamailio as a dispatcher. >> > >> > And I've already tried adding the PATH module with the >> > use_received >> > parameter and add_path() and add_path_received() functions. That >> > didn't >> > help. >> > >> > On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen misak@misak.dk >> > wrote: >> >> >> >> Hi, >> >> >> >> You need to handle in dialog routing - check one of the configs >> >> that >> >> ships with kamailio. Right now Kamailio forwards all SIP >> >> packets >> >> to >> >> freeswitch, even the ones that freeswitch sends to Kamailio. >> >> >> >> /Morten >> >> >> >> 2011/10/5 Henrik Aagaard Sørensen >> >> henrikaagaardsorensen@gmail.com: >> >> > I have a setup with Kamailio as dispatcher in front of a >> >> > FreeSwitch >> >> > server. >> >> > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >> >> > >> >> > I'm currently getting "Too many hops" when calling between >> >> > SIP >> >> > clients. >> >> > I am >> >> > able to call to FreeSwitch and listen to voicemail, hold >> >> > music >> >> > etc. >> >> > >> >> > After a long conversation with a FreeSwitch expert, and some >> >> > tests, I >> >> > was >> >> > told that Kamailio delivers the wrong IP (NAT problems) to >> >> > FreeSwitch. >> >> > >> >> > I've also run tshark on both FreeSwitch and Kamailio and when >> >> > calling >> >> > between clients they just send the packets between each >> >> > other. >> >> > >> >> > Can anyone help me out? I've tried to Google a lot for this >> >> > problem >> >> > and >> >> > asked in several IRC channels, mailing lists and forums. >> >> > Without >> >> > any >> >> > luck. >> >> > >> >> > _______________________________________________ >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> > mailing >> >> > list >> >> > sr-users@lists.sip-router.org >> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Morten Isaksen >> >> >> >> _______________________________________________ >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> mailing >> >> list >> >> sr-users@lists.sip-router.org >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> > mailing >> > list >> > sr-users@lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Morten Isaksen >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
When trying to dial 101 this is a tshark output on the Kamailio:
0.000000 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:101@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Temporarily Unavailable 0.447130 95.214.24.165 -> 215.183.255.142 SIP Request: ACK sip:101@sip.my-domain.com
On Sun, Oct 9, 2011 at 12:14 PM, Morten Isaksen misak@misak.dk wrote:
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening
to
voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is
wrong
with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've tested it a lot know, your latest config-example. At it actually works when I connect 2 devices, 1 iPhone and 1 Android. But when connecting 1 phone and my laptop with SFLPhone or Linphone I cannot call the laptop. Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk wrote: > > Try this one http://pastebin.com/mahKECAw > > /Morten > > 2011/10/6 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
:
> > Hi Morten. > > > > I've tried to add that part: http://pastebin.com/MmKnbKLz > > > > But now it won't even register. Do you know any config-example
for
> > a > > working > > dispatcher for Kamailio? > > > > On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk > > wrote: > >> > >> This part > >> > >> # handle requests within SIP dialogs > >> route(WITHINDLG); > >> > >> 2011/10/6 Henrik Aagaard Sørensen > >> henrikaagaardsorensen@gmail.com: > >> > Hi Morten. > >> > > >> > Do you mean anything specific in the standard config: > >> > http://pastebin.com/Aj4mHAJq > >> > > >> > Because that handles registrations, subscriber list etc.
etc...
> >> > I'm > >> > only > >> > interested in Kamailio as a dispatcher. > >> > > >> > And I've already tried adding the PATH module with the > >> > use_received > >> > parameter and add_path() and add_path_received() functions.
That
> >> > didn't > >> > help. > >> > > >> > On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen <
misak@misak.dk>
> >> > wrote: > >> >> > >> >> Hi, > >> >> > >> >> You need to handle in dialog routing - check one of the
configs
> >> >> that > >> >> ships with kamailio. Right now Kamailio forwards all SIP > >> >> packets > >> >> to > >> >> freeswitch, even the ones that freeswitch sends to Kamailio. > >> >> > >> >> /Morten > >> >> > >> >> 2011/10/5 Henrik Aagaard Sørensen > >> >> henrikaagaardsorensen@gmail.com: > >> >> > I have a setup with Kamailio as dispatcher in front of a > >> >> > FreeSwitch > >> >> > server. > >> >> > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD > >> >> > > >> >> > I'm currently getting "Too many hops" when calling between > >> >> > SIP > >> >> > clients. > >> >> > I am > >> >> > able to call to FreeSwitch and listen to voicemail, hold > >> >> > music > >> >> > etc. > >> >> > > >> >> > After a long conversation with a FreeSwitch expert, and
some
> >> >> > tests, I > >> >> > was > >> >> > told that Kamailio delivers the wrong IP (NAT problems) to > >> >> > FreeSwitch. > >> >> > > >> >> > I've also run tshark on both FreeSwitch and Kamailio and
when
> >> >> > calling > >> >> > between clients they just send the packets between each > >> >> > other. > >> >> > > >> >> > Can anyone help me out? I've tried to Google a lot for this > >> >> > problem > >> >> > and > >> >> > asked in several IRC channels, mailing lists and forums. > >> >> > Without > >> >> > any > >> >> > luck. > >> >> > > >> >> > _______________________________________________ > >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > >> >> > mailing > >> >> > list > >> >> > sr-users@lists.sip-router.org > >> >> >
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> >> > > >> >> > > >> >> > >> >> > >> >> > >> >> -- > >> >> Morten Isaksen > >> >> > >> >> _______________________________________________ > >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > >> >> mailing > >> >> list > >> >> sr-users@lists.sip-router.org > >> >>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> > > >> > > >> > _______________________________________________ > >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > >> > mailing > >> > list > >> > sr-users@lists.sip-router.org > >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > > >> > > >> > >> > >> > >> -- > >> Morten Isaksen > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
> >> list > >> sr-users@lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
> > list > > sr-users@lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > Morten Isaksen > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Sorry for jumping in, it seems to me that its Domain name issue. are you sure sip.my-domain.com resolves to your Kamailio Server. Is this domain added in domain table and in SIP_DOMAIN env variable !!?
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
When trying to dial 101 this is a tshark output on the Kamailio:
0.000000 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:101@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Temporarily Unavailable 0.447130 95.214.24.165 -> 215.183.255.142 SIP Request: ACK sip:101@sip.my-domain.com
On Sun, Oct 9, 2011 at 12:14 PM, Morten Isaksen misak@misak.dk wrote:
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening
to
voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is
wrong
with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've tested it a lot know, your latest config-example. At it
actually
works when I connect 2 devices, 1 iPhone and 1 Android. But when connecting 1 phone and my laptop with SFLPhone or Linphone I cannot call the laptop. Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com > > Still getting "Too Many Hops" :( > > On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk > wrote: >> >> Try this one http://pastebin.com/mahKECAw >> >> /Morten >> >> 2011/10/6 Henrik Aagaard Sørensen <
henrikaagaardsorensen@gmail.com>:
>> > Hi Morten. >> > >> > I've tried to add that part: http://pastebin.com/MmKnbKLz >> > >> > But now it won't even register. Do you know any config-example
for
>> > a >> > working >> > dispatcher for Kamailio? >> > >> > On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen <misak@misak.dk
>> > wrote: >> >> >> >> This part >> >> >> >> # handle requests within SIP dialogs >> >> route(WITHINDLG); >> >> >> >> 2011/10/6 Henrik Aagaard Sørensen >> >> henrikaagaardsorensen@gmail.com: >> >> > Hi Morten. >> >> > >> >> > Do you mean anything specific in the standard config: >> >> > http://pastebin.com/Aj4mHAJq >> >> > >> >> > Because that handles registrations, subscriber list etc.
etc...
>> >> > I'm >> >> > only >> >> > interested in Kamailio as a dispatcher. >> >> > >> >> > And I've already tried adding the PATH module with the >> >> > use_received >> >> > parameter and add_path() and add_path_received() functions.
That
>> >> > didn't >> >> > help. >> >> > >> >> > On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen <
misak@misak.dk>
>> >> > wrote: >> >> >> >> >> >> Hi, >> >> >> >> >> >> You need to handle in dialog routing - check one of the
configs
>> >> >> that >> >> >> ships with kamailio. Right now Kamailio forwards all SIP >> >> >> packets >> >> >> to >> >> >> freeswitch, even the ones that freeswitch sends to Kamailio. >> >> >> >> >> >> /Morten >> >> >> >> >> >> 2011/10/5 Henrik Aagaard Sørensen >> >> >> henrikaagaardsorensen@gmail.com: >> >> >> > I have a setup with Kamailio as dispatcher in front of a >> >> >> > FreeSwitch >> >> >> > server. >> >> >> > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >> >> >> > >> >> >> > I'm currently getting "Too many hops" when calling between >> >> >> > SIP >> >> >> > clients. >> >> >> > I am >> >> >> > able to call to FreeSwitch and listen to voicemail, hold >> >> >> > music >> >> >> > etc. >> >> >> > >> >> >> > After a long conversation with a FreeSwitch expert, and
some
>> >> >> > tests, I >> >> >> > was >> >> >> > told that Kamailio delivers the wrong IP (NAT problems) to >> >> >> > FreeSwitch. >> >> >> > >> >> >> > I've also run tshark on both FreeSwitch and Kamailio and
when
>> >> >> > calling >> >> >> > between clients they just send the packets between each >> >> >> > other. >> >> >> > >> >> >> > Can anyone help me out? I've tried to Google a lot for
this
>> >> >> > problem >> >> >> > and >> >> >> > asked in several IRC channels, mailing lists and forums. >> >> >> > Without >> >> >> > any >> >> >> > luck. >> >> >> > >> >> >> > _______________________________________________ >> >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> >> > mailing >> >> >> > list >> >> >> > sr-users@lists.sip-router.org >> >> >> >
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >> >> > >> >> >> > >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Morten Isaksen >> >> >> >> >> >> _______________________________________________ >> >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> >> mailing >> >> >> list >> >> >> sr-users@lists.sip-router.org >> >> >>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >> > >> >> > >> >> > _______________________________________________ >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> > mailing >> >> > list >> >> > sr-users@lists.sip-router.org >> >> >
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Morten Isaksen >> >> >> >> _______________________________________________ >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
>> >> list >> >> sr-users@lists.sip-router.org >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
>> > list >> > sr-users@lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Morten Isaksen >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Sammy.
SIP_DOMAIN is already set in kamctlrc to sip.my-domain.com.
On Mon, Oct 10, 2011 at 10:00 AM, Sammy Govind govoiper@gmail.com wrote:
Sorry for jumping in, it seems to me that its Domain name issue. are you sure sip.my-domain.com resolves to your Kamailio Server. Is this domain added in domain table and in SIP_DOMAIN env variable !!?
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
When trying to dial 101 this is a tshark output on the Kamailio:
0.000000 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:101@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Temporarily Unavailable 0.447130 95.214.24.165 -> 215.183.255.142 SIP Request: ACK sip:101@sip.my-domain.com
On Sun, Oct 9, 2011 at 12:14 PM, Morten Isaksen misak@misak.dk wrote:
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk
wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for
debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for
listening to
voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is
wrong
with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com > > Hi Morten. > > I've tested it a lot know, your latest config-example. At it
actually
> works when I connect 2 devices, 1 iPhone and 1 Android. But when > connecting > 1 phone and my laptop with SFLPhone or Linphone I cannot call the > laptop. > Does that make any sense? > > 2011/10/6 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
>> >> Still getting "Too Many Hops" :( >> >> On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk >> wrote: >>> >>> Try this one http://pastebin.com/mahKECAw >>> >>> /Morten >>> >>> 2011/10/6 Henrik Aagaard Sørensen <
henrikaagaardsorensen@gmail.com>:
>>> > Hi Morten. >>> > >>> > I've tried to add that part: http://pastebin.com/MmKnbKLz >>> > >>> > But now it won't even register. Do you know any config-example
for
>>> > a >>> > working >>> > dispatcher for Kamailio? >>> > >>> > On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen <
misak@misak.dk>
>>> > wrote: >>> >> >>> >> This part >>> >> >>> >> # handle requests within SIP dialogs >>> >> route(WITHINDLG); >>> >> >>> >> 2011/10/6 Henrik Aagaard Sørensen >>> >> henrikaagaardsorensen@gmail.com: >>> >> > Hi Morten. >>> >> > >>> >> > Do you mean anything specific in the standard config: >>> >> > http://pastebin.com/Aj4mHAJq >>> >> > >>> >> > Because that handles registrations, subscriber list etc.
etc...
>>> >> > I'm >>> >> > only >>> >> > interested in Kamailio as a dispatcher. >>> >> > >>> >> > And I've already tried adding the PATH module with the >>> >> > use_received >>> >> > parameter and add_path() and add_path_received() functions.
That
>>> >> > didn't >>> >> > help. >>> >> > >>> >> > On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen <
misak@misak.dk>
>>> >> > wrote: >>> >> >> >>> >> >> Hi, >>> >> >> >>> >> >> You need to handle in dialog routing - check one of the
configs
>>> >> >> that >>> >> >> ships with kamailio. Right now Kamailio forwards all SIP >>> >> >> packets >>> >> >> to >>> >> >> freeswitch, even the ones that freeswitch sends to
Kamailio.
>>> >> >> >>> >> >> /Morten >>> >> >> >>> >> >> 2011/10/5 Henrik Aagaard Sørensen >>> >> >> henrikaagaardsorensen@gmail.com: >>> >> >> > I have a setup with Kamailio as dispatcher in front of a >>> >> >> > FreeSwitch >>> >> >> > server. >>> >> >> > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >>> >> >> > >>> >> >> > I'm currently getting "Too many hops" when calling
between
>>> >> >> > SIP >>> >> >> > clients. >>> >> >> > I am >>> >> >> > able to call to FreeSwitch and listen to voicemail, hold >>> >> >> > music >>> >> >> > etc. >>> >> >> > >>> >> >> > After a long conversation with a FreeSwitch expert, and
some
>>> >> >> > tests, I >>> >> >> > was >>> >> >> > told that Kamailio delivers the wrong IP (NAT problems)
to
>>> >> >> > FreeSwitch. >>> >> >> > >>> >> >> > I've also run tshark on both FreeSwitch and Kamailio and
when
>>> >> >> > calling >>> >> >> > between clients they just send the packets between each >>> >> >> > other. >>> >> >> > >>> >> >> > Can anyone help me out? I've tried to Google a lot for
this
>>> >> >> > problem >>> >> >> > and >>> >> >> > asked in several IRC channels, mailing lists and forums. >>> >> >> > Without >>> >> >> > any >>> >> >> > luck. >>> >> >> > >>> >> >> > _______________________________________________ >>> >> >> > SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users
>>> >> >> > mailing >>> >> >> > list >>> >> >> > sr-users@lists.sip-router.org >>> >> >> >
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>> >> >> > >>> >> >> > >>> >> >> >>> >> >> >>> >> >> >>> >> >> -- >>> >> >> Morten Isaksen >>> >> >> >>> >> >> _______________________________________________ >>> >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>> >> >> mailing >>> >> >> list >>> >> >> sr-users@lists.sip-router.org >>> >> >>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>> >> > >>> >> > >>> >> > _______________________________________________ >>> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>> >> > mailing >>> >> > list >>> >> > sr-users@lists.sip-router.org >>> >> >
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>> >> > >>> >> > >>> >> >>> >> >>> >> >>> >> -- >>> >> Morten Isaksen >>> >> >>> >> _______________________________________________ >>> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
>>> >> list >>> >> sr-users@lists.sip-router.org >>> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> > >>> > >>> > _______________________________________________ >>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
>>> > list >>> > sr-users@lists.sip-router.org >>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> > >>> > >>> >>> >>> >>> -- >>> Morten Isaksen >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
>>> list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Please send the full capture.
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
When trying to dial 101 this is a tshark output on the Kamailio:
0.000000 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:101@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Temporarily Unavailable 0.447130 95.214.24.165 -> 215.183.255.142 SIP Request: ACK sip:101@sip.my-domain.com
On Sun, Oct 9, 2011 at 12:14 PM, Morten Isaksen misak@misak.dk wrote:
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening to voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is wrong with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've tested it a lot know, your latest config-example. At it actually works when I connect 2 devices, 1 iPhone and 1 Android. But when connecting 1 phone and my laptop with SFLPhone or Linphone I cannot call the laptop. Does that make any sense?
2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com > > Still getting "Too Many Hops" :( > > On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk > wrote: >> >> Try this one http://pastebin.com/mahKECAw >> >> /Morten >> >> 2011/10/6 Henrik Aagaard Sørensen >> henrikaagaardsorensen@gmail.com: >> > Hi Morten. >> > >> > I've tried to add that part: http://pastebin.com/MmKnbKLz >> > >> > But now it won't even register. Do you know any config-example >> > for >> > a >> > working >> > dispatcher for Kamailio? >> > >> > On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk >> > wrote: >> >> >> >> This part >> >> >> >> # handle requests within SIP dialogs >> >> route(WITHINDLG); >> >> >> >> 2011/10/6 Henrik Aagaard Sørensen >> >> henrikaagaardsorensen@gmail.com: >> >> > Hi Morten. >> >> > >> >> > Do you mean anything specific in the standard config: >> >> > http://pastebin.com/Aj4mHAJq >> >> > >> >> > Because that handles registrations, subscriber list etc. >> >> > etc... >> >> > I'm >> >> > only >> >> > interested in Kamailio as a dispatcher. >> >> > >> >> > And I've already tried adding the PATH module with the >> >> > use_received >> >> > parameter and add_path() and add_path_received() functions. >> >> > That >> >> > didn't >> >> > help. >> >> > >> >> > On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen >> >> > misak@misak.dk >> >> > wrote: >> >> >> >> >> >> Hi, >> >> >> >> >> >> You need to handle in dialog routing - check one of the >> >> >> configs >> >> >> that >> >> >> ships with kamailio. Right now Kamailio forwards all SIP >> >> >> packets >> >> >> to >> >> >> freeswitch, even the ones that freeswitch sends to Kamailio. >> >> >> >> >> >> /Morten >> >> >> >> >> >> 2011/10/5 Henrik Aagaard Sørensen >> >> >> henrikaagaardsorensen@gmail.com: >> >> >> > I have a setup with Kamailio as dispatcher in front of a >> >> >> > FreeSwitch >> >> >> > server. >> >> >> > This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >> >> >> > >> >> >> > I'm currently getting "Too many hops" when calling between >> >> >> > SIP >> >> >> > clients. >> >> >> > I am >> >> >> > able to call to FreeSwitch and listen to voicemail, hold >> >> >> > music >> >> >> > etc. >> >> >> > >> >> >> > After a long conversation with a FreeSwitch expert, and >> >> >> > some >> >> >> > tests, I >> >> >> > was >> >> >> > told that Kamailio delivers the wrong IP (NAT problems) to >> >> >> > FreeSwitch. >> >> >> > >> >> >> > I've also run tshark on both FreeSwitch and Kamailio and >> >> >> > when >> >> >> > calling >> >> >> > between clients they just send the packets between each >> >> >> > other. >> >> >> > >> >> >> > Can anyone help me out? I've tried to Google a lot for >> >> >> > this >> >> >> > problem >> >> >> > and >> >> >> > asked in several IRC channels, mailing lists and forums. >> >> >> > Without >> >> >> > any >> >> >> > luck. >> >> >> > >> >> >> > _______________________________________________ >> >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> >> > mailing >> >> >> > list >> >> >> > sr-users@lists.sip-router.org >> >> >> > >> >> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> > >> >> >> > >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Morten Isaksen >> >> >> >> >> >> _______________________________________________ >> >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> >> mailing >> >> >> list >> >> >> sr-users@lists.sip-router.org >> >> >> >> >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > >> >> > >> >> > _______________________________________________ >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> > mailing >> >> > list >> >> > sr-users@lists.sip-router.org >> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Morten Isaksen >> >> >> >> _______________________________________________ >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> >> mailing >> >> list >> >> sr-users@lists.sip-router.org >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> > mailing >> > list >> > sr-users@lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Morten Isaksen >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
That is the complete capture. Without registration though. I'll send another capture later with registrations.
Also, this failed with "Time out". I'll try to create an event with "Too many loops" as well.
On 10/10/2011, at 17.17, Morten Isaksen misak@misak.dk wrote:
Please send the full capture.
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
When trying to dial 101 this is a tshark output on the Kamailio:
0.000000 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:101@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Temporarily Unavailable 0.447130 95.214.24.165 -> 215.183.255.142 SIP Request: ACK sip:101@sip.my-domain.com
On Sun, Oct 9, 2011 at 12:14 PM, Morten Isaksen misak@misak.dk wrote:
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
Dear Morten and everyone else.
I'm still struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far (with help from Morten): http://pastebin.com/nBPSpe6S
Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops makes calls between them produce the error "Too many hops".
With all of them I'm able to call in to the Freeswitch, for listening to voicemail, hold music etc.
So I guess it's still NAT problems or similar?
Can anyone spot the error, missing thing or something else that is wrong with the config?
P.S. Adding phones, laptops etc. directly to FreeSwitch, without Kamailio, makes everything works.
2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com > > Hi Morten. > > I've tested it a lot know, your latest config-example. At it > actually > works when I connect 2 devices, 1 iPhone and 1 Android. But when > connecting > 1 phone and my laptop with SFLPhone or Linphone I cannot call the > laptop. > Does that make any sense? > > 2011/10/6 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com >> >> Still getting "Too Many Hops" :( >> >> On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk >> wrote: >>> >>> Try this one http://pastebin.com/mahKECAw >>> >>> /Morten >>> >>> 2011/10/6 Henrik Aagaard Sørensen >>> henrikaagaardsorensen@gmail.com: >>>> Hi Morten. >>>> >>>> I've tried to add that part: http://pastebin.com/MmKnbKLz >>>> >>>> But now it won't even register. Do you know any config-example >>>> for >>>> a >>>> working >>>> dispatcher for Kamailio? >>>> >>>> On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen misak@misak.dk >>>> wrote: >>>>> >>>>> This part >>>>> >>>>> # handle requests within SIP dialogs >>>>> route(WITHINDLG); >>>>> >>>>> 2011/10/6 Henrik Aagaard Sørensen >>>>> henrikaagaardsorensen@gmail.com: >>>>>> Hi Morten. >>>>>> >>>>>> Do you mean anything specific in the standard config: >>>>>> http://pastebin.com/Aj4mHAJq >>>>>> >>>>>> Because that handles registrations, subscriber list etc. >>>>>> etc... >>>>>> I'm >>>>>> only >>>>>> interested in Kamailio as a dispatcher. >>>>>> >>>>>> And I've already tried adding the PATH module with the >>>>>> use_received >>>>>> parameter and add_path() and add_path_received() functions. >>>>>> That >>>>>> didn't >>>>>> help. >>>>>> >>>>>> On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen >>>>>> misak@misak.dk >>>>>> wrote: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> You need to handle in dialog routing - check one of the >>>>>>> configs >>>>>>> that >>>>>>> ships with kamailio. Right now Kamailio forwards all SIP >>>>>>> packets >>>>>>> to >>>>>>> freeswitch, even the ones that freeswitch sends to Kamailio. >>>>>>> >>>>>>> /Morten >>>>>>> >>>>>>> 2011/10/5 Henrik Aagaard Sørensen >>>>>>> henrikaagaardsorensen@gmail.com: >>>>>>>> I have a setup with Kamailio as dispatcher in front of a >>>>>>>> FreeSwitch >>>>>>>> server. >>>>>>>> This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >>>>>>>> >>>>>>>> I'm currently getting "Too many hops" when calling between >>>>>>>> SIP >>>>>>>> clients. >>>>>>>> I am >>>>>>>> able to call to FreeSwitch and listen to voicemail, hold >>>>>>>> music >>>>>>>> etc. >>>>>>>> >>>>>>>> After a long conversation with a FreeSwitch expert, and >>>>>>>> some >>>>>>>> tests, I >>>>>>>> was >>>>>>>> told that Kamailio delivers the wrong IP (NAT problems) to >>>>>>>> FreeSwitch. >>>>>>>> >>>>>>>> I've also run tshark on both FreeSwitch and Kamailio and >>>>>>>> when >>>>>>>> calling >>>>>>>> between clients they just send the packets between each >>>>>>>> other. >>>>>>>> >>>>>>>> Can anyone help me out? I've tried to Google a lot for >>>>>>>> this >>>>>>>> problem >>>>>>>> and >>>>>>>> asked in several IRC channels, mailing lists and forums. >>>>>>>> Without >>>>>>>> any >>>>>>>> luck. >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>> mailing >>>>>>>> list >>>>>>>> sr-users@lists.sip-router.org >>>>>>>> >>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Morten Isaksen >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>> mailing >>>>>>> list >>>>>>> sr-users@lists.sip-router.org >>>>>>> >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>> mailing >>>>>> list >>>>>> sr-users@lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Morten Isaksen >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>> mailing >>>>> list >>>>> sr-users@lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>> mailing >>>> list >>>> sr-users@lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >>> >>> >>> -- >>> Morten Isaksen >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>> list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Morten.
I've got the output from tshark when "Too many hops" occurs. And it's no wonder, as my FreeSwitch are sending the SIP packets to Kamailio and Kamailio are sending them back to Freeswitch.
What to do?
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
That is the complete capture. Without registration though. I'll send another capture later with registrations.
Also, this failed with "Time out". I'll try to create an event with "Too many loops" as well.
On 10/10/2011, at 17.17, Morten Isaksen misak@misak.dk wrote:
Please send the full capture.
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
When trying to dial 101 this is a tshark output on the Kamailio:
0.000000 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:101@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Temporarily Unavailable 0.447130 95.214.24.165 -> 215.183.255.142 SIP Request: ACK sip:101@sip.my-domain.com
On Sun, Oct 9, 2011 at 12:14 PM, Morten Isaksen misak@misak.dk wrote:
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk
wrote:
Can you capture one of the calls that fails with tcpdump.
Also try to add some xlog lines in the configuration file for
debuging.
What does the log from rtpproxy show?
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com: > Dear Morten and everyone else. > > I'm still struggling with Kamailio as a simple dispatcher for > FreeSwitch. > This is my configuration so far (with help from Morten): > http://pastebin.com/nBPSpe6S > > Connecting an iPhone and an Android makes the calls between them > timeout. > Connecting one of the phones and my laptops makes calls between them > produce > the error "Too many hops". > > With all of them I'm able to call in to the Freeswitch, for
listening
> to > voicemail, hold music etc. > > So I guess it's still NAT problems or similar? > > Can anyone spot the error, missing thing or something else that is > wrong > with the config? > > P.S. Adding phones, laptops etc. directly to FreeSwitch, without > Kamailio, > makes everything works. > > 2011/10/7 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com >> >> Hi Morten. >> >> I've tested it a lot know, your latest config-example. At it >> actually >> works when I connect 2 devices, 1 iPhone and 1 Android. But when >> connecting >> 1 phone and my laptop with SFLPhone or Linphone I cannot call the >> laptop. >> Does that make any sense? >> >> 2011/10/6 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
>>> >>> Still getting "Too Many Hops" :( >>> >>> On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk >>> wrote: >>>> >>>> Try this one http://pastebin.com/mahKECAw >>>> >>>> /Morten >>>> >>>> 2011/10/6 Henrik Aagaard Sørensen >>>> henrikaagaardsorensen@gmail.com: >>>>> Hi Morten. >>>>> >>>>> I've tried to add that part: http://pastebin.com/MmKnbKLz >>>>> >>>>> But now it won't even register. Do you know any config-example >>>>> for >>>>> a >>>>> working >>>>> dispatcher for Kamailio? >>>>> >>>>> On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen <misak@misak.dk
>>>>> wrote: >>>>>> >>>>>> This part >>>>>> >>>>>> # handle requests within SIP dialogs >>>>>> route(WITHINDLG); >>>>>> >>>>>> 2011/10/6 Henrik Aagaard Sørensen >>>>>> henrikaagaardsorensen@gmail.com: >>>>>>> Hi Morten. >>>>>>> >>>>>>> Do you mean anything specific in the standard config: >>>>>>> http://pastebin.com/Aj4mHAJq >>>>>>> >>>>>>> Because that handles registrations, subscriber list etc. >>>>>>> etc... >>>>>>> I'm >>>>>>> only >>>>>>> interested in Kamailio as a dispatcher. >>>>>>> >>>>>>> And I've already tried adding the PATH module with the >>>>>>> use_received >>>>>>> parameter and add_path() and add_path_received() functions. >>>>>>> That >>>>>>> didn't >>>>>>> help. >>>>>>> >>>>>>> On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen >>>>>>> misak@misak.dk >>>>>>> wrote: >>>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> You need to handle in dialog routing - check one of the >>>>>>>> configs >>>>>>>> that >>>>>>>> ships with kamailio. Right now Kamailio forwards all SIP >>>>>>>> packets >>>>>>>> to >>>>>>>> freeswitch, even the ones that freeswitch sends to Kamailio. >>>>>>>> >>>>>>>> /Morten >>>>>>>> >>>>>>>> 2011/10/5 Henrik Aagaard Sørensen >>>>>>>> henrikaagaardsorensen@gmail.com: >>>>>>>>> I have a setup with Kamailio as dispatcher in front of a >>>>>>>>> FreeSwitch >>>>>>>>> server. >>>>>>>>> This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >>>>>>>>> >>>>>>>>> I'm currently getting "Too many hops" when calling between >>>>>>>>> SIP >>>>>>>>> clients. >>>>>>>>> I am >>>>>>>>> able to call to FreeSwitch and listen to voicemail, hold >>>>>>>>> music >>>>>>>>> etc. >>>>>>>>> >>>>>>>>> After a long conversation with a FreeSwitch expert, and >>>>>>>>> some >>>>>>>>> tests, I >>>>>>>>> was >>>>>>>>> told that Kamailio delivers the wrong IP (NAT problems) to >>>>>>>>> FreeSwitch. >>>>>>>>> >>>>>>>>> I've also run tshark on both FreeSwitch and Kamailio and >>>>>>>>> when >>>>>>>>> calling >>>>>>>>> between clients they just send the packets between each >>>>>>>>> other. >>>>>>>>> >>>>>>>>> Can anyone help me out? I've tried to Google a lot for >>>>>>>>> this >>>>>>>>> problem >>>>>>>>> and >>>>>>>>> asked in several IRC channels, mailing lists and forums. >>>>>>>>> Without >>>>>>>>> any >>>>>>>>> luck. >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>>> mailing >>>>>>>>> list >>>>>>>>> sr-users@lists.sip-router.org >>>>>>>>> >>>>>>>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Morten Isaksen >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>> mailing >>>>>>>> list >>>>>>>> sr-users@lists.sip-router.org >>>>>>>> >>>>>>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>> mailing >>>>>>> list >>>>>>> sr-users@lists.sip-router.org >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Morten Isaksen >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>> mailing >>>>>> list >>>>>> sr-users@lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>> mailing >>>>> list >>>>> sr-users@lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>> >>>> >>>> >>>> -- >>>> Morten Isaksen >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
>>>> list >>>> sr-users@lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >> > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Dear all,
Hope you will help us with this issue.
In the INVITE message there is a field name CONTACT.
1. Is it possible that the 'port' in the CONTACT field is different from the UDP port?
2. What shall be the action in case the ports are different?
3. Is it true that the UDP port in BYE message shall be same as that of INVITE-CONTACT port?
4. Is there any clear reference in the RFC for treatment in this kind of situation?
Thanks and regards,
Amar Tuladhar.
On 10/11/2011 02:45 AM, Amar Tuladhar wrote:
Hope you will help us with this issue.
In the INVITE message there is a field name CONTACT.
1.Is it possible that the ‘port’ in the CONTACT field is different from the UDP port?
What's "the UDP port"? The source port from which the INVITE request originated? If so, then yes.
2.What shall be the action in case the ports are different?
The action according to what? According to RFC 3261, the network and transport-level reachability information in the Contact URI should be used to reach that UA.
3.Is it true that the UDP port in BYE message shall be same as that of INVITE-CONTACT port?
Again, what's "the UDP port"? The destination port?
Assuming yes, your question is really a specific instantiation of a generality. A BYE is a type of request known as a sequential request. Sequential requests take place within a dialog. A dialog is created by the initial INVITE request, and torn down by either a CANCEL (if not yet established) or a BYE (if established). While the dialog is up, one or more sequential requests pertaining to it may occur, such as a reinvite or a BYE.
The request URI of sequential/in-dialog requests is set to the Contact URI of either the initial INVITE request or the final response of the UAS in the transaction that initially set up the dialog.
So in short, yes, the request URI of the BYE going toward UA A would be equal to the Contact of UA A.
4.Is there any clear reference in the RFC for treatment in this kind of situation?
Which RFC? There are many RFCs. There is no reference in 3261 to this situation, but there are other RFCs that deal with far-end NAT traversal strategies. Ignoring the network and transport-layer reachability information in SIP messages--and using the actual received source IP address and port instead--is the basis of the most common far-end (that is, server-side) NAT traversal strategy.
So yes, using received source address and port instead of what the Contact says is actually quite common when dealing with NAT'd endpoints, but it is not 3261-compliant behaviour.
El Tue, 11 Oct 2011 12:30:48 +0545 "Amar Tuladhar" amar@smarttel.com.np escribió:
Dear all,
Hope you will help us with this issue.
Please, do not use the "answer" button to create a new topic. You break thread views.
thanks,
jon
Hi Morten and everyone else.
The problem with Kamailio in front of FreeSWITCH still exists. After a lot of debugging etc. I think the problem occurs due to the IP FreeSWITCH receives from Kamailio.
My users are registrered with the IP from Kamailio in FreeSWITCH and not the IP address they comes from (which Kamailio sees).
So is it possible to setu Kamailio so FreeSWITCH thinks it receives the SIP packets directly from the SIP clients and not Kamailio?
Or should this not be the problem?
Because I think that's why FreeSWITCH send packets back to Kamailio instead of the SIP clients.
I guess FreeSWITCH shouldn't send packets back to Kamailio as Kamailio just is a dispatcher.
2011/10/11 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
Hi Morten.
I've got the output from tshark when "Too many hops" occurs. And it's no wonder, as my FreeSwitch are sending the SIP packets to Kamailio and Kamailio are sending them back to Freeswitch.
What to do?
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com
That is the complete capture. Without registration though. I'll send another capture later with registrations.
Also, this failed with "Time out". I'll try to create an event with "Too many loops" as well.
On 10/10/2011, at 17.17, Morten Isaksen misak@misak.dk wrote:
Please send the full capture.
2011/10/10 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
When trying to dial 101 this is a tshark output on the Kamailio:
0.000000 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:101@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Temporarily Unavailable 0.447130 95.214.24.165 -> 215.183.255.142 SIP Request: ACK sip:101@sip.my-domain.com
On Sun, Oct 9, 2011 at 12:14 PM, Morten Isaksen misak@misak.dk
wrote:
From Kamailio.
2011/10/8 Henrik Aagaard Sørensen henrikaagaardsorensen@gmail.com:
From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen misak@misak.dk
wrote:
> > Can you capture one of the calls that fails with tcpdump. > > Also try to add some xlog lines in the configuration file for
debuging.
> > What does the log from rtpproxy show? > > 2011/10/8 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
:
>> Dear Morten and everyone else. >> >> I'm still struggling with Kamailio as a simple dispatcher for >> FreeSwitch. >> This is my configuration so far (with help from Morten): >> http://pastebin.com/nBPSpe6S >> >> Connecting an iPhone and an Android makes the calls between them >> timeout. >> Connecting one of the phones and my laptops makes calls between
them
>> produce >> the error "Too many hops". >> >> With all of them I'm able to call in to the Freeswitch, for
listening
>> to >> voicemail, hold music etc. >> >> So I guess it's still NAT problems or similar? >> >> Can anyone spot the error, missing thing or something else that is >> wrong >> with the config? >> >> P.S. Adding phones, laptops etc. directly to FreeSwitch, without >> Kamailio, >> makes everything works. >> >> 2011/10/7 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
>>> >>> Hi Morten. >>> >>> I've tested it a lot know, your latest config-example. At it >>> actually >>> works when I connect 2 devices, 1 iPhone and 1 Android. But when >>> connecting >>> 1 phone and my laptop with SFLPhone or Linphone I cannot call the >>> laptop. >>> Does that make any sense? >>> >>> 2011/10/6 Henrik Aagaard Sørensen <
henrikaagaardsorensen@gmail.com>
>>>> >>>> Still getting "Too Many Hops" :( >>>> >>>> On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen misak@misak.dk >>>> wrote: >>>>> >>>>> Try this one http://pastebin.com/mahKECAw >>>>> >>>>> /Morten >>>>> >>>>> 2011/10/6 Henrik Aagaard Sørensen >>>>> henrikaagaardsorensen@gmail.com: >>>>>> Hi Morten. >>>>>> >>>>>> I've tried to add that part: http://pastebin.com/MmKnbKLz >>>>>> >>>>>> But now it won't even register. Do you know any config-example >>>>>> for >>>>>> a >>>>>> working >>>>>> dispatcher for Kamailio? >>>>>> >>>>>> On Thu, Oct 6, 2011 at 10:54 AM, Morten Isaksen <
misak@misak.dk>
>>>>>> wrote: >>>>>>> >>>>>>> This part >>>>>>> >>>>>>> # handle requests within SIP dialogs >>>>>>> route(WITHINDLG); >>>>>>> >>>>>>> 2011/10/6 Henrik Aagaard Sørensen >>>>>>> henrikaagaardsorensen@gmail.com: >>>>>>>> Hi Morten. >>>>>>>> >>>>>>>> Do you mean anything specific in the standard config: >>>>>>>> http://pastebin.com/Aj4mHAJq >>>>>>>> >>>>>>>> Because that handles registrations, subscriber list etc. >>>>>>>> etc... >>>>>>>> I'm >>>>>>>> only >>>>>>>> interested in Kamailio as a dispatcher. >>>>>>>> >>>>>>>> And I've already tried adding the PATH module with the >>>>>>>> use_received >>>>>>>> parameter and add_path() and add_path_received() functions. >>>>>>>> That >>>>>>>> didn't >>>>>>>> help. >>>>>>>> >>>>>>>> On Wed, Oct 5, 2011 at 7:41 PM, Morten Isaksen >>>>>>>> misak@misak.dk >>>>>>>> wrote: >>>>>>>>> >>>>>>>>> Hi, >>>>>>>>> >>>>>>>>> You need to handle in dialog routing - check one of the >>>>>>>>> configs >>>>>>>>> that >>>>>>>>> ships with kamailio. Right now Kamailio forwards all SIP >>>>>>>>> packets >>>>>>>>> to >>>>>>>>> freeswitch, even the ones that freeswitch sends to Kamailio. >>>>>>>>> >>>>>>>>> /Morten >>>>>>>>> >>>>>>>>> 2011/10/5 Henrik Aagaard Sørensen >>>>>>>>> henrikaagaardsorensen@gmail.com: >>>>>>>>>> I have a setup with Kamailio as dispatcher in front of a >>>>>>>>>> FreeSwitch >>>>>>>>>> server. >>>>>>>>>> This is my kamailio.cfg: http://pastebin.com/8PR2GFBD >>>>>>>>>> >>>>>>>>>> I'm currently getting "Too many hops" when calling between >>>>>>>>>> SIP >>>>>>>>>> clients. >>>>>>>>>> I am >>>>>>>>>> able to call to FreeSwitch and listen to voicemail, hold >>>>>>>>>> music >>>>>>>>>> etc. >>>>>>>>>> >>>>>>>>>> After a long conversation with a FreeSwitch expert, and >>>>>>>>>> some >>>>>>>>>> tests, I >>>>>>>>>> was >>>>>>>>>> told that Kamailio delivers the wrong IP (NAT problems) to >>>>>>>>>> FreeSwitch. >>>>>>>>>> >>>>>>>>>> I've also run tshark on both FreeSwitch and Kamailio and >>>>>>>>>> when >>>>>>>>>> calling >>>>>>>>>> between clients they just send the packets between each >>>>>>>>>> other. >>>>>>>>>> >>>>>>>>>> Can anyone help me out? I've tried to Google a lot for >>>>>>>>>> this >>>>>>>>>> problem >>>>>>>>>> and >>>>>>>>>> asked in several IRC channels, mailing lists and forums. >>>>>>>>>> Without >>>>>>>>>> any >>>>>>>>>> luck. >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>>>> mailing >>>>>>>>>> list >>>>>>>>>> sr-users@lists.sip-router.org >>>>>>>>>> >>>>>>>>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Morten Isaksen >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>>> mailing >>>>>>>>> list >>>>>>>>> sr-users@lists.sip-router.org >>>>>>>>> >>>>>>>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>> mailing >>>>>>>> list >>>>>>>> sr-users@lists.sip-router.org >>>>>>>>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Morten Isaksen >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>> mailing >>>>>>> list >>>>>>> sr-users@lists.sip-router.org >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>> mailing >>>>>> list >>>>>> sr-users@lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Morten Isaksen >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing
>>>>> list >>>>> sr-users@lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > > -- > Morten Isaksen > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
> sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Morten Isaksen
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Alex Balashov -
Amar Tuladhar -
Henrik Aagaard Sørensen -
Jon Bonilla -
Morten Isaksen -
Sammy Govind