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
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
> That is the complete capture. Without registra
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 wrote:
> Please send the full capture.
>
> 2011/10
Please send the full capture.
2011/10/10 Henrik Aagaard Sørensen :
> When trying to dial 101 this is a tshark output on the Kamailio:
>
> 0.00 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:1...@si
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 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 tabl
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
> When trying to dial 101 this is a tshark output on the Ka
When trying to dial 101 this is a tshark output on the Kamailio:
0.00 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:1...@sip.my-domain.com
0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Tem
>From Kamailio.
2011/10/8 Henrik Aagaard Sørensen :
> From Kamailio or FreeSwitch?
>
> On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen 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 do
>From Kamailio or FreeSwitch?
On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen 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 Søre
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 :
> Dear Morten and everyone else.
>
> I'm still struggling with Kamailio as a simple dispatch
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
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
> St
Still getting "Too Many Hops" :(
On Thu, Oct 6, 2011 at 10:19 PM, Morten Isaksen wrote:
> Try this one http://pastebin.com/mahKECAw
>
> /Morten
>
> 2011/10/6 Henrik Aagaard Sørensen :
> > Hi Morten.
> >
> > I've tried to add that part: http://pastebin.com/MmKnbKLz
> >
> > But now it won't even r
Try this one http://pastebin.com/mahKECAw
/Morten
2011/10/6 Henrik Aagaard Sørensen :
> 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 1
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 wrote:
> This part
>
> # handle requests within SIP dialogs
> route(WITHIND
This part
# handle requests within SIP dialogs
route(WITHINDLG);
2011/10/6 Henrik Aagaard Sørensen :
> 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 K
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_pat
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 :
> I have a setup with Kamailio as dispatcher in fr
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 conversat
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