i was thinking of changing the record-route before sending out.
I tried it, but it's not working
subst_hf("Record-route", "/^<sips:1.2.3.4:6055\(.*\)$/<sip:
1.2.3.4:6055\1/", "a");
Kelvin Chua
On Wed, Mar 4, 2015 at 2:52 PM, Kelvin Chua <kel...@gmail.com> wrote:
> just an idea, will it work if i used subst_hf?
>
> Kelvin Chua
>
> On Tue, Mar 3, 2015 at 5:16 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>> We need to review this in rr.
>>
>> Meanwhile you can use s.substr transformation to get whats after sips and
>> prefix it with sip in r-uri.
>>
>> Cheers,
>> Daniel
>>
>>
>> On Tuesday, March 3, 2015, Kelvin Chua <kel...@gmail.com> wrote:
>>
>>> Found the problem, on the 200 OK, I have this record route list
>>>
>>> Record-Route: <sips:<K2
>>> IP>:6056;transport=tls;r2=on;lr;ftag=as620b910c;did=242.fd92;nat=yes>.
>>> Record-Route: <sips:<K2
>>> IP>:6055;r2=on;lr;ftag=as620b910c;did=242.fd92;nat=yes>.
>>> Record-Route: <sip:<K1 IP>:5080;lr=on;did=242.968>
>>>
>>> the second entry is wrong. it should be sip: and not sips:
>>> how can we force this? protocol pv is read-only
>>>
>>>
>>> Kelvin Chua
>>>
>>> On Tue, Mar 3, 2015 at 2:43 PM, Daniel-Constantin Mierla <
>>> mico...@gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> doesn't the ACK have a Route header for K1 and double Route headers for
>>>> K2? K1 should use the first Route of K2 for routing, not the R-URI.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>> On 03/03/15 05:14, Kelvin Chua wrote:
>>>>
>>>> I have 2 kamailio servers and 1 asterisk server.
>>>>
>>>> 1. asterisk calls kamailio1
>>>> 2. kamailio1 relays INVITE to kamailio2
>>>> 3. kamailio2 relays INVITE to client registered using TLS
>>>> 4. client answers with 200 OK, sends to kamailio2
>>>> 5. kamailio2 relays 200 OK to kamailio1
>>>> 6. kamailio1 relays 200 OK to asterisk
>>>> 7. asterisk sends ACK to kamailio1
>>>> 8. kamailio1 complains "forward_request(): forward_req: ERROR: cannot
>>>> forward to af 2, proto 3 no corresponding listening socket" which is
>>>> because 200 OK has a contact header with tls transport from client
>>>> 9. call drops in 30 secs which is expected because client never
>>>> received the ACK
>>>>
>>>> any ideas on a fix or workaround?
>>>>
>>>>
>>>> Kelvin Chua
>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>> --
>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
>>>> http://www.linkedin.com/in/miconda
>>>> Kamailio World Conference, May 27-29, 2015
>>>> Berlin, Germany - http://www.kamailioworld.com
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
>> <http://www.linkedin.com/in/miconda>
>>
>>
>
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