Asunto: Re: [SR-Users] ACK not sent and rr-enforced
Hello,
can you post the ngrep trace of such call (fron incoming invite, to the bye,
taken on your server)? That will help to see what could be mismatching there.
Cheers,
Daniel
On Thu, Mar 10, 2011 at 11:06 AM, Dominguez Jover, Ricardodjo
-users-boun...@lists.sip-router.org] En nombre de Alex
Balashov
Enviado el: martes, 15 de marzo de 2011 17:25
Para: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] ACK not sent and rr-enforced
On 03/15/2011 08:28 AM, Dominguez Jover, Ricardo wrote:
Should I infer IPTEL.org is not implementing SIP
Am 16.03.2011 10:55, schrieb Dominguez Jover, Ricardo:
Thanks Klaus,
I just was wondering if there was something I could configure in my proxy.
I don't think so.
Of course you could do dirty hacks, e.g. storing the clients contact in
a record-route cookie and restoring the RURI from this
to me...
Cheers,
Ricardo
De: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Enviado el: lunes, 14 de marzo de 2011 10:58
Para: Dominguez Jover, Ricardo
CC: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] ACK not sent and rr-enforced
Hello,
I will look over it very soon. As a hint
On 03/15/2011 08:28 AM, Dominguez Jover, Ricardo wrote:
Should I infer IPTEL.org is not implementing SIP RFC 3261 in the
right way? It seems odd to me...
No, Ricardo, that is not the correct inference. First, if the ACK is an
end-to-end ACK (as for a 200 OK), it is generated by the sending
Enviado el: lunes, 07 de marzo de 2011 20:03
Para: sr-users@lists.sip-router.org
Asunto: [SR-Users] ACK not sent and rr-enforced
Hi everybody.
I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external
test accounts to check if the calls are established and the media flow is ok
Hello Daniel, here it is.
Thanks
Ricardo
De: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Enviado el: jueves, 10
de marzo de 2011 12:49
Para: Dominguez Jover, Ricardo
CC: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] ACK not sent and rr-enforced
Hello,
can you post the ngrep
Hi everybody.
I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external
test accounts to check if the calls are established and the media flow is ok.
When I use a sip2sip.info or VoIP Talk accounts, then all is working fine
between my internal and these external
