[OpenSIPS-Users] Polycom REGISTER loop

[Nick Khamis]

Duane,

Thank you so much for your response! I temporarily commented out

"server_header="example.test.com", and we have two way communication
happening now. Not quite sure why sine the "server_header" is not
"opensips.example.com", and is actually a FQDN (i.e, known to public
DNSs).
Will look into it further and keep the list posted, and enrich google
in the process.

N.

On 1/9/13, Duane Larson <duane.larson at gmail.com> wrote:
> Sorry.  After looking at this more I see that this is a siptrace from the
> phone itself so the 200 OK is obviously making it to the phone.  So yeah
> your "opensips.example.com." might be whats messing it up.
>
> Look at this example of a 200 OK
> http://www.rfc-ref.org/RFC-TEXTS/3261/chapter24.html
>
> What is adding that line?  It needs to go away I think.
>
>
>
> On Wed, Jan 9, 2013 at 4:59 PM, Nick Khamis <symack at gmail.com> wrote:
>
>> Hahah, trying so hard to make everyone (in this case internal/external
>> UAs) Happy! The external (i.e., outside of the network) phones are all
>> working nicely. It's just the phones within the network that are
>> complaining. Looking at the NAT box, I see that they have 5060 port
>> forwarded to the OpenSIPS box.
>> Pinging the phone from the proxy is ok.
>>
>> Any thoughts?
>>
>> Nick.
>>
>>
>>
>> On 1/9/13, Duane Larson <duane.larson at gmail.com> wrote:
>> > So you say you "closed gaping holes".  Are you saying that the local
>> > polycom phone are behind a firewall?  If they are behind the firewall
>> then
>> > you need to figure out why the 200 OK is not making it to the polycom
>> > phones.  It could be because Polycom doesn't support rport (just a shot
>> in
>> > the dark from the info you have provided).
>> >
>> >
>> > On Wed, Jan 9, 2013 at 4:05 PM, Nick Khamis <symack at gmail.com> wrote:
>> >
>> >> Hello Everyone,
>> >>
>> >> After scrambling to close up some gaping holes and getting RTP
>> >> relaying some media,
>> >> all of which is playing so nicely together. I have been accused of
>> >> breaking the "REGISTER"
>> >> functionality on the local Polycom phones. The remote ones all seem to
>> >> be ok but for some reason, I have the following SIP trace from the
>> >> local phones:
>> >>
>> >> I suspect it's the "opensips.example.com." (which the actual domain
>> >> name has been replaced for security reasons) at the bottom of the
>> >> response causing the problem?
>> >>
>> >> U 1999/12/18 17:42:11.890804 192.168.2.11:5060 -> 192.168.2.5:5060
>> >> REGISTER sip:192.168.2.5:5060 SIP/2.0.
>> >> Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK5dfc96b3ADE4AD6E.
>> >> From: "1001" <sip:1001 at 192.168.2.5>;tag=8D956A9-11D1CBB2.
>> >> To: <sip:1001 at 192.168.2.5>.
>> >> CSeq: 1 REGISTER.
>> >> Call-ID: 1496d055-3a3afc27-1f9da518 at 192.168.2.11.
>> >> Contact: <sip:1001 at 192.168.2.11:5060>;methods="INVITE, ACK, BYE,
>> >> CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
>> >> REFER".
>> >> User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.8.0070.
>> >> Accept-Language: en.
>> >> Max-Forwards: 70.
>> >> Expires: 3600.
>> >> Content-Length: 0.
>> >> .
>> >>
>> >>
>> >> U 1999/12/18 17:42:11.891532 192.168.2.5:5060 -> 192.168.2.11:5060
>> >> SIP/2.0 200 OK.
>> >> Via: SIP/2.0/UDP
>> >> 192.168.2.11:5060
>> >> ;received=192.168.2.11;rport=5060;branch=z9hG4bK5dfc96b3ADE4AD6E.
>> >> From: "1001" <sip:1001 at 192.168.2.5>;tag=8D956A9-11D1CBB2.
>> >> To: <sip:1001 at 192.168.2.5>;tag=b3499e451a8d3922338ff1f455481e64.34b3.
>> >> CSeq: 1 REGISTER.
>> >> Call-ID: 1496d055-3a3afc27-1f9da518 at 192.168.2.11.
>> >> Contact: <sip:1001 at 192.168.2.11:5060>;expires=3600.
>> >> opensips.example.com.
>> >> Content-Length: 0.
>> >> Warning: 392 192.168.2.5:5060 "Noisy feedback tells:  pid=2910
>> >> req_src_ip=192.168.2.11 req_src_port=5060 in_uri=sip:192.168.2.5:5060
>> >> out_uri=sip:192.168.2.5:5060 via_cnt==1".
>> >> .
>> >>
>> >>
>> >> U 1999/12/18 17:42:12.860355 192.168.2.11:5060 -> 192.168.2.5:5060
>> >> REGISTER sip:192.168.2.5:5060 SIP/2.0.
>> >> Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK52241620779FEF4F.
>> >> From: "1001" <sip:1001 at 192.168.2.5>;tag=DC560FE-72CEA503.
>> >> To: <sip:1001 at 192.168.2.5>.
>> >> CSeq: 1 REGISTER.
>> >> Call-ID: df32c17a-de5b3c44-2c79b191 at 192.168.2.11.
>> >> Contact: <sip:1001 at 192.168.2.11:5060>;methods="INVITE, ACK, BYE,
>> >> CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
>> >> REFER".
>> >> User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.8.0070.
>> >> Accept-Language: en.
>> >> Max-Forwards: 70.
>> >> Expires: 3600.
>> >> Content-Length: 0.
>> >> .
>> >>
>> >>
>> >> U 1999/12/18 17:42:12.860750 192.168.2.5:5060 -> 192.168.2.11:5060
>> >> SIP/2.0 200 OK.
>> >> Via: SIP/2.0/UDP
>> >> 192.168.2.11:5060
>> >> ;received=192.168.2.11;rport=5060;branch=z9hG4bK52241620779FEF4F.
>> >> From: "1001" <sip:1001 at 192.168.2.5>;tag=DC560FE-72CEA503.
>> >> To: <sip:1001 at 192.168.2.5>;tag=b3499e451a8d3922338ff1f455481e64.2efb.
>> >> CSeq: 1 REGISTER.
>> >> Call-ID: df32c17a-de5b3c44-2c79b191 at 192.168.2.11.
>> >> Contact: <sip:1001 at 192.168.2.11:5060>;expires=3600.
>> >> opensips.example.com.
>> >> Content-Length: 0.
>> >> Warning: 392 192.168.2.5:5060 "Noisy feedback tells:  pid=2912
>> >> req_src_ip=192.168.2.11 req_src_port=5060 in_uri=sip:192.168.2.5:5060
>> >> out_uri=sip:192.168.2.5:5060 via_cnt==1".
>> >>
>> >> As you can see the expire is set to 3600 on the UA, and the ok is
>> >> being sent back however, Poly does not seem to be receiving the 200?
>> >>
>> >> Thanks in Advance,
>> >>
>> >> Nick.
>> >> .
>> >>
>> >> _______________________________________________
>> >> Users mailing list
>> >> Users at lists.opensips.org
>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >>
>> >
>> >
>> >
>> > --
>> > --
>> > *--*--*--*--*--*
>> > Duane
>> > *--*--*--*--*--*
>> > --
>> >
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>