asterisk users - Feb 2015 - having trouble to register cisco 7975 with pjsip

Ok after I added tcp transport and disable force_rport phone get registered, but
still have issues with calls,

when I call from cisco from, it work except hangup.

when I call to cisco phone asterisk return congested

debug of call
<--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 --->
INVITE sip:111 at 192.168.1.61:51179;transport=tcp SIP/2.0
Via: SIP/2.0/TCP
192.168.1.4:55246;rport;branch=z9hG4bKPjcb9ec9ba-0c77-4530-a3b7-44209357f3a0;alias
From: <sip:502 at 192.168.1.4>;tag=abebd75c-501a-4b4f-ad69-ee98175b8dbd
To: <sip:111 at 192.168.1.61>
Contact: <sip:28552048-b20b-4e7c-8454-f7d1486fd8ef at
192.168.1.4:55246;transport=TCP>
Call-ID: bb515935-7292-47b4-890d-6f82eb335815
CSeq: 25333 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   283

v=0
o=- 1231372975 1231372975 IN IP4 192.168.1.4
s=Asterisk
c=IN IP4 192.168.1.4
t=0 0
m=audio 17856 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[Feb 24 05:47:01] WARNING[16179]: pjsip:0 <?>: 	tsx0x7f1aa0157 Failed to
send Request msg INVITE/cseq=12216 (tdta0x7f1aa00e41c0)! err=120111 (Connection
refused)
[Feb 24 05:47:01] ERROR[16179]: pjsip:0 <?>: 	tcpc0x7f1aa01c TCP connect()
error: Connection refused [code=120111]
[Feb 24 05:47:01] WARNING[16179]: pjsip:0 <?>: 	tsx0x7f1aa01c3 Failed to
send Request msg INVITE/cseq=25333 (tdta0x7f1aa00ad810)! err=120111 (Connection
refused)

> On 24 Feb 2015, at 15:05, Joshua Colp <jcolp at digium.com> wrote:
> 
> Nick Awesome wrote:
>> Hay guys, got trouble with registration with cisco 7975
> 
> The "force_rport" option is incompatible with Cisco, it needs to
be explicitly set to no in the endpoint.
> 
> Cheers,
> 
> -- 
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
> 
> -- 
> _____________________________________________________________________
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Nick Awesome wrote:
> Ok after I added tcp transport and disable force_rport phone get
> registered, but still have issues with calls,
>
> when I call from cisco from, it work except hangup.
>
> when I call to cisco phone asterisk return congested
If you use UDP with force_rport=no it'll work.
If you use TCP then set rewrite_contact=yes so it'll reuse the 
established TCP connection.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

Oh god it works !

to switch cisco to upd I used config:
<transportLayerProtocol>2</transportLayerProtocol>

with udp it works well, thanks for your help :)
> On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote:
> 
> If you use UDP with force_rport=no it'll work.
> If you use TCP then set rewrite_contact=yes so it'll reuse the
established TCP connection.
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