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

Hay guys, got trouble with registration with cisco 7975

Here is the debug :

<--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 --->
REGISTER sip:192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381
From: <sip:111 at 192.168.1.4>;tag=0c8525a68961001f44d503e2-d9359bd3
To: <sip:111 at 192.168.1.4>
Call-ID: 0c8525a6-89610004-b972d038-5864c98e at 192.168.1.61
Max-Forwards: 70
Date: Tue, 24 Feb 2015 07:13:42 GMT
CSeq: 110 REGISTER
User-Agent: Cisco-CP7975G/8.5.3
Contact: <sip:111 at
192.168.1.61:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0c8525a68961>";+u.sip!model.ccm.cisco.com="437"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Expires: 3600


<--- Transmitting SIP response (481 bytes) to UDP:192.168.1.61:49531 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.61:5060;rport=49531;received=192.168.1.61;branch=z9hG4bKd16b1eb7
Call-ID: 0c8525a6-89610002-845d0080-f3559596 at 192.168.1.61
From: <sip:111 at 192.168.1.4>;tag=0c8525a68961001d53245ebc-a1b56549
To: <sip:111 at 192.168.1.4>;tag=z9hG4bKd16b1eb7
CSeq: 110 REGISTER
WWW-Authenticate: Digest 
realm="asterisk",nonce="1424762038/41d5874af9ea9408c257949c309c8aa0",opaque="7f15d8c2312c7b0d",algorithm=md5,qop="auth"
Content-Length:  0


username and password are correct, this phone was working with 3CX just fine but
won?t work with asterisk for some reason. (

any idea what may cause the problem?
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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

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
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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