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Nick Awesome

2015-Feb-24 14:21 UTC

[asterisk-users] having trouble to register cisco 7975 with pjsip

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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Nick Awesome

2015-Feb-26 06:00 UTC

head link

[asterisk-users] having trouble to register cisco 7975 with pjsip

another issues with cisco 7975

I have phone registered on asterisk

have 2 different issues on different versions of firmware, 

on 9-4-2-1S I have not working 3way conference, when I trying to connect second
call, phone says ?unable to set up conference?
and sending some cisco xml data to asterisk which cannot be handled, thats the
problem,

I know on firmware 8-5-4 3way conference works just fine 3cx phone system so
must be same with asterisk,

but with asterisk when I do ANY call from cisco phone with fw 8-5-4

cisco hangup call after channels connect, debug

<--- Received SIP request (1003 bytes) from UDP:192.168.1.61:49163 --->
INVITE sip:*777 at 192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKa67a2ab7
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at 192.168.1.4;user=phone>
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
Max-Forwards: 70
Date: Thu, 26 Feb 2015 05:52:42 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7975G/8.5.3
Contact: <sip:111 at 192.168.1.61:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: kpml,dialog
Content-Length: 322
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 626 0 IN IP4 192.168.1.61
s=SIP Call
t=0 0
m=audio 30354 RTP/AVP 0 8 18 116 101
c=IN IP4 192.168.1.61
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (485 bytes) to UDP:192.168.1.61:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.61:5060;received=192.168.1.61;branch=z9hG4bKa67a2ab7
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at 192.168.1.4;user=phone>;tag=z9hG4bKa67a2ab7
CSeq: 101 INVITE
WWW-Authenticate: Digest 
realm="asterisk",nonce="1424929962/9af5af19e633c82d2a9e17ec97afb72b",opaque="2776507e426bda2b",algorithm=md5,qop="auth"
Content-Length:  0

<--- Received SIP request (368 bytes) from UDP:192.168.1.61:49174 --->
ACK sip:*777 at 192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKa67a2ab7
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at 192.168.1.4;user=phone>;tag=z9hG4bKa67a2ab7
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
Date: Thu, 26 Feb 2015 05:52:42 GMT
CSeq: 101 ACK
Content-Length: 0

<--- Received SIP request (1271 bytes) from UDP:192.168.1.61:49163 --->
INVITE sip:*777 at 192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK4affb043
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at 192.168.1.4;user=phone>
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
Max-Forwards: 70
Date: Thu, 26 Feb 2015 05:52:42 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7975G/8.5.3
Contact: <sip:111 at 192.168.1.61:5060;transport=udp>
Authorization: Digest
username="111",realm="asterisk",uri="sip:*777 at
192.168.1.4;user=phone",response="8b90970d8fc724893e876263ce8c2cd3",nonce="1424929962/9af5af19e633c82d2a9e17ec97afb72b",opaque="2776507e426bda2b",cnonce="945bf4a1",qop=auth,nc=00000001,algorithm=md5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: kpml,dialog
Content-Length: 322
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 626 0 IN IP4 192.168.1.61
s=SIP Call
t=0 0
m=audio 30354 RTP/AVP 0 8 18 116 101
c=IN IP4 192.168.1.61
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (312 bytes) to UDP:192.168.1.61:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.61:5060;received=192.168.1.61;branch=z9hG4bK4affb043
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at 192.168.1.4;user=phone>
CSeq: 102 INVITE
Content-Length:  0

<--- Transmitting SIP response (738 bytes) to UDP:192.168.1.61:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.61:5060;received=192.168.1.61;branch=z9hG4bK4affb043
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at
192.168.1.4;user=phone>;tag=916a8d96-8a85-4474-b404-e30615c6c963
CSeq: 102 INVITE
Contact: <sip:192.168.1.4:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   163

v=0
o=- 626 2 IN IP4 192.168.1.4
s=Asterisk
c=IN IP4 192.168.1.4
t=0 0
m=audio 10474 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (697 bytes) from UDP:192.168.1.61:49163 --->
ACK sip:192.168.1.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK22ad7045
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at
192.168.1.4;user=phone>;tag=916a8d96-8a85-4474-b404-e30615c6c963
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
Max-Forwards: 70
Date: Thu, 26 Feb 2015 05:52:42 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7975G/8.5.3
Authorization: Digest
username="111",realm="asterisk",uri="sip:*777 at
192.168.1.4;user=phone",response="8b90970d8fc724893e876263ce8c2cd3",nonce="1424929962/9af5af19e633c82d2a9e17ec97afb72b",opaque="2776507e426bda2b",cnonce="945bf4a1",qop=auth,nc=00000001,algorithm=md5
Content-Length: 0


<--- Received SIP request (686 bytes) from UDP:192.168.1.61:49163 --->
BYE sip:192.168.1.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKf9a5d51f
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at
192.168.1.4;user=phone>;tag=916a8d96-8a85-4474-b404-e30615c6c963
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
Max-Forwards: 70
Date: Thu, 26 Feb 2015 05:52:42 GMT
CSeq: 103 BYE
User-Agent: Cisco-CP7975G/8.5.3
Content-Length: 0
Authorization: Digest
username="111",realm="asterisk",uri="sip:192.168.1.4:5060",response="6ab95be6adc870723154d7e0fb6f7cd4",nonce="1424929962/9af5af19e633c82d2a9e17ec97afb72b",opaque="2776507e426bda2b",cnonce="884cb6e9",qop=auth,nc=00000002,algorithm=md5


<--- Transmitting SIP response (346 bytes) to UDP:192.168.1.61:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.61:5060;received=192.168.1.61;branch=z9hG4bKf9a5d51f
Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0 at 192.168.1.61
From: "111" <sip:111 at
192.168.1.4>;tag=0c8525a689610012e85fd91b-ee689f06
To: <sip:*777 at
192.168.1.4;user=phone>;tag=916a8d96-8a85-4474-b404-e30615c6c963
CSeq: 103 BYE
Content-Length:  0
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Nick Awesome

2015-Feb-28 06:53 UTC

head link

[asterisk-users] having trouble to register cisco 7975 with pjsip

success! just replaced MeetMe to Bridge in softkey.xml and conf works now with
the latest fw!


On Feb 26, 2015, at 9:00 AM, Nick Awesome <jleed at me.com>
wrote:> 
> I have not working 3way conference, when I trying to connect second call,
phone says ?unable to set up conference?
> and sending some cisco xml data to asterisk which cannot be handled, thats
the problem,
>

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