I have had 0 success with getting this to work with Trixbox 2.6 or 2.8 any ideas? I have tried using the endpoint manager and setting it as a 2010. I have tried manually configuring it, etc. Just no luck getting it to authenticate.
Grandstream GXP2110
Submitted by dudemcdudedude on Fri, 12/31/2010 - 11:55am.
Nothing Special About This Phone
Fri, 12/31/2010 - 12:28pm
I have no idea what you are doing wrong, you didn't provide specifics. There is nothing special about this phone. You need to follow the same basic configuration steps as any other SIP phone.
In trixbox setup a new extension configuring, at a minimum; extension number and secret.
On the phone you must configure, at a minimum; IP address and subnet mask, SIP server(trixbox IP), SIP user ID(extension number), Authenticate Password(secret).
Here's the Grandstream manual.
I have a GXP2010 registered
Tue, 01/04/2011 - 8:57am
Ok the phone registers but I
Tue, 01/04/2011 - 12:29pm
Although I don't think it's
Tue, 01/04/2011 - 12:47pm
Although I don't think it's SW related, try Grandstream latest f/w 1.2.5.3. http://grandstream.com/firmware
You can point your phone to: 72.172.83.110 for automatic upgrade
Sip Debug from the phone.
Thu, 01/06/2011 - 9:36am
Sip Debug from the phone. Also to mention GXP2010s work just fine with any codec.
--- (11 headers 0 lines) ---
Really destroying SIP dialog '7f00e7b733f42e80611612fc5934c3af@74.218.14.75' Method: OPTIONS
voip*CLI>
INVITE sip:*98@192.168.10.16 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.224:5060;branch=z9hG4bK1143721111;rport
Route:
From: ""
To:
Call-ID: 464021992-5060-4@BJC.BGI.BA.CCE
CSeq: 30 INVITE
Contact:
X-Grandstream-PBX: true
Max-Forwards: 70
Proxy-Require: 192.168.10.16
User-Agent: Grandstream GXP2110 1.0.1.26
Privacy: none
P-Asserted-Identity: ""
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 560
v=0
o=205 8000 8000 IN IP4 192.168.10.224
s=SIP Call
c=IN IP4 192.168.10.224
t=0 0
m=audio 5004 RTP/SAVP 18 0 4 9 97 2 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5iRwS71bOjJZHWQc4a/jyFNQz3rONOOYlJ3V635+|2^32
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:8mSiYq9fvemRFgf2MuilFbD4Zn9zNLRWzEj0ojRy|2^32
--- (19 headers 20 lines) ---
Sending to 192.168.10.224 : 5060 (NAT)
Using INVITE request as basis request - 464021992-5060-4@BJC.BGI.BA.CCE
voip*CLI>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.224:5060;branch=z9hG4bK1143721111;received=192.168.10.224;rport=5060
From: ""
To:
Call-ID: 464021992-5060-4@BJC.BGI.BA.CCE
CSeq: 30 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78d122f7"
Content-Length: 0
Scheduling destruction of SIP dialog '464021992-5060-4@BJC.BGI.BA.CCE' in 32000 ms (Method: INVITE)
Found user '205'
voip*CLI>
ACK sip:*98@192.168.10.16 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.224:5060;branch=z9hG4bK1143721111;rport
Route:
From: ""
To:
Call-ID: 464021992-5060-4@BJC.BGI.BA.CCE
CSeq: 30 ACK
Content-Length: 0
--- (8 headers 0 lines) ---
INVITE sip:*98@192.168.10.16 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.224:5060;branch=z9hG4bK25324028;rport
Route:
From: ""
To:
Call-ID: 464021992-5060-4@BJC.BGI.BA.CCE
CSeq: 31 INVITE
Contact:
Proxy-Authorization: Digest username="205", realm="asterisk", nonce="78d122f7", uri="sip:*98@192.168.10.16", response="923c345ca25c1e3ad0e7007914c45c67", algorithm=MD5
X-Grandstream-PBX: true
Max-Forwards: 70
Proxy-Require: 192.168.10.16
User-Agent: Grandstream GXP2110 1.0.1.26
Privacy: none
P-Asserted-Identity: ""
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 560
v=0
o=205 8000 8000 IN IP4 192.168.10.224
s=SIP Call
c=IN IP4 192.168.10.224
t=0 0
m=audio 5004 RTP/SAVP 18 0 4 9 97 2 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5iRwS71bOjJZHWQc4a/jyFNQz3rONOOYlJ3V635+|2^32
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:8mSiYq9fvemRFgf2MuilFbD4Zn9zNLRWzEj0ojRy|2^32
--- (20 headers 20 lines) ---
Sending to 192.168.10.224 : 5060 (NAT)
Using INVITE request as basis request - 464021992-5060-4@BJC.BGI.BA.CCE
Found user '205'
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.224:5060;branch=z9hG4bK25324028;received=192.168.10.224;rport=5060
From: ""
To:
Call-ID: 464021992-5060-4@BJC.BGI.BA.CCE
CSeq: 31 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Scheduling destruction of SIP dialog '464021992-5060-4@BJC.BGI.BA.CCE' in 32000 ms (Method: INVITE)
voip*CLI>
ACK sip:*98@192.168.10.16 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.224:5060;branch=z9hG4bK25324028;rport
Route:
From: ""
To:
Call-ID: 464021992-5060-4@BJC.BGI.BA.CCE
CSeq: 31 ACK
Content-Length: 0
--- (8 headers 0 lines) ---
Member Since:
2007-12-03