Answer the question
In order to leave comments, you need to log in
Why does the CISCO 7906G sewn on SIP not understand the on-hook?
There are CISCO 7906G phones, re-sewn to SIP. Phones do not want to understand in any way that the subscriber on the other end of the wire has hung up. This applies to both internal calls and incoming calls from mobile and landline phones. The Asterisk server itself is located behind NAT, but I also tried to place Asterisk and these phones on the same network - it did not help.
There are also other phone models on the network, incl. Cisco SPA303 - there are no problems with it.
Tell me where to dig?
Answer the question
In order to leave comments, you need to log in
Spread a specific version of the firmware - in the same situation, the same phones work perfectly for me.
My version is SIP11.9-0-3S
Password entered,
<--- SIP read from TCP:MY_REAL_IP:49176 --->
REGISTER sip:10.10.10.5:5060 SIP/2.0
Via: SIP/2.0/TCP MY_REAL_IP:49176;branch=z9hG4bKc7a53e87
From: <sip:[email protected]:5060>;tag=00070e36ee480003e64d4b0e-f0245601
To: <sip:[email protected]:5060>
Call-ID: [email protected]_REAL_IP
Max-Forwards: 70
Date: Fri, 29 Apr 2016 08:12:37 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7906G/9.2.1
Contact: <sip:[email protected]_REAL_IP:49176;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00070e36ee48>";+u.sip!devicename.ccm.cisco.com="SEP00070E36EE48";+u.sip!model.ccm.cisco.com="369"
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:24 Name=SEP00070E36EE48 Load=SIP11.9-2-1SR2S Last=phone-reg-rej"
Expires: 3600
<------------->
--- (14 headers 0 lines) ---
Sending to MY_REAL_IP:49176 (no NAT)
Sending to MY_REAL_IP:49176 (no NAT)
<--- Transmitting (no NAT) to MY_REAL_IP:49176 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP MY_REAL_IP:49176;branch=z9hG4bKc7a53e87;received=MY_REAL_IP
From: <sip:[email protected]:5060>;tag=00070e36ee480003e64d4b0e-f0245601
To: <sip:[email protected]:5060>;tag=as506b1cee
Call-ID: [email protected]_REAL_IP
CSeq: 101 REGISTER
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c6955e2"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]_REAL_IP' in 32000 ms (Method: REGISTER)
<--- SIP read from TCP:MY_REAL_IP:49176 --->
REGISTER sip:10.10.10.5:5060 SIP/2.0
Via: SIP/2.0/TCP MY_REAL_IP:49176;branch=z9hG4bK0ceca61c
From: <sip:[email protected]:5060>;tag=00070e36ee480003e64d4b0e-f0245601
To: <sip:[email protected]:5060>
Call-ID: [email protected]_REAL_IP
Max-Forwards: 70
Date: Fri, 29 Apr 2016 08:12:37 GMT
CSeq: 102 REGISTER
User-Agent: Cisco-CP7906G/9.2.1
Contact: <sip:[email protected]_REAL_IP:49176;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00070e36ee48>";+u.sip!devicename.ccm.cisco.com="SEP00070E36EE48";+u.sip!model.ccm.cisco.com="369"
Authorization: Digest username="921",realm="asterisk",uri="sip:ASTERISK_REAL_IP",response="43f07556b2590fd1b6bbdcaf2d8b000b",nonce="4c6955e2",algorithm=MD5
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:24 Name=SEP00070E36EE48 Load=SIP11.9-2-1SR2S Last=phone-reg-rej"
Expires: 3600
<------------->
--- (15 headers 0 lines) ---
Sending to MY_REAL_IP:49176 (no NAT)
<--- Transmitting (no NAT) to MY_REAL_IP:49176 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP MY_REAL_IP:49176;branch=z9hG4bK0ceca61c;received=MY_REAL_IP
From: <sip:[email protected]:5060>;tag=00070e36ee480003e64d4b0e-f0245601
To: <sip:[email protected]:5060>;tag=as506b1cee
Call-ID: [email protected]_REAL_IP
CSeq: 102 REGISTER
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Date: Fri, 29 Apr 2016 08:12:31 GMT
Content-Length: 0
Didn't find what you were looking for?
Ask your questionAsk a Question
731 491 924 answers to any question