Hi,
Many times i found that my extension says busy or unavailable and it goes to voice mail, whether my X-lite phone is registered properly and its free. I have checked with Linksys PAP2 as well but same, sometimes it works fine. I don’t understand what could be problem with that,
One more thing i noticed that on X-Lite all codes work like *43(echo) *79(DND disable) but same never works on Linksys PAP2
Please help
could be nat mappings. set qualify=yes in sip.conf to keep it open.
as for the star codes, that is the pap2 thinking you are trying to control it… set them to be normal (non-star) extension or modify the pap2 dialplan to includ the star codes.
HI,
Thanks for the posting reply,
but still issue is not resolved i don’t understand what is the problem.
try sip show peers and see what it says for the sip peer?
also do a sip debug peer peername and see what happens when you try to call it… post htat here…
HI,
10.10.10.10 : IP address of my Asterisk box
192.168.14.3 : My local IP adress
0091123456789 : PSTN number from which i dialed number
One more thing i would like to mention here that i have only one ISP connection and from same router i am using Port forwarding
to run my Asterisk and same LAN i have my extensions registered, some of extensions have NAT= always and some has
NAT=yes, but both conditions it says extension is busy, following is logs for extension, Please help!!!
[color=red]asteriskCLI> sip debug peer 201
SIP Debugging Enabled for IP: 10.10.10.10:63546
recordingcheck|20060911-100015|1157949015.826: No DB Entry AMPUSER//recording - Not recording
recordingcheck|20060911-100016|1157949016.827: No DB Entry AMPUSER//recording - Not recording
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘0091123456789’ number is '0091123456789’
dialparties.agi: Methodology of ring is 'ringall’
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘0091123456789’ number is '0091123456789’
dialparties.agi: Methodology of ring is 'ringall’
recordingcheck|20060911-100016|1157949015.826: Inbound recording not enabled
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘0091123456789’ number is '0091123456789’
dialparties.agi: Methodology of ring is 'none’
recordingcheck|20060911-100017|1157949016.827: Inbound recording not enabled
dialparties.agi: Extension 201 is not available to be called
dialparties.agi: Extension 201 has call waiting disabled
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘0091123456789’ number is '0091123456789’
dialparties.agi: Methodology of ring is 'none’
dialparties.agi: Extension 201 is not available to be called
dialparties.agi: Extension 201 has call waiting disabled
asteriskCLI>
<-- SIP read from 10.10.10.10:63546:
REGISTER sip:10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.3:5060;branch=z9hG4bK-2bba1322
From: VoIP Telecom sip:201@10.10.10.10;tag=3f1526ebae86c671o0
To: VoIP Telecom sip:201@10.10.10.10
Call-ID: 4feb2853-cf67429@192.168.14.3
CSeq: 7 REGISTER
Max-Forwards: 70
Authorization: Digest username=“201”,realm=“asterisk”,nonce=“3317e0fe”,uri="sip:201@10.10.10.10",algorithm=MD5,response="60f29d6c2e6dc5241ac3247d512b05c4"
Contact: VoIP Telecom sip:201@192.168.14.3:5060;expires=180
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
— (13 headers 0 lines)—
Using latest REGISTER request as basis request
Transmitting (NAT) to 10.10.10.10:63546:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.3:5060;branch=z9hG4bK-2bba1322;received=10.10.10.10
From: VoIP Telecom sip:201@10.10.10.10;tag=3f1526ebae86c671o0
To: VoIP Telecom sip:201@10.10.10.10
Call-ID: 4feb2853-cf67429@192.168.14.3
CSeq: 7 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:201@10.10.10.10
Content-Length: 0
Transmitting (NAT) to 10.10.10.10:63546:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.14.3:5060;branch=z9hG4bK-2bba1322;received=10.10.10.10
From: VoIP Telecom sip:201@10.10.10.10;tag=3f1526ebae86c671o0
To: VoIP Telecom sip:201@10.10.10.10;tag=as517ab026
Call-ID: 4feb2853-cf67429@192.168.14.3
CSeq: 7 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:201@10.10.10.10
WWW-Authenticate: Digest realm=“asterisk”, nonce="7223bda3"
Content-Length: 0
Scheduling destruction of call ‘4feb2853-cf67429@192.168.14.3’ in 15000 ms
asterisk*CLI>
<-- SIP read from 10.10.10.10:63546:
REGISTER sip:10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.3:5060;branch=z9hG4bK-e895d9f7
From: VoIP Telecom sip:201@10.10.10.10;tag=3f1526ebae86c671o0
To: VoIP Telecom sip:201@10.10.10.10
Call-ID: 4feb2853-cf67429@192.168.14.3
CSeq: 8 REGISTER
Max-Forwards: 70
Authorization: Digest username=“201”,realm=“asterisk”,nonce=“7223bda3”,uri="sip:201@10.10.10.10",algorithm=MD5,response="2fa977c9f75e232684e52883c4695b63"
Contact: VoIP Telecom sip:201@192.168.14.3:5060;expires=180
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
— (13 headers 0 lines)—
Using latest REGISTER request as basis request
Sending to 192.168.14.3 : 5060 (NAT)
Transmitting (NAT) to 10.10.10.10:63546:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.3:5060;branch=z9hG4bK-e895d9f7;received=10.10.10.10
From: VoIP Telecom sip:201@10.10.10.10;tag=3f1526ebae86c671o0
To: VoIP Telecom sip:201@10.10.10.10
Call-ID: 4feb2853-cf67429@192.168.14.3
CSeq: 8 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:201@10.10.10.10
Content-Length: 0
Transmitting (NAT) to 10.10.10.10:63546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.3:5060;branch=z9hG4bK-e895d9f7;received=10.10.10.10
From: VoIP Telecom sip:201@10.10.10.10;tag=3f1526ebae86c671o0
To: VoIP Telecom sip:201@10.10.10.10;tag=as517ab026
Call-ID: 4feb2853-cf67429@192.168.14.3
CSeq: 8 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 180
Contact: sip:201@192.168.14.3:5060;expires=180
Date: Mon, 11 Sep 2006 04:30:25 GMT
Content-Length: 0
Scheduling destruction of call ‘4feb2853-cf67429@192.168.14.3’ in 15000 ms
12 headers, 3 lines
Reliably Transmitting (NAT) to 10.10.10.10:63546:
NOTIFY sip:201@192.168.14.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7b9cddb6;rport
From: “Private” sip:Private@10.10.10.10;tag=as5529bc71
To: sip:201@192.168.14.3:5060
Contact: sip:Private@10.10.10.10
Call-ID: 19f6f31767d41c915628899669c35527@10.10.10.10
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93
Messages-Waiting: no
Message-Account: sip:asterisk@10.10.10.10
Voice-Message: 0/0 (0/0)
Scheduling destruction of call ‘19f6f31767d41c915628899669c35527@10.10.10.10’ in 15000 ms
asterisk*CLI>
<-- SIP read from 10.10.10.10:63546:
SIP/2.0 200 OK
To: sip:201@192.168.14.3:5060;tag=274d83f3973ecd09i0
From: “Private” sip:Private@10.10.10.10;tag=as5529bc71
Call-ID: 19f6f31767d41c915628899669c35527@10.10.10.10
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7b9cddb6
Server: Linksys/PAP2-2.0.12(LS)
Content-Length: 0
— (8 headers 0 lines)—
Destroying call ‘19f6f31767d41c915628899669c35527@10.10.10.10’
– x=0, open writing: /var/spool/asterisk/voicemail/technical/201/tmp/epnipb format: wav49, 0x8c9eab8
– x=1, open writing: /var/spool/asterisk/voicemail/technical/201/tmp/epnipb format: wav, 0x8ceceb0
Destroying call '4feb2853-cf67429@192.168.14.3’
asterisk*CLI>[/color]