[asterisk-users] server and 2 uniden phones no ringing
Jerry Geis
geisj at pagestation.com
Fri Sep 26 12:25:56 CDT 2008
I have a box running asterisk 1.4.17 that had been working.
it has 2 uniden phones connected on it.
This was working and now the phones dont ring when calling each other.
below is the sip debug. I cant see why the other phone does not ring?
I also tried changing the canreinvite for no to yes but that made no
difference after restarting.
Very simple network. server, linksys router and 2 phones. 192.168.1.X
for everything.
Any ideas?
Jerry
[522]
type=friend
username=522
secret=522
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid="522 522" <522>
qualify=no
canreinvite=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[532]
type=friend
username=532
secret=532
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
canreinvite=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
demobox*CLI>
<--- SIP read from 192.168.1.75:5060 --->
INVITE sip:522 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKz8184a8a520f49fe1a53a4d9647a51fb7
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
CSeq: 420456 INVITE
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
To: <sip:522 at 192.168.1.150>
Contact: <sip:532 at 192.168.1.75:5060>
Session-Expires: 300
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
S
[Kdemobox*CLI>
upported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 269
v=0
o=- 1794556993 298723 IN IP4 192.168.1.75
s=-
c=IN IP4 192.168.1.75
t=0 0
m=audio 30006 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=sendrecv
a=ptime:20
<------------->
?--- (14 headers 13 lines) ---
?
[Kdemobox*CLI>
Sending to 192.168.1.75 : 5060 (no NAT)
?
[Kdemobox*CLI>
Using INVITE request as basis request - 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
?
[Kdemobox*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.75:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKz8184a8a520f49fe1a53a4d9647a51fb7;received=192.168.1.75
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
To: <sip:522 at 192.168.1.150>;tag=as15ac0056
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
CSeq: 420456 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ebfd7ed"
Content-Length: 0
<------------>
?
[Kdemobox*CLI>
Scheduling destruction of SIP dialog '1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150' in 32000 ms (Method: INVITE)
?Found user '532'
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.75:5060 --->
ACK sip:522 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKz8184a8a520f49fe1a53a4d9647a51fb7
CSeq: 420456 ACK
To: <sip:522 at 192.168.1.150>;tag=as15ac0056
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
User-Agent: Uniden SIP Phone p2 Ver BS4.77
<------------->
?--- (7 headers 0 lines) ---
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.75:5060 --->
INVITE sip:522 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKvf10df6048844205ddfc1a15bace4c673
CSeq: 420457 INVITE
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
To: <sip:522 at 192.168.1.150>
Contact: <sip:532 at 192.168.1.75:5060>
Session-Expires: 300
Content-Type: application/sdp
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
S
[Kdemobox*CLI>
upported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 269
Proxy-Authorization: Digest realm="asterisk", nonce="2ebfd7ed", algorithm=MD5, uri="sip:522 at 192.168.1.150", username="532", response="301dfbf68f00b164f64effa90188bf58"
v=0
o=- 1794556993 298723 IN IP4 192.168.1.75
s=-
c=IN IP4 192.168.1.75
t=0 0
m=audio 30006 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=sendrecv
a=ptime:20
<------------->
?--- (15 headers 13 lines) ---
?
[Kdemobox*CLI>
Sending to 192.168.1.75 : 5060 (no NAT)
?Using INVITE request as basis request - 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
?
[Kdemobox*CLI>
Found user '532'
?
[Kdemobox*CLI>
Found RTP audio format 0
?
[Kdemobox*CLI>
Found RTP audio format 8
?Found RTP audio format 18
?Found RTP audio format 101
?Peer audio RTP is at port 192.168.1.75:30006
?
[Kdemobox*CLI>
Found audio description format PCMU for ID 0
?Found audio description format PCMA for ID 8
?Found audio description format G729 for ID 18
?Found audio description format telephone-event for ID 101
?
[Kdemobox*CLI>
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
?Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
?
[Kdemobox*CLI>
Peer audio RTP is at port 192.168.1.75:30006
?Looking for 522 in smvoice-sip (domain 192.168.1.150)
?
[Kdemobox*CLI>
list_route: hop: <sip:532 at 192.168.1.75:5060>
?
[Kdemobox*CLI>
<--- Transmitting (no NAT) to 192.168.1.75:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKvf10df6048844205ddfc1a15bace4c673;received=192.168.1.75
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
To: <sip:522 at 192.168.1.150>
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
CSeq: 420457 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:522 at 192.168.1.150>
Content-Length: 0
<------------>
?
[Kdemobox*CLI>
-- Executing [522 at smvoice-sip:1] NoOp("SIP/532-009a1120", "5xx") in new stack
?
[Kdemobox*CLI>
-- Executing [522 at smvoice-sip:2] Set("SIP/532-009a1120", "SMVOICE_CONTEXT_EXTEN=522") in new stack
?
[Kdemobox*CLI>
-- Executing [522 at smvoice-sip:3] AGI("SIP/532-009a1120", "smvoice|-digium_asterisk|-asterisk_callat_forwarding|522") in new stack
?
[Kdemobox*CLI>
-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
?
[Kdemobox*CLI>
-- AGI Script smvoice completed, returning 0
?
[Kdemobox*CLI>
-- Executing [522 at smvoice-sip:4] GotoIf("SIP/532-009a1120", "0?INVALID|1") in new stack
? -- Executing [522 at smvoice-sip:5] GotoIf("SIP/532-009a1120", "0?_5XX-NOANSWER|1") in new stack
?
[Kdemobox*CLI>
-- Executing [522 at smvoice-sip:6] Dial("SIP/532-009a1120", "SIP/522|20|") in new stack
?
[Kdemobox*CLI>
Audio is at 192.168.1.150 port 10010
?
[Kdemobox*CLI>
Adding codec 0x4 (ulaw) to SDP
?Adding codec 0x8 (alaw) to SDP
?Adding codec 0x2 (gsm) to SDP
?
[Kdemobox*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
?
[Kdemobox*CLI>
Reliably Transmitting (no NAT) to 192.168.1.99:5060:
INVITE sip:522 at 192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK386c8038;rport
From: "532" <sip:532 at 192.168.1.150>;tag=as769c48e6
To: <sip:522 at 192.168.1.99:5060>
Contact: <sip:532 at 192.168.1.150>
Call-ID: 715fe67f020f7a2c7035ecc668354736 at 192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Sep 2008 17:08:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 9808 9808 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 10010 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
?
[Kdemobox*CLI>
-- Called 522
?
[Kdemobox*CLI>
Retransmitting #1 (no NAT) to 192.168.1.99:5060:
INVITE sip:522 at 192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK386c8038;rport
From: "532" <sip:532 at 192.168.1.150>;tag=as769c48e6
To: <sip:522 at 192.168.1.99:5060>
Contact: <sip:532 at 192.168.1.150>
Call-ID: 715fe67f020f7a2c7035ecc668354736 at 192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Sep 2008 17:08:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 9808 9808 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 10010 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
?
[Kdemobox*CLI>
Retransmitting #2 (no NAT) to 192.168.1.99:5060:
INVITE sip:522 at 192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK386c8038;rport
From: "532" <sip:532 at 192.168.1.150>;tag=as769c48e6
To: <sip:522 at 192.168.1.99:5060>
Contact: <sip:532 at 192.168.1.150>
Call-ID: 715fe67f020f7a2c7035ecc668354736 at 192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Sep 2008 17:08:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 9808 9808 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 10010 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
?
[Kdemobox*CLI>
Retransmitting #3 (no NAT) to 192.168.1.99:5060:
INVITE sip:522 at 192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK386c8038;rport
From: "532" <sip:532 at 192.168.1.150>;tag=as769c48e6
To: <sip:522 at 192.168.1.99:5060>
Contact: <sip:532 at 192.168.1.150>
Call-ID: 715fe67f020f7a2c7035ecc668354736 at 192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Sep 2008 17:08:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 9808 9808 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 10010 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
?
[Kdemobox*CLI>
Really destroying SIP dialog '2d0eeb2d37a1f17e1b151f1c3fd070e5 at 192.168.1.150' Method: INVITE
?
[Kdemobox*CLI>
Retransmitting #4 (no NAT) to 192.168.1.99:5060:
INVITE sip:522 at 192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK386c8038;rport
From: "532" <sip:532 at 192.168.1.150>;tag=as769c48e6
To: <sip:522 at 192.168.1.99:5060>
Contact: <sip:532 at 192.168.1.150>
Call-ID: 715fe67f020f7a2c7035ecc668354736 at 192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Sep 2008 17:08:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 9808 9808 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 10010 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.75:5060 --->
CANCEL sip:522 at 192.168.1.150 SIP/2.0
CSeq: 420457 CANCEL
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKvf10df6048844205ddfc1a15bace4c673
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
To: <sip:522 at 192.168.1.150>
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Proxy-Authorization: Digest realm="asterisk", nonce="2ebfd7ed", algorithm=MD5, uri="sip:522 at 192.168.1.150", username="532", response="301dfbf68f00b164f64effa90188bf58"
<------------->
?--- (10 headers 0 lines) ---
?
[Kdemobox*CLI>
Sending to 192.168.1.75 : 5060 (no NAT)
?
[Kdemobox*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.75:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKvf10df6048844205ddfc1a15bace4c673;received=192.168.1.75
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
To: <sip:522 at 192.168.1.150>;tag=as10f53df8
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
CSeq: 420457 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
?
[Kdemobox*CLI>
<--- Transmitting (no NAT) to 192.168.1.75:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKvf10df6048844205ddfc1a15bace4c673;received=192.168.1.75
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
To: <sip:522 at 192.168.1.150>;tag=as10f53df8
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
CSeq: 420457 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:522 at 192.168.1.150>
Content-Length: 0
<------------>
?
[Kdemobox*CLI>
Scheduling destruction of SIP dialog '715fe67f020f7a2c7035ecc668354736 at 192.168.1.150' in 32000 ms (Method: INVITE)
?
[Kdemobox*CLI>
== Spawn extension (smvoice-sip, 522, 6) exited non-zero on 'SIP/532-009a1120'
?
[Kdemobox*CLI>
<--- SIP read from 192.168.1.75:5060 --->
ACK sip:522 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKvf10df6048844205ddfc1a15bace4c673
CSeq: 420457 ACK
To: <sip:522 at 192.168.1.150>;tag=as10f53df8
Call-ID: 1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150
From: 532 <sip:532 at 192.168.1.150>;tag=6619ac3b4bbd705d7102c4565d72e1bc
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Proxy-Authorization: Digest realm="asterisk", nonce="2ebfd7ed", algorithm=MD5, uri="sip:522 at 192.168.1.150", username="532", response="301dfbf68f00b164f64effa90188bf58"
<------------->
?--- (8 headers 0 lines) ---
?
[Kdemobox*CLI>
Really destroying SIP dialog '1e35a5b8b01e838b5f192b29044d17e4-55ae664765b4 at 192.168.1.150' Method: ACK
?
[Kdemobox*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
[0m
More information about the asterisk-users
mailing list