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Jerry Geis

2008-Sep-26 17:25 UTC

[asterisk-users] server and 2 uniden phones no ringing

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).

Mark Michelson

2008-Sep-26 19:44 UTC

head link

[asterisk-users] server and 2 uniden phones no ringing

Jerry Geis wrote:> 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
<snip>

Based on the SIP debug included here, it appears that Asterisk is not receiving 
a response to the INVITE it is sending to 522 (192.168.1.99). Since the phone is
not ringing, it makes me suspect that for some reason the linksys is preventing 
the INVITE from reaching the phone. If you can look at a packet capture on the 
linksys, you may want to verify that the linksys isn't modifying or blocking
the
INVITE that Asterisk is sending.

Mark Michelson

Jerry Geis

2008-Sep-26 20:04 UTC

head link

[asterisk-users] server and 2 uniden phones no ringing

>
> <snip>
>
> Based on the SIP debug included here, it appears that Asterisk is not
receiving
> a response to the INVITE it is sending to 522 (192.168.1.99). Since the
phone is
> not ringing, it makes me suspect that for some reason the linksys is
preventing
> the INVITE from reaching the phone. If you can look at a packet capture on
the
> linksys, you may want to verify that the linksys isn't modifying or
blocking the
> INVITE that Asterisk is sending.
>
> Mark Michelson
>
>   Mark,

Thanks, I looked at the Linksys WRT54G wireless router.
I see nothing that would stop the invite.
I have disabled the firewall on the server.
I have updated to 1.4.21.1
Still same behavior.

I can call into the dialplay and hear audio of some playback wave file.
I just cant call another phone. it never rings.

The linksys router is set as the default and DHCP is turned off as my 
server is
providing DHCP.

Any other toughts?

Jerry

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asterisk users - Sep 2008 - server and 2 uniden phones no ringing

[asterisk-users] server and 2 uniden phones no ringing

[asterisk-users] server and 2 uniden phones no ringing

[asterisk-users] server and 2 uniden phones no ringing

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