[Asterisk-Users] RE: SIP Peers unreachable

Chris chris_veaudry at sympatico.ca
Fri May 2 14:46:55 MST 2003 

(Sorry if this breaks the thread. I receive the list mail by the daily
digest.)

I removed the qualifier from all my SIP extensions in the sip.conf as
suggested by both John and Mark. Still the same issue.

It's interesting, Mark, that you mentioned that the clients may not support
OPTIONS. When I turn on SIP debug, I get 3 retransmitions of CSeq: 102
OPTIONS in the console for each extension. Not sure what this means.

Here is the dump from tethereal while calling 421 from 422. It does look
like the SIP request are making it to the destination.

[root at homegw asterisk requirements]# tethereal port 5060
Capturing on eth0
  0.000000     pc-00033 -> homegw.defrag.homeip.net SIP/SDP Request: INVITE
sip:
421 at 192.200.14.251, with session description
  0.036408 homegw.defrag.homeip.net -> pc-00033     SIP Status: 407 Proxy
Authen
tication Required
  0.042191     pc-00033 -> homegw.defrag.homeip.net SIP Request: ACK
sip:421 at 172
.20.14.251
  0.045378 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  0.049398 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  0.252140     pc-00033 -> homegw.defrag.homeip.net SIP/SDP Request: INVITE
sip:
421 at 192.200.14.251, with session description
  0.363390 homegw.defrag.homeip.net -> pc-00033     SIP Status: 100 Trying
  0.710967 homegw.defrag.homeip.net -> pc-00033     SIP/SDP Status: 200 OK,
with
 session description
  0.772759     pc-00033 -> homegw.defrag.homeip.net SIP Request: ACK
sip:421 at 172
.20.14.251
  1.050840 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  1.051029 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  2.060714 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  2.060900 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  3.070620 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  3.070807 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  4.734937     pc-00033 -> homegw.defrag.homeip.net SIP Request: BYE
sip:421 at 172
.20.14.251
  4.737003 homegw.defrag.homeip.net -> pc-00033     SIP Status: 200 OK
  4.759445     pc-00033 -> homegw.defrag.homeip.net SIP Request: ACK
sip:421 at 172
.20.14.251
 14.081490 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 14.085292 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
 15.089388 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 15.089571 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
 16.099288 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 16.099471 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
 17.109195 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 17.109388 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
[root at homegw asterisk requirements]#

Here is a capture from the console with SIP Debug turned on.

 to 192.200.14.33:5060
Sip read: >
ACK sip:421 at 192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>;tag=as23f24230
Contact: <sip:sippc at 192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21726 ACK
Max-Forwards: 70
Content-Length: 0


9 headers, 0 lines
Sip read: >
INVITE sip:421 at 192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>
Contact: <sip:sippc at 192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21727 INVITE
Proxy-Authorization: Digest
username="sippc",realm="asterisk",nonce="788a7172",response="1b6ff26ec4f095b
723d429306edca5d9",uri="sip:421 at 192.200.14.251"
Content-Type: application/sdp
Content-Length: 229

v=0
o=sippc 39160289 39160289 IN IP4 192.200.14.33
s=X-Lite
c=IN IP4 192.200.14.33
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:126 x-pro-encrypted/8000

10 headers, 10 lines
Using latest request as basis request
Sending to 192.200.14.33 : 5060 (non-NAT)
Capabilities: us - 14, them - 4, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 421 in default
list_route: hop: <sip:sippc at 192.200.14.33:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>;tag=as011374f6
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21727 INVITE
User-Agent: Asterisk PBX
Contact: <sip:421 at 192.200.14.251>
Content-Length: 0


 to 192.200.14.33:5060
    -- Executing Dial("SIP/sippc-040b", "SIP/sipset") in new stack
  == Everyone is busy at this time
    -- Executing VoiceMail("SIP/sippc-040b", "b421") in new stack
We're at 192.200.14.251 port 11980
Answering with capability 4
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>;tag=as011374f6
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21727 INVITE
User-Agent: Asterisk PBX
Contact: <sip:421 at 192.200.14.251>
Content-Type: application/sdp
Content-Length: 191

v=0
o=root 24171 24171 IN IP4 192.200.14.251
s=session
c=IN IP4 192.200.14.251
t=0 0
m=audio 11980 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 192.200.14.33:5060
  == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
'/etc/asterisk/voicemail.conf': Found
Sip read: >
ACK sip:421 at 192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>;tag=as011374f6
Contact: <sip:sippc at 192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21727 ACK
Max-Forwards: 70
Content-Length: 0


9 headers, 0 lines
    -- Playing 'vm-theperson'
Retransmitting #1 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK18327382
From: "asterisk" <sip:asterisk at 192.200.14.251>;tag=as4dbb6c0a
To: <sip:>
Contact: <sip:asterisk at 192.200.14.251>
Call-ID: 4683962a3c7615a76bf2643b0823b915 at 192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.31:5060
Retransmitting #1 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK0ab60e7d
From: "asterisk" <sip:asterisk at 192.200.14.251>;tag=as26d1d313
To: <sip:>
Contact: <sip:asterisk at 192.200.14.251>
Call-ID: 62829f7e5be8b4292b9760a477045992 at 192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.33:5060
Retransmitting #2 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK18327382
From: "asterisk" <sip:asterisk at 192.200.14.251>;tag=as4dbb6c0a
To: <sip:>
Contact: <sip:asterisk at 192.200.14.251>
Call-ID: 4683962a3c7615a76bf2643b0823b915 at 192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.31:5060
Retransmitting #2 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK0ab60e7d
From: "asterisk" <sip:asterisk at 192.200.14.251>;tag=as26d1d313
To: <sip:>
Contact: <sip:asterisk at 192.200.14.251>
Call-ID: 62829f7e5be8b4292b9760a477045992 at 192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.33:5060
    -- Playing 'digits/4'
Sip read: >
BYE sip:421 at 192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>;tag=as011374f6
Contact: <sip:sippc at 192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21728 BYE
Content-Length: 0


8 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>;tag=as011374f6
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21728 BYE
User-Agent: Asterisk PBX
Contact: <sip:421 at 192.200.14.251>
Content-Length: 0


 to 192.200.14.33:5060
  == Spawn extension (default, 421, 102) exited non-zero on 'SIP/sippc-040b'
Sip read:
ACK sip:421 at 192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc at 192.200.14.251>;tag=939212736
To: <sip:421 at 192.200.14.251>;tag=as011374f6
Contact: <sip:sippc at 192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C at 192.200.14.33
CSeq: 21728 ACK
Max-Forwards: 70
Content-Length: 0


9 headers, 0 lines
Retransmitting #3 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK18327382
From: "asterisk" <sip:asterisk at 192.200.14.251>;tag=as4dbb6c0a
To: <sip:>
Contact: <sip:asterisk at 192.200.14.251>
Call-ID: 4683962a3c7615a76bf2643b0823b915 at 192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.31:5060
Retransmitting #3 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK0ab60e7d
From: "asterisk" <sip:asterisk at 192.200.14.251>;tag=as26d1d313
To: <sip:>
Contact: <sip:asterisk at 192.200.14.251>
Call-ID: 62829f7e5be8b4292b9760a477045992 at 192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.33:5060
homegw*CLI>

Thanks for your assitance.

Cheers,

Chris

On Fri, 2 May 2003, Chris wrote:

> Hi Everyone,
>
> I'm new to * and I'm trying to setup a small configuration of SIP clients.
> Eventually when I get this working I plan on expanding with a Digium
> developers kit to add analog phones and PSTN access.
>
> My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
> peers seem to register with * but I cannot call to one another. When I
dial
> the associated extension, the call goes to the programmed voicemail
> extension (busy) yet if I create an extension to call out through the
proxy
> (IX66), I can still reach my destination. It's just calling within * there
> is a problem. I suspect it's because the status is unreachable but I'm not
> sure how to fix it.
>
> Here is the sip show peers output.
>
> Name/username    Host                 Mask             Port     Status
> sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060
UNREACHABLE
> sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060
UNREACHABLE
>
> Here is the sip.conf settings:
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = default
> register => 9055551212 at somewhere.homeip.net
>
> [sippc]
> type=friend
> username=sippc
> secret=blah
> host=dynamic
> qualify=3000
>
> [sipset]
> type=friend
> username=sipset
> secret=blah
> host=dynamic
> qualify=3000
>
> Here is the extensions.conf settings:
> exten => 421,1,Dial(SIP/sipset)  	  ; Mitel 5055 SIP Phone
> exten => 421,2,Voicemail(u421)
> exten => 421,102,Voicemail(b421)
> exten => 422,1,Dial(SIP/sippc)  	  ; Xten client
> exten => 422,2,Voicemail(u422)
> exten => 422,102,Voicemail(b422)
>
> exten => 444,1,Dial(SIP/tony at somewhere.homeip.net)  ; friends MSN (4.6)
> account registered to IX66
>
>
> These are the console messages when I dial 421 from 422
>
>     -- Executing Dial("SIP/sippc-b5f6", "SIP/sipset") in new stack
>   == Everyone is busy at this time
>     -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in new stack
>   == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
> '/etc/asterisk/voicemail.conf': Found
>     -- Playing 'vm-theperson'
>     -- Playing 'digits/4'
>     -- Playing 'digits/2'
>     -- Playing 'digits/1'
>     -- Playing 'vm-isonphone'
>     -- Playing 'vm-intro'
>   == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-b5f6'
>
>
> Any help is appreciated.
>
> Thanks.
>
> Chris
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>



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