[asterisk-users] adding a second extension
Stephen Reese
rsreese at gmail.com
Thu Oct 23 12:47:21 CDT 2008
On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez <jerdguez at gmail.com> wrote:
> And this phone are connected in a local LAN??
> Because I see Asterisk receiving a "Bad request" from 68.156.63.118
> If those phones are not in your local LAN, try with a soft phone first.
> Could be Zoiper or Xlite.
> Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101
> sending a "400 Bad request" back to Asterisk.
>
Both of these phones are on my local lan but the Asterisk server is at
a colo facility on the internet outside of the local lan. The local
lan does use NAT/PAT. I see an error "Warning: 399 Bad Request -
'Malformed/Missing FROM: field'. Is this a problem?
Thanks
---
ns1*CLI>
<--- SIP read from 68.156.63.118:1082 --->
INVITE sip:101 at neocipher.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
To: <sip:101 at neocipher.net;user=phone>
Call-ID: 1279503304 at 172.16.2.18
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:102 at 68.156.63.118:1083;user=phone;transport=udp>
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Expires: 300
Content-Length: 274
Content-Type: application/sdp
v=0
o=102 157742 157742 IN IP4 172.16.2.18
s=Cisco 7912 SIP Call
c=IN IP4 68.156.63.118
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 12 lines) ---
Sending to 68.156.63.118 : 1083 (no NAT)
Using INVITE request as basis request - 1279503304 at 172.16.2.18
<--- Reliably Transmitting (NAT) to 68.156.63.118:1082 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118
From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
To: <sip:101 at neocipher.net;user=phone>;tag=as355e0f84
Call-ID: 1279503304 at 172.16.2.18
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="ns1.neocipher.net",
nonce="7c2e1ba9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1279503304 at 172.16.2.18' in 32000
ms (Method: INVITE)
Found user '102'
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: <sip:rsreese at inbound18.vitelity.net>;tag=as401a34d4
To: <sip:rsreese at inbound18.vitelity.net>
Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:rsreese at 64.2.142.116>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: <sip:rsreese at inbound18.vitelity.net>;tag=as401a34d4
To: <sip:rsreese at inbound18.vitelity.net>;tag=as7a2f92a1
Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="575628ec"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name inbound18.vitelity.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.2.142.116:5060:
REGISTER sip:inbound18.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport
From: <sip:rsreese at inbound18.vitelity.net>;tag=as751cb0af
To: <sip:rsreese at inbound18.vitelity.net>
Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="rsreese", realm="asterisk",
algorithm=MD5, uri="sip:inbound18.vitelity.net", nonce="575628ec",
response="b765dbdebba8af18b19707efe651d65d"
Expires: 120
Contact: <sip:rsreese at 209.251.157.91>
Event: registration
Content-Length: 0
---
<--- SIP read from 68.156.63.118:1082 --->
ACK sip:101 at neocipher.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
To: <sip:101 at neocipher.net;user=phone>;tag=as355e0f84
Call-ID: 1279503304 at 172.16.2.18
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
ns1*CLI>
<--- SIP read from 68.156.63.118:1082 --->
INVITE sip:101 at neocipher.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2
From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
To: <sip:101 at neocipher.net;user=phone>
Call-ID: 1279503304 at 172.16.2.18
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:102 at 68.156.63.118:1083;user=phone;transport=udp>
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest
username="102",realm="ns1.neocipher.net",nonce="7c2e1ba9",uri="sip:101 at neocipher.net",response="105dfec593cbcfac83380461870c3a07"
Expires: 300
Content-Length: 274
Content-Type: application/sdp
v=0
o=102 157750 157750 IN IP4 172.16.2.18
s=Cisco 7912 SIP Call
c=IN IP4 68.156.63.118
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 12 lines) ---
Sending to 68.156.63.118 : 1082 (NAT)
Using INVITE request as basis request - 1279503304 at 172.16.2.18
Found user '102'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 68.156.63.118:16384
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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)
Peer audio RTP is at port 68.156.63.118:16384
Looking for 101 in default (domain neocipher.net)
list_route: hop: <sip:102 at 68.156.63.118:1083;user=phone;transport=udp>
<--- Transmitting (NAT) to 68.156.63.118:1082 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2;received=68.156.63.118
From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
To: <sip:101 at neocipher.net;user=phone>
Call-ID: 1279503304 at 172.16.2.18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101 at 209.251.157.91>
Content-Length: 0
<------------>
-- Executing [101 at default:1] Dial("SIP/102-081e4968",
"SIP/101&SIP/9046260705 at vitel-outbound|30") in new stack
Audio is at 209.251.157.91 port 10532
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 68.156.63.118:1038:
INVITE sip:101 at 68.156.63.118:1039;transport=udp SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
From: "Stephen\" <sip:102 at 209.251.157.91>;tag=as5b299b78
To: <sip:101 at 68.156.63.118:1039;transport=udp>
Contact: <sip:102 at 209.251.157.91>
Call-ID: 7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 23 Oct 2008 17:39:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 5235 5235 IN IP4 209.251.157.91
s=session
c=IN IP4 209.251.157.91
t=0 0
m=audio 10532 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 101
Audio is at 209.251.157.91 port 18610
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.2.142.17:5060:
INVITE sip:9046260705 at outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK1b46ea9b;rport
From: "Stephen\" <sip:rsreese at 209.251.157.91>;tag=as0f273e60
To: <sip:9046260705 at outbound.vitelity.net>
Contact: <sip:rsreese at 209.251.157.91>
Call-ID: 02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Stephen\" <sip:102 at 209.251.157.91>;privacy=off;screen=no
Date: Thu, 23 Oct 2008 17:39:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 428
v=0
o=root 5235 5235 IN IP4 209.251.157.91
s=session
c=IN IP4 209.251.157.91
t=0 0
m=audio 18610 RTP/AVP 0 3 8 112 5 10 7 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 9046260705 at vitel-outbound
ns1*CLI>
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK6245e988;received=209.251.157.91;rport=5060
From: <sip:rsreese at inbound18.vitelity.net>;tag=as751cb0af
To: <sip:rsreese at inbound18.vitelity.net>
Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:rsreese at 64.2.142.116>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
ns1*CLI>
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK6245e988;received=209.251.157.91;rport=5060
From: <sip:rsreese at inbound18.vitelity.net>;tag=as751cb0af
To: <sip:rsreese at inbound18.vitelity.net>;tag=as7a2f92a1
Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:rsreese at 209.251.157.91>;expires=60
Date: Thu, 23 Oct 2008 17:27:25 GMT
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog
'44561e1779828d6773b538cd497220d0 at 209.251.157.91' in 32000 ms (Method:
REGISTER)
[Oct 23 13:39:52] NOTICE[5264]: chan_sip.c:12682
handle_response_register: Outbound Registration: Expiry for
inbound18.vitelity.net is 60 sec (Scheduling reregistration in 45 s)
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK1b46ea9b;received=209.251.157.91;rport=5060
From: "Stephen\" <sip:rsreese at 209.251.157.91>;tag=as0f273e60
To: <sip:9046260705 at outbound.vitelity.net>;tag=as0d5d9875
Call-ID: 02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 64.2.142.17:5060:
ACK sip:9046260705 at outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK1b46ea9b;rport
From: "Stephen\" <sip:rsreese at 209.251.157.91>;tag=as0f273e60
To: <sip:9046260705 at outbound.vitelity.net>;tag=as0d5d9875
Contact: <sip:rsreese at 209.251.157.91>
Call-ID: 02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Stephen\" <sip:102 at 209.251.157.91>;privacy=off;screen=no
Content-Length: 0
---
-- SIP/vitel-outbound-081f9240 is circuit-busy
ns1*CLI>
<--- SIP read from 68.156.63.118:1038 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
From: "Stephen\" <sip:102 at 209.251.157.91>;tag=as5b299b78
To: <sip:101 at 68.156.63.118:1039;transport=udp>
Call-ID: 7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91
Warning: 399 Bad Request - 'Malformed/Missing FROM: field'
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 400 "Bad Request" back from 68.156.63.118
Transmitting (NAT) to 68.156.63.118:1038:
ACK sip:101 at 68.156.63.118:1039;transport=udp SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
From: "Stephen\" <sip:102 at 209.251.157.91>;tag=as5b299b78
To: <sip:101 at 68.156.63.118:1039;transport=udp>
Contact: <sip:102 at 209.251.157.91>
Call-ID: 7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog
'02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91' Method: INVITE
-- SIP/101-08195e68 is circuit-busy
== Everyone is busy/congested at this time (2:0/2/0)
-- Executing [101 at default:2] GotoIf("SIP/102-081e4968",
"0?lbl_default_1:") in new stack
-- Executing [101 at default:3] GotoIf("SIP/102-081e4968",
"0?lbl_default_1:") in new stack
-- Executing [101 at default:4] Hangup("SIP/102-081e4968", "") in new stack
== Spawn extension (default, 101, 4) exited non-zero on 'SIP/102-081e4968'
Scheduling destruction of SIP dialog '1279503304 at 172.16.2.18' in 32000
ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 68.156.63.118:1082 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2;received=68.156.63.118
From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
To: <sip:101 at neocipher.net;user=phone>;tag=as74c3be83
Call-ID: 1279503304 at 172.16.2.18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101 at 209.251.157.91>
Content-Length: 0
<------------>
Really destroying SIP dialog
'7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91' Method: INVITE
ns1*CLI>
<--- SIP read from 68.156.63.118:1082 --->
ACK sip:101 at neocipher.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2
From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
To: <sip:101 at neocipher.net;user=phone>;tag=as74c3be83
Call-ID: 1279503304 at 172.16.2.18
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Proxy-Authorization: Digest
username="102",realm="ns1.neocipher.net",nonce="7c2e1ba9",uri="sip:101 at neocipher.net",response="105dfec593cbcfac83380461870c3a07"
Content-Length: 0
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