[Asterisk-Users] * and Sipgate (UK)
Robert P. McKenzie
rmckenzi at rpmdp.com
Wed Apr 27 16:04:41 MST 2005
I'd given up trying to get sipgate working with my * server, but given some problems I've had with a voip provider I
need to revisit this again. I've tried to set things up as per sipgate's website, I've read the info on voip-info and
from several other postings from other lists (and this one). Still, no luck at all. It has to be something simple.
Any help or suggestions is greatly appreciated. There has to be something very simple missing/wrong.
Cheers!!!
Here are my configs, and the errors I get follow:
Stats from the server:
*CLI> sip show registry
Host Username Refresh State
sipgate.co.uk:5060 1438645 105 Registered
*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
sipgate_london/ 217.10.79.219 255.255.255.255 5060 Unmonitored
cottenham_line2 81.xxx.yyy.zzz D N 255.255.255.255 5061 Unmonitored
cottenham_line1 81.xxx.yyy.zzz D N 255.255.255.255 5060 Unmonitored
sip.conf:
register => 1438645:XXXXXXXX at sipgate.co.uk/1438645
[sipgate_london]
type=friend
username=1438645
secret=XXXXXXXX
host=sipgate.co.uk
fromuser=1438645
fromdomain=sipgate.co.uk
nat=no ; dedicated hosted server, no NAT in use
authuser=1438645
dtmfmode=info
context=rob
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
extensions.conf
; Inbound sipgate calls -- This works fine
[in_sipgate]
; London Number
exten => 1438645,1,Answer
exten => 1438645,2,Dial(${ROBSPHONES}|60|t)
exten => 1438645,3,Voicemail(u50)
; Outbound sipgate calls -- This does not work
[out_sipgate_london]
exten => _8.,1,SetCallerID(02070438645)
exten => _8.,2,Dial(SIP/${EXTEN:1}@sipgate_london,20,tr)
exten => _8.,3,Congestion
exten => _8.,4,Busy
exten => _8.,5,Hangup
Here are the short logs, non-debug:
-- Executing SetCallerID("SIP/sipura_line1-3d72", "02070438645") in new stack
-- Executing Dial("SIP/sipura_line1-3d72", "SIP/447733322998 at sipgate_london|20|tr") in new stack
-- Called 447733322998 at sipgate_london
-- Got SIP response 404 "Not Found" back from 217.10.79.219
-- SIP/sipgate_london-c497 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Congestion("SIP/sipura_line1-3d72", "") in new stack
== Spawn extension (rob, 8447733322998, 3) exited non-zero on 'SIP/cottenham_line1-3d72'
-- Executing Hangup("SIP/sipura_line1-3d72", "") in new stack
== Spawn extension (rob, h, 1) exited non-zero on 'SIP/sipura_line1-3d72'
The errors I get are (sorry, long debug log):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5061;branch=z9hG4bK-75c8e83d;received=81.xxx.yyy.zzz;rport=5061
From: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=as60a78da7
Call-ID: 4cadaa5-c2411537 at 172.xxx.yyy.zzz
CSeq: 28950 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:sipura_line2 at 81.xxx.yyy.zzz:5061>;expires=60
Date: Wed, 27 Apr 2005 22:45:18 GMT
Content-Length: 0
to 81.xxx.yyy.zzz:5061
Scheduling destruction of call '4cadaa5-c2411537 at 172.xxx.yyy.zzz' in 15000 ms
partdeux*CLI>
Sip read:
NOTIFY sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5061;branch=z9hG4bK-bf246b9a;rport
From: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>
Call-ID: 2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz
CSeq: 57229 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5061;branch=z9hG4bK-bf246b9a
From: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>;tag=as19389899
Call-ID: 2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz
CSeq: 57229 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 81.xxx.yyy.zzz:5061
Destroying call '2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz'
partdeux*CLI>
Sip read:
NOTIFY sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-b19cf813;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>
Call-ID: b503c1bd-3d9b98ff at 172.xxx.yyy.zzz
CSeq: 57229 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-b19cf813
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>;tag=as4e6334ec
Call-ID: b503c1bd-3d9b98ff at 172.xxx.yyy.zzz
CSeq: 57229 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Destroying call 'b503c1bd-3d9b98ff at 172.xxx.yyy.zzz'
Destroying call '4cadaa5-c2411537 at 172.xxx.yyy.zzz'
partdeux*CLI>
Sip read:
NOTIFY sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5061;branch=z9hG4bK-b4b8b3a5;rport
From: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>
Call-ID: 2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz
CSeq: 57230 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5061;branch=z9hG4bK-b4b8b3a5
From: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>;tag=as74f0e15a
Call-ID: 2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz
CSeq: 57230 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 81.xxx.yyy.zzz:5061
Destroying call '2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz'
partdeux*CLI>
Sip read:
NOTIFY sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-434dd6d;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>
Call-ID: b503c1bd-3d9b98ff at 172.xxx.yyy.zzz
CSeq: 57230 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-434dd6d
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>;tag=as7b31891b
Call-ID: b503c1bd-3d9b98ff at 172.xxx.yyy.zzz
CSeq: 57230 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Destroying call 'b503c1bd-3d9b98ff at 172.xxx.yyy.zzz'
partdeux*CLI>
Sip read:
NOTIFY sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5061;branch=z9hG4bK-f44d83fc;rport
From: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>
Call-ID: 2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz
CSeq: 57231 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5061;branch=z9hG4bK-f44d83fc
From: Robert McKenzie <sip:sipura_line2 at sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>;tag=as48fe39be
Call-ID: 2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz
CSeq: 57231 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 81.xxx.yyy.zzz:5061
Destroying call '2b8ccbcd-7f4f962f at 172.xxx.yyy.zzz'
partdeux*CLI>
Sip read:
NOTIFY sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-d476f6eb;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>
Call-ID: b503c1bd-3d9b98ff at 172.xxx.yyy.zzz
CSeq: 57231 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
10 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-d476f6eb
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>;tag=as650ffffd
Call-ID: b503c1bd-3d9b98ff at 172.xxx.yyy.zzz
CSeq: 57231 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Destroying call 'b503c1bd-3d9b98ff at 172.xxx.yyy.zzz'
partdeux*CLI>
Sip read:
REGISTER sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-213f095c;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>
Call-ID: 83f4a715-fd543387 at 172.xxx.yyy.zzz
CSeq: 29151 REGISTER
Max-Forwards: 70
Authorization: Digest
username="sipura_line1",realm="asterisk",nonce="1d6b1dfd",uri="sip:sip.thehostedbox.com",algorithm=MD5,response="3b15c3c92da8b15ebfc60cb0690744f7"
Contact: Robert McKenzie <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>;expires=60
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
13 headers, 0 lines
Using latest request as basis request
Sending to 81.xxx.yyy.zzz : 5060 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-213f095c;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387 at 172.xxx.yyy.zzz
CSeq: 29151 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:sipura_line1 at 69.60.122.0>
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-213f095c;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387 at 172.xxx.yyy.zzz
CSeq: 29151 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:sipura_line1 at 69.60.122.0>
WWW-Authenticate: Digest realm="asterisk", nonce="27b55da7"
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Scheduling destruction of call '83f4a715-fd543387 at 172.xxx.yyy.zzz' in 15000 ms
partdeux*CLI>
Sip read:
REGISTER sip:sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-9f60a59d;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>
Call-ID: 83f4a715-fd543387 at 172.xxx.yyy.zzz
CSeq: 29152 REGISTER
Max-Forwards: 70
Authorization: Digest
username="sipura_line1",realm="asterisk",nonce="27b55da7",uri="sip:sip.thehostedbox.com",algorithm=MD5,response="6c9ab8d5b1e5364261008a2f2958f411"
Contact: Robert McKenzie <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>;expires=60
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
13 headers, 0 lines
Using latest request as basis request
Sending to 81.xxx.yyy.zzz : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-9f60a59d;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387 at 172.xxx.yyy.zzz
CSeq: 29152 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:sipura_line1 at 69.60.122.0>
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-9f60a59d;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387 at 172.xxx.yyy.zzz
CSeq: 29152 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>;expires=60
Date: Wed, 27 Apr 2005 22:46:00 GMT
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Scheduling destruction of call '83f4a715-fd543387 at 172.xxx.yyy.zzz' in 15000 ms
partdeux*CLI>
Sip read:
INVITE sip:8447733322998 at sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-8a1e2b6e;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Robert McKenzie <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 426
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 87021836 87021836 IN IP4 81.xxx.yyy.zzz
s=-
c=IN IP4 81.xxx.yyy.zzz
t=0 0
m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
14 headers, 19 lines
Using latest request as basis request
Sending to 81.xxx.yyy.zzz : 5060 (non-NAT)
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-8a1e2b6e;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>;tag=as68f701f3
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998 at 69.60.122.0>
Proxy-Authenticate: Digest realm="asterisk", nonce="6fe33842"
Content-Length: 0
to 81.xxx.yyy.zzz:5060
Scheduling destruction of call '5a00da92-bdf6a221 at 172.xxx.yyy.zzz' in 15000 ms
Found user 'sipura_line1'
partdeux*CLI>
Sip read:
ACK sip:8447733322998 at sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-8a1e2b6e;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>;tag=as68f701f3
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 101 ACK
Max-Forwards: 70
Contact: Robert McKenzie <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
10 headers, 0 lines
partdeux*CLI>
Sip read:
INVITE sip:8447733322998 at sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-732dd872;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="sipura_line1",realm="asterisk",nonce="6fe33842",uri="sip:8447733322998 at sip.thehostedbox.com",algorithm=MD5,response="b0c4b5664faa2365ca66bb62d276793e"
Contact: Robert McKenzie <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 426
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 87021836 87021836 IN IP4 81.xxx.yyy.zzz
s=-
c=IN IP4 81.xxx.yyy.zzz
t=0 0
m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 19 lines
Using latest request as basis request
Sending to 81.xxx.yyy.zzz : 5060 (NAT)
Found user 'sipura_line1'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 81.xxx.yyy.zzz:16424
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined -
0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 8447733322998 in rob
list_route: hop: <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-732dd872;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998 at 69.60.122.0>
Content-Length: 0
to 81.xxx.yyy.zzz:5060
-- Executing SetCallerID("SIP/sipura_line1-bddb", "02070438645") in new stack
-- Executing Dial("SIP/sipura_line1-bddb", "SIP/447733322998 at sipgate_london|20|tr") in new stack
We're at 69.60.122.0 port 14522
Answering/Requesting with root capability 0x8 (alaw)
Answering with capability 0x4 (ulaw)
12 headers, 9 lines
Reliably Transmitting:
INVITE sip:447733322998 at sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 69.60.122.0:5060;branch=z9hG4bK0184f59d
From: "02070438645" <sip:1438645 at sipgate.co.uk>;tag=as52a9297c
To: <sip:447733322998 at sipgate.co.uk>
Contact: <sip:1438645 at 69.60.122.0>
Call-ID: 74f4c91c5d916a822080d6821863c5be at sipgate.co.uk
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 27 Apr 2005 22:46:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 16184 16184 IN IP4 69.60.122.0
s=session
c=IN IP4 69.60.122.0
t=0 0
m=audio 14522 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT) to 217.10.79.219:5060
-- Called 447733322998 at sipgate_london
Transmitting (NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-732dd872;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998 at 69.60.122.0>
Content-Length: 0
to 81.xxx.yyy.zzz:5060
partdeux*CLI>
Sip read:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.60.122.0:5060;branch=z9hG4bK0184f59d
From: "02070438645" <sip:1438645 at sipgate.co.uk>;tag=as52a9297c
To: <sip:447733322998 at sipgate.co.uk>;tag=019efdbfb68dd123f382ae5ce73ea92d.e770
Call-ID: 74f4c91c5d916a822080d6821863c5be at sipgate.co.uk
CSeq: 102 INVITE
Server: sipgate ser
Content-Length: 0
Warning: 392 217.10.79.219:5060 "Noisy feedback tells: pid=13369 req_src_ip=69.60.122.0 req_src_port=5060
in_uri=sip:447733322998 at sipgate.co.uk out_uri=sip:447733322998 at sipgate.co.uk via_cnt==1"
9 headers, 0 lines
-- Got SIP response 404 "Not Found" back from 217.10.79.219
Transmitting:
ACK sip:447733322998 at sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 69.60.122.0:5060;branch=z9hG4bK0184f59d
From: "02070438645" <sip:1438645 at sipgate.co.uk>;tag=as52a9297c
To: <sip:447733322998 at sipgate.co.uk>;tag=019efdbfb68dd123f382ae5ce73ea92d.e770
Contact: <sip:1438645 at 69.60.122.0>
Call-ID: 74f4c91c5d916a822080d6821863c5be at sipgate.co.uk
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 217.10.79.219:5060
-- SIP/sipgate_london-c9e9 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Congestion("SIP/sipura_line1-bddb", "") in new stack
Transmitting (NAT):
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-732dd872;received=81.xxx.yyy.zzz;rport=5060
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998 at 69.60.122.0>
Content-Length: 0
to 81.xxx.yyy.zzz:5060
== Spawn extension (rob, 8447733322998, 3) exited non-zero on 'SIP/sipura_line1-bddb'
-- Executing Hangup("SIP/sipura_line1-bddb", "") in new stack
== Spawn extension (rob, h, 1) exited non-zero on 'SIP/sipura_line1-bddb'
partdeux*CLI>
Sip read:
ACK sip:8447733322998 at sip.thehostedbox.com SIP/2.0
Via: SIP/2.0/UDP 81.xxx.yyy.zzz:5060;branch=z9hG4bK-732dd872;rport
From: Robert McKenzie <sip:sipura_line1 at sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998 at sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221 at 172.xxx.yyy.zzz
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="sipura_line1",realm="asterisk",nonce="6fe33842",uri="sip:8447733322998 at sip.thehostedbox.com",algorithm=MD5,response="7db1d54ca86500688f2e34f74d89984c"
Contact: Robert McKenzie <sip:sipura_line1 at 81.xxx.yyy.zzz:5060>
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
11 headers, 0 lines
Destroying call '74f4c91c5d916a822080d6821863c5be at sipgate.co.uk'
Destroying call '5a00da92-bdf6a221 at 172.xxx.yyy.zzz'
--
Robert P. McKenzie | GammaRay Technical Services Ltd
rmckenzi at rpmdp.com | rob at gammaray-tech.com
http://www.uk-experience.com | http://www.gammaray-tech.com
Ecademy Profile: http://www.ecademy.com/account.php?op=view&id=64014
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