thr3ads.net - asterisk users - [Asterisk-Users] * and Sipgate (UK) [Apr 2005]

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Robert P. McKenzie

2005-Apr-27 16:04 UTC

[Asterisk-Users] * and Sipgate (UK)

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@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@sipgate_london|20|tr") in new stack
     -- Called 447733322998@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@sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: Robert McKenzie <sip:sipura_line2@sip.thehostedbox.com>;tag=as60a78da7
Call-ID: 4cadaa5-c2411537@172.xxx.yyy.zzz
CSeq: 28950 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:sipura_line2@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@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@sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>
Call-ID: 2b8ccbcd-7f4f962f@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@sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>;tag=as19389899
Call-ID: 2b8ccbcd-7f4f962f@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@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>
Call-ID: b503c1bd-3d9b98ff@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>;tag=as4e6334ec
Call-ID: b503c1bd-3d9b98ff@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@172.xxx.yyy.zzz'
Destroying call '4cadaa5-c2411537@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@sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>
Call-ID: 2b8ccbcd-7f4f962f@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@sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>;tag=as74f0e15a
Call-ID: 2b8ccbcd-7f4f962f@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@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>
Call-ID: b503c1bd-3d9b98ff@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>;tag=as7b31891b
Call-ID: b503c1bd-3d9b98ff@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@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@sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>
Call-ID: 2b8ccbcd-7f4f962f@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@sip.thehostedbox.com>;tag=6d7b40ad6d38a6cfo1
To: <sip:sip.thehostedbox.com>;tag=as48fe39be
Call-ID: 2b8ccbcd-7f4f962f@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@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>
Call-ID: b503c1bd-3d9b98ff@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: <sip:sip.thehostedbox.com>;tag=as650ffffd
Call-ID: b503c1bd-3d9b98ff@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@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1@sip.thehostedbox.com>
Call-ID: 83f4a715-fd543387@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@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1@sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387@172.xxx.yyy.zzz
CSeq: 29151 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:sipura_line1@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1@sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387@172.xxx.yyy.zzz
CSeq: 29151 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:sipura_line1@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@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1@sip.thehostedbox.com>
Call-ID: 83f4a715-fd543387@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@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1@sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387@172.xxx.yyy.zzz
CSeq: 29152 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:sipura_line1@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@sip.thehostedbox.com>;tag=9448389d95568f9fo0
To: Robert McKenzie <sip:sipura_line1@sip.thehostedbox.com>;tag=as53205cb6
Call-ID: 83f4a715-fd543387@172.xxx.yyy.zzz
CSeq: 29152 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:sipura_line1@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@172.xxx.yyy.zzz' in
15000 ms
partdeux*CLI>

Sip read:
INVITE sip:8447733322998@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Robert McKenzie <sip:sipura_line1@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>;tag=as68f701f3
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998@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@172.xxx.yyy.zzz' in
15000 ms
Found user 'sipura_line1'
partdeux*CLI>

Sip read:
ACK sip:8447733322998@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>;tag=as68f701f3
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 101 ACK
Max-Forwards: 70
Contact: Robert McKenzie <sip:sipura_line1@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@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest 
username="sipura_line1",realm="asterisk",nonce="6fe33842",uri="sip:8447733322998@sip.thehostedbox.com",algorithm=MD5,response="b0c4b5664faa2365ca66bb62d276793e"
Contact: Robert McKenzie <sip:sipura_line1@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@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998@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@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@sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 69.60.122.0:5060;branch=z9hG4bK0184f59d
From: "02070438645" <sip:1438645@sipgate.co.uk>;tag=as52a9297c
To: <sip:447733322998@sipgate.co.uk>
Contact: <sip:1438645@69.60.122.0>
Call-ID: 74f4c91c5d916a822080d6821863c5be@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@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998@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@sipgate.co.uk>;tag=as52a9297c
To:
<sip:447733322998@sipgate.co.uk>;tag=019efdbfb68dd123f382ae5ce73ea92d.e770
Call-ID: 74f4c91c5d916a822080d6821863c5be@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@sipgate.co.uk out_uri=sip:447733322998@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@sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 69.60.122.0:5060;branch=z9hG4bK0184f59d
From: "02070438645" <sip:1438645@sipgate.co.uk>;tag=as52a9297c
To:
<sip:447733322998@sipgate.co.uk>;tag=019efdbfb68dd123f382ae5ce73ea92d.e770
Contact: <sip:1438645@69.60.122.0>
Call-ID: 74f4c91c5d916a822080d6821863c5be@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8447733322998@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@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@sip.thehostedbox.com>;tag=127d67de994c9545o0
To: <sip:8447733322998@sip.thehostedbox.com>;tag=as741753ba
Call-ID: 5a00da92-bdf6a221@172.xxx.yyy.zzz
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest 
username="sipura_line1",realm="asterisk",nonce="6fe33842",uri="sip:8447733322998@sip.thehostedbox.com",algorithm=MD5,response="7db1d54ca86500688f2e34f74d89984c"
Contact: Robert McKenzie <sip:sipura_line1@81.xxx.yyy.zzz:5060>
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


11 headers, 0 lines
Destroying call '74f4c91c5d916a822080d6821863c5be@sipgate.co.uk'
Destroying call '5a00da92-bdf6a221@172.xxx.yyy.zzz'



-- 
Robert P. McKenzie                 |   GammaRay Technical Services Ltd
rmckenzi@rpmdp.com                 |             rob@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

Luki

2005-Apr-27 19:22 UTC

head link

[Asterisk-Users] * and Sipgate (UK)

Robert,

It looks like you're dialing 447733322998, 44 for UK, then the area
code, etc. I have sipgate.de setup to dial local numbers (any German
number) as 0+AREA CODE+NUMBER. Always dial the area code, even if you
sipgate number is in the same city. For international numbers you need
to dial 00+COUNTRY CODE+AREA CODE+NUMBER. I think similar rules apply
for sipate.co.uk, so try dialing the above as: 07733322998 or
00447733322998.

Besides that, maybe a stupid question, but do you have money in your
sipgate account?

--Luki

Paul Tyreman

2005-Apr-29 15:20 UTC

head link

[Asterisk-Users] * and Sipgate (UK)

What are you using instead of SIPGATE in the UK ?

I also have this problem with DTMF tones not being passed to Asterisk from a 
PSTN line and my e-mails are being ignored too !

If only they sorted that problem out, it would be a great service.

Thanks, Paul.


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com 
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rafal 
Kaniewski
Posted At: 29 April 2005 22:35
Posted To: Asterisk-Users
Conversation: [Asterisk-Users] * and Sipgate (UK)
Subject: RE: [Asterisk-Users] * and Sipgate (UK)

I gave up with sipgate after dtmp tone recognition didn't work - and found 
other who also have this problem and emails to sipgate are ignored..
------------------------------------
Rafal Kaniewski
------------------------------------

Matthew Walster

2005-May-02 13:21 UTC

head link

[Asterisk-Users] * and Sipgate (UK)

On Friday 29 April 2005 23:20, Paul Tyreman wrote:> What are you using instead of SIPGATE in the UK ?
Gossiptel.com do an excellent service as well.

Matthew Walster
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Gary Stimson

2005-May-02 15:46 UTC

head link

[Asterisk-Users] * and Sipgate (UK)

On Friday 29 April 2005 23:20, Paul Tyreman wrote:> What are you using instead of SIPGATE in the UK ?
>
> I also have this problem with DTMF tones not being passed to Asterisk from
> a PSTN line and my e-mails are being ignored too !
Both are often reported problems with Sipgate. I have the DTMF problem with 
them. They took a couple of weeks to reply to an email about something else.

Now I use voiptalk.org

Gary

-- 
Gary Stimson
Zedcore Systems
UK t/f: 0114 238 1828
USA/Canada toll-free t/f: 1-877-ZEDCORE
Global t/f: +44 114 238 1828

asterisk users - Apr 2005 - * and Sipgate (UK)

[Asterisk-Users] * and Sipgate (UK)

[Asterisk-Users] * and Sipgate (UK)

[Asterisk-Users] * and Sipgate (UK)

[Asterisk-Users] * and Sipgate (UK)

[Asterisk-Users] * and Sipgate (UK)