[SOLVED] Asterisk using ip-address in invite

tronds · July 20, 2006, 8:27pm

I have problems calling out with my provider. The reason is that asterisk is using the ip-address instead of the domain name in the INVITE message. My provider doesn’t manage to get the routing correct when the somain name is missing, so I get a “number not registered message” when I try to call out.
Is it any way I can prevent asterisk from converting the domain name to the corresponding ip-address when placing the INVITE message?

I know this is the exact cause of the problem becauase both I and my provider has traced a lot of messages and discovered the the error occurs when the is is used instead of the domain name.

Trond.

IronHelix · July 20, 2006, 11:25pm

in sip.conf for the provider’s entry add

fromdomain=sip.provider.net

(or whatever their domain is)

tronds · July 21, 2006, 5:36am

IronHelix wrote:

in sip.conf for the provider’s entry add

fromdomain=sip.provider.net

I have tried that. Doesn’t matter. Asterisk replaces the domain name with the corresponding IP address anyway.

IronHelix · July 21, 2006, 5:52am

are you sure that sip.conf entry is being used then? paste the relevant sip.conf and extensions.conf part that you use to dial…

tronds · July 21, 2006, 6:06am

This is output from asterisk sip debug:

-- Executing SetVar("SIP/102-4c10", "number=5553322") in new stack

Jul 21 07:57:31 WARNING[14811]: pbx.c:5970 pbx_builtin_setvar_old: SetVar is deprecated, please use Set instead.
– Executing NoOp(“SIP/102-4c10”, “Number to dial is 5553322”) in new stack
– Executing Dial(“SIP/102-4c10”, “SIP/5553322@itelefon|60”) in new stack
We’re at 195.159.197.22 port 17054
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 62.97.243.50:5060:
INVITE sip:5553322@62.97.243.50 SIP/2.0 [color=red]<---- Problem![/color]
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK3b5e34d1;rport
From: “5553344” sip:5553344@itelefon.no;tag=as3aee1353
To: sip:5553322@62.97.243.50 [color=red]<---- Problem![/color]
Contact: sip:5553344@195.159.197.22
Call-ID: 4baae5f07184655942af6ff059062071@itelefon.no
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Fri, 21 Jul 2006 05:57:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 235

The marked lines are the problem lines. It should use the domain itelefon.no instead of the IP-address.

tronds · July 21, 2006, 6:12am

[color=green]sip.conf:[/color]
[itelefon]
type=peer
callerid="my phone no"
username=username
secret=secret
host=dynamic
fromuser=my phone no
fromdomain=itelefon.no
port=5060
nat=no
tos=lowdelay
insecure=very
context=incoming
canreinvite=no
dtmfmode=inband
outboundproxy=itelefon.no
externhost=my.domain.name
localnet=192.168.1.0/255.255.255.0
externip=my.domain.name
qualify=yes

[color=green]extensions.conf:[/color]
exten => _0.,1,SetVar(number=${EXTEN:1})
exten => _0.,2,NoOp(Number to dial is ${number})
exten => _0.,3,Dial(SIP/${number}@itelefon,60)
exten => _0.,4,Congestion
exten => _0.,104,Busy

IronHelix · July 21, 2006, 6:46am

hmmm. post the entire sip dialog (including their rejection?)

tronds · July 21, 2006, 1:44pm

This is what I get with asterisk -rvvv, sip debug

mir*CLI> sip debug
SIP Debugging enabled
Destroying call 'b1fbfbb0-d5b9ef82@192.168.1.161’
Jul 21 15:38:38 NOTICE[14808]: chan_sip.c:5241 sip_reregister: – Re-registration for 51290089@itelefon.no
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 62.97.243.50:5060:
REGISTER sip:itelefon.no SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK54ab1725;rport
From: sip:51290089@itelefon.no;tag=as5139d456
To: sip:51290089@itelefon.no
Call-ID: 451115b97cd611146d3c6e263f2310ce@itelefon.no
CSeq: 460 REGISTER
User-Agent: 51290089
Max-Forwards: 70
Authorization: Digest username=“username”, realm=“itelefon.no”, algorithm=MD5, uri=“sip:itelefon.no”, nonce=“MTE1MzQ4OTIwMDgxNGVjMmQ1MGJiYWMyYWRjMzEwNWVmZWJiYWY3MGQ2MDM5”, response=“5415342589653f953b908c0a9447c106”, opaque=“98578820c7431841990971c26ccffaa9”, qop=auth, cnonce=“6e3c070a”, nc=00000002
Expires: 120
Contact: sip:51290089@195.159.197.22
Event: registration
Content-Length: 0

mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK54ab1725;rport=63910
To: sip:51290089@itelefon.no;tag=54c4ad-10c914c4dd7–294e
From: sip:51290089@itelefon.no;tag=as5139d456
Call-ID: 451115b97cd611146d3c6e263f2310ce@itelefon.no
CSeq: 460 REGISTER
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0
Contact: sip:51290089@195.159.197.22;expires=120

— (9 headers 0 lines)—
Scheduling destruction of call '451115b97cd611146d3c6e263f2310ce@itelefon.no’ in 32000 ms
Jul 21 15:38:38 NOTICE[14808]: chan_sip.c:9668 handle_response_register: Outbound Registration: Expiry for itelefon.no is 120 sec (Scheduling reregistration in 105 s)
mir*CLI>
<-- SIP read from 192.168.1.161:5061:
INVITE sip:046680452@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-498d9353
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Anonymous sip:102@192.168.1.161:5061
Expires: 240
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 430
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 241483363 241483363 IN IP4 192.168.1.161
s=-
c=IN IP4 192.168.1.161
t=0 0
m=audio 16388 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 INVITE request as basis request - 1f0c9b98-c1304a2a@localhost
Sending to 192.168.1.161 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.161:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-498d9353;received=192.168.1.161
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100;tag=as14456bb4
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 101 INVITE
User-Agent: 51290089
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:046680452@192.168.1.100
Proxy-Authenticate: Digest realm=“asterisk”, nonce="3958e7e0"
Content-Length: 0

Scheduling destruction of call ‘1f0c9b98-c1304a2a@localhost’ in 15000 ms
Found user '102’
mir*CLI>
<-- SIP read from 192.168.1.161:5061:
ACK sip:046680452@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-498d9353
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100;tag=as14456bb4
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 101 ACK
Max-Forwards: 70
Contact: Anonymous sip:102@192.168.1.161:5061
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0

— (10 headers 0 lines)—
mir*CLI>
<-- SIP read from 192.168.1.161:5061:
INVITE sip:046680452@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-2dc7071b
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“102”,realm=“asterisk”,nonce=“3958e7e0”,uri="sip:046680452@192.168.1.100",algorithm=MD5,response="2df740371fcefe7ba85567a44c87f9cd"
Contact: Anonymous sip:102@192.168.1.161:5061
Expires: 240
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 430
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 241483363 241483363 IN IP4 192.168.1.161
s=-
c=IN IP4 192.168.1.161
t=0 0
m=audio 16388 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 INVITE request as basis request - 1f0c9b98-c1304a2a@localhost
Sending to 192.168.1.161 : 5061 (non-NAT)
Found user '102’
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 192.168.1.161:16388
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 - 0xf (g723|gsm|ulaw|alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xd (g723|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 046680452 in default (domain 192.168.1.100)
list_route: hop: sip:102@192.168.1.161:5061
Transmitting (no NAT) to 192.168.1.161:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-2dc7071b;received=192.168.1.161
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 102 INVITE
User-Agent: 51290089
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:046680452@192.168.1.100
Content-Length: 0

-- Executing SetVar("SIP/102-f00d", "number=46680452") in new stack
-- Executing NoOp("SIP/102-f00d", "Number to dial is 46680452") in new stack
-- Executing Dial("SIP/102-f00d", "SIP/46680452@itelefon|60") in new stack

We’re at 195.159.197.22 port 19032
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 62.97.243.50:5060:
INVITE sip:46680452@62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK26e95518;rport
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50
Contact: sip:51290089@195.159.197.22
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 102 INVITE
User-Agent: 51290089
Max-Forwards: 70
Date: Fri, 21 Jul 2006 13:38:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 14783 14783 IN IP4 195.159.197.22
s=session
c=IN IP4 195.159.197.22
t=0 0
m=audio 19032 RTP/AVP 8 0 3 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=silenceSupp:off - - - -

-- Called 46680452@itelefon

mir*CLI>
<-- SIP read from 64.34.224.29:5060:

— (0 headers 0 lines) Nat keepalive —
mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK26e95518;rport=63910
To: sip:46680452@62.97.243.50;tag=54c4ad-10c914c4dd7–2736
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 102 INVITE
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0
Proxy-Authenticate: Digest qop=“auth”,stale=false,realm=“itelefon.no”,opaque=“5218b27ce6844118484ab04e551bbb76”,nonce=“MTE1MzQ4OTQxNzE0ODY1NjAyNzEwM2UzYjg4NmY1OGVkMzI1OWM2Mjg0YmUw”

— (9 headers 0 lines)—
Transmitting (no NAT) to 62.97.243.50:5060:
ACK sip:46680452@62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK26e95518;rport
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50;tag=54c4ad-10c914c4dd7–2736
Contact: sip:51290089@195.159.197.22
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 102 ACK
User-Agent: 51290089
Max-Forwards: 70
Content-Length: 0

We’re at 195.159.197.22 port 19032
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
Reliably Transmitting (no NAT) to 62.97.243.50:5060:
INVITE sip:46680452@62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK22311435;rport
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50
Contact: sip:51290089@195.159.197.22
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 103 INVITE
User-Agent: 51290089
Max-Forwards: 70
Proxy-Authorization: Digest username=“username”, realm=“itelefon.no”, algorithm=MD5, uri="sip:46680452@62.97.243.50", nonce=“MTE1MzQ4OTQxNzE0ODY1NjAyNzEwM2UzYjg4NmY1OGVkMzI1OWM2Mjg0YmUw”, response=“6f40131dcf7b2c84b8a336500f01f75d”, opaque=“5218b27ce6844118484ab04e551bbb76”, qop=auth, cnonce=“10338102”, nc=00000001
Date: Fri, 21 Jul 2006 13:38:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 14783 14784 IN IP4 195.159.197.22
s=session
c=IN IP4 195.159.197.22
t=0 0
m=audio 19032 RTP/AVP 8 0 3 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=silenceSupp:off - - - -

12 headers, 0 lines
Reliably Transmitting (no NAT) to 62.97.243.50:5060:
OPTIONS sip:62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK58a32d43;rport
From: “asterisk” sip:asterisk@itelefon.no;tag=as6de890c4
To: sip:62.97.243.50
Contact: sip:asterisk@195.159.197.22
Call-ID: 39b0649b3764f2ef0a0e7a371c2c754f@itelefon.no
CSeq: 102 OPTIONS
User-Agent: 51290089
Max-Forwards: 70
Date: Fri, 21 Jul 2006 13:38:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

12 headers, 0 lines
Reliably Transmitting (no NAT) to 62.97.243.50:5060:
OPTIONS sip:62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK131450ae;rport
From: “asterisk” sip:asterisk@itelefon.no;tag=as2a86d05a
To: sip:62.97.243.50
Contact: sip:asterisk@195.159.197.22
Call-ID: 2ae44f043f8cc86427faa1c95a2bb084@itelefon.no
CSeq: 102 OPTIONS
User-Agent: 51290089
Max-Forwards: 70
Date: Fri, 21 Jul 2006 13:38:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK26e95518;rport=63910
To: sip:46680452@62.97.243.50;tag=54c4ad-10c914c4dd7–2736
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 102 INVITE
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0
Proxy-Authenticate: Digest qop=“auth”,stale=false,realm=“itelefon.no”,opaque=“5218b27ce6844118484ab04e551bbb76”,nonce=“MTE1MzQ4OTQxNzE0ODY1NjAyNzEwM2UzYjg4NmY1OGVkMzI1OWM2Mjg0YmUw”

— (9 headers 0 lines)—
Transmitting (no NAT) to 62.97.243.50:5060:
ACK sip:46680452@62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK22311435;rport
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50;tag=54c4ad-10c914c4dd7–2736
Contact: sip:51290089@195.159.197.22
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 102 ACK
User-Agent: 51290089
Max-Forwards: 70
Content-Length: 0

Retransmitting #1 (no NAT) to 62.97.243.50:5060:
OPTIONS sip:62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK58a32d43;rport
From: “asterisk” sip:asterisk@itelefon.no;tag=as6de890c4
To: sip:62.97.243.50
Contact: sip:asterisk@195.159.197.22
Call-ID: 39b0649b3764f2ef0a0e7a371c2c754f@itelefon.no
CSeq: 102 OPTIONS
User-Agent: 51290089
Max-Forwards: 70
Date: Fri, 21 Jul 2006 13:38:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Retransmitting #1 (no NAT) to 62.97.243.50:5060:
OPTIONS sip:62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK131450ae;rport
From: “asterisk” sip:asterisk@itelefon.no;tag=as2a86d05a
To: sip:62.97.243.50
Contact: sip:asterisk@195.159.197.22
Call-ID: 2ae44f043f8cc86427faa1c95a2bb084@itelefon.no
CSeq: 102 OPTIONS
User-Agent: 51290089
Max-Forwards: 70
Date: Fri, 21 Jul 2006 13:38:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK22311435;rport=63910
To: sip:46680452@62.97.243.50
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 103 INVITE
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0

— (8 headers 0 lines)—
mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK58a32d43;rport=63910
To: sip:62.97.243.50;tag=54c4ad-10c914c4dd7–21a6
From: “asterisk” sip:asterisk@itelefon.no;tag=as6de890c4
Call-ID: 39b0649b3764f2ef0a0e7a371c2c754f@itelefon.no
CSeq: 102 OPTIONS
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0

— (8 headers 0 lines)—
Destroying call '39b0649b3764f2ef0a0e7a371c2c754f@itelefon.no’
mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK131450ae;rport=63910
To: sip:62.97.243.50;tag=54c4ad-10c914c4dd7–219f
From: “asterisk” sip:asterisk@itelefon.no;tag=as2a86d05a
Call-ID: 2ae44f043f8cc86427faa1c95a2bb084@itelefon.no
CSeq: 102 OPTIONS
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0

— (8 headers 0 lines)—
Destroying call '2ae44f043f8cc86427faa1c95a2bb084@itelefon.no’
mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK58a32d43;rport=63910
To: sip:62.97.243.50;tag=54c4ad-10c914c4dd7–21a6
From: “asterisk” sip:asterisk@itelefon.no;tag=as6de890c4
Call-ID: 39b0649b3764f2ef0a0e7a371c2c754f@itelefon.no
CSeq: 102 OPTIONS
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0

— (8 headers 0 lines)—
Destroying call '39b0649b3764f2ef0a0e7a371c2c754f@itelefon.no’
mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK131450ae;rport=63910
To: sip:62.97.243.50;tag=54c4ad-10c914c4dd7–219f
From: “asterisk” sip:asterisk@itelefon.no;tag=as2a86d05a
Call-ID: 2ae44f043f8cc86427faa1c95a2bb084@itelefon.no
CSeq: 102 OPTIONS
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0

— (8 headers 0 lines)—
Destroying call '2ae44f043f8cc86427faa1c95a2bb084@itelefon.no’
mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK22311435;rport=63910
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50;tag=5F752E14-538
Date: Fri, 21 Jul 2006 13:38:38 GMT
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: sip:46680452@82.134.95.94:5060
Record-Route: sip:62.97.243.50:5060;transport=UDP;lr
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 179

v=0
o=CiscoSystemsSIP-GW-UserAgent 9329 4085 IN IP4 82.134.95.94
s=SIP Call
c=IN IP4 82.134.95.94
t=0 0
m=audio 18136 RTP/AVP 8
c=IN IP4 82.134.95.94
a=rtpmap:8 PCMA/8000

— (15 headers 8 lines)—
Found RTP audio format 8
Peer audio RTP is at port 82.134.95.94:18136
Found description format PCMA
Capabilities: us - 0xf (g723|gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
– SIP/itelefon-7dff is making progress passing it to SIP/102-f00d
We’re at 192.168.1.100 port 16502
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.1.161:5061:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-2dc7071b;received=192.168.1.161
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100;tag=as0460f5e7
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 102 INVITE
User-Agent: 51290089
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:046680452@192.168.1.100
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 14783 14783 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 16502 RTP/AVP 8 0 3 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK22311435;rport=63910
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50;tag=5F752E14-538
Date: Fri, 21 Jul 2006 13:38:39 GMT
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: sip:46680452@82.134.95.94:5060
Record-Route: sip:62.97.243.50:5060;transport=UDP;lr
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 179

v=0
o=CiscoSystemsSIP-GW-UserAgent 9329 4085 IN IP4 82.134.95.94
s=SIP Call
c=IN IP4 82.134.95.94
t=0 0
m=audio 18136 RTP/AVP 8
c=IN IP4 82.134.95.94
a=rtpmap:8 PCMA/8000

— (15 headers 8 lines)—
Found RTP audio format 8
Peer audio RTP is at port 82.134.95.94:18136
Found description format PCMA
Capabilities: us - 0xf (g723|gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
– SIP/itelefon-7dff is making progress passing it to SIP/102-f00d
mir*CLI>
<-- SIP read from 192.168.1.161:5061:
CANCEL sip:046680452@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-2dc7071b
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username=“102”,realm=“asterisk”,nonce=“3958e7e0”,uri="sip:046680452@192.168.1.100",algorithm=MD5,response="bf213e4a023057cc8de3cfbbf36f6067"
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0

— (10 headers 0 lines)—
Sending to 192.168.1.161 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.161:5061:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-2dc7071b;received=192.168.1.161
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100;tag=as0460f5e7
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 102 INVITE
User-Agent: 51290089
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:046680452@192.168.1.100
Content-Length: 0

Transmitting (no NAT) to 192.168.1.161:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-2dc7071b;received=192.168.1.161
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100;tag=as0460f5e7
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 102 CANCEL
User-Agent: 51290089
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:046680452@192.168.1.100
Content-Length: 0

Reliably Transmitting (no NAT) to 62.97.243.50:5060:
CANCEL sip:46680452@62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK22311435;rport
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50
Contact: sip:51290089@195.159.197.22
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 103 CANCEL
User-Agent: 51290089
Max-Forwards: 70
Proxy-Authorization: Digest username=“username”, realm=“itelefon.no”, algorithm=MD5, uri="sip:46680452@62.97.243.50", nonce=“MTE1MzQ4OTQxNzE0ODY1NjAyNzEwM2UzYjg4NmY1OGVkMzI1OWM2Mjg0YmUw”, response=“65bfbb6787d96256c43822178741b71c”, opaque=“5218b27ce6844118484ab04e551bbb76”, qop=auth, cnonce=“1463a2b4”, nc=00000002
Content-Length: 0

Scheduling destruction of call '270b46997d34f9a90b76c3854046dce5@itelefon.no’ in 15000 ms
== Spawn extension (default, 046680452, 3) exited non-zero on 'SIP/102-f00d’
mir*CLI>
<-- SIP read from 192.168.1.161:5061:
ACK sip:046680452@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5061;branch=z9hG4bK-2dc7071b
From: Anonymous sip:102@192.168.1.100;tag=68d1cc0a117b22o1
To: sip:046680452@192.168.1.100;tag=as0460f5e7
Call-ID: 1f0c9b98-c1304a2a@localhost
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“102”,realm=“asterisk”,nonce=“3958e7e0”,uri="sip:046680452@192.168.1.100",algorithm=MD5,response="1b8962f5b5e55404996c54733fad7764"
Contact: Anonymous sip:102@192.168.1.161:5061
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0

— (11 headers 0 lines)—
Destroying call ‘1f0c9b98-c1304a2a@localhost’

<-- SIP read from 62.97.243.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK22311435;rport=63910
To: sip:46680452@62.97.243.50
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 103 CANCEL
Server: hotsip-transactron/core-3.0.6.211
Content-Length: 0

— (8 headers 0 lines)—
mir*CLI>
<-- SIP read from 62.97.243.50:5060:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 195.159.197.22:5060;received=195.159.197.22;branch=z9hG4bK22311435;rport=63910
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50;tag=5F752E14-538
Date: Fri, 21 Jul 2006 13:38:43 GMT
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow-Events: telephone-event
Content-Length: 0

— (10 headers 0 lines)—
Transmitting (no NAT) to 62.97.243.50:5060:
ACK sip:46680452@62.97.243.50 SIP/2.0
Via: SIP/2.0/UDP 195.159.197.22:5060;branch=z9hG4bK22311435;rport
From: “51290089” sip:51290089@itelefon.no;tag=as634e8a32
To: sip:46680452@62.97.243.50;tag=5F752E14-538
Contact: sip:51290089@195.159.197.22
Call-ID: 270b46997d34f9a90b76c3854046dce5@itelefon.no
CSeq: 103 ACK
User-Agent: 51290089
Max-Forwards: 70
Content-Length: 0

spyke · July 21, 2006, 3:53pm

Why don’t you set:
host=sip.provider.net

C.

tronds · July 21, 2006, 7:33pm

Why don’t you set:
host=sip.provider.net

Because I tought that host was my host name.

Well, that did the trick.
I’m able to call out again after nearly seven weeks.

Thanks.