Problem configuring a certain Sip-provider

Hello
I try to use the SIP-provider GulfSip. I have no problems to use it with a softphone. However, all efforts to use it with the Asterisk failed so far. I provide below the entries made in sip.conf and a debug log from establishing a test call. Could someone of you please check why the call is rejected and how it possibly could be avoided? Note that my Asterisk is installed on a vServer at an external provider (IP_Asterisk) and I am connected to it via a Linksys adaptor (IP_Linksys). I have about ten different providers configured in sip.conf and they all work without any problems.

sip.conf:[code]
[general]
alwaysauthreject=yes
context=default
bindport=5060
bindaddr=IP_Asterisk
srvlookup=yes
useragent=MyDevice
tos_audio=0xb0
tos_sip=0xb0
disallow=all
allow=ulaw
allow=alaw
allow=ilbc

register => User_Nr:is_secret@sip.gulfsip.com:6321/GulfSip

[GulfSip]
type=peer
port=6321
username=User_Nr
fromuser=+…
secret=is_secret
host=sip.gulfsip.com
nat=yes
insecure=port,invite
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=ilbc

[GulfSip_in]
type=peer
host=sip.gulfsip.com
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
context=incoming[/code]

Debug log when making a call:[code]User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
vs8709*CLI>
<— SIP read from IP_Linksys:5061 —>
NOTIFY sip:IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030
From: sip:10b@IP_Asterisk;tag=210a5427b3032177o1
To: sip:IP_Asterisk
Call-ID: 33253ea7-e66138f7@127.0.0.1
CSeq: 585 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.1.42 : 5061 (no NAT)
vs8709*CLI>
<— Transmitting (no NAT) to 192.168.1.42:5061 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030;received=IP_Linksys
From: sip:10b@IP_Asterisk;tag=210a5427b3032177o1
To: sip:IP_Asterisk;tag=as73692944
Call-ID: 33253ea7-e66138f7@127.0.0.1
CSeq: 585 NOTIFY
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
vs8709*CLI>
<— SIP read from IP_Linksys:5061 —>
NOTIFY sip:IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030
From: sip:10b@IP_Asterisk;tag=210a5427b3032177o1
To: sip:IP_Asterisk
Call-ID: 33253ea7-e66138f7@127.0.0.1
CSeq: 585 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.1.42 : 5061 (no NAT)

<— Transmitting (no NAT) to 192.168.1.42:5061 —>
IP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030;received=IP_Linksys
From: sip:10b@IP_Asterisk;tag=210a5427b3032177o1
To: sip:IP_Asterisk;tag=as5a33e10b
Call-ID: 33253ea7-e66138f7@127.0.0.1
CSeq: 585 NOTIFY
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
vs8709*CLI>
<— SIP read from IP_Linksys:5062 —>
NOTIFY sip:IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-6baf9108
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=829651a7b1eb41f7o0
To: sip:IP_Asterisk
Call-ID: 17b83467-f414be37@127.0.0.1
CSeq: 575 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.1.42 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.42:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-6baf9108;received=IP_Linksys
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=829651a7b1eb41f7o0
To: sip:IP_Asterisk;tag=as2d786770
Call-ID: 17b83467-f414be37@127.0.0.1
CSeq: 575 NOTIFY
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
vs8709*CLI>
<— SIP read from IP_Linksys:5062 —>
INVITE sip:00NR_TO_CALL@IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-1c79e8f0
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=d90b1a2ca4f54990o0
To: sip:00NR_TO_CALL@IP_Asterisk
Remote-Party-ID: “+NR_TO_CALL” sip:10a@IP_Asterisk;screen=yes;party=calling
Call-ID: 43abab5c-1c126a20@192.168.1.42
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “+NR_TO_CALL” sip:10a@192.168.1.42:5060
Expires: 240
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 1198925 1198925 IN IP4 192.168.1.42
s=-
c=IN IP4 192.168.1.42
t=0 0
m=audio 16460 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
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=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 20 lines) —
Sending to 192.168.1.42 : 5060 (no NAT)
Using INVITE request as basis request - 43abab5c-1c126a20@192.168.1.42

<— Reliably Transmitting (NAT) to IP_Linksys:5062 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-1c79e8f0;received=IP_Linksys
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=d90b1a2ca4f54990o0
To: sip:00NR_TO_CALL@IP_Asterisk;tag=as149d3d12
Call-ID: 43abab5c-1c126a20@192.168.1.42
CSeq: 101 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="13829fd9"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘43abab5c-1c126a20@192.168.1.42’ in 32000 ms (Method: INVITE)
Found user '10a’
vs8709*CLI>
<— SIP read from IP_Linksys:5062 —>
ACK sip:00NR_TO_CALL@IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-1c79e8f0
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=d90b1a2ca4f54990o0
To: sip:00NR_TO_CALL@IP_Asterisk;tag=as149d3d12
Call-ID: 43abab5c-1c126a20@192.168.1.42
CSeq: 101 ACK
Max-Forwards: 70
Contact: “+NR_TO_CALL” sip:10a@192.168.1.42:5060
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
vs8709*CLI>
<— SIP read from IP_Linksys:5062 —>
INVITE sip:00NR_TO_CALL@IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e0ee288c
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=d90b1a2ca4f54990o0
To: sip:00NR_TO_CALL@IP_Asterisk
Remote-Party-ID: “+NR_TO_CALL” sip:10a@IP_Asterisk;screen=yes;party=calling
Call-ID: 43abab5c-1c126a20@192.168.1.42
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“10a”,realm=“asterisk”,nonce=“13829fd9”,uri=“sip:00NR_TO_CALL@IP_Asterisk”,algorithm=MD5,response="61794f75ff9ee497e0d1967177766998"
Contact: “+NR_TO_CALL” sip:10a@192.168.1.42:5060
Expires: 240
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 1198925 1198925 IN IP4 192.168.1.42
s=-
c=IN IP4 192.168.1.42
t=0 0
m=audio 16460 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
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=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (16 headers 20 lines) —
Sending to IP_Linksys : 5062 (NAT)
Using INVITE request as basis request - 43abab5c-1c126a20@192.168.1.42
Found user '10a’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
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
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.42:16460
Looking for 00NR_TO_CALL in app10 (domain IP_Asterisk)
list_route: hop: sip:10a@192.168.1.42:5060
vs8709*CLI>
<— Transmitting (NAT) to IP_Linksys:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e0ee288c;received=IP_Linksys
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=d90b1a2ca4f54990o0
To: sip:00NR_TO_CALL@IP_Asterisk
Call-ID: 43abab5c-1c126a20@192.168.1.42
CSeq: 102 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:00NR_TO_CALL@IP_Asterisk
Content-Length: 0

<------------>
– Executing [00NR_TO_CALL@app10:1] Dial(“SIP/10a-0000074f”, “SIP/00NR_TO_CALL@9975120|45|r”) in new stack
Audio is at IP_Asterisk port 10276
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 37.58.67.5:6321:
INVITE sip:00NR_TO_CALL@sip.gulfsip.com:6321 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk:5060;branch=z9hG4bK6d173508;rport
From: “Home” sip:+NR_fromuser@IP_Asterisk;tag=as52572b92
To: sip:00NR_TO_CALL@sip.gulfsip.com:6321
Contact: sip:+NR_fromuser@IP_Asterisk
Call-ID: 1044041b2cb5783f01d1d6371d0ab938@IP_Asterisk
CSeq: 102 INVITE
User-Agent: MyDevice
Max-Forwards: 70
Date: Fri, 14 Sep 2012 15:49:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 16787 16787 IN IP4 IP_Asterisk
s=session
c=IN IP4 IP_Asterisk
t=0 0
m=audio 10276 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendr

============================================================================

            • Continues here from an other test call (same number) * * * * * *
              ============================================================================

v=0
o=root 16787 16787 IN IP4 IP_Asterisk
s=session
c=IN IP4 IP_Asterisk
t=0 0
m=audio 18412 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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 00NR_TO_CALL@9975120

vs8709*CLI>
<— Transmitting (NAT) to IP_Linksys:5062 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e6291fe8;received=IP_Linksys
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=e335ad4a72f4390o0
To: sip:00NR_TO_CALL@IP_Asterisk;tag=as099fc966
Call-ID: 7f5cb352-6420b528@192.168.1.42
CSeq: 102 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:00NR_TO_CALL@IP_Asterisk
Content-Length: 0

<------------>
vs8709*CLI>
<— SIP read from 37.58.67.5:6321 —>
SIP/2.0 403 No relaying
Via: SIP/2.0/UDP IP_Asterisk:5060;received=IP_Asterisk;branch=z9hG4bK31b7ebf3;rport=5060
From: “Home” sip:+NR_fromuser@IP_Asterisk;tag=as5591a135
To: sip:00NR_TO_CALL@sip.gulfsip.com:6321;tag=3c0de42e4be99a5add523ae5e2615b5d.5030
Call-ID: 299870370ba8a8e93cddf33973e3d32c@IP_Asterisk
CSeq: 102 INVITE
Server: OpenSIPS (1.7.1-tls (x86_64/linux))
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 37.58.67.5:6321:
ACK sip:00NR_TO_CALL@sip.gulfsip.com:6321 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk:5060;branch=z9hG4bK31b7ebf3;rport
From: “Home” sip:+NR_fromuser@IP_Asterisk;tag=as5591a135
To: sip:00NR_TO_CALL@sip.gulfsip.com:6321;tag=3c0de42e4be99a5add523ae5e2615b5d.5030
Contact: sip:+NR_fromuser@IP_Asterisk
Call-ID: 299870370ba8a8e93cddf33973e3d32c@IP_Asterisk
CSeq: 102 ACK
User-Agent: MyDevice
Max-Forwards: 70
Content-Length: 0


[2012-09-14 17:45:06] WARNING[17168]: chan_sip.c:13053 handle_response_invite: Received response: “Forbidden” from ‘“Home” sip:+NR_fromuser@IP_Asterisk;tag=as5591a135’
– SIP/9975120-0000074e is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘SIP/10a-0000074d’ status is 'CONGESTION’
vs8709*CLI>
<— Reliably Transmitting (NAT) to IP_Linksys:5062 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e6291fe8;received=IP_Linksys
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=e335ad4a72f4390o0
To: sip:00NR_TO_CALL@IP_Asterisk;tag=as099fc966
Call-ID: 7f5cb352-6420b528@192.168.1.42
CSeq: 102 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21

<------------>
Really destroying SIP dialog ‘299870370ba8a8e93cddf33973e3d32c@IP_Asterisk’ Method: INVITE
vs8709*CLI>
<— SIP read from IP_Linksys:5062 —>
ACK sip:00NR_TO_CALL@IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e6291fe8
From: “+NR_TO_CALL” sip:10a@IP_Asterisk;tag=e335ad4a72f4390o0
To: sip:00NR_TO_CALL@IP_Asterisk;tag=as099fc966
Call-ID: 7f5cb352-6420b528@192.168.1.42
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“10a”,realm=“asterisk”,nonce=“17bd5247”,uri=“sip:00NR_TO_CALL@IP_Asterisk”,algorithm=MD5,response="5a874f03246b9bc85a1fd2753f23902f"
Contact: “+NR_TO_CALL” sip:10a@192.168.1.42:5060
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘7f5cb352-6420b528@192.168.1.42’ Method: ACK
vs8709*CLI>[/code]

SIP/2.0 403 No relaying

It seems to think that the destination address isn’t local to it.

[quote=“david55”]It seems to think that the destination address isn’t local to it.[/quote]Or, could it be that they compare the countries of the mobile phone which had to be provided at registration and the IP_Asterisk which is located in a different country? To check this I would have to run the softphone on an IP from the country where the vServer of the Asterisk is located.