[asterisk-users] sipgate doesn't work with sipgate anymore

[Michael Obster]

Hi,

looks like I've found the solution by myself. The sipgate_out context
needs the parameter
insecure=invite
also I missed to set the context for the dialplan.

So in sip.conf using
------
[sipgate_out]
type=friend
context=extern
insecure=invite
nat=yes
username=1234567
fromuser=1234567
fromdomain=sipgate.de
secret=secret
host=sipgate.de
qualify=yes
------

works. Hope this information helps other people because I looked into 5
forums/links on google found 5 question on this topic but no answer ;-).

Regards,
Michael

Michael Obster schrieb:
> Hi,
> 
> have some problem with incoming calls from sipgate. This was working in
> 1.4 but in 1.6 I get a 401 Unauthorized :-(.
> 
> Sipgate has mentioned that I have to change the type to friend, but it
> is already friend, so what's wrong?
> 
> Kind regards,
> Michael
> 
> Here is the sip.conf:
> [sipgate_out]
> type=friend
> nat=yes
> username=1234567
> fromuser=1234567
> fromdomain=sipgate.de
> secret=secret
> host=sipgate.de
> qualify=yes
> 
> 
> Here is the SIP trace:
> <------------->
> --- (18 headers 19 lines) ---
>   == Using SIP RTP CoS mark 5
> Sending to 217.10.79.9 : 5060 (no NAT)
> Using INVITE request as basis request -
> 40fa80331421fa800e4633bd497e1cd8 at sipgate.de
> No user '015122633153' in SIP users list
> Found peer 'sipgate_out' for '015122633153' from 217.10.79.9:5060
> 
> <--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 217.10.79.9:5060;branch=z9hG4bK14af.633376b5.0;received=217.10.79.9
> Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK14af.633376b5.0
> Via: SIP/2.0/UDP
> 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK6263330b
> Via: SIP/2.0/UDP 217.10.67.8:5060;branch=z9hG4bK6263330b;rport=5060
> From: "015122633153" <sip:015122633153 at sipgate.de>;tag=as2931c3cc
> To: <sip:00498411111111 at sipgate.de>;tag=as795f5a0d
> Call-ID: 40fa80331421fa800e4633bd497e1cd8 at sipgate.de
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.0.9
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="041f4025"
> Content-Length: 0
> 
> 
> <------------>
> Scheduling destruction of SIP dialog
> '40fa80331421fa800e4633bd497e1cd8 at sipgate.de' in 6400 ms (Method: INVITE)
> netmaster*CLI>
> <--- SIP read from UDP://217.10.79.9:5060 --->
> ACK sip:1234567 at 192.168.173.2:5060 SIP/2.0
> Max-Forwards: 10
> Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK14af.633376b5.0
> Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK14af.633376b5.0
> From: "015122633153" <sip:015122633153 at sipgate.de>;tag=as2931c3cc
> Call-ID: 40fa80331421fa800e4633bd497e1cd8 at sipgate.de
> To: <sip:00498411111111 at sipgate.de>;tag=as795f5a0d
> CSeq: 102 ACK
> Content-Length: 0
> X-hint: rr-enforced
> 
> 
> 
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