(incoming SIP) Call rejected because extension not found

Asterisk Asterisk Support
1

Hi,

I have a Asterisk Appliance AA50 and a lot of difficulties to receive incoming SIP calls from sipgate.de

First some debug output:

[code]<— SIP read from 217.10.79.9:5060 —>
INVITE sip:yyyyyyyy@192.168.17.200 SIP/2.0
Record-Route: sip:217.10.79.9;lr=on;ftag=as08cea9e1
Record-Route: sip:172.20.40.3;lr=on
Record-Route: sip:217.10.79.9;lr=on;ftag=as08cea9e1
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf86f.7f2f98b2.3
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKf86f.7f2f98b2.3
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK25e124d5
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK25e124d5;rport=5060
From: “001902xxxxxxx” sip:001902xxxxxxx@sipgate.de;tag=as08cea9e1
To: sip:0049931xxxxxxx@sipgate.de
Contact: sip:001902xxxxxxx@217.10.67.141
Call-ID: 26c485d417220fbd3aca8b804f4b0a2d@sipgate.de
CSeq: 102 INVITE
Max-Forwards: 67
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 404

<— Reliably Transmitting (NAT) to 217.10.79.9:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf86f.7f2f98b2.3;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKf86f.7f2f98b2.3
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK25e124d5
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK25e124d5;rport=5060
From: “001902xxxxxxx” sip:001902xxxxxxx@sipgate.de;tag=as08cea9e1
To: sip:0049931xxxxxxx@sipgate.de;tag=as49226b6d
Call-ID: 26c485d417220fbd3aca8b804f4b0a2d@sipgate.de
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
[Nov 14 22:35:02] NOTICE[354]: chan_sip.c:14527 handle_request_invite: Call from ‘yyyyyyyy’ to extension ‘yyyyyyyy’ rejected because extension not found.
Scheduling destruction of SIP dialog ‘26c485d417220fbd3aca8b804f4b0a2d@sipgate.de’ in 32000 ms (Method: INVITE)

[/code]

Ok, I understand that * can’t find the extension ‘yyyyyyyy’

The appliance has been configured through the GUI.

There is a incoming call rule for this trunk with pattern ‘s’. In my opinion this should match.

Maybe I am blind after 3 days debugging but I can’t see in which context * is looking for this extension.

Anybody who can help?

Thanks,
Ralf

users.conf:

context=DID_yyyyyyyy
  host=sipgate.de
  username=yyyyyyyy
  secret=xxxxxxx
  fromuser=yyyyyyyy
  fromdomain=sipgate.de
  canreinvite=no
  nat=yes
  type=friend
  insecure=very
  trunkname=sipgate e1
  hasiax=no
  registeriax=no
  hassip=yes
  registersip=yes
  trunkstyle=voip
  hasexten=no
  disallow=all
  allow=all

extensions.conf

[DID_yyyyyyyy]
include=DID_yyyyyyyy_default

[DID_yyyyyyyy_default]
exten=s,1,Goto(default|201|1)

sip.conf

context=default
  bindport=5060
  bindaddr=0.0.0.0
  srvlookup=yes
  limitonpeers=yes
  allowexternaldomains=no
  allowexternalinvites=yes
  allowguest=yes
  allowoverlap=yes
  allowsubscribe=no
  allowtransfer=yes
  alwaysauthreject=no
  autodomain=no
  callevents=no
  canreinvite=nonat
  checkmwi=10
  compactheaders=no
  defaultexpiry=
  domain=192.168.17.200
  dtmfmode=rfc2833
  dumphistory=no
  externhost=xyz.dnsalias.org
  externip=
  externrefresh=60
  fromdomain=
  g726nonstandard=no
  jbenable=no
  jbforce=no
  jbimpl=
  jblog=no
  jbmaxsize=
  jbresyncthreshold=
  language=
  localnet=192.168.17.0/24
  maxcallbitrate=384
  maxexpiry=3600
  minexpiry=60
  mohinterpret=
  mohsuggest=
  nat=yes
  notifymimetype=
  notifyringing=no
  pedantic=no
  progressinband=
  promiscredir=no
  realm=
  recordhistory=no
  registerattempts=0
  registertimeout=60
  relaxdtmf=no
  rtpholdtimeout=
  rtptimeout=
  sendrpid=no
  sipdebug=no
  subscribecontext=
  t1min=100
  t38pt_udptl=no
  tos_audio=ef
  tos_sip=CS3
  tos_video=AF41
  trustrpid=no
  useragent=
  usereqphone=yes
  videosupport=yes
  register=yyyyyyyy:xxxxxxx@sipgate.de/yyyyyyyy
  disallow=all
  allow=ulaw,alaw,gsm,g726,g722
2

Hi

[quote]There is a incoming call rule for this trunk with pattern ‘s’. In my opinion this should match.
[/quote]

S is not a match all, its a match nothing realy.

to match all use _x. for example. not _.

Ian

3

Hi Ian,

I tried _x. as well. Doesn’t make any difference.

I also tried ‘yyyyyyyy’ (the extension mentioned in debug output) and my actual phone number ‘0049931xxxxxxxx’. Nothing works.

I am assuming * is looking in a different context but I don’t know in which one because I added the extension to ‘default’ as well.

Ralf

4

in your context for the incoming sip trunk you need to put
context = whatever it is

right now you have it using context = default from the general settings

5

My users.conf contains a context:

context=DID_yyyyyyyy

I am wondering if there is a way to get the context in the debug output?

6

I notice that you have nat turned on. Is there a reason? If you have a firewall, that needs to be checked.

7

[quote][DID_yyyyyyyy_default]
exten=s,1,Goto(default|201|1) [/quote]

Should be
[DID_yyyyyyyy_default]
exten => s,1,Goto(default|201|1)

Ian

8

You mean I should replace “=” by “=>” ?

This line was written by the Asterisk GUI and it works well for an IAX trunk…

:confused:

9

According to voip-info.org/wiki-Asterisk+ … sions.conf the => in the dialplan can be replaced by =, but I have never tried it. It might be worth a try if you think that this step is not being executed.

Ian.