Topic: Inbound calls failing

[ReetP]

In addition to the link to my Telrad PSTN box which works I have also got two VOIP accounts which I have inherited, one with VOIPStunt (same company as similar to VOIPBuster I think) and one with Draytek. They're more for testing than serious use.

I can get outbound calls on both of them, but I cannot get inbound calls.

I setup two trunks. I used the VOIPBuster template for the VOIPStunt one with details as follows :

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Registration string - John****:password@sip.voipstunt.com

type=peer
host=sip.voipstunt.com
fromdomain=sip.voipstunt.com
qualify=3000
canreinvite=no
username=John****
fromuser=John****
secret=password
disallow=all
allow=alaw
allow=ulaw
Draytel I made an educated guess from the General SIP template:

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Registration string - 824xxxx:password@draytel.org

type=peer
host=draytel.org
qualify=3000
canreinvite=no
username=824xxxx
fromuser=824xxxx
secret=password
insecure=very
disallow=all
allow=alaw
allow=ulaw

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faxserver*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
VoipStunt/John****        194.120.0.198               5060     OK (66 ms)
DraytelVOIP/824****        217.14.132.183              5060     OK (45 ms)
Telrad                     10.0.0.180                  5060     OK (1 ms)
5000/John                  10.0.0.48        D          5060     OK (171 ms)


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Host                            Username       Refresh State                Reg.Time
sip.voipstunt.com:5060          John****          145 Registered           Wed, 16 Dec 2009 17:53:11
draytel.org:5060                824****            145 Registered           Wed, 16 Dec 2009 17:53:11


For an incoming VOIPStunt call I get a trace as follows :

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Capturing on eth0
  0.000000 194.120.0.198 -> 10.0.0.2     SIP/SDP Request: INVITE sip:s@213.123.128.243, with session description
  0.000761     10.0.0.2 -> 194.120.0.198 SIP Status: 407 Proxy Authentication Required
  0.071443 194.120.0.198 -> 10.0.0.2     SIP Request: ACK sip:s@213.123.128.243
 13.151150     10.0.0.2 -> 194.120.0.198 SIP Request: OPTIONS sip:sip.voipstunt.com
 13.220071 194.120.0.198 -> 10.0.0.2     SIP Status: 200 Ok
 28.227592     10.0.0.2 -> 194.120.0.198 SIP Request: REGISTER sip:sip.voipstunt.com
 28.300599 194.120.0.198 -> 10.0.0.2     SIP Status: 401 Unauthorized    (1 bindings)
 28.301001     10.0.0.2 -> 194.120.0.198 SIP Request: REGISTER sip:sip.voipstunt.com
 28.380895 194.120.0.198 -> 10.0.0.2     SIP Status: 200 Ok    (1 bindings)

From Draytel I get this :

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Capturing on eth0
  0.000000 217.14.132.183 -> 10.0.0.2     SIP/SDP Request: INVITE sip:s@10.0.0.2, with session description
  0.001133     10.0.0.2 -> 217.14.132.183 SIP Status: 404 Not Found
  0.050057 217.14.132.183 -> 10.0.0.2     SIP Request: ACK sip:s@10.0.0.2
 24.243026     10.0.0.2 -> 217.14.132.183 SIP Request: OPTIONS sip:draytel.org
 24.288014 217.14.132.183 -> 10.0.0.2     SIP Status: 484 Address Incomplete
 64.411533     10.0.0.2 -> 217.14.132.183 SIP Request: REGISTER sip:draytel.org
 64.457354 217.14.132.183 -> 10.0.0.2     SIP Status: 100 Trying    (0 bindings)
 64.461554 217.14.132.183 -> 10.0.0.2     SIP Status: 401 Unauthorized    (0 bindings)
 64.461947     10.0.0.2 -> 217.14.132.183 SIP Request: REGISTER sip:draytel.org
 64.510564 217.14.132.183 -> 10.0.0.2     SIP Status: 100 Trying    (0 bindings)
 64.518463 217.14.132.183 -> 10.0.0.2     SIP Status: 200 OK    (1 bindings)
 84.289311     10.0.0.2 -> 217.14.132.183 SIP Request: OPTIONS sip:draytel.org
 84.336323 217.14.132.183 -> 10.0.0.2     SIP Status: 484 Address Incomplete

The box is behind a router but it is set in a DMZ

I have set an external IP which can be seen in the VOIPStunt details but doesn't show for the Draytek though I'm not sure why.

Any ideas why these calls do not come through ? Guess I've missed something stupidly simple but can't figure out what.

[SARK devs]

Your invites are malformed causing them to drive the s extension in Asterisk.  SAIL won't accept INVITES into the s extension (at least not without modification).

INVITE sip:s@213.123.128.243

This probabaly due to the fact that you haven't tagged your registration strings with the DDI (DiD) you wish to drive.

John****:password@sip.voipstunt.com

Usually better to drive a DDI like this...

John****:password@sip.voipstunt.com/myddi

Then, (if you haven't already specified the DiD in the SIP trunk) create a PTT_DiD_Group DDI to receive the call into.

Usually this


Kind Regards

S


 

[ReetP]

Thanks for the reply.

Your invites are malformed causing them to drive the s extension in Asterisk.  SAIL won't accept INVITES into the s extension (at least not without modification).

This probabaly due to the fact that you haven't tagged your registration strings with the DDI (DiD) you wish to drive.
Usually better to drive a DDI like this...

John****:password@sip.voipstunt.com/mydddi

Ok, changed that. The Draytel is now fine and accepts incoming calls. The VOIPStunt doesn't - details below

Then, (if you haven't already specified the DiD in the SIP trunk) create a PTT_DiD_Group DDI to receive the call into.

I had the extensions setup before for the connection to my Telrad box. Still not really clear why I need a trunk and a PTT_DiD_Group if I have registered an 'Extension' !!!!! Well beyond my newbie level of understanding. I have 3 extensions in there.


WRT the VOIPStunt account, I get a number unobtainable when I dial it. I wasn't sure if it was to do with the registration and this :

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4910.890422     10.0.0.2 -> 77.72.169.129 SIP Request: REGISTER sip:sip.voipstunt.com
4910.953210 77.72.169.129 -> 10.0.0.2     SIP Status: 401 Unauthorized    (1 bindings)

Alterantively is it because I haven't got the trunk settings right and it doesn't how to deal with the call ??

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5307.436536 77.72.169.129 -> 10.0.0.2     SIP/SDP Request: INVITE sip:05601566xxx@213.123.128.xxx, with session description
5307.437388     10.0.0.2 -> 77.72.169.129 SIP Status: 407 Proxy Authentication Required
5307.498794 77.72.169.129 -> 10.0.0.2     SIP Request: ACK sip:05601566xxx@213.123.128.xxx


------------->
--- (11 headers 14 lines) ---
Sending to 77.72.169.129 : 5060 (no NAT)
Using INVITE request as basis request - f330f9ea60cb481b88099e65a1bfa848
Found peer 'VoipStunt'                                                 

<--- Reliably Transmitting (no NAT) to 77.72.169.129:5060 --->
SIP/2.0 407 Proxy Authentication Required                     
Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK03589370528e4a8985a16156294fce68;received=77.72.169.129
From: <sip:00441621783xxx@sip.voipstunt.com:5060>;tag=110113ac4af9b8e4528429                             
To: <sip:Johnxxx@213.123.128.xxx>;tag=as6800fed1                                                       
Call-ID: f330f9ea60cb481b88099e65a1bfa848                                                               
CSeq: 7 INVITE                                                                                           
User-Agent: Asterisk PBX                                                                                 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY                                       
Supported: replaces                                                                                     
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40f1deed"                             
Content-Length: 0                                                                                       


<------------>
Scheduling destruction of SIP dialog 'f330f9ea60cb481b88099e65a1bfa848' in 6400 ms (Method: INVITE)
faxserver*CLI>                                                                                     
<--- SIP read from 77.72.169.129:5060 --->
ACK sip:05601566429@213.123.128.xxx SIP/2.0
Via: SIP/2.0/UDP 77.72.169.129:5060;branch=z9hG4bK03589370528e4a8985a16156294fce68
From: <sip:00441621783xxx@sip.voipstunt.com:5060>;tag=110113ac4af9b8e4528429
To: <sip:Johnxxx@213.123.128.xxx>;tag=as6800fed1
Contact: sip:00441621783xxx@77.72.169.129:5060
Call-ID: f330f9ea60cb481b88099e65a1bfa848
CSeq: 7 ACK
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

I would guess it is  the incoming call not being matched to an extension on the system, but I'm blowed if I know what too change !!!!!!!

Any help much appreciated - I'm pleased I have the Draytel settings working. Would be nice to get the VOIPStunt ones going as well.

[SARK devs]

you've missed the insecure=very couplet from the voipstunt sip peer in sip.conf

Kind Regards

S

[ReetP]

you've missed the insecure=very couplet from the voipstunt sip peer in sip.conf

Complete genius !!! Solved in one. Thank you so very very much.

B. Rgds
John

P.S. any idea why it still shows a

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SIP Status: 401 Unauthorized error in the tethereal ?
It doesn't seem to affect calls.

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IP Request: OPTIONS sip:draytel.org
SIP Status: 484 Address Incomplete

I wondered if that was because it is missing a hostname ??

It doesn't seem to affect things but I was just curious