pstn can't find my registered voip numbers

Hi,

We have a 12 VOIP extension installation in our office. 11 extensions are registerd to PBXes, one directly to the SIP provider. The phones (Grandstream) are behind a NAT router.

"Internal calling" between PBXes extensions works fine.

The directly registered (soft)phone to one of my SIP accounts works fine; Using that (soft)phone, I can then call all registered PBXes trunks that have an inbound route defined via the full DID number. The directly registered DID number is reachable both via Voip as well as PSTN.

However, if I try to call that same PBXes registered trunk from the PSTN network, the DID number appears to be unregistered or something like that, the call is immediatly answered by the provider's voicemail and there is no log entry in PBXes.

Ant ideas?

Maarten

Hi Maarten,

On the donzee account there is currently 1 Classic and 6 SIP extensions. Are the rest of the extensions registered on another account?

At the time you tried to call the PBXes registered DID from the PSTN, was your soft-phone registered on the same account?

Since there is no log entry on your PBXes account, what does the DID provider's log indicate happened on these calls?

Does Ritstele require using the 38383 port or can their SIP Proxy function on the regular 5060 port?

Diafora, thanks for your reply. First I'll answer your questions, then I have some new information,...

1. The other extensions are either not yet registered or directly from the terminal (without PBex in between).

2. The softphone (Express Talk) is registered to a unique account at ritstele. No double registrations exist.

3. I did not yet contact the DID provider but do have some more info from the calling party logfile. See hereunder.

4. I tried to register at port 5060, but that does not work.

When I call 0207074647 (or any other PBex registered number) from "outside" It works a few times but at the 3rd (?) attempt I get a busy reply. This indicates to me that the "Hangup call" is not understood by ritstele.

I am absolutely no expert on this but when I analyze the PBXes log I read:

&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
Jul 14 13:51:27 VERBOSE[24095] chan_sip.c: Hangup call SIP/donzee-647-9dd0, SIP callid 5196e6e14ca25ffc5f2d8df949eae1bd@91.121.136.13
Jul 14 13:51:27 VERBOSE[24095] chan_sip.c: Hangup call SIP/31207074697-026a, SIP callid 069ab0bf37d0287a38860f4a1bc031fc@sip.ritstele.com
&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&

To me it sounds like the hangup is send to the 697 trunk in stead of the 647 trunk and therefore the lines are not disconnected an after the 2nd attempt, there are no more incoming lines. Does this make sense? Apparently we have unlimited ritstele to ritstele lines.

I also recorded a SIP session using voipbuster where I get the "busy" reply:

&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
13:33:18 UDP Packet Received from 194.221.62.198:5060 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 192.168.0.66:38383;branch=z9hG4bK143560;rport
From: "0654730243" <sip:donzee@sip.voipstunt.com>;tag=5117
To: <sip:0207074668@sip.voipstunt.com>;tag=c11710acc12b10ac48708bc612206e
Contact: sip:0207074668@194.221.62.198:5060
Call-ID: 1216034408-3560-MARGERET@192.168.0.66
CSeq: 751 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 201

v=0
o=donzee 1216035198 1216035198 IN IP4 194.221.62.163
s=SIP Call
c=IN IP4 194.221.62.163
t=0 0
m=audio 57824 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

----------------------------------------------------------------

13:33:21 UDP Packet Received from 194.221.62.198:5060 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 192.168.0.66:38383;branch=z9hG4bK143560;rport
From: "0654730243" <sip:donzee@sip.voipstunt.com>;tag=5117
To: <sip:0207074668@sip.voipstunt.com>;tag=c11710acc12b10ac48708bc612206e
Contact: sip:0207074668@194.221.62.198:5060
Call-ID: 1216034408-3560-MARGERET@192.168.0.66
CSeq: 751 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


----------------------------------------------------------------

13:33:21 UDP Packet Sent to 194.221.62.198:5060 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
ACK sip:0207074668@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.66:38383;rport;branch=z9hG4bK143560
To: <sip:0207074668@sip.voipstunt.com>;tag=c11710acc12b10ac48708bc612206e
From: "0654730243" <sip:donzee@sip.voipstunt.com>;tag=5117
Call-ID: 1216034408-3560-MARGERET@192.168.0.66
CSeq: 751 ACK
Max-Forwards: 20
User-Agent: NCH Swift Sound Express Talk 3.08
Content-Length: 0
&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&

Another thought:

Since all trunks register at sip.ritstele.com but under a unique username/password it looks like PBXes does not know which trunk to send a "Hangup" to. It looks like the first defined ritstele trunck gets a hangup instead of the one over which the call comes in.

In my inbound route I refer to the "Trunk" with the "Trunk Name" defined under trunks, the "Caller ID Number" is left blank.

What I want to establish this way is that every trunk with a DID number is routed to another extension based on on the incoming trunk.

Thanks in advance for your reaction,

Maarten

update:

When I register my softclient to a trunk that returned busy via PBSex, all works flawless.

It does look like PBXes looses registration and can'r re-register for some reason.

Any debugging tips? Is there a debugging proxy available for SIP somewhere?

Thanks,

Maarten

Maarten,

In order to facilitate troubleshooting this issue, delete all Ritstele trunks except one, so your tests will be concentrated on a single trunk.

If the single Ritstele trunk and its' associated Inbound Routes work properly, then add another Ritstele trunk and test the functionality of the Inbound Routes with 2 trunks. I believe it will be easier to isolate the issue this way.

It is strange, that your PBXes account can register to the Ritstele trunk once, but cannot re-register later on.

On another note, please inquire with Ritstele, whether their SIP Proxy can accept registration requests on 5060 or another standard port, instead of on the non-standard 38383 port.