Topic: Call rejected because extension not found

[ReetP]

Ok, I give up. I can't find the answer to this ! I'm no expert in this field but have read as much as I can.

I have a Telrad system connected to an Asterisk box, both on a 10.0.0.x network. The extensions are running on a 'localnetwork' 192.168.10.x connected via VPN

The Telrad has one incoming DDI number directed to the Asterisk box.

The Asterisk box is on an otherwise clean 7.4 server with the Asterisk 1.4.25 contribs


Things I can do :

Call any other Asterisk extension
Call any Telrad extension
Dial out via the Telrad box to the PSTN

What I can't do :

Dial any Asterisk extension from the Telrad box
Receive any incoming calls from the Telrad box to the Asterisk box

I can see the incoming calls on the Asterisk box, but it comes up as 'extension not found'

I thought that it might be because the extensions are on the 192.168.10.x network, but having tried an extension on the 10.0.0.x network, the same problem is still there.

The incoming calls from the public network come via a DDI number.

The first log below is a call made from an extenson on the Telrad system to an extension on the Asterisk. The extension is actually via a VOIP phone situated in Spain (a long & complicated story), hence the extension number, but the same occurs if I make a call from a local Telrad extension

Below that I have pasted an Asterisk extension to extension call

Below that is an incoming call from off site to the DDI number

I am not sure how the Inbound Routes should be set in Trunks. I have tried to a single extension and to a group but it doesn't make any difference and I did wonder how it should be set correctly ??

If anyone has any ideas I would be grateful !!!

B. Rgds
John

Logs - can't get them all in one post.......... will post others in follow up

Telrad Extension to Asterisk Extension

Code: [Select]

faxserver*CLI>                                                                         
<--- SIP read from 10.0.0.180:5060 --->                                               
INVITE sip:5000@10.0.0.2:5060;user=phone SIP/2.0                                       
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b412089                           
From: "SPAIN" <sip:270@10.0.0.180:5060;user=phone>;tag=8f35210                         
To: <sip:5000@10.0.0.2:5060;user=phone>                                               
Call-ID: 991108288@10.0.0.180                                                         
CSeq: 1 INVITE                                                                         
Contact: "SPAIN" <sip:270@10.0.0.180:5060;user=phone>                                 
Max-Forwards: 70                                                                       
Organization: Telrad_Connegy                                                           
Connegy: <NSI-B=5A 53 2A 54 2A 37 >                                                   
Content-Type: application/sdp                                                         
Content-Length: 190                                                                   

v=0
o=Telrad_Connegy 138607856 138607856 IN IP4 10.0.0.180
s=SIP Call                                           
c=IN IP4 192.168.10.183                               
t=0 0                                                 
m=audio 6000 RTP/AVP 0 101                           
a=rtpmap:0 PCMU/8000                                 
a=rtpmap:101 telephone-event/8000                     

<------------->
--- (12 headers 8 lines) ---
Sending to 10.0.0.180 : 5060 (no NAT)
Using INVITE request as basis request - 991108288@10.0.0.180
Found peer 'Telrad'                                         
Found RTP audio format 0                                   
Found RTP audio format 101                                 
Peer audio RTP is at port 192.168.10.183:6000               
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.10.183:6000
Looking for 5000 in mainmenu (domain 10.0.0.2)

<--- Reliably Transmitting (no NAT) to 10.0.0.180:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b412089;received=10.0.0.180
From: "SPAIN" <sip:270@10.0.0.180:5060;user=phone>;tag=8f35210
To: <sip:5000@10.0.0.2:5060;user=phone>;tag=as3dd748ed
Call-ID: 991108288@10.0.0.180
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[Dec  1 16:42:22] NOTICE[8292]: chan_sip.c:14703 handle_request_invite: Call from 'Telrad' to extension '5000' rejected because extension not found.
Scheduling destruction of SIP dialog '991108288@10.0.0.180' in 6400 ms (Method: INVITE)
faxserver*CLI>
<--- SIP read from 10.0.0.180:5060 --->
ACK sip:5000@10.0.0.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b412089;received=10.0.0.180
From: "SPAIN" <sip:270@10.0.0.180:5060;user=phone>;tag=8f35210
To: <sip:5000@10.0.0.2:5060;user=phone>;tag=as3dd748ed
Call-ID: 991108288@10.0.0.180
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '991108288@10.0.0.180' Method: ACK
Really destroying SIP dialog 'Conn1381484881804289383' Method: OPTIONS
faxserver*CLI>
<--- SIP read from 10.0.0.180:5060 --->
OPTIONS sip:10.0.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b4100
To: <sip:10.0.0.2:5060>
From: <sip:10.0.0.180:5060>;tag=83bfa88
CSeq: 1 OPTIONS
Call-ID: Conn1381484881804289383
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Looking for s in mainmenu (domain 10.0.0.2)

<--- Transmitting (no NAT) to 10.0.0.180:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b4100;received=10.0.0.180
From: <sip:10.0.0.180:5060>;tag=83bfa88
To: <sip:10.0.0.2:5060>;tag=as53f8bcb9
Call-ID: Conn1381484881804289383
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Conn1381484881804289383' in 32000 ms (Method: OPTIONS)

[ReetP]

Tried to post the other bits but they are too long

I have put them here :

http://www.reetspetit.net/mybay/Other/Sip_debug_1.txt

Not sure why I have a 'null null' below ???

[SARK devs]

Hello John

You don't say if you are running a workbench but it looks like SAIL or SARK.  Here is your problem...

Code: [Select]

Using INVITE request as basis request - 991108288@10.0.0.180
Found peer 'Telrad'                                         
Found RTP audio format 0                                   
Found RTP audio format 101                                 
Peer audio RTP is at port 192.168.10.183:6000               
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.10.183:6000
Looking for 5000 in mainmenu (domain 10.0.0.2)            <== RIGHT HERE!!!!!

<--- Reliably Transmitting (no NAT) to 10.0.0.180:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b412089;received=10.0.0.180
From: "SPAIN" <sip:270@10.0.0.180:5060;user=phone>;tag=8f35210
To: <sip:5000@10.0.0.2:5060;user=phone>;tag=as3dd748ed
Call-ID: 991108288@10.0.0.180
CSeq: 1 INVITE
Your call has arrived with an invite of 5000@10.0.0.2, which is correct... unfortunately, the extension 5000 does not exist in the context "mainmenu" (which is the inbound call path; usually used to process external calls coming in on DDI  numbers).  You are sending in an invite for an internal extension, which lives under the context "internal".  Asterisk can't find extension 5000 in the mainmenu context so it drops the call with a 404 Notfound.

So,  how to fix it...  Simply define a PTT_DiD trunk with a DiD (DDI for the Brits) number of 5000 and route it to extension 5000 (you shouldn't actually need to do the routing, SAIL is usually smart enough to figure it out automatically for you).

Should get you back on the road, at least as far as inbound is concerned.

Kind Regards

S

 

[ReetP]

AAAAAAAAaahhhhhhhhhhh !!!!!!!!!!!!!!

Platform is home built SME + Asterisk + SAIL

Think I get it now. Can't access it from here but will be in the office in the morrow - had to fly back last night for another emergency.

Many thanks for the pointer - I'm sure you have it spot on. Will file a report asap.

Many thanks !!!!!!

[ReetP]

Hmmm.

If I try to create a trunk with a DID of 5000 I get the following error :

Operation status report - Keyfield already exists in the Database - Choose another name for the Trunk

I can create one called 5000@10.0.0.2 with inbound routes to 5000 but I'm not sure that's right.

If I do, when I call from a Telrad extension I don't get the 'Extension not found' but do get the following :

Code: [Select]

<--- SIP read from 10.0.0.180:5060 --->
OPTIONS sip:10.0.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b4100
To: <sip:10.0.0.2:5060>
From: <sip:10.0.0.180:5060>;tag=83bfa88
CSeq: 1 OPTIONS
Call-ID: Conn1381484881804289383
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Looking for s in mainmenu (domain 10.0.0.2)

<--- Transmitting (no NAT) to 10.0.0.180:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b4100;received=10.0.0.180
From: <sip:10.0.0.180:5060>;tag=83bfa88
To: <sip:10.0.0.2:5060>;tag=as56b52852
Call-ID: Conn1381484881804289383
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Conn1381484881804289383' in 32000 ms (Method: OPTIONS)
faxserver*CLI>
<--- SIP read from 10.0.0.180:5060 --->
INVITE sip:5000@10.0.0.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b416252
From: "SARAH" <sip:22@10.0.0.180:5060;user=phone>;tag=9323b5c
To: <sip:5000@10.0.0.2:5060;user=phone>
Call-ID: -1249937032@10.0.0.180
CSeq: 1 INVITE
Contact: "SARAH" <sip:22@10.0.0.180:5060;user=phone>
Max-Forwards: 70
Organization: Telrad_Connegy
Connegy: <NSI-B=5A 53 2A 54 2A 37 >
Content-Type: application/sdp
Content-Length: 186

v=0
o=Telrad_Connegy 138750984 138750984 IN IP4 10.0.0.180
s=SIP Call
c=IN IP4 10.0.0.181
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (12 headers 8 lines) ---
Sending to 10.0.0.180 : 5060 (no NAT)
Using INVITE request as basis request - -1249937032@10.0.0.180
Found peer 'Telrad'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.181:6000
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.0.181:6000
Looking for 5000 in mainmenu (domain 10.0.0.2)

<--- Reliably Transmitting (no NAT) to 10.0.0.180:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b416252;received=10.0.0.180
From: "SARAH" <sip:22@10.0.0.180:5060;user=phone>;tag=9323b5c
To: <sip:5000@10.0.0.2:5060;user=phone>;tag=as4584326b
Call-ID: -1249937032@10.0.0.180
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '-1249937032@10.0.0.180' in 6400 ms (Method: INVITE)
faxserver*CLI>
<--- SIP read from 10.0.0.180:5060 --->
ACK sip:5000@10.0.0.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180:5060;branch=z9hG4bKa0000b416252;received=10.0.0.180
From: "SARAH" <sip:22@10.0.0.180:5060;user=phone>;tag=9323b5c
To: <sip:5000@10.0.0.2:5060;user=phone>;tag=as4584326b
Call-ID: -1249937032@10.0.0.180
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

I must be doing something stupidly wrong with a very easy solution but I'm blowed if I can see it !

Any help appreciated.

[SARK devs]

What kind of trunk?

It need sto be a PTT_DiD_Group and you need to be running a fairly late release of SAIL (2.2.4 is best).

Kind REgards

S

[ReetP]

Wrong type of trunk !

Changed it to your suggestion.

Yeee ha - seems to have done it. Had a quick look but in I a rush, but will look in depth on Monday

Many thanks

[ReetP]

I can confirm this all works OK now thank you very much.

I have other queries now but will post a new thread.

Many thanks again