Tirxbox (asterix) inbound (recieve calls) trunk

Kramer8u
Kramer8u used Ask the Experts™
on
Hi All ,
Just began to setup my shiny new trixbox CE ( asterix is cool , I am impressed so far..

I have a book (trixbox CE 2.6 Kerry Garrison) + have been reading alot of forum data but im stuck/unsure about how to configure the inbound call settings so the DID works on my test bed.

basically i have :
1 Trixbox (2.8) ( vmwared).
1 softfone X-lite.
1 voip acc with gotalk.
1 DID from gotalk (02804 xxxx)

Config:
I have setup 1 extension (505) with the defaults. and configured X-lite to connect with this.

I have setup 1 trunk (gotalk-SIP) with the following :

PEER details:
allow=alaw
disallow=all
dtmfmode=inband
fromuser=094xxxxx (your account no. NOT DID)
host=sip.gotalk.com
qualify=yes
secret=password (account password)
type=peer
username=094xxxxx

USER context: 094xxxxx
USER details:
context=from-trunk
fromuser=094xxxxx
host=dynamic
insecure=very
nat=yes
secret=password
type=user
username=094xxxxx

Register String:
094xxxxx:password@sip.gotalk.com/094xxxxx

I can call out find and am registering sucessfully

But when I try to dial my DID it goes straight to the gotalk voicemail ...

My question !!
What have i missed in the incoimming setup/ context setup/ DID inbound setup??
the book doesn't detail inbound enough at all.

Why is it going straight to voicemail ?
How do i get the extension ( 505) to ring when an inbound call is made to my DID?

thankyou in advance

Kramer
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Commented:
One thing that you must change is the host setting for the trunk USER details. It needs to be the same as the PEER settings and must not be set to "dynamic".
USER details:
context=from-trunk
fromuser=094xxxxx
host=sip.gotalk.com
insecure=very
...

Depending what you have set as the "Inbound DID" for extension 505, you may also need to change your register string:
Register String:
094xxxxx:password@sip.gotalk.com/505

The number on the end of the register string (after the slash) is what gotalk will send to Trixbox as the DID number of any inbound call so it needs to match the DID number of the extension you want to ring.

Try those changes and let us know if it fixes your problem.

Author

Commented:
thankyou , but still going straight to voicemail :
in extension setup for 505  what do I put here:

Assigned DID/CID
-------------------------------
DID description:
ADD DID:
ADD DID CID:

Is it ( ADD DID) the same number i add after the trailing slash in the registration string???

I changed the registration string and put the extension number as you instructed.
then in the above tried in ADD DID: 505 applyed and restarted but still straight to gotalk voicemail


Commented:
hmm. I thought this was going to be easy! Yes, the "Add Inbound DID" and the number after the trailing slash in the registration string are what I would expect to match, but different VoIP service providers may do things in slightly different ways. Have you tried using the DID number 02804xxxx instead of 505 in those two places?

If that doesn't work, I will need you to collect some sip packets and post them back here. First, can you please just check that the basics are right (they probably are, but its best to be sure).
1. Run the CLI command "sip show peers". You should see something like this:
 Name/username          Host                   Dyn   Nat   ACL   Port       Status
 gotalk-SIP/505            202.169.178.10                            5060      OK (115 ms)
 094xxxxxx                   202.169.178.10                            5060     Unmonitored
 505/505                      192.168.x.y         D                   A  5060     OK (28 ms)

2. Confirm that Trixbox is registered with gotalk.com: CLI command "sip show registry":
 Host                               Username      Refresh State             Reg.Time
 sip.gotalk.com:5060       094xxxx            335   Registered     Mon, 26 Oct 2009 08:47:14

To collect the SIP packets, use the CLI command "sip set debug peer gotalk-SIP". (The name of the peer must be the name shown in the "sip show peers" list). Now make a call to the DID number and you should see a load of data scroll up the screen. Can you capture that data and post it back here as an attached code snippet or file.

To switch off the sip debug, the command is "sip set debug off".
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Author

Commented:
Crikey! ( I'm Aussie, but don't wrestle crocodiles)
I did try my did after the trailing slash then added
it to the ADD did section of the extension 505 setup
but still to voicemail...
I have logged into the asterisks console and did run
some sip command I think show register was on the book
anyways I will cap some packets for you as per your instructions and get back to you
once again thankyou well nail it in the end

Author

Commented:
CLI command Show peers:
Name/username                                  Host              DYN  Nat ACL Port              Status
Goltalk-SIP/09474159           202.169.178.10                                    5060           unmonitored
505/505                                 192.168.1.192                D     N     A    5061            OK (5 ms)
2 sip peers monitored: 1 online,  unmonitored: 1 online
EOF--------------------------------------

CLI command Show registry:
Host                            Username             Refresh         State           Reg
sip.gotalk.com:5060       09474159                 105          Registered    Tue, 27 oct 2009
EOF--------------------------------------

Collected packets:

<--- SIP read from 202.169.178.10:5060 --->
INVITE sip:0280141063@123.243.199.233:5076;user=phone SIP/2.0
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4ae66b00-8341-0
Max-Forwards: 70
Contact: <sip:0405194914@202.169.178.10:5060>
To: <sip:0280141063@192.168.1.192:5060>
From: "0405194914"<sip:0405194914@202.169.178.10:5060>;tag=51ec33ee-co336-INS001
Call-ID: 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
CSeq: 33601 INVITE
Content-Type: application/sdp
Supported: 100rel
User-Agent: ENSR2.5.4
Content-Length: 426

v=0
o=- 1374434286 1374434286 IN IP4 202.169.178.10
s=ENSResip
c=IN IP4 202.169.178.12
t=0 0
m=audio 15108 RTP/AVP 96 97 18 4 8 0 101
a=fmtp:96 mode=30
a=fmtp:97 mode=20
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:96 iLBC/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=sendrecv

<------------->
--- (12 headers 19 lines) ---
Sending to 202.169.178.10 : 5060 (no NAT)
Using INVITE request as basis request - 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
Found peer 'Gotalk-SIP'

<--- Reliably Transmitting (no NAT) to 202.169.178.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4ae66b00-8341-0;received=202.169.178.10
From: "0405194914"<sip:0405194914@202.169.178.10:5060>;tag=51ec33ee-co336-INS001
To: <sip:0280141063@192.168.1.192:5060>;tag=as3cca2a71
Call-ID: 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
CSeq: 33601 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fd4e20b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202' in 32000 ms (Method: INVITE)
trixbox1*CLI>
<--- SIP read from 202.169.178.10:5060 --->
ACK sip:0280141063@123.243.199.233:5076;user=phone SIP/2.0
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4ae66b00-8341-0
To: <sip:0280141063@192.168.1.192:5060>;tag=as3cca2a71
From: "0405194914"<sip:0405194914@202.169.178.10:5060>;tag=51ec33ee-co336-INS001
Call-ID: 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
CSeq: 33601 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 202.169.178.10:5060 --->
INVITE sip:0280141063@123.243.199.233:5076;user=phone SIP/2.0
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4ae66b00-8342-0
Max-Forwards: 70
Contact: <sip:0405194914@202.169.178.10:5060>
To: <sip:0280141063@192.168.1.192:5060>
From: "0405194914"<sip:0405194914@202.169.178.10:5060>;tag=51ec33ee-co336-INS001
Call-ID: 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
CSeq: 33602 INVITE
Content-Type: application/sdp
Proxy-Authorization: Digest username="09474159",realm="asterisk",nonce="2fd4e20b",uri="sip:0280141063@123.243.199.233:5076;user=phone",response="355e113e7460d0227fb9c61d411d1f04",algorithm=MD5
Supported: 100rel
User-Agent: ENSR2.5.4
Content-Length: 426

v=0
o=- 1374434286 1374434286 IN IP4 202.169.178.10
s=ENSResip
c=IN IP4 202.169.178.12
t=0 0
m=audio 15108 RTP/AVP 96 97 18 4 8 0 101
a=fmtp:96 mode=30
a=fmtp:97 mode=20
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:96 iLBC/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=sendrecv

<------------->
--- (13 headers 19 lines) ---
Sending to 202.169.178.10 : 5060 (no NAT)
Using INVITE request as basis request - 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
Found peer 'Gotalk-SIP'

<--- Reliably Transmitting (no NAT) to 202.169.178.10:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4ae66b00-8342-0;received=202.169.178.10
From: "0405194914"<sip:0405194914@202.169.178.10:5060>;tag=51ec33ee-co336-INS001
To: <sip:0280141063@192.168.1.192:5060>;tag=as3cca2a71
Call-ID: 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
CSeq: 33602 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202' in 32000 ms (Method: INVITE)
trixbox1*CLI>
<--- SIP read from 202.169.178.10:5060 --->
ACK sip:0280141063@123.243.199.233:5076;user=phone SIP/2.0
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4ae66b00-8342-0
To: <sip:0280141063@192.168.1.192:5060>;tag=as3cca2a71
From: "0405194914"<sip:0405194914@202.169.178.10:5060>;tag=51ec33ee-co336-INS001
Call-ID: 4c80-4cc-022197061034-TCP-IMG03-3-210.80.190.202
CSeq: 33602 ACK
Content-Length: 0


<------------->





Commented:
Add this line to the PEER details for the gotalk trunk:
insecure=invite

That should fix it, although I do not understand why your trunk USER details are being ignored.

Commented:
I forgot something. The DID number being sent is 0280141063 so you will need to set this as the DID associated with the extension you want to ring.

Author

Commented:
ok its ringing now on the phone that i Dial my DID , but my extension( softphone)  isn't..

I have tried the DID in the registration string and also  the gotalk member ID ..

Also tried both of the above in the ADD DID section of the extension setup (set to send to extension 505) ..

I will play around a bit more and see if i can get the softphone to ring but can you re iterate the inbound setup?
I should sove it myself but yeah , progress !!!! i love progress

Author

Commented:
Hrm...

When I used the extension after the registration astring trailing slash , i can see the inbould call light( Trixbox system status)  up, my soft phone doesn't ring then the call is dropped after 5 seconds.

If u use my DID after the trailing slash of registration string . i can hear it ringing but no recognition in the trixbox system status and my softphone also doesn;t ring.

If i use my gotalk id same as above ...


does this give you any clues?

Commented:
I need you to capture some debug info. Please try the following:
At the CLI command "sip set debug on" (similar to before, but now captures all SIP packets)
At the CLI command "core set verbose 3"
Make a call to your DID number.

Switch off the debug using "sip set debug off" and "core set verbose 1".

It is preferable if you post the results back here using "Attach Code Snippet". Also, please say which combination of trailing number on the registration string and "Add DID" were in use at the time.

Author

Commented:
Below is the data you requested i have run it on the 3 registarion string trailing slash values I.E. DID, userID, Extension.

I have also mapped all 3 of these inbound routes to my extension 505 ..
WITH REGISTRATION STRING: /0280141063
 
    -- Executing [0280141063@from-sip-external:1] NoOp("SIP/09474159-09c0d2d8", "Received incoming SIP connection from unknown peer to 0280141063") in new stack
    -- Executing [0280141063@from-sip-external:2] Set("SIP/09474159-09c0d2d8", "DID=0280141063") in new stack
    -- Executing [0280141063@from-sip-external:3] Goto("SIP/09474159-09c0d2d8", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/09474159-09c0d2d8", "0?from-trunk|0280141063|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/09474159-09c0d2d8", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-10-30 06:58:25 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/09474159-09c0d2d8", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/09474159-09c0d2d8", "2") in new stack
 
    -- Executing [s@from-sip-external:5] Playback("SIP/09474159-09c0d2d8", "ss-noservice") in new stack
    -- <SIP/09474159-09c0d2d8> Playing 'ss-noservice' (language 'en')
 
    -- Executing [s@from-sip-external:6] PlayTones("SIP/09474159-09c0d2d8", "congestion") in new stack
  == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/09474159-09c0d2d8'
    -- Executing [h@from-sip-external:1] NoOp("SIP/09474159-09c0d2d8", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/09474159-09c0d2d8[0    ;37;40m", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/09474159-09c0d2d8", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/09474159-09c0d2d8", "0?from-trunk|s|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/09474159-09c0d2d8[0;3    7;40m", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-10-30 06:58:32 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/09474159-09c0d2d8", "") in new stack
  == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/09474159-09c0d2d8'
   --------------------------------------------------------------
 
WITH REGISTRATION STRING: /09474159
 
-- Executing [09474159@from-sip-external:1] NoOp("SIP/09474159-09cf5c68", "Received incoming SIP connection from unknown peer to 09474159") in new stack
    -- Executing [09474159@from-sip-external:2] Set("SIP/09474159-09cf5c68", "DID=09474159") in new stack
    -- Executing [09474159@from-sip-external:3] Goto("SIP/09474159-09cf5c68", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/09474159-09cf5c68", "0?from-trunk|09474159|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/09474159-09cf5c68", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-10-30 07:06:12 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/09474159-09cf5c68", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/09474159-09cf5c68", "2") in new stack
 
    -- Executing [s@from-sip-external:5] Playback("SIP/09474159-09cf5c68", "ss-noservice") in new stack
    -- <SIP/09474159-09cf5c68> Playing 'ss-noservice' (language 'en')
    -- Executing [s@from-sip-external:6] PlayTones("SIP/09474159-09cf5c68", "congestion") in new stack
  == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/09474159-09cf5c68'
    -- Executing [h@from-sip-external:1] NoOp("SIP/09474159-09cf5c68", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/09474159-09cf5c68[0    ;37;40m", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/09474159-09cf5c68", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/09474159-09cf5c68", "0?from-trunk|s|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/09474159-09cf5c68[0;3    7;40m", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-10-30 07:06:19 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/09474159-09cf5c68", "") in new stack
  == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/09474159-09cf5c68'
 
 
   ----------------------------------------------------------
 
WITH REGISTRATION STRING: /505
 
-- Executing [505@from-sip-external:1] NoOp("SIP/09474159-09cf5c68", "Received incoming SIP connection from unknown peer to 505") in new stack
    -- Executing [505@from-sip-external:2] Set("SIP/09474159-09cf5c68", "DID=505") in new stack
    -- Executing [505@from-sip-external:3] Goto("SIP/09474159-09cf5c68", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/09474159-09cf5c68", "0?from-trunk|505|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/09474159-09cf5c68", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-10-30 07:09:17 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/09474159-09cf5c68", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/09474159-09cf5c68", "2") in new stack
 
    -- Executing [s@from-sip-external:5] Playback("SIP/09474159-09cf5c68", "ss-noservice") in new stack
    -- <SIP/09474159-09cf5c68> Playing 'ss-noservice' (language 'en')
 
    -- Executing [s@from-sip-external:6] PlayTones("SIP/09474159-09cf5c68", "congestion") in new stack
  == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/09474159-09cf5c68'
    -- Executing [h@from-sip-external:1] NoOp("SIP/09474159-09cf5c68", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/09474159-09cf5c68[0    ;37;40m", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/09474159-09cf5c68", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/09474159-09cf5c68", "0?from-trunk|s|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/09474159-09cf5c68[0;3    7;40m", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-10-30 07:09:24 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/09474159-09cf5c68", "") in new stack
  == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/09474159-09cf5c68'

Open in new window

Commented:
Inbound calls from GoTalk should not be reported as "incoming SIP connection from unknown peer". It is supposed to be a known peer. Because the peer sending the calls is regarded as unknown, it means the call is being handled in the context [from-sip-external] - it should be handled in [from-trunk].

I think this must be connected to the fact that your trunk USER details do not appear in the list when you type the command "sip show peers". Please try changing the line "type=user" to "type=friend" in the USER details for the GoTalk trunk. Then reload and see if something appears for the USER context when you type "sip show peers".

Author

Commented:
since changing to friend i get
:
Connections
IP Phones Online 2
IP Trunks Online  1

Name/username              Host            Dyn Nat ACL Port     Status              
Gotalk-SIP/09474159        202.169.178.10              5060     Unmonitored          
501/501                    192.168.1.182    D   N   A  5060     OK (1 ms)            
09474159/09474159          202.169.178.10              5060     Unmonitored  

Why 2 IP phones online ?
before i only had the gotalk and the extension when i run show peers now i have 3
in any case still not ringing on my softphone


Author

Commented:
Just to check again this is what i got: outbound calls only

SIP-Trunk
 
Peer detail :SIP-Gotalk
type=friend
allow=alaw&g729
host=sip.gotalk.com
username=09xxxxx
secret=xxxxxxx
fromdomain=sip.gotalk.com
insecure=invite
-----------------------------------------------------------
User Context: 09xxxxx
type=friend
username=09xxxxx
secret=xxxxx
host=sip.gotalk.com
insecure=very
fromuser=09xxxxx

Reg string: 09xxxxx:xxxxxx@sip.gotalk.com/09xxxxx

-----------------------------
Extension : 501

Assigned DID/CID
 0280141063
 09xxxxx
 501
---------------------------
Default device options :
No voicemail

---------------------------
twinkle SIP phone :
Alex: you have the following registrations
<sip:501@192.168.1.192>;expires=2430
sip show peers:\
Name/username              Host            Dyn Nat ACL Port     Status              
Gotalk-SIP/09xxxxx        202.x.x.10              5060     Unmonitored          
501/501                    192.168.1.182    D   N   A  5060     OK (1 ms)            
09xxxxx/09xxxxx          202.x.x.10              5060     Unmonitored
Commented:
What happened to "context=from-trunk"? That line needs to be in both the Peer and the User details. Please also try changing the name for User Context from "09474159" to "gotalk-in". You should then see a list like this when you type "sip show peers":
Name/username              Host            Dyn Nat ACL Port     Status              
Gotalk-SIP/09474159        202.169.178.10              5060     Unmonitored          
501/501                    192.168.1.182    D   N   A  5060     OK (1 ms)            
gotalk-in/09474159          202.169.178.10              5060     Unmonitored  

This list shows you all the peers that Trixbox knows about - it includes external peers such as gotalk and local peers such as your softphone. In the case of a sip trunk like gotalk, there should be two entries in the list - one for outbound connections and the other for inbound (hence the use of the name gotalk-in).

P.S. Ask gotalk to change the password on your account a.s.a.p. You have posted more info here than you should have; the account could be hijacked.

Author

Commented:
grr i thought i got that damn PW...

I will get password changed ASAP , anyways its only 5 dollar pre-paid that i have to top up at the moment , not going on a plan till i iron out all the setup IVR etc....
 will try these , thankyou again !!!

Author

Commented:
grr i thought i got that damn PW...

I will get password changed ASAP , anyways its only 5 dollar pre-paid that i have to top up at the moment , not going on a plan till i iron out all the setup IVR etc....
 will try these , thankyou again !!!

Author

Commented:
You'll be happy to know my extension now rings , have a little problem with sound , cant seem to hear the other caller , but i think thats my end ..........

Once again , thankyou for you patience , I would have given up on me along time ago !!!

Im glad you didn't : )

Commented:
Great that the extension is now ringing. I never give up unless the questioner is being uncooperative and is ignoring the questions and suggestions I make... or I just simply run out of ideas.

By the way, do you know which change finally fixed the problem - so others can see the solution.

If you need help with the 1-way audio, it would be best to post as a new question now.

Author

Commented:
From trunk  line + type friend I guess hard to say after everthing we went through
I can post the complete working config. If you feel for others to use  

Commented:
No, don't bother - you might post a password again! Others can read the comments and try the different suggestions for themselves.

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