Kramer8u
asked on
Tirxbox (asterix) inbound (recieve calls) trunk
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.gota
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
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.gota
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
ASKER CERTIFIED SOLUTION
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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".
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".
ASKER
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
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
ASKER
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
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
Max-Forwards: 70
Contact: <sip:0405194914@202.169.17
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
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-
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
From: "0405194914"<sip:040519491
To: <sip:0280141063@192.168.1.
Call-ID: 4c80-4cc-022197061034-TCP-
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
trixbox1*CLI>
<--- SIP read from 202.169.178.10:5060 --->
ACK sip:0280141063@123.243.199
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
CSeq: 33601 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from 202.169.178.10:5060 --->
INVITE sip:0280141063@123.243.199
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
Max-Forwards: 70
Contact: <sip:0405194914@202.169.17
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
CSeq: 33602 INVITE
Content-Type: application/sdp
Proxy-Authorization: Digest username="09474159",realm=
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-
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
From: "0405194914"<sip:040519491
To: <sip:0280141063@192.168.1.
Call-ID: 4c80-4cc-022197061034-TCP-
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
trixbox1*CLI>
<--- SIP read from 202.169.178.10:5060 --->
ACK sip:0280141063@123.243.199
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
CSeq: 33602 ACK
Content-Length: 0
<------------->
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
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
Max-Forwards: 70
Contact: <sip:0405194914@202.169.17
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
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-
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
From: "0405194914"<sip:040519491
To: <sip:0280141063@192.168.1.
Call-ID: 4c80-4cc-022197061034-TCP-
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
trixbox1*CLI>
<--- SIP read from 202.169.178.10:5060 --->
ACK sip:0280141063@123.243.199
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
CSeq: 33601 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from 202.169.178.10:5060 --->
INVITE sip:0280141063@123.243.199
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
Max-Forwards: 70
Contact: <sip:0405194914@202.169.17
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
CSeq: 33602 INVITE
Content-Type: application/sdp
Proxy-Authorization: Digest username="09474159",realm=
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-
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
From: "0405194914"<sip:040519491
To: <sip:0280141063@192.168.1.
Call-ID: 4c80-4cc-022197061034-TCP-
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
trixbox1*CLI>
<--- SIP read from 202.169.178.10:5060 --->
ACK sip:0280141063@123.243.199
Via: SIP/2.0/UDP 202.169.178.10:5060;branch
To: <sip:0280141063@192.168.1.
From: "0405194914"<sip:040519491
Call-ID: 4c80-4cc-022197061034-TCP-
CSeq: 33602 ACK
Content-Length: 0
<------------->
SOLUTION
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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.
ASKER
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
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
ASKER
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?
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?
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.
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.
ASKER
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 ..
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'
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".
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".
ASKER
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
:
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
ASKER
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.
--------------------------
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>;ex
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
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.
--------------------------
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>;ex
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
SOLUTION
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ASKER
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 !!!
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 !!!
ASKER
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 !!!
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 !!!
ASKER
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 : )
Once again , thankyou for you patience , I would have given up on me along time ago !!!
Im glad you didn't : )
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.
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.
ASKER
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
I can post the complete working config. If you feel for others to use
No, don't bother - you might post a password again! Others can read the comments and try the different suggestions for themselves.
ASKER
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