[WXC/VFX] Problems with incoming calls

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foufee

35 posts

Geek

Topic # 28560 4-Dec-2008 12:57

Hi,
Hopefully someone will be able to help, for some reason incoming calls do not work on my setup (Trixbox 2.5), I have a incoming route (with no DID), and allow anonymous calls enabled. If I ring 7777 it all works as expected, but if I call from the landline (POTS), it doesn't, so far I have found that if I point the inbound route at an extension (XLite client, connected, registered and working (as I can dial out with it)), it goes straight to voice mail, if it's a PAP2t device as the extension, I get "Channel unavailable" displayed by the asterix client on the command line.
What is the best way forward and how do I work out what's wrong with the setup?

Cheers

Matthew

rphenix

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Reply # 182053 4-Dec-2008 16:25

whats the output of sip show peers?

foufee

35 posts

Geek

Reply # 182058 4-Dec-2008 16:45

Name/username              Host            Dyn Nat ACL Port     Status
VFX SIP Trunk/XXXXXX 58.28.20.150         N      5060     Unmonitored
501                        (Unspecified)    D   N      0        UNKNOWN
500/500                    192.168.0.221    D   N      5060     OK (17 ms)
103                        (Unspecified)    D   N      0        UNKNOWN
102/102                    192.168.0.220    D   N      5060     OK (18 ms)
101/101                    192.168.0.207    D   N      28772    OK (105 ms)
100/100                    (Unspecified)    D   N      0        UNKNOWN
7 sip peers [Monitored: 3 online, 3 offline Unmonitored: 1 online, 0 offline]

Where 100 and 101 are Xlite clients, 102 is a Pap2T as is 500, and 501is line 2 on one of the Pap2t

sbiddle

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Reply # 182059 4-Dec-2008 16:45

What voicemail? VFX or your Asterisk voicemail?

If it's the VFX voicemail it's pretty much guaranteed that your Asterisk box is not properly registered with VFX.

What do you get if you see if you do a sip debug and make an incoming call?

sbiddle

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Reply # 182062 4-Dec-2008 16:49

sip show peers isn't that helpful

try " sip show registry " and see if VFX is succesfully registered

foufee

35 posts

Geek

Reply # 182065 4-Dec-2008 16:53

The output of sip show registry is
trixbox1*CLI> sip show registry
Host Username Refresh State Reg.Time
pan.wxnz.net:5060 48311266 165 Registered Thu, 04 Dec 2008 16:51:37

As for the debug stuff, I'll have to play with that later, can't do it from here as I don't have an external line to call on.

sbiddle

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Reply # 182068 4-Dec-2008 17:03

What is your only incoming route? Any DID / Any CID?

foufee

35 posts

Geek

Reply # 182076 4-Dec-2008 17:18

Here is the output of an incoming call,
I note that there is
"- Executing [s@macro-dial:7] Dial("SIP/XXXXXXXXXXXXXXX",
"SIP/100|15|Ttr") in new
stack

-- Couldn't call
100

Scheduling destruction of SIP dialog
'34dddaf3558781ce6835090c4a023222@192.168.2.1' in 6400 ms (Method:
INVITE)

== Everyone is busy/congested at this time
(0:0/0/0)
"
which looks like the problem, but I don't know why...

My incoming route has no DID set or CID and is set to go straight to extension 100

Cheers

Matthew

trixbox1*CLI> sip debug
SIP Debugging re-enabled
trixbox1*CLI> sip show registry
Host Username Refresh State Reg.Time
pan.wxnz.net:5060 48311266 165 Registered Thu, 04 Dec 2008 17:10:58
trixbox1*CLI>
trixbox1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
VFX SIP Trunk/cWjyXAiXjsJ 58.28.20.150 N 5060 Unmonitored
500/500 192.168.0.221 D N 5060 OK (17 ms)
102/102 192.168.0.220 D N 5060 OK (13 ms)
101/101 192.168.0.207 D N 28772 OK (105 ms)
100/100 192.168.0.166 D N 5060 OK (3 ms)
5 sip peers [Monitored: 4 online, 0 offline Unmonitored: 1 online, 0 offline]

Incoming Call

INVITE sip:48311266@192.168.2.1;transport=udp SIP/2.0
From: "Anonymous";tag=96141c3a-13c4-49375888-2f84e342-19a126dc
To: "Matthew Huck"
Call-ID: BW1711255080412081159991393@10.251.1.11
CSeq: 1 INVITE
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-18a48-49375888-2f84e342-6d9cc3e5
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 9
Contact:
Content-Type: application/sdp
Content-Length: 336

v=0
o=BroadWorks 14403104 1 IN IP4 58.28.20.150
s=-
c=IN IP4 58.28.20.150
t=0 0
m=audio 37130 RTP/AVP 18 8 0 100 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:100 X-NSE/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:101 0-16
a=fmtp:100 192-194
a=fmtp:18 annexb=yes
a=bsoft:1 image udptl t38

--- (12 headers 15 lines) ---
Sending to 58.28.20.150 : 5060 (no NAT)
Using INVITE request as basis request - BW1711255080412081159991393@10.251.1.11
Found peer 'VFX SIP Trunk'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 58.28.20.150:37130
Found audio description format telephone-event for ID 101
Found unknown media description format X-NSE for ID 100
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 58.28.20.150:37130
Looking for 48311266 in from-trunk (domain 192.168.2.1)
list_route: hop:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-18a48-49375888-2f84e342-6d9cc3e5;received=58.28.20.150
From: "Anonymous";tag=96141c3a-13c4-49375888-2f84e342-19a126dc
To: "Matthew Huck"
Call-ID: BW1711255080412081159991393@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

-- Executing [48311266@from-trunk:1] NoOp("SIP/XXXXXXXXXXXXXXX", "Catch-All DID Match - Found 48311266 - You probably want a DID for this.") in new stack
-- Executing [48311266@from-trunk:2] Goto("SIP/XXXXXXXXXXXXXXX", "ext-did|s|1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/XXXXXXXXXXXXXXX", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/XXXXXXXXXXXXXXX", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/XXXXXXXXXXXXXXX", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/XXXXXXXXXXXXXXX", "") in new stack
-- Executing [s@ext-did:3] ExecIf("SIP/XXXXXXXXXXXXXXX", "0 |Set|CALLERID(name)=anonymous") in new stack
-- Executing [s@ext-did:4] Set("SIP/XXXXXXXXXXXXXXX", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:5] SetCallerPres("SIP/XXXXXXXXXXXXXXX", "allowed_not_screened") in new stack
-- Executing [s@ext-did:6] Goto("SIP/XXXXXXXXXXXXXXX", "from-did-direct|100|1") in new stack
-- Goto (from-did-direct,100,1)
-- Executing [100@from-did-direct:1] Macro("SIP/XXXXXXXXXXXXXXX", "exten-vm|100|100") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/XXXXXXXXXXXXXXX", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSER=anonymous") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/XXXXXXXXXXXXXXX", "1|Set|REALCALLERIDNUM=anonymous") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/XXXXXXXXXXXXXXX", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/XXXXXXXXXXXXXXX", "Using CallerID "Anonymous" ") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/XXXXXXXXXXXXXXX", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/XXXXXXXXXXXXXXX", "VMBOX=100") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/XXXXXXXXXXXXXXX", "EXTTOCALL=100") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/XXXXXXXXXXXXXXX", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/XXXXXXXXXXXXXXX", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/XXXXXXXXXXXXXXX", "RT=15") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/XXXXXXXXXXXXXXX", "record-enable|100|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/XXXXXXXXXXXXXXX", "recordingcheck|20081204-171152|1228363912.51") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20081204-171152|1228363912.51: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/XXXXXXXXXXXXXXX", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/XXXXXXXXXXXXXXX", "dial|15|Ttr|100") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/XXXXXXXXXXXXXXX", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Anonymous' number is 'anonymous'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 100 to extension map
-- dialparties.agi: Extension 100 cf is disabled
-- dialparties.agi: Extension 100 do not disturb is disabled
-- dialparties.agi: DbDel CALLTRACE/100 - Caller ID is not defined
-- dialparties.agi: Filtered ARG3: 100
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/XXXXXXXXXXXXXXX", "SIP/100|15|Ttr") in new stack
-- Couldn't call 100
Scheduling destruction of SIP dialog '34dddaf3558781ce6835090c4a023222@192.168.2.1' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dial:8] Set("SIP/XXXXXXXXXXXXXXX", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/XXXXXXXXXXXXXXX", "0?CHANUNAVAIL|1") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?exit|return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/XXXXXXXXXXXXXXX", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/XXXXXXXXXXXXXXX", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/XXXXXXXXXXXXXXX", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/XXXXXXXXXXXXXXX", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/XXXXXXXXXXXXXXX", "Voicemail is 100") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?s-CHANUNAVAIL|1") in new stack
-- Executing [s@macro-exten-vm:17] NoOp("SIP/XXXXXXXXXXXXXXX", "Sending to Voicemail box 100") in new stack
-- Executing [s@macro-exten-vm:18] Macro("SIP/XXXXXXXXXXXXXXX", "vm|100|CHANUNAVAIL|") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/XXXXXXXXXXXXXXX", "user-callerid|SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSER=anonymous") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/XXXXXXXXXXXXXXX", "0|Set|REALCALLERIDNUM=anonymous") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/XXXXXXXXXXXXXXX", "Using CallerID "Anonymous" ") in new stack
-- Executing [s@macro-vm:2] Set("SIP/XXXXXXXXXXXXXXX", "VMGAIN=""") in new stack
-- Executing [s@macro-vm:3] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?vmx|1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?s-CHANUNAVAIL|1") in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/XXXXXXXXXXXXXXX", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:3] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,5)
-- Executing [vmx@macro-vm:5] NoOp("SIP/XXXXXXXXXXXXXXX", "Checking if ext 100 is enabled: ") in new stack
-- Executing [vmx@macro-vm:6] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-vm:1] Macro("SIP/XXXXXXXXXXXXXXX", "get-vmcontext|100") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/XXXXXXXXXXXXXXX", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/XXXXXXXXXXXXXXX", "") in new stack
-- Executing [s-CHANUNAVAIL@macro-vm:2] VoiceMail("SIP/XXXXXXXXXXXXXXX", "100@default|u") in new stack
Audio is at 192.168.2.1 port 15438
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
trixbox1*CLI>

SIP/2.0 200 OK
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-18a48-49375888-2f84e342-6d9cc3e5;received=58.28.20.150
From: "Anonymous";tag=96141c3a-13c4-49375888-2f84e342-19a126dc
To: "Matthew Huck";tag=as05f44083
Call-ID: BW1711255080412081159991393@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 2357 2357 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 15438 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/XXXXXXXXXXXXXXX' in macro 'vm'
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/XXXXXXXXXXXXXXX' in macro 'exten-vm'
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/XXXXXXXXXXXXXXX'
Scheduling destruction of SIP dialog 'BW1711255080412081159991393@10.251.1.11' in 32000 ms (Method: INVITE)
trixbox1*CLI>

ACK sip:48311266@192.168.2.1;transport=udp SIP/2.0
From: "Anonymous";tag=96141c3a-13c4-49375888-2f84e342-19a126dc
To: "Matthew Huck";tag=as05f44083
Call-ID: BW1711255080412081159991393@10.251.1.11
CSeq: 1 ACK
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-18a4a-49375889-2f84e7be-590fa23
Max-Forwards: 9
Contact:
Content-Length: 0

--- (9 headers 0 lines) ---
set_destination: Parsing for address/port to send to
set_destination: set destination to 58.28.20.150, port 5060
Reliably Transmitting (NAT) to 58.28.20.150:5060:
BYE sip:anonymous@as.wxcnz.net:5060;maddr=58.28.20.150;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK1a0bcf65;rport
From: "Matthew Huck";tag=as05f44083
To: "Anonymous";tag=96141c3a-13c4-49375888-2f84e342-19a126dc
Call-ID: BW1711255080412081159991393@10.251.1.11
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
Scheduling destruction of SIP dialog 'BW1711255080412081159991393@10.251.1.11' in 32000 ms (Method: ACK)
trixbox1*CLI>

SIP/2.0 200 OK
From: "Matthew Huck";tag=as05f44083
To: "Anonymous";tag=96141c3a-13c4-49375888-2f84e342-19a126dc
Call-ID: BW1711255080412081159991393@10.251.1.11
CSeq: 102 BYE
Via: SIP/2.0/UDP 192.168.2.1:5060;received=202.174.163.26;rport=5060;branch=z9hG4bK1a0bcf65
Content-Length: 0

--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'BW1711255080412081159991393@10.251.1.11' Method: ACK
trixbox1*CLI>

--- (0 headers 0 lines) Nat keepalive ---
Really destroying SIP dialog '34dddaf3558781ce6835090c4a023222@192.168.2.1' Method: INVITE

foufee

35 posts

Geek

Reply # 182078 4-Dec-2008 17:26

This is when the inbound route is forwarded to a Pap2T, it goes to the VFX VoiceMail system instead.

<--- SIP read from 192.168.0.166:5060 --->

<------------->
--- (0 headers 0 lines) Nat keepalive ---
trixbox1*CLI>
<--- SIP read from 58.28.20.150:5060 --->
INVITE sip:48311266@192.168.2.1;transport=udp SIP/2.0
From: ;tag=96141c3a-13c4-49375af1-2f8e4f32-51404f05
To: "Matthew Huck"
Call-ID: BW172143077041208-260954667@10.251.1.11
CSeq: 1 INVITE
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-190ca-49375af1-2f8e4f32-13535b5a
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 9
Contact:
Content-Type: application/sdp
Content-Length: 336

v=0
o=BroadWorks 14405150 1 IN IP4 58.28.20.150
s=-
c=IN IP4 58.28.20.150
t=0 0
m=audio 35514 RTP/AVP 18 8 0 100 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:100 X-NSE/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:101 0-16
a=fmtp:100 192-194
a=fmtp:18 annexb=yes
a=bsoft:1 image udptl t38

<------------->
--- (12 headers 15 lines) ---
Sending to 58.28.20.150 : 5060 (no NAT)
Using INVITE request as basis request - BW172143077041208-260954667@10.251.1.11
Found peer 'VFX SIP Trunk'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 58.28.20.150:35514
Found audio description format telephone-event for ID 101
Found unknown media description format X-NSE for ID 100
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 58.28.20.150:35514
Looking for 48311266 in from-trunk (domain 192.168.2.1)
list_route: hop:

<--- Transmitting (NAT) to 58.28.20.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-190ca-49375af1-2f8e4f32-13535b5a;received=58.28.20.150
From: ;tag=96141c3a-13c4-49375af1-2f8e4f32-51404f05
To: "Matthew Huck"
Call-ID: BW172143077041208-260954667@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

<------------>
-- Executing [48311266@from-trunk:1] NoOp("SIP/XXXXXXXXXXXXXXX", "Catch-All DID Match - Found 48311266 - You probably want a DID for this.") in new stack
-- Executing [48311266@from-trunk:2] Goto("SIP/XXXXXXXXXXXXXXX", "ext-did|s|1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/XXXXXXXXXXXXXXX", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/XXXXXXXXXXXXXXX", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/XXXXXXXXXXXXXXX", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/XXXXXXXXXXXXXXX", "") in new stack
-- Executing [s@ext-did:3] ExecIf("SIP/XXXXXXXXXXXXXXX", "1 |Set|CALLERID(name)=0000") in new stack
-- Executing [s@ext-did:4] Set("SIP/XXXXXXXXXXXXXXX", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:5] SetCallerPres("SIP/XXXXXXXXXXXXXXX", "allowed_not_screened") in new stack
-- Executing [s@ext-did:6] Goto("SIP/XXXXXXXXXXXXXXX", "from-did-direct|102|1") in new stack
-- Goto (from-did-direct,102,1)
-- Executing [102@from-did-direct:1] Macro("SIP/XXXXXXXXXXXXXXX", "exten-vm|novm|102") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/XXXXXXXXXXXXXXX", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSER=0000") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/XXXXXXXXXXXXXXX", "1|Set|REALCALLERIDNUM=0000") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/XXXXXXXXXXXXXXX", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/XXXXXXXXXXXXXXX", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/XXXXXXXXXXXXXXX", "Using CallerID "0000" <0000>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/XXXXXXXXXXXXXXX", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/XXXXXXXXXXXXXXX", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/XXXXXXXXXXXXXXX", "EXTTOCALL=102") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/XXXXXXXXXXXXXXX", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/XXXXXXXXXXXXXXX", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/XXXXXXXXXXXXXXX", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/XXXXXXXXXXXXXXX", "record-enable|102|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/XXXXXXXXXXXXXXX", "recordingcheck|20081204-172209|1228364529.55") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20081204-172209|1228364529.55: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/XXXXXXXXXXXXXXX", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/XXXXXXXXXXXXXXX", "dial||Ttr|102") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/XXXXXXXXXXXXXXX", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '0000' number is '0000'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 102 to extension map
-- dialparties.agi: Extension 102 cf is disabled
-- dialparties.agi: Extension 102 do not disturb is disabled
-- dialparties.agi: dbset CALLTRACE/102 to 0000
-- dialparties.agi: Filtered ARG3: 102
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/XXXXXXXXXXXXXXX", "SIP/102||Ttr") in new stack
-- Couldn't call 102
Scheduling destruction of SIP dialog '2eeb2d1a083e6c063b5a42f917fe0f0c@192.168.2.1' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dial:8] Set("SIP/XXXXXXXXXXXXXXX", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/XXXXXXXXXXXXXXX", "0?CHANUNAVAIL|1") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?exit|return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/XXXXXXXXXXXXXXX", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/XXXXXXXXXXXXXXX", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/XXXXXXXXXXXXXXX", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/XXXXXXXXXXXXXXX", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/XXXXXXXXXXXXXXX", "Voicemail is novm") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/XXXXXXXXXXXXXXX", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/XXXXXXXXXXXXXXX", "IVR_RETVM: IVR_CONTEXT: ") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/XXXXXXXXXXXXXXX", "0?exit|1") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/XXXXXXXXXXXXXXX", "congestion") in new stack
Audio is at 192.168.2.1 port 18232
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
trixbox1*CLI>
<--- Transmitting (NAT) to 58.28.20.150:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-190ca-49375af1-2f8e4f32-13535b5a;received=58.28.20.150
From: ;tag=96141c3a-13c4-49375af1-2f8e4f32-51404f05
To: "Matthew Huck";tag=as7fe1d4ec
Call-ID: BW172143077041208-260954667@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 2357 2357 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 18232 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/XXXXXXXXXXXXXXX' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/XXXXXXXXXXXXXXX'
Scheduling destruction of SIP dialog 'BW172143077041208-260954667@10.251.1.11' in 32000 ms (Method: INVITE)
trixbox1*CLI>
<--- Reliably Transmitting (NAT) to 58.28.20.150:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-190ca-49375af1-2f8e4f32-13535b5a;received=58.28.20.150
From: ;tag=96141c3a-13c4-49375af1-2f8e4f32-51404f05
To: "Matthew Huck";tag=as7fe1d4ec
Call-ID: BW172143077041208-260954667@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

<------------>
Retransmitting #1 (NAT) to 58.28.20.150:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-190ca-49375af1-2f8e4f32-13535b5a;received=58.28.20.150
From: ;tag=96141c3a-13c4-49375af1-2f8e4f32-51404f05
To: "Matthew Huck";tag=as7fe1d4ec
Call-ID: BW172143077041208-260954667@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

---
Retransmitting #2 (NAT) to 58.28.20.150:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-190ca-49375af1-2f8e4f32-13535b5a;received=58.28.20.150
From: ;tag=96141c3a-13c4-49375af1-2f8e4f32-51404f05
To: "Matthew Huck";tag=as7fe1d4ec
Call-ID: BW172143077041208-260954667@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

---
Really destroying SIP dialog 'BW172113379041208-1339800337@10.251.1.11' Method: ACK
Retransmitting #3 (NAT) to 58.28.20.150:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-190ca-49375af1-2f8e4f32-13535b5a;received=58.28.20.150
From: ;tag=96141c3a-13c4-49375af1-2f8e4f32-51404f05
To: "Matthew Huck";tag=as7fe1d4ec
Call-ID: BW172143077041208-260954667@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

---
Really destroying SIP dialog '2eeb2d1a083e6c063b5a42f917fe0f0c@192.168.2.1' Method: INVITE
trixbox1*CLI>
<--- SIP read from 192.168.0.166:5060 --->

<------------->
--- (0 headers 0 lines) Nat keepalive ---
Reliably Transmitting (NAT) to 192.168.0.221:5060:
OPTIONS sip:500@192.168.0.221:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK73b507d0;rport
From: "Unknown" ;tag=as0948e05f
To:
Contact:
Call-ID: 068009e9787deb8b6b2ef81e641dafd3@192.168.2.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 04 Dec 2008 04:22:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

---

foufee

35 posts

Geek

Reply # 183232 10-Dec-2008 06:31

Is there anything else that would help in giving someone a hint as to where to give me a hint to look?

Cheers

Matthew

sbiddle

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Reply # 183236 10-Dec-2008 07:58

It's definately something with your extension settings and not a VFX issue. Calls are being passed through OK and being routed by the ALL/ALL inbound route.

What exactly do you have in your extension settings and what have you changed in the ATA configuration? Have you forced any codec selections anywhere?

foufee

35 posts

Geek

Reply # 183247 10-Dec-2008 08:42

I just deleted all the extensions, and started from scratch.
I've got one extension now, and the only things with any values are

displayname=OfficePAP2t
secret=blah
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5600
qualify=yes
dial=SIP/100
mailbox=100@device

And as for the PAP2t, is has a preferred codec of G711U and "Use Preferred Codec only" is set to "No"

I can still make out going calls with the Pap2t.

Matthew

sbiddle

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Reply # 183254 10-Dec-2008 09:01

Is this PAP2T onsite and using the same IP range as the trixbox machine?

Try disabling g729 in the trunk settings (allow=ulaw&alaw) and see if this does anything. Asterisk doesn't natively support g.729 without installing a g.729 codec, it does support passthru fine but can't transcode.

foufee

35 posts

Geek

Reply # 183257 10-Dec-2008 09:11

Awesome, disabling g729 and now the extension rings.....

Thanks for all your help.

Cheers

Matthew

sbiddle

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+1 received by user: 6250

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Reply # 183261 10-Dec-2008 09:21

If you do want to use G.729 you need to install the codec. It is not a open source or freely available codec.

Digium sell G.729 software & licences at US$10 each you can get more info here $10 ain't bad as a donation to Digium for creating a great product (Asterisk) but this is a per channel licence.

Or a another alternative here which is of the same quality but allows concurrent calls. Obviously just read the terms & conditions of using the product.