calls hangup immediately on pickup.. odd

[agidi1]

Dear Gurus

I have a first time problem.

Fresh install of Elastix 1.6.2-37 with OpenVox with 4 FXO modules (tested two cards)

Cards recognized, ports are green.

When we dial in and send the call to an ext. The call hang-up con cli immediately on pickup.

Then rings again, and hangs up if we pickup. (tried it on SIP and iax2 endpoints, same issue)

Intra extensions are working fine.
We can't dial out either. All lines are busy.

Is is a SIP issue or a dahdi issue?

I turned on sip debug for on ext and did a trial for you to see.

All ideas are welcomed, this is the first time we have this issue.
thanks

pbx*CLI> help sip debug
Usage: sip set debug
Enables dumping of SIP packets for debugging purposes

sip set debug ip <host[ORT]>
Enables dumping of SIP packets to and from host.

sip set debug peer <peername>
Enables dumping of SIP packets to and from host.
Require peer to be registered.
pbx*CLI> sip set debug peer 3026
SIP Debugging Enabled for IP: 192.168.0.165:5060
-- Starting simple switch on 'DAHDI/5-1'
-- Executing [s@from-pstn:1] Set("DAHDI/5-1", "__FROM_DID=s") in new stack
-- Executing [s@from-pstn:2] Gosub("DAHDI/5-1", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("DAHDI/5-1", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("DAHDI/5-1", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("DAHDI/5-1", "") in new stack
-- Executing [s@from-pstn:3] ExecIf("DAHDI/5-1", "1 |Set|CALLERID(name)=") in new stack
-- Executing [s@from-pstn:4] Set("DAHDI/5-1", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@from-pstn:5] SetCallerPres("DAHDI/5-1", "allowed_not_screened") in new stack
-- Executing [s@from-pstn:6] Goto("DAHDI/5-1", "from-did-direct|3026|1") in new stack
-- Goto (from-did-direct,3026,1)
-- Executing [3026@from-did-direct:1] Macro("DAHDI/5-1", "exten-vm|novm|3026") in new stack
-- Executing [s@macro-exten-vm:1] Macro("DAHDI/5-1", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("DAHDI/5-1", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/5-1", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/5-1", "1|Set|REALCALLERIDNUM=") in new stack
-- Executing [s@macro-user-callerid:4] Set("DAHDI/5-1", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("DAHDI/5-1", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/5-1", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("DAHDI/5-1", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("DAHDI/5-1", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("DAHDI/5-1", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("DAHDI/5-1", "Using CallerID "" <>") in new stack
-- Executing [s@macro-exten-vm:2] Set("DAHDI/5-1", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("DAHDI/5-1", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("DAHDI/5-1", "EXTTOCALL=3026") in new stack
-- Executing [s@macro-exten-vm:5] Set("DAHDI/5-1", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("DAHDI/5-1", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("DAHDI/5-1", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("DAHDI/5-1", "record-enable|3026|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("DAHDI/5-1", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("DAHDI/5-1", "recordingcheck|20120421-150257|1335038576.6") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20120421-150257|1335038576.6: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("DAHDI/5-1", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("DAHDI/5-1", "dial||tr|3026") in new stack
-- Executing [s@macro-dial:1] GotoIf("DAHDI/5-1", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("DAHDI/5-1", "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 'unknown' number is 'unknown'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 3026 to extension map
> dialparties.agi: Extension 3026 has call screening off
-- dialparties.agi: Extension 3026 cf is disabled
-- dialparties.agi: Extension 3026 do not disturb is disabled
> dialparties.agi: extnum 3026 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: ExtensionState: 0
-- dialparties.agi: DbDel CALLTRACE/3026 - Caller ID is not defined
-- dialparties.agi: Filtered ARG3: 3026
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("DAHDI/5-1", "SIP/3026||tr") in new stack
Audio is at 192.168.0.54 port 15630
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.165:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK774dd7de;rport
From: "Unknown" <sip:[email protected]>;tag=as57f4abd4
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 21 Apr 2012 20:02:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 524

v=0
o=root 2843 2843 IN IP4 192.168.0.54
s=session
c=IN IP4 192.168.0.54
t=0 0
m=audio 15630 RTP/AVP 0 8 3 112 5 10 7 18 110 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSuppff - - - -
a=ptime:20
a=sendrecv

---
-- Called 3026
pbx*CLI>
<--- SIP read from 192.168.0.165:5060 --->
SIP/2.0 100 Trying
To: <sip:[email protected]:5060>
From: "Unknown" <sip:[email protected]>;tag=as57f4abd4
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK774dd7de
Server: Cisco/SPA502G-7.4.8a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 192.168.0.165:5060 --->
SIP/2.0 180 Ringing
To: <sip:[email protected]:5060>;tag=9a4186c5fb451902i0
From: "Unknown" <sip:[email protected]>;tag=as57f4abd4
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK774dd7de
Contact: "3026" <sip:[email protected]:5060>
Server: Cisco/SPA502G-7.4.8a
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
-- SIP/3026-08e9f270 is ringing
pbx*CLI>
<--- SIP read from 192.168.0.165:5060 --->
SIP/2.0 200 OK
To: <sip:[email protected]:5060>;tag=9a4186c5fb451902i0
From: "Unknown" <sip:[email protected]>;tag=as57f4abd4
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK774dd7de
Contact: "3026" <sip:[email protected]:5060>
Server: Cisco/SPA502G-7.4.8a
Content-Length: 210
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 7849604 7849604 IN IP4 192.168.0.165
s=-
c=IN IP4 192.168.0.165
t=0 0
m=audio 16512 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.165:16512
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), 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 192.168.0.165:16512
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.165, port 5060
Transmitting (NAT) to 192.168.0.165:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK63e4c840;rport
From: "Unknown" <sip:[email protected]>;tag=as57f4abd4
To: <sip:[email protected]:5060>;tag=9a4186c5fb451902i0
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/3026-08e9f270 answered DAHDI/5-1
-- Executing [h@macro-dial:1] Macro("DAHDI/5-1", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("DAHDI/5-1", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("DAHDI/5-1", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("DAHDI/5-1", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] NoOp("DAHDI/5-1", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:13] GotoIf("DAHDI/5-1", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("DAHDI/5-1", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("DAHDI/5-1", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp("DAHDI/5-1", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("DAHDI/5-1", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,22)
-- Executing [s@macro-hangupcall:22] GotoIf("DAHDI/5-1", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] GotoIf("DAHDI/5-1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,27)
-- Executing [s@macro-hangupcall:27] Hangup("DAHDI/5-1", "") in new stack
== Spawn extension (macro-hangupcall, s, 27) exited non-zero on 'DAHDI/5-1' in macro 'hangupcall'
== Spawn h extension (macro-dial, h, 1) exited non-zero on 'DAHDI/5-1'
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.165, port 5060
Reliably Transmitting (NAT) to 192.168.0.165:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK14955437;rport
From: "Unknown" <sip:[email protected]>;tag=as57f4abd4
To: <sip:[email protected]:5060>;tag=9a4186c5fb451902i0
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== Spawn extension (macro-dial, s, 7) exited non-zero on 'DAHDI/5-1' in macro 'dial'
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'DAHDI/5-1' in macro 'exten-vm'
== Spawn extension (from-did-direct, 3026, 1) exited non-zero on 'DAHDI/5-1'
-- Hungup 'DAHDI/5-1'
pbx*CLI>
<--- SIP read from 192.168.0.165:5060 --->
SIP/2.0 200 OK
To: <sip:[email protected]:5060>;tag=9a4186c5fb451902i0
From: "Unknown" <sip:[email protected]>;tag=as57f4abd4
Call-ID: [email protected]
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK14955437
Server: Cisco/SPA502G-7.4.8a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
-- Starting simple switch on 'DAHDI/5-1'
-- Executing [s@from-pstn:1] Set("DAHDI/5-1", "__FROM_DID=s") in new stack
-- Executing [s@from-pstn:2] Gosub("DAHDI/5-1", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("DAHDI/5-1", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("DAHDI/5-1", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("DAHDI/5-1", "") in new stack
-- Executing [s@from-pstn:3] ExecIf("DAHDI/5-1", "1 |Set|CALLERID(name)=") in new stack
-- Executing [s@from-pstn:4] Set("DAHDI/5-1", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@from-pstn:5] SetCallerPres("DAHDI/5-1", "allowed_not_screened") in new stack
-- Executing [s@from-pstn:6] Goto("DAHDI/5-1", "from-did-direct|3026|1") in new stack
-- Goto (from-did-direct,3026,1)
-- Executing [3026@from-did-direct:1] Macro("DAHDI/5-1", "exten-vm|novm|3026") in new stack
-- Executing [s@macro-exten-vm:1] Macro("DAHDI/5-1", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("DAHDI/5-1", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/5-1", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/5-1", "1|Set|REALCALLERIDNUM=") in new stack
-- Executing [s@macro-user-callerid:4] Set("DAHDI/5-1", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("DAHDI/5-1", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/5-1", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("DAHDI/5-1", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("DAHDI/5-1", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("DAHDI/5-1", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("DAHDI/5-1", "Using CallerID "" <>") in new stack
-- Executing [s@macro-exten-vm:2] Set("DAHDI/5-1", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("DAHDI/5-1", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("DAHDI/5-1", "EXTTOCALL=3026") in new stack
-- Executing [s@macro-exten-vm:5] Set("DAHDI/5-1", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("DAHDI/5-1", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("DAHDI/5-1", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("DAHDI/5-1", "record-enable|3026|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("DAHDI/5-1", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("DAHDI/5-1", "recordingcheck|20120421-150307|1335038586.8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20120421-150307|1335038586.8: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("DAHDI/5-1", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("DAHDI/5-1", "dial||tr|3026") in new stack
-- Executing [s@macro-dial:1] GotoIf("DAHDI/5-1", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("DAHDI/5-1", "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 'unknown' number is 'unknown'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 3026 to extension map
> dialparties.agi: Extension 3026 has call screening off
-- dialparties.agi: Extension 3026 cf is disabled
-- dialparties.agi: Extension 3026 do not disturb is disabled
> dialparties.agi: extnum 3026 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: ExtensionState: 0
-- dialparties.agi: DbDel CALLTRACE/3026 - Caller ID is not defined
-- dialparties.agi: Filtered ARG3: 3026
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("DAHDI/5-1", "SIP/3026||tr") in new stack
Audio is at 192.168.0.54 port 10752
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.165:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK6ce8afea;rport
From: "Unknown" <sip:[email protected]>;tag=as5953644d
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 21 Apr 2012 20:03:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 524

v=0
o=root 2843 2843 IN IP4 192.168.0.54
s=session
c=IN IP4 192.168.0.54
t=0 0
m=audio 10752 RTP/AVP 0 8 3 112 5 10 7 18 110 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSuppff - - - -
a=ptime:20
a=sendrecv

---
-- Called 3026
pbx*CLI>
<--- SIP read from 192.168.0.165:5060 --->
SIP/2.0 100 Trying
To: <sip:[email protected]:5060>
From: "Unknown" <sip:[email protected]>;tag=as5953644d
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK6ce8afea
Server: Cisco/SPA502G-7.4.8a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 192.168.0.165:5060 --->
SIP/2.0 180 Ringing
To: <sip:[email protected]:5060>;tag=58ab8c1fe42e4755i0
From: "Unknown" <sip:[email protected]>;tag=as5953644d
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK6ce8afea
Contact: "3026" <sip:[email protected]:5060>
Server: Cisco/SPA502G-7.4.8a
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
-- SIP/3026-08e9f270 is ringing
Reliably Transmitting (NAT) to 192.168.0.165:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK78495dab;rport
From: "Unknown" <sip:[email protected]>;tag=as5a16b674
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 21 Apr 2012 20:03:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
pbx*CLI>
<--- SIP read from 192.168.0.165:5060 --->
SIP/2.0 200 OK
To: <sip:[email protected]:5060>;tag=180d1eb7ce54565bi0
From: "Unknown" <sip:[email protected]>;tag=as5a16b674
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.54:5060;branch=z9hG4bK78495dab
Server: Cisco/SPA502G-7.4.8a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
Scheduling

 

[agidi1]

We have had no luck on this one.
Any ideas.
We rebooted, re loaded, re ran dahdi setup. No luck.
All pointers and suggestions are welcomed.

 

[Bob1]

I had a brief look when you posted and immediately thought of codec....

However after reading your logs, it does not appear so....

Generally calls dropping like that are due to two common areas

1) Context
2) Codec (in particular G729 codec implemented incorrectly or not at all).

I assume that this is a stock standard Elastix setup with no addons, and no changes to context, so it wipes out the context issue

Codec, when you call extension to extension and both phones have G729, Asterisk performs a pass through function (meaning that it does not touch down on the Asterisk box, therefore codec on the asterisk box plays no part).

However when you come in through on Dahdi, it is touching down on the Asterisk box.

So just confirm that you have nothing set on the asterisk box or the phones to use G729 exclusively, just in case i have read the logs wrong...

Anyhow, just some feedback...as it seems very strange....

Regards

Bob

 

[agidi1]

Bob i apareciste your reply

Good morning. I yum updated dahdi with no luck.

But the only thing I did add to this install was a g729 codec, I might have gotten the wrong gone, aldo it does works with remote extensions.
I will remove it, reboot, and test.
Will get back to you.
Thanks again.

 

[agidi1]

1) i apologize for the typos on the first reply, damn autocorrect.. replied from phone.

2) I eliminated the g729 codec, rebooted, no luck.
Im testing with an IAX2 ext all codecs allowed, and a SIP ext same scenario.

I tried dahdi_monitor and I can see it ring, but still cannot pick up the call or make a call.

This is a good intel motherboard, I've never had issues with this brand. Install went smooth after an OK linux mediacheck

What other debug can i try? dahdi? Try do determine why/where the call can't be passed on?

all ideas appreciated. thanks

 

[Bob1]

agidi,

I know it is painful, but if you could do the following, just to eliminate the codec altogether

1) Set the Inbound route to go to one extension only (the one we are going to hard set)
2) In that extension configuration enter disallow ALL, allow ULAW
3) On your Extension Hardware (check that you do not have it to exclusively use a codec e.g. G729

Under the Asterisk CLI, perform the following command (replace 201 with whatever your extension number is)
sip show peer 201
and post the full result

Next make a call in on your system to that extension (let me know what the extension number is), and post the log from /var/log/asterisk/full (basically from when the call comes in to where it hangs up).

Don't post anything else, so that we can focus on those two items...

Regards

Bob