INCOMING CALL FROM SIP PROBLEM :(

RafaelMorenoBR

Joined
Dec 16, 2009
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#1
Hi Guys,

I work in a big ISP and Telecom provider in Brazil, and I'm trying to make Elastix work on my SIP server, but I have some issues getting the calls from my SIP EXPRESS ROUTER to my Elastix PBX I don't know if anyone has seen this problem but I really want some help to solve this. And then I can say to my costumers that Elastix is the best PBX to work on our SIP.

Anyways let's get to the problem

I did this tutorial from freepbx:
http://www.freepbx.org/support/document ... nd-it-and-

But is still not working,

I got some logs from my sip server and my Elastix Server

SIP:


Code:
U 200.196.28.101:51961 -> 200.202.12.42:5060
  INVITE sip:551133236214@sps2.matrix.net.br:5060 SIP/2.0..Via: SIP/2.0/UDP  200.196.28.101:5060..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:5511332362
  14@sps2.matrix.net.br>..Date: Wed, 16 Dec 2009 15:35:57 GMT..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..Supported: timer,100rel..Min-SE:  1800..Cisco-Gui
  d: 2313931478-3918467550-2935210755-1931747413..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, IN
  FO..CSeq: 101 INVITE..Max-Forwards: 6..Remote-Party-ID: <sip:1156410879@200.196.28.101>;party=calling;screen=no;privacy=off..Timestamp: 1260977757..Contact: <sip:115641087
  9@200.196.28.101:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 386....v=0..o=CiscoSystemsSIP-GW-UserAgent 4299 6350 IN
   IP4 200.196.28.101..s=SIP Call..c=IN IP4 200.196.28.101..t=0 0..m=audio 16668 RTP/AVP 18 0 8 100 101 19..c=IN IP4 200.196.28.101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=
  yes..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:19 CN/8000..    

U 200.202.12.42:5060 -> 200.196.28.101:51961
  SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP  200.196.28.101:5060;rport=51961..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:551
  133236214@sps2.matrix.net.br>..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..CSeq: 101 INVITE..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Len
  gth: 0..Warning: 392 200.202.12.42:5060 "Noisy feedback tells:  pid=18526 req_src_ip=200.196.28.101 req_src_port=51961 in_uri=sip:551133236214@sps2.matrix.net.br:5060 out_
  uri=sip:MS00142489@200.202.12.52:5060 via_cnt==1"....                                                                                                                      

U 200.202.12.42:5060 -> 200.202.12.52:5060
  INVITE sip:MS00142489@200.202.12.52:5060 SIP/2.0..Record-Route: <sip:551133236214@200.202.12.42;ftag=1FA85A94-945;lr=on>..Via: SIP/2.0/UDP 200.202.12.42;branch=z9hG4bK24fa
  .5db8466.0..Via: SIP/2.0/UDP  200.196.28.101:5060;rport=51961..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:551133236214@sps2.matrix.net.br>..Date: Wed
  , 16 Dec 2009 15:35:57 GMT..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..Supported: timer,100rel..Min-SE:  1800..Cisco-Guid: 2313931478-3918467550-29352107
  55-1931747413..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forward
  s: 5..Remote-Party-ID: <sip:1156410879@200.196.28.101>;party=calling;screen=no;privacy=off..Timestamp: 1260977757..Contact: <sip:1156410879@200.196.28.101:51961>..Expires:
   180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 402..P-hint: fixed NAT contact for request..P-hint: request forced to rtp proxy....v=0.
  .o=CiscoSystemsSIP-GW-UserAgent 4299 6350 IN IP4 200.196.28.101..s=SIP Call..c=IN IP4 200.202.12.42..t=0 0..m=audio 52058 RTP/AVP 18 0 8 100 101 19..c=IN IP4 200.202.12.42
  ..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=yes..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000..
  a=fmtp:101 0-16..a=rtpmap:19 CN/8000..a=nortpproxy:yes..                                                                                                                   

U 200.202.12.52:5060 -> 200.202.12.42:5060
  SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 200.202.12.42;branch=z9hG4bK24fa.5db8466.0;received=200.202.12.42..Via: SIP/2.0/UDP  200.196.28.101:5060;rport=51961..From: <sip:11
  56410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:551133236214@sps2.matrix.net.br>;tag=as6702ba88..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..CSeq: 101
   INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0....                    

U 200.202.12.42:5060 -> 200.202.12.52:5060
  ACK sip:MS00142489@200.202.12.52:5060 SIP/2.0..Via: SIP/2.0/UDP 200.202.12.42;branch=z9hG4bK24fa.5db8466.0..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..Call-ID
  : 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..To: <sip:551133236214@sps2.matrix.net.br>;tag=as6702ba88..CSeq: 101 ACK..User-Agent: Sip EXpress router(0.8.14 (i386/
  linux))..Content-Length: 0....                                                                                                                                             

U 200.202.12.42:5060 -> 200.196.28.101:51961
  SIP/2.0 404 Not Found..Via: SIP/2.0/UDP  200.196.28.101:5060;rport=51961..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:551133236214@sps2.matrix.net.br>
  ;tag=as6702ba88..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, 
  SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0..P-hint: fixed NAT contact for response....                                                                

U 200.196.28.101:51961 -> 200.202.12.42:5060
  ACK sip:551133236214@sps2.matrix.net.br:5060 SIP/2.0..Via: SIP/2.0/UDP  200.196.28.101:5060..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:551133236214@
  sps2.matrix.net.br>;tag=as6702ba88..Date: Wed, 16 Dec 2009 15:35:57 GMT..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..Max-Forwards: 6..Content-Length: 0..C
  Seq: 101 ACK....
PBX:

Code:
U 200.202.12.42:5060 -> 200.202.12.52:5060
  INVITE sip:MS00142489@200.202.12.52:5060 SIP/2.0..Record-Route: <sip:551133236214@200.202.12.42;ftag=1FA85A94-945;lr=on>..Via: SIP/2.0/UDP 200.202.12.42;branch=z9hG4bK24fa
  .5db8466.0..Via: SIP/2.0/UDP  200.196.28.101:5060;rport=51961..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:551133236214@sps2.matrix.net.br>..Date: Wed
  , 16 Dec 2009 15:35:57 GMT..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..Supported: timer,100rel..Min-SE:  1800..Cisco-Guid: 2313931478-3918467550-29352107
  55-1931747413..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forward
  s: 5..Remote-Party-ID: <sip:1156410879@200.196.28.101>;party=calling;screen=no;privacy=off..Timestamp: 1260977757..Contact: <sip:1156410879@200.196.28.101:51961>..Expires:
   180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 402..P-hint: fixed NAT contact for request..P-hint: request forced to rtp proxy....v=0.
  .o=CiscoSystemsSIP-GW-UserAgent 4299 6350 IN IP4 200.196.28.101..s=SIP Call..c=IN IP4 200.202.12.42..t=0 0..m=audio 52058 RTP/AVP 18 0 8 100 101 19..c=IN IP4 200.202.12.42
  ..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=yes..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000..
  a=fmtp:101 0-16..a=rtpmap:19 CN/8000..a=nortpproxy:yes..                                                                                                                   

U 200.202.12.52:5060 -> 200.202.12.42:5060
  SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 200.202.12.42;branch=z9hG4bK24fa.5db8466.0;received=200.202.12.42..Via: SIP/2.0/UDP  200.196.28.101:5060;rport=51961..From: <sip:11
  56410879@200.196.28.101>;tag=1FA85A94-945..To: <sip:551133236214@sps2.matrix.net.br>;tag=as6702ba88..Call-ID: 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..CSeq: 101
   INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0....                    

U 200.202.12.42:5060 -> 200.202.12.52:5060
  ACK sip:MS00142489@200.202.12.52:5060 SIP/2.0..Via: SIP/2.0/UDP 200.202.12.42;branch=z9hG4bK24fa.5db8466.0..From: <sip:1156410879@200.196.28.101>;tag=1FA85A94-945..Call-ID
  : 89EE3C0E-E98F11DE-AEF6C303-73242055@200.196.28.101..To: <sip:551133236214@sps2.matrix.net.br>;tag=as6702ba88..CSeq: 101 ACK..User-Agent: Sip EXpress router(0.8.14 (i386/
  linux))..Content-Length: 0....
This "NOT FOUND" is my problem, the SIP Trunk is set correctly and one of my extensions are working perfectly and does have the DID set on it.

If anyone has ever seen this problem I really appreciate a little help :)

Anyways,

Thanks,
 

jgutierrez

Joined
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#2
Some quick questions:

1. Do you need a register string? If so, what is the result of
asterisk -rx "sip show registry"

2. Have you tried using on the trunk definition the flag
insecure=very

3. I will recommend you to set an inbound route for ANY DID / ANY CID (this can be done if you leave blank the DID and CID fields)

4. Are you sure that you don't have any issues with your codecs? What codec should use your SIP trunk?

NOTES:
* By the way, when someone calls you, what do they get? A busy signal, a message from your provider, a message from your elastix server telling the person that the line is not in service?

* Paste the configuration of your sip trunk

* Enable extra log information on the CLI
edit /etc/asterisk/logger.conf
add the following line
console => notice,warning,error,debug,verbose
then from the shell execute
asterisk -rx "module reload"
Generate an inbound call, and copy the log that you got from the CLI
 

RafaelMorenoBR

Joined
Dec 16, 2009
Messages
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#3
jgutierrez said:
Some quick questions:

1. Do you need a register string? If so, what is the result of
asterisk -rx "sip show registry"

2. Have you tried using on the trunk definition the flag
insecure=very

3. I will recommend you to set an inbound route for ANY DID / ANY CID (this can be done if you leave blank the DID and CID fields)

4. Are you sure that you don't have any issues with your codecs? What codec should use your SIP trunk?

NOTES:
* By the way, when someone calls you, what do they get? A busy signal, a message from your provider, a message from your elastix server telling the person that the line is not in service?

* Paste the configuration of your sip trunk

* Enable extra log information on the CLI
edit /etc/asterisk/logger.conf
add the following line
console => notice,warning,error,debug,verbose
then from the shell execute
asterisk -rx "module reload"
Generate an inbound call, and copy the log that you got from the CLI

Hello my Country Neighbor

Many thanks for the Help

1-)Yes it's registered.
Code:
Host                            Username       Refresh State                Reg.Time                 
sps2.matrix.net.br:5060         MS00142489         105 Registered           Wed, 16 Dec 2009 15:35:28
2-)Yes I did that :)

3-)I also did that :)

4-)My sip server can receive calls as G729a and G711, I tried to call from PSTN and from my ATA that is logged on my sip server and make calls to everywhere.

NOTES:

I receive a busy signal because my SIP server give me a code that number was not found, I have also setup this DID and the account to a PAP2 ATA and the incoming call came fine.

SIP Tunk:

Code:
host=sps2.matrix.net.br
username=myuser
secret=mypass
type=peer
port=5060
domain=sps2.matrix.net.br
fromuser=myuser
fromdomain=sps2.matrix.net.br
insecury=very
qualify=no
nat=no
disallow=all
allow=g729&alaw&ulaw
dtmfmode=rfc2833
reinvite=no
canreinvite=no
insecure=invite
context=custom-get-did-from-sip
INCOMING

secret=mysecret
username=myuser
type=user
context=custom-get-did-from-sip
insecure=invite
The CLI:

Code:
[Dec 16 15:44:38] NOTICE[3257]: chan_sip.c:14847 handle_request_invite: Call from 'MS00142489' to extension 'MS00142489' rejected because extension not found.
 

RafaelMorenoBR

Joined
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#4
tried now using "context=from-trunk"

Code:
    -- Executing [MS00142489@from-trunk:1] NoOp("SIP/MatrixOut-b7c09f58", "Catch-All DID Match - Found MS00142489 - You probably want a DID for this.") in new stack
    -- Executing [MS00142489@from-trunk:2] Goto("SIP/MatrixOut-b7c09f58", "ext-did|s|1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] Set("SIP/MatrixOut-b7c09f58", "__FROM_DID=s") in new stack
    -- Executing [s@ext-did:2] Gosub("SIP/MatrixOut-b7c09f58", "app-blacklist-check|s|1") in new stack
    -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/MatrixOut-b7c09f58", "") in new stack
    -- Executing [s@app-blacklist-check:2] GotoIf("SIP/MatrixOut-b7c09f58", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/MatrixOut-b7c09f58", "") in new stack
    -- Executing [s@ext-did:3] ExecIf("SIP/MatrixOut-b7c09f58", "0 |Set|CALLERID(name)=MS00140838") in new stack
    -- Executing [s@ext-did:4] Set("SIP/MatrixOut-b7c09f58", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [s@ext-did:5] SetCallerPres("SIP/MatrixOut-b7c09f58", "allowed_not_screened") in new stack
    -- Executing [s@ext-did:6] Goto("SIP/MatrixOut-b7c09f58", "from-did-direct|100|1") in new stack
    -- Goto (from-did-direct,100,1)
    -- Executing [100@from-did-direct:1] Macro("SIP/MatrixOut-b7c09f58", "exten-vm|novm|100") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/MatrixOut-b7c09f58", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/MatrixOut-b7c09f58", "AMPUSER=MS00140838") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/MatrixOut-b7c09f58", "0?report") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/MatrixOut-b7c09f58", "1|Set|REALCALLERIDNUM=MS00140838") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: ExecIf
[Dec 16 16:21:18] DEBUG[4179]: func_db.c:70 function_db_read: DB: DEVICE/MS00140838/user not found in database.
    -- Executing [s@macro-user-callerid:4] Set("SIP/MatrixOut-b7c09f58", "AMPUSER=") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
[Dec 16 16:21:18] DEBUG[4179]: func_db.c:70 function_db_read: DB: AMPUSER//cidname not found in database.
    -- Executing [s@macro-user-callerid:5] Set("SIP/MatrixOut-b7c09f58", "AMPUSERCIDNAME=") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/MatrixOut-b7c09f58", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/MatrixOut-b7c09f58", "0?continue") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s@macro-user-callerid:11] Set("SIP/MatrixOut-b7c09f58", "__TTL=64") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/MatrixOut-b7c09f58", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/MatrixOut-b7c09f58", "Using CallerID "Rafael" <MS00140838>") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Noop
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Macro
    -- Executing [s@macro-exten-vm:2] Set("SIP/MatrixOut-b7c09f58", "RingGroupMethod=none") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-exten-vm:3] Set("SIP/MatrixOut-b7c09f58", "VMBOX=novm") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-exten-vm:4] Set("SIP/MatrixOut-b7c09f58", "EXTTOCALL=100") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
[Dec 16 16:21:18] DEBUG[4179]: func_db.c:70 function_db_read: DB: CFU/100 not found in database.
    -- Executing [s@macro-exten-vm:5] Set("SIP/MatrixOut-b7c09f58", "CFUEXT=") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
[Dec 16 16:21:18] DEBUG[4179]: func_db.c:70 function_db_read: DB: CFB/100 not found in database.
    -- Executing [s@macro-exten-vm:6] Set("SIP/MatrixOut-b7c09f58", "CFBEXT=") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-exten-vm:7] Set("SIP/MatrixOut-b7c09f58", "RT=""") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-exten-vm:8] Macro("SIP/MatrixOut-b7c09f58", "record-enable|100|IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/MatrixOut-b7c09f58", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s@macro-record-enable:4] AGI("SIP/MatrixOut-b7c09f58", "recordingcheck|20091216-162118|1260987678.2") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20091216-162118|1260987678.2: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: AGI
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/MatrixOut-b7c09f58", "") in new stack
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Macro
    -- Executing [s@macro-exten-vm:9] Macro("SIP/MatrixOut-b7c09f58", "dial||tr|100") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/MatrixOut-b7c09f58", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
[Dec 16 16:21:18] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s@macro-dial:3] AGI("SIP/MatrixOut-b7c09f58", "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
[Dec 16 16:21:19] WARNING[4182]: config.c:765 process_text_line: Unknown directive '#permit=192.168.1.0/255.255.255.0' at line 18 of /etc/asterisk/manager_custom.conf
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: Caller ID name is 'Rafael' number is 'MS00140838'
  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: ExtensionState: 0
  dialparties.agi: Extension 100 has ExtensionState: 0
    --  dialparties.agi: Checking CW and CFB status for extension 100
    --  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
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: AGI
    -- Executing [s@macro-dial:7] Dial("SIP/MatrixOut-b7c09f58", "SIP/100||tr") in new stack
[Dec 16 16:21:19] NOTICE[4179]: app_dial.c:1185 dial_exec_full: Hey! chan SIP/MatrixOut-b7c09f58's context='macro-dial', and exten='s'
[Dec 16 16:21:19] WARNING[4179]: chan_sip.c:3103 sip_call: No audio format found to offer. Cancelling call to 100
    -- Couldn't call 100
  == Everyone is busy/congested at this time (0:0/0/0)
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Dial
    -- Executing [s@macro-dial:8] Set("SIP/MatrixOut-b7c09f58", "DIALSTATUS=CHANUNAVAIL") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-dial:9] GosubIf("SIP/MatrixOut-b7c09f58", "0?CHANUNAVAIL|1") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GosubIf
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Macro
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/MatrixOut-b7c09f58", "0?exit|return") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s@macro-exten-vm:11] Set("SIP/MatrixOut-b7c09f58", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/MatrixOut-b7c09f58", "0?docfu|1") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GosubIf
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/MatrixOut-b7c09f58", "0?docfb|1") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GosubIf
    -- Executing [s@macro-exten-vm:14] Set("SIP/MatrixOut-b7c09f58", "DIALSTATUS=CHANUNAVAIL") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Set
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/MatrixOut-b7c09f58", "Voicemail is novm") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: NoOp
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/MatrixOut-b7c09f58", "1?s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/MatrixOut-b7c09f58", "IVR_RETVM:  IVR_CONTEXT: ") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: Noop
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/MatrixOut-b7c09f58", "0?exit|1") in new stack
[Dec 16 16:21:19] DEBUG[4179]: app_macro.c:379 _macro_exec: Executed application: GotoIf
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/MatrixOut-b7c09f58", "congestion") in new stack
[Dec 16 16:21:19] WARNING[4179]: channel.c:3055 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[Dec 16 16:21:19] WARNING[4179]: indications.c:121 playtones_alloc: Unable to set 'SIP/MatrixOut-b7c09f58' to signed linear format (write)
[Dec 16 16:21:19] NOTICE[4179]: res_indications.c:212 handle_playtones: Unable to start playtones
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/MatrixOut-b7c09f58' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 100, 1) exited non-zero on 'SIP/MatrixOut-b7c09f58'
 

dicko

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#5
[snip]
[Dec 16 16:21:19] WARNING[4179]: chan_sip.c:3103 sip_call: No audio format found to offer. Cancelling call to 100
-- Couldn't call 100
[/snip]

check your codecs.
 

jgutierrez

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#6
By the way, g729 is not installed by default. Seems to me that the key remains on the codec, try using alaw, instead of g729.
On you trunk configuration, you must have something like:

disallow=all
allow=alaw&ulaw
 

RafaelMorenoBR

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#7
Thanks Everyone,

The problem was the Codecs and my sip server was asking for the did on the end of the register, so i got to register username:password@server.mysip.com.br/DIDNUMBER.

Again many thanks,
 

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