Asterisk 11.3 problem

ebt · May 22, 2013, 4:01pm

Hi

I installed a new Centos6 with Asterisk 11.3 Connected to a SIP trunk.
When i dial in a number 0731234567 the asterisk reports rejected because extension 0731234567 not found in context default. when i change in extension the extension to _0731234567. everything works fine.
any idea?

dejanst · May 23, 2013, 7:26am

Please copy/paste the content of default context from extensions.conf and the copy/paste from the Asterisk CLI (verbosity atleast 3) when you make an incoming call. Also please do a “sip set debug on” in Asterisk CLI and copy/paste the output.

ebt · May 23, 2013, 7:38am

hi

<— SIP read from UDP:172.16.10.43:5060 —>
INVITE sip:0737687007;cic=BARAK@10.57.13.141:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B12974ec30a35bd8f
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021d50ec
To: sip:0737687007@10.57.13.141
Call-ID: 151151832_58731307@172.16.10.43
CSeq: 24406 INVITE
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: “Eyal Bentamar” sip:240@172.16.10.43:5060
P-Asserted-Identity: “Eyal Bentamar” sip:Anonymous@172.16.10.43:5060
Proxy-Require: privacy
Privacy: critical; id
Supported: timer,100rel,replaces
Session-Expires: 1800
Min-SE: 90
Content-Length: 307
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 19159 24390 IN IP4 172.16.10.43
s=SIP Media Capabilities
c=IN IP4 172.16.10.33
t=0 0
m=audio 21214 RTP/AVP 0 8 18 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
<------------->
— (19 headers 14 lines) —
Sending to 172.16.10.43:5060 (no NAT)
Using INVITE request as basis request - 151151832_58731307@172.16.10.43
Found peer ‘013-out’ for ‘240’ from 172.16.10.43:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 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
Found audio description format telephone-event for ID 100
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.10.33:21214
Looking for 0737687007;cic=BARAK in default (domain 10.57.13.141)

<— Reliably Transmitting (no NAT) to 172.16.10.43:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B12974ec30a35bd8f;received=172.16.10.43
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021d50ec
To: sip:0737687007@10.57.13.141;tag=as2bf32cb7
Call-ID: 151151832_58731307@172.16.10.43
CSeq: 24406 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[May 23 10:36:14] NOTICE[28359][C-0000002c]: chan_sip.c:25251 handle_request_invite: Call from ‘013-out’ (172.16.10.43:5060) to extension ‘0737687007’ rejected because extension not found in context ‘default’.
Scheduling destruction of SIP dialog ‘151151832_58731307@172.16.10.43’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:172.16.10.43:5060 —>
ACK sip:0737687007;cic=BARAK@10.57.13.141:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B12974ec30a35bd8f
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021d50ec
To: sip:0737687007@10.57.13.141;tag=as2bf32cb7
Call-ID: 151151832_58731307@172.16.10.43
CSeq: 24406 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘151151832_58731307@172.16.10.43’ Method: ACK

<— SIP read from UDP:172.16.10.43:5060 —>
INVITE sip:0737687007;cic=SUBS@10.57.13.141:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK05Bdfc05420891ce8c2
From: sip:089363272@172.16.10.43;pstn-params=9081828088;tag=gK05473edd
To: sip:737687007@10.57.13.141
Call-ID: 151343654_49453381@172.16.10.43
CSeq: 6297 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: sip:089363272@172.16.10.43:5060
P-Asserted-Identity: sip:089363272@172.16.10.43:5060
Supported: timer,100rel,replaces
Session-Expires: 1800
Min-SE: 90
Content-Length: 305
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 7763 27271 IN IP4 172.16.10.43
s=SIP Media Capabilities
c=IN IP4 172.16.10.33
t=0 0
m=audio 9784 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
— (17 headers 14 lines) —
Sending to 172.16.10.43:5060 (no NAT)
Using INVITE request as basis request - 151343654_49453381@172.16.10.43
Found peer ‘013-out’ for ‘089363272’ from 172.16.10.43:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.10.33:9784
Looking for 0737687007;cic=SUBS in default (domain 10.57.13.141)

<— Reliably Transmitting (no NAT) to 172.16.10.43:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK05Bdfc05420891ce8c2;received=172.16.10.43
From: sip:089363272@172.16.10.43;pstn-params=9081828088;tag=gK05473edd
To: sip:737687007@10.57.13.141;tag=as3638b861
Call-ID: 151343654_49453381@172.16.10.43
CSeq: 6297 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[May 23 10:36:22] NOTICE[28359][C-0000002d]: chan_sip.c:25251 handle_request_invite: Call from ‘013-out’ (172.16.10.43:5060) to extension ‘0737687007’ rejected because extension not found in context ‘default’.
Scheduling destruction of SIP dialog ‘151343654_49453381@172.16.10.43’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:172.16.10.43:5060 —>
ACK sip:0737687007;cic=SUBS@10.57.13.141:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK05Bdfc05420891ce8c2
From: sip:089363272@172.16.10.43;pstn-params=9081828088;tag=gK05473edd
To: sip:737687007@10.57.13.141;tag=as3638b861
Call-ID: 151343654_49453381@172.16.10.43
CSeq: 6297 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘151343654_49453381@172.16.10.43’ Method: ACK
propus*CLI>

thanks

ebt · May 23, 2013, 8:28am

The defualt context is

[default]

exten => s,1,Answer()
exten => s,n,Goto(timewatch,1000,1);; PRODUCTION

exten => t,1,Hangup()

exten => i,1,Goto(default,s,2)

;; Incoming 013

exten =>_0737687000.,1,Answer()
exten =>_0737687000.,n,Set(CHANNEL(language)=he)
exten =>_0737687000.,n,Goto(timewatch,1000,1);; PRODUCTION

exten =>_0737687006.,1,Answer()
exten =>_0737687006.,n,Set(CHANNEL(language)=ar)
exten =>_0737687006.,n,Goto(timewatch,1000,1);; PRODUCTION

exten =>0737687007,1,Noop(${CALLERID(num)})
exten =>0737687007,n,hangup(17)

exten => h,1,NoOp(Starting Hangup sequence)
;;;;exten => h,n,system(/bin/bash /opt/support/bin/moveCallFile ${CALLERID(num)})
exten => h,n,Hangup()
;; End CallBack

cli output when calling a number defined in _073XXXXXX. format is:

<— SIP read from UDP:172.16.10.43:5060 —>
INVITE sip:0737687000;cic=BARAK@10.57.13.141:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B11f87901a71d9873
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021bee76
To: sip:0737687000@10.57.13.141
Call-ID: 151151497_23667784@172.16.10.43
CSeq: 4462 INVITE
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: “Eyal Bentamar” sip:240@172.16.10.43:5060
P-Asserted-Identity: “Eyal Bentamar” sip:Anonymous@172.16.10.43:5060
Proxy-Require: privacy
Privacy: critical; id
Supported: timer,100rel,replaces
Session-Expires: 1800
Min-SE: 90
Content-Length: 306
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 29079 5778 IN IP4 172.16.10.43
s=SIP Media Capabilities
c=IN IP4 172.16.10.33
t=0 0
m=audio 25668 RTP/AVP 0 8 18 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
<------------->
— (19 headers 14 lines) —
Sending to 172.16.10.43:5060 (no NAT)
Using INVITE request as basis request - 151151497_23667784@172.16.10.43
Found peer ‘013-out’ for ‘240’ from 172.16.10.43:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 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
Found audio description format telephone-event for ID 100
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.10.33:25668
Looking for 0737687000;cic=BARAK in default (domain 10.57.13.141)
list_route: hop: sip:240@172.16.10.43:5060

<— Transmitting (no NAT) to 172.16.10.43:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B11f87901a71d9873;received=172.16.10.43
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021bee76
To: sip:0737687000@10.57.13.141
Call-ID: 151151497_23667784@172.16.10.43
CSeq: 4462 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:0737687000;cic=BARAK@10.57.13.141:5060
Content-Length: 0

<------------>
Audio is at 18484
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 172.16.10.43:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B11f87901a71d9873;received=172.16.10.43
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021bee76
To: sip:0737687000@10.57.13.141;tag=as36f0f332
Call-ID: 151151497_23667784@172.16.10.43
CSeq: 4462 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:0737687000;cic=BARAK@10.57.13.141:5060
Content-Type: application/sdp
Require: timer
Content-Length: 259

v=0
o=root 1408244904 1408244904 IN IP4 10.57.13.141
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.13.141
t=0 0
m=audio 18484 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:172.16.10.43:5060 —>
ACK sip:0737687000;cic=BARAK@10.57.13.141:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B11f93480a71d9873
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021bee76
To: sip:0737687000@10.57.13.141;tag=as36f0f332
Call-ID: 151151497_23667784@172.16.10.43
CSeq: 4462 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:172.16.10.43:5060 —>
BYE sip:0737687000;cic=BARAK@10.57.13.141:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B12003679a71d9873
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021bee76
To: sip:0737687000@10.57.13.141;tag=as36f0f332
Call-ID: 151151497_23667784@172.16.10.43
CSeq: 4463 BYE
Max-Forwards: 70
Supported: 100rel
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 172.16.10.43:5060 (no NAT)
Scheduling destruction of SIP dialog ‘151151497_23667784@172.16.10.43’ in 6400 ms (Method: BYE)

<— Transmitting (no NAT) to 172.16.10.43:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.43:5060;branch=z9hG4bK02B12003679a71d9873;received=172.16.10.43
From: “Eyal Bentamar” sip:240@10.57.13.141;pstn-params=9083828088;tag=gK021bee76
To: sip:0737687000@10.57.13.141;tag=as36f0f332
Call-ID: 151151497_23667784@172.16.10.43
CSeq: 4463 BYE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘151151497_23667784@172.16.10.43’ Method: BYE
propus*CLI>

CLI output when i call a number which is properly defined : 073787007