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Home » Kerio User Forums » Kerio Operator » Call Routing Fails for Incoming Calls (Call from XX to extension YY rejected because extension not found.)
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zinian

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Hi

Call Routing of incoming calls is giving me tough trouble. I have configured a range of external numbers by 41445260XXX and I use a rewirte rule "strip 8 from left". Then I get the following debug message:


[15/Jun/2011 12:24:13] {asterisk} operator asterisk[1727]: NOTICE[1743]: chan_sip.c:15519 in handle_request_invite: Call from \'41445260389\' to extension \'41445260380\' rejected because extension not found.


But extension 380 does exist and is registered! Is the rewrite rule not applied?!

I have also tried with an explicit mapping (no XXX but a list of coma-separated number) - no success!

Any suggestions... Highly appreciated!! And... I am using v1.1.0beta2

Here is a more complete Debug excerpt:



[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: <--- SIP read from 212.117.203.34:5060 ---> INVITE sip:41445260380@212.126.164.102[/email] SIP/2.0 Via: SIP/2.0/UDP 212.117.203.34:5060;branch=z9hG4bK-d8754z-0ec12c56c348ad74-1 ---d8754z-;rport Via: SIP/2.0/UDP 212.117.203.34:5061;branch=z9hG4bK-l3gkp3dbigjclfpa;rport=50 61 Max-Forwards: 69 Record-Route: <sip:212.117.203.34;lr> Contact: \"Anonymous\"<sip:212.117.203.34:5061> To: <sip:41445260380@212.117.203.34[/email]> From: <sip:0445260412<_at_>212.117.203.34>;tag=jfotrbnwnrnfjjw3.o Call-ID: 26A36E69C61DC5C29D09D7D0FBCB914053DD9399~2o~1o CSeq: 873 INVITE Expires: 300 Content-Disposition: session Content-Type: application/sdp User-Agent: Sippy cisco-GUID: 1251233545-2114105336-1441096176-4215432486 h323-conf-id: 1251233545-2114105336-1441096176-4215432486 Portabilling-notify: aor=41445260389 Content-Length: 188 v=0 o=Sippy 237206920 0 IN IP4 212.117.203.34 s=yyirktn t=0 0 m=audio 40028 RTP/AVP 3 0 8 100 c=IN IP4 212.117.203.31 a=rt
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: --- (18 headers 9 lines) ---
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Sending to 212.117.203.34 : 5060 (no NAT)
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Using INVITE request as basis request - 26A36E69C61DC5C29D09D7D0FBCB914053DD9399~2o~1o
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: NOTICE[1706]: securityModule.h:252 in int resKerioSecurityCheck(sockaddr_in*, char*): KerioSecurity check 212.117.203.34 (uri:<sip:41445260380<_at_>212.117.203.34>)
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Found peer \'i-ZI\'
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: NOTICE[1706]: chan_sip.c:20153 in securityPostAuth: Kerio security: securityPostAuth - AUTH_SUCCESSFUL.
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: NOTICE[1706]: securityModule.h:360 in void resKerioSecurityAuthSuccess(sockaddr_in*, char*): Asterisk: Host 212.117.203.34 authenticated (uri:<sip:41445260380<_at_>212.117.203.34>).
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Found RTP audio format 3
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Found RTP audio format 0
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Found RTP audio format 8
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Found RTP audio format 100
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Found audio description format telephone-event for ID 100
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Capabilities: us - 0xa0e (gsm|ulaw|alaw|g726|speex), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Peer audio RTP is at port 212.117.203.31:40028
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Looking for 41445260380 in i-pro1-voipgateway-org (domain 212.126.164.102)
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: <--- Reliably Transmitting (NAT) to 212.117.203.34:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 212.117.203.34:5060;branch=z9hG4bK-d8754z-0ec12c56c348ad74-1 ---d8754z-;received=212.117.203.34;rport=5060 Via: SIP/2.0/UDP 212.117.203.34:5061;branch=z9hG4bK-l3gkp3dbigjclfpa;rport=50 61 From: <sip:0445260412@212.117.203.34[/email]>;tag=jfotrbnwnrnfjjw3.o To: <sip:41445260380<_at_>212.117.203.34>;tag=as04a434f1 Call-ID: 26A36E69C61DC5C29D09D7D0FBCB914053DD9399~2o~1o CSeq: 873 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------>
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: NOTICE[1706]: chan_sip.c:15519 in handle_request_invite: Call from \'41445260389\' to extension \'41445260380\' rejected because extension not found.
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Scheduling destruction of SIP dialog \'26A36E69C61DC5C29D09D7D0FBCB914053DD9399~2o~1o\' in 32000 ms (Method: INVITE)
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: <--- SIP read from 212.117.203.34:5060 ---> ACK sip:41445260380@212.126.164.102[/email] SIP/2.0 Via: SIP/2.0/UDP 212.117.203.34:5060;branch=z9hG4bK-d8754z-0ec12c56c348ad74-1 ---d8754z-;rport Max-Forwards: 70 To: <sip:41445260380@212.117.203.34[/email]>;tag=as04a434f1 From: <sip:0445260412<_at_>212.117.203.34>;tag=jfotrbnwnrnfjjw3.o Call-ID: 26A36E69C61DC5C29D09D7D0FBCB914053DD9399~2o~1o CSeq: 873 ACK Content-Length: 0 <------------->
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: --- (8 headers 0 lines) ---
[17/Jun/2011 11:53:04] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Really destroying SIP dialog \'26A36E69C61DC5C29D09D7D0FBCB914053DD9399~2o~1o\' Method: ACK
[17/Jun/2011 11:53:09] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Really destroying SIP dialog \'98430ba1-4e9035b5<_at_>10.42.3.13\' Method: REGISTER
[17/Jun/2011 11:53:09] {asterisk} operator asterisk[1690]: VERBOSE[1706]: Really destroying SIP dialog \'4200f9e5-68cf7c41<_at_>10.42.3.13\' Method: REGISTER

[Updated on: Fri, 17 June 2011 14:36]

  •  
Vladimir Toncar (Kerio)

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zinian wrote on Fri, 17 June 2011 14:28

to extension \'41445260380\' rejected because extension not found


Hi,

It looks that you are not rewriting the number to '380' but to '41445260380'. Can you post the full content (or a screenshot) of your rewriting rule?

Vladimir
  •  
zinian

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That's what I thought as well. It somehow seems the rewriting is not applied for some reasons. I have played with the rule but couldn't observe any effect.

See screenshot in the attachment

  • Attachment: rewrite.png
    (Size: 38.25KB, Downloaded 454 times)
  •  
Vladimir Toncar (Kerio)

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I guess you do not need to string digits from the calling number.

Can you send me your 'supportInfo' file? You will find it in the System Health page. Please send it to me in a private message.
  •  
Jakub Zeman (Kerio)

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Hi,

this problem is connected with bug in Kerio Operator which occurs when Caller ID is enabled.

The bug will be fixed in Kerio Operator 1.1.0 RC2

Jakub
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