[asterisk-users] the lenght of the uri affects on dialplan?

Sun Aug 26 17:42:51 CDT 2012

mention the complete scnario and your sip.conf.

Regards,

Faisal 
(sent from phone)

Rafael Visser <rafael_visser at hotmail.com> wrote:

>
>Hi Gurus..
>I use asterisk for just for ivr.
>My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call is rejected with "No matching peer" and the "handle_request_invite: Sending fake auth rejection for device x". It doesn't match it's own default context. 
>
>Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong.
>Do you know what am i missing?
>Thanks in advance.
>
>
>
>
>
>
>
>
>
>Debug with long hostname (B is considered as an '*')
>================================
><--- SIP read from TCP:10.146.9.70:6240 --->
>INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone SIP/2.0
>From: <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
>To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
>Max-Forwards: 70
>Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096
>Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
>CSeq: 7313 INVITE
>P-Asserted-Identity: <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY;user=phone>
>Accept: application/sdp
>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
>P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
>Supported: 100rel
>Content-Type: application/sdp
>Contact: <sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP>
>Content-Length: 414
>
>v=0
>o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY
>s=-
>t=0 0
>a=sendrecv
>m=audio 13802 RTP/AVP 8 96 18 97
>c=IN IP4 10.143.1.67
>b=RR:0
>b=RS:0
>a=rtpmap:8 PCMA/8000
>a=rtpmap:96 AMR/8000
>a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
>a=rtpmap:18 G729/8000
>a=fmtp:18 annexb=yes
>a=rtpmap:97 telephone-event/8000
>a=fmtp:97 0-15
>a=maxptime:40
><------------->
>--- (15 headers 17 lines) ---
>Sending to 10.146.9.70:5060 (no NAT)
>Using INVITE request as basis request - 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
>################
>No matching peer for '971200152' from '10.146.9.70:6240'
>[Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: Sending fake auth rej
>ection for device <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
>#################
><--- Reliably Transmitting (no NAT) to 10.146.9.70:5060 --->
>SIP/2.0 401 Unauthorized
>Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096;received=10.146.9.70
>From: <sip:971200152 at MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695
>To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>;tag=as4cfd0d54
>Call-ID: 9caX8060616182201-AAAABOPA- at MSSASU1.MYDOMAIN.COM.PY
>CSeq: 7313 INVITE
>Server: Asterisk PBX 1.8.7.0
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
>Supported: replaces, timer
>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35ff0feb"
>Content-Length: 0
>
>
>
>
>Short hostname on switch
>===============
>Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1 (pid = 25430)
>fdosis-ims1*CLI> core set verbose 1
>Verbosity was 0 and is now 1
>
><--- SIP read from UDP:10.146.9.70:5060 --->
>INVITE sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone SIP/2.0
>From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
>To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
>Max-Forwards: 70
>Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982
>Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
>CSeq: 14481 INVITE
>P-Asserted-Identity: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>
>Accept: application/sdp
>llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
>P-Charging-Vector: icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
>Supported: 100rel
>Content-Type: application/sdp
>Contact: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>
>Content-Length: 407
>
>v=0
>o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN
>s=-
>t=0 0
>a=sendrecv
>m=audio 30838 RTP/AVP 8 96 18 97
>c=IN IP4 10.143.1.68
>b=RR:0
>b=RS:0
>a=rtpmap:8 PCMA/8000
>a=rtpmap:96 AMR/8000
>a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
>a=rtpmap:18 G729/8000
>a=fmtp:18 annexb=yes
>a=rtpmap:97 telephone-event/8000
>a=fmtp:97 0-15
>a=maxptime:40
><------------->
>--- (15 headers 17 lines) ---
>Sending to 10.146.9.70:5060 (no NAT)
>Using INVITE request as basis request - qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
>Found peer 'sip.ericsson' for '971200152' from 10.146.9.70:5060
>Found RTP audio format 8
>Found RTP audio format 96
>Found RTP audio format 18
>Found RTP audio format 97
>Found audio description format PCMA for ID 8
>Found unknown media description format AMR for ID 96
>Found audio description format G729 for ID 18
>Found audio description format telephone-event for ID 97
>Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
>Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
>Peer audio RTP is at port 10.143.1.68:30838
>Looking for B56510123456789012345 in incoming-sip-ericsson (domain SISIVR03.MYDOMAIN.COM.PY)
>list_route: hop: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP>
>
><--- Transmitting (no NAT) to 10.146.9.70:5060 --->
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982;received=10.146.9.70
>From: <sip:971200152 at MSSASU1.MYDOMAIN;user=phone>;tag=0046120455
>To: <sip:B56510123456789012345 at SISIVR03.MYDOMAIN.COM.PY;user=phone>
>Call-ID: qDaQ1240646182201-AAAAAKDE- at MSSASU1.MYDOMAIN
>CSeq: 14481 INVITE
>Server: Asterisk PBX 1.8.7.0
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
>Supported: replaces, timer
>Contact: <sip:B56510123456789012345 at 10.146.9.132:5060>
>Content-Length: 0
>
>
> 		 	   		  
>--
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