[asterisk-users] the lenght of the uri affects on dialplan?
shengbin zhu
zhu.shengbin2003 at gmail.com
Sun Aug 26 22:28:34 CDT 2012
Hi,Rafael
I found some difference in INVITE message when the switch is long name, see
inline mark in red.
I guess the root cause is the server
MSSSASU1.MYDOMAiN.COM.PY<http://msssasu1.mydomain.com.py/> doesn't
support tcp.
Best Regards
Ben
2012/8/27 Rafael Visser <rafael_visser at hotmail.com>
> 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
> [Ben] --here indicate the SIP use TCP as transport protocol, but the
> Contact header field is UDP protocol, they are mismatch, you can check the
> same INVITE message when the switch is short name,[ they are matched.
> 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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