[asterisk-users] the lenght of the uri affects on dialplan?
Rafael Visser
rafael_visser at hotmail.com
Sun Aug 26 18:52:43 CDT 2012
Ok...
sip.conf
[general]
context=default ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=yes ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
relaxdtmf=yes
dtmfmode=inband
;rfc2833compensate=yes
users.conf
[general]
fullname = New User
userbase = 6000
hasvoicemail = yes
vmsecret = 1234
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
[sip.ericsson]
;cambios allowguest hosts
;allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
type=friend
calllimit=200
fromuser=ivr1
dtmfmode=inband
username=administrador
context=incoming-sip-ericsson
host=10.146.9.70
host=ericsson
host=MSSASU1.MYDOMAIN.COM.PY
port=5060
disallow=all
allow=alaw
allow=gsm
allow=ulaw
qualify=yes
insecure=no
> Date: Mon, 27 Aug 2012 03:42:51 +0500
> From: faisal at vopium.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan?
>
> 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
> >
> >
> >
> >--
> >_____________________________________________________________________
> >-- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
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> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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