Hi gurus!!!
I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn.
Every minute asterisk sends an OPTION Request, i beleived that it’s related to qualify functions.
Then every minute annoyng answer of the pstn is “403 Forbidden”.
Some people from PSTN coments that asterisk is not sending the username in the OPTION Request, required by the pstn.
Is it wright?
How can i instruct FREEPBX to send the username in the option request?
Thanks in advance.
rv
Sip Debug of the peer
Reliably Transmitting (NAT) to 201.217.31.XX:5060:
OPTIONS sip:201.217.31.10 SIP/2.0 <---- here
Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected]:6060;tag=as4491c6af
To: sip:201.217.31.10
Contact: sip:[email protected]:6060
Call-ID: [email protected]:6060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(1.8.25.0)
Date: Wed, 25 Jun 2014 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:201.217.31.XX:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060
From: “Unknown” sip:[email protected]:6060;tag=as4491c6af
To: sip:201.217.31.XX;tag=aprqngfrt-nm50ea10000c6
Call-ID: [email protected]:6060
CSeq: 102 OPTIONS
This is the peer.
- Name : desde-XopaXo-2376XXX
Secret :
MD5Secret :
Remote Secret:
Context : from-trunk
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : “” <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 201.217.31.10
Addr->IP : 201.217.31.10:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 595212376458
SIP Options : timer
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (ulaw:20,alaw:20,gsm:20)
Auto-Framing : No
Status : OK (36 ms)
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No - Name : desde-XopaXo-2376XXX
Secret :
MD5Secret :
Remote Secret:
Context : from-trunk
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : “” <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 201.217.31.XX
Addr->IP : 201.217.31.XX:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 59X212376XXX
SIP Options : timer
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (ulaw:20,alaw:20,gsm:20)
Auto-Framing : No
Status : OK (36 ms)
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Taking a look of the example of rfc3261.txt (pg 67), we found “carol”, so it makingme see that i am missing some config.
OPTIONS sip:[email protected] SIP/2.0 <---- here we found username in the uri
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
Max-Forwards: 70
To: <sip:[email protected]>
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