Asterisk 13 PJSIP : "Contact" field in SIP is randomized

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Asterisk 13 PJSIP : "Contact" field in SIP is randomized

Postby nauliv » Tue Feb 02, 2016 5:55 pm

Hello,

We are using Asterisk 13.1-cert2 and PJSIP, with Nextiva as the SIP Trunk provider.
Incoming calls are working, but dialing out is not working. The carrier says that there are 2 reasons why dialing out is not working :

1) If you look at the SIP trace below, the carrier is indicating that the reason for not working is because the "Contact:" line has a cryptic-randomized number in there, instead of the actual contact.

Right now it sends: Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d@107.207.90.90:5060>
But it should be sending: Contact: <sip:7143861212@107.207.90.90:5060>


2) We are unable to send the X-P-Asserted-Identity, because while the dialplan runs PJSIP_HEADER without error, the option doesn't make it into the SIP packet.

Thanks a lot in advance for your help !!

(hackers: don't waste time, private information have been changed)

******** SIP TRACE FOR OUTBOUND CALL **********

<--- Transmitting SIP request (891 bytes) to UDP:208.73.140.70:5060 --->
INVITE sip:9492345566@208.73.140.70:5060 SIP/2.0
Via: SIP/2.0/UDP 107.207.90.90:5060;rport;branch=z9hG4bKPja-5qMYTknEobutnWKzOTPpSrCg3swCz
From: "Manager Office"
<sip:7143861212@107.207.90.90>;tag=slIrplpaQV6S.tpB64u4BEbK-.7HBEis
To: <sip:9492345566@208.73.140.70>
Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d@107.207.90.90:5060>
Call-ID: QmF69h0tpNPtZhE6IrwYqtoQ67kcJ5pn
CSeq: 9147 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
PRACK, REFER, REGISTER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 455864548 455864548 IN IP4 107.207.90.90
s=Asterisk
c=IN IP4 107.207.90.90
t=0 0
m=audio 19628 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv



******** DIALPLAN FOR OUTBOUND CALL *********************

[dial-out]
exten => _X.,1,NoOp()
same => n,Set(PJSIP_HEADER(add,X-P-Asserted-Identity)=sip:7147890101)
same => n,Set(CALLERID(num)=7143861212)
same => n,Dial(PJSIP/${EXTEN}@nextiva1,45,r)
same => n,Hangup()


******** PJSIP CONFIGURATION FOR SIP TRUNK ***********

[nextiva1]
type=endpoint
transport=transport-udp
context=from-nextiva
disallow=all
allow=ulaw
aors=nextiva1-aor

[nextiva1-auth]
type=auth
auth_type=userpass
username=7143861212
password=NotMyRealPassword

[nextiva1-reg]
type=registration
outbound_auth=nextiva1-auth
server_uri=sip:bt.voipdnsservers.com
client_uri=sip:7143861212@bt.voipdnsservers.com


[nextiva1]
type=identify
endpoint=nextiva1
match=208.73.140.70

[nextiva1-aor]
type=aor
contact=sip:bt.voipdnsservers.com:5060
nauliv
Newsterisk
 
Posts: 8
Joined: Wed Jul 09, 2008 12:22 am

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