[asterisk-bugs] [JIRA] (ASTERISK-25741) res_pjsip: "Contact" contains UUID for user portion
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Wed Feb 10 17:12:33 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25741?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-25741:
------------------------------------
Description:
Hello,
Using Asterisk 13.1-cert2 and PJSIP, with Nextiva as the SIP Trunk provider. 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.
It sends:
Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 107.207.90.90:5060>
Instead of sending:
Contact: <sip:7143861212 at 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.
h2. SIP TRACE FOR OUTBOUND CALL
{noformat}
<--- Transmitting SIP request (891 bytes) to UDP:208.73.140.70:5060 --->
INVITE sip:9492345566 at 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 at 107.207.90.90>;tag=slIrplpaQV6S.tpB64u4BEbK-.7HBEis
To: <sip:9492345566 at 208.73.140.70>
Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 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
{noformat}
h2. DIALPLAN FOR OUTBOUND CALL
{noformat}
[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()
{noformat}
h2. PJSIP CONFIGURATION FOR SIP TRUNK
{noformat}
[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 at bt.voipdnsservers.com
[nextiva1]
type=identify
endpoint=nextiva1
match=208.73.140.70
[nextiva1-aor]
type=aor
contact=sip:bt.voipdnsservers.com:5060
{noformat}
was:
Hello,
Using Asterisk 13.1-cert2 and PJSIP, with Nextiva as the SIP Trunk provider. 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.
It sends:
Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 107.207.90.90:5060>
Instead of sending:
Contact: <sip:7143861212 at 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.
h.2 SIP TRACE FOR OUTBOUND CALL
{noformat}
<--- Transmitting SIP request (891 bytes) to UDP:208.73.140.70:5060 --->
INVITE sip:9492345566 at 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 at 107.207.90.90>;tag=slIrplpaQV6S.tpB64u4BEbK-.7HBEis
To: <sip:9492345566 at 208.73.140.70>
Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 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
{noformat}
h.2 DIALPLAN FOR OUTBOUND CALL
{noformat}
[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()
{noformat}
h.2 PJSIP CONFIGURATION FOR SIP TRUNK
{noformat}
[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 at bt.voipdnsservers.com
[nextiva1]
type=identify
endpoint=nextiva1
match=208.73.140.70
[nextiva1-aor]
type=aor
contact=sip:bt.voipdnsservers.com:5060
{noformat}
> res_pjsip: "Contact" contains UUID for user portion
> ---------------------------------------------------
>
> Key: ASTERISK-25741
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25741
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_pjsip
> Affects Versions: 13.1.0
> Environment: CentOS 2.6.32-504.23.4.el6.i686
> Reporter: Oliver Nauliv
> Assignee: Unassigned
> Severity: Minor
>
> Hello,
> Using Asterisk 13.1-cert2 and PJSIP, with Nextiva as the SIP Trunk provider. 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.
> It sends:
> Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 107.207.90.90:5060>
> Instead of sending:
> Contact: <sip:7143861212 at 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.
> h2. SIP TRACE FOR OUTBOUND CALL
> {noformat}
> <--- Transmitting SIP request (891 bytes) to UDP:208.73.140.70:5060 --->
> INVITE sip:9492345566 at 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 at 107.207.90.90>;tag=slIrplpaQV6S.tpB64u4BEbK-.7HBEis
> To: <sip:9492345566 at 208.73.140.70>
> Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 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
> {noformat}
> h2. DIALPLAN FOR OUTBOUND CALL
> {noformat}
> [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()
> {noformat}
> h2. PJSIP CONFIGURATION FOR SIP TRUNK
> {noformat}
> [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 at bt.voipdnsservers.com
> [nextiva1]
> type=identify
> endpoint=nextiva1
> match=208.73.140.70
> [nextiva1-aor]
> type=aor
> contact=sip:bt.voipdnsservers.com:5060
> {noformat}
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