[asterisk-users] PJSIP configuration question

Sun Dec 14 17:50:15 CST 2014

I am running PJPROJECT 2.3 and Asterisk 13.0.0.

I answer the call, about 15 seconds later, vitality hangs up on my cell phone.
However, Asterisk is never notified
When the OK (for the answer) occurs, the ACK seems to never be accepted.
The OK recvd with ACK sent occurs several times.

Here are the pjsip.conf settings...
[global]
type = global
debug = yes

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93

[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no

[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93

When I Originate a call via AMI...

Action: Originate
ActionID: S8
Channel: PJSIP/8005555555 at outbound.vitelity.net<mailto:PJSIP/8005555555 at outbound.vitelity.net>
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: John Doe <1234>
Variable: CALLERID(num-pres)=allowed_passed_screen
Async: true
The call goes through to my cell phone.  I answer on my cell phone and Asterisk sees the call being answered.  However, Vitelity disconnects the cell phone about 15 seconds later.
When looking at the PJSIP trace, the ACK repsonse to the 200 OK (Answer) are missing the Contact header.  From what I understand that is likely the reason Vitelity doesn't seem to process the ACK.

*CLI>     -- Called 8005555555 at outbound.vitelity.net<mailto:8005555555 at outbound.vitelity.net>
<--- Transmitting SIP request (1018 bytes) to UDP:64.2.142.93:5060 --->
INVITE sip:8005555555 at 64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.166:5060;rport;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <sip:8005555555 at 64.2.142.93>
Contact: <sip:63240147-e592-47ae-9fbd-4f1cfbb5c5a6 at 192.168.11.166:5060>
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 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
Remote-Party-ID: "John Doe" <sip:1234 at 192.168.11.166>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 2134048799 2134048799 IN IP4 192.168.11.166
s=Asterisk
c=IN IP4 192.168.11.166
t=0 0
m=audio 16262 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

<--- Received SIP response (384 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.11.166:5060;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <sip:8005555555 at 64.2.142.93>
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 INVITE
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0

<--- Received SIP response (851 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.166:5060;received=192.168.11.166;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2
Record-Route: <sip:64.2.142.93;lr=on>
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <sip:8005555555 at 64.2.142.93>;tag=as466c6135
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 66.241.99.145>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3457 3457 IN IP4 66.241.99.145
s=session
c=IN IP4 66.241.99.145
t=0 0
m=audio 11872 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

    -- PJSIP/outbound.vitelity.net-00000000 is making progress
       > 0x60fc840 -- Probation passed - setting RTP source address to 66.241.99.145:11872
<--- Received SIP response (837 bytes) from UDP:64.2.142.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.166:5060;received=192.168.11.166;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2
Record-Route: <sip:64.2.142.93;lr=on>
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <sip:8005555555 at 64.2.142.93>;tag=as466c6135
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:18005555555 at 66.241.99.145>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3457 3458 IN IP4 66.241.99.145
s=session
c=IN IP4 66.241.99.145
t=0 0
m=audio 11872 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<--- Transmitting SIP request (443 bytes) to UDP:64.2.142.93:5060 --->
ACK sip:18005555555 at 64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.166:5060;rport;branch=z9hG4bKPj5fa1c45c-fb91-4d87-aa6b-9ae451dcd211
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=340be959-af87-4b77-a7a7-8aaf83940b03
To: <sip:8005555555 at 64.2.142.93>;tag=as466c6135
Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125
CSeq: 10207 ACK
Route: <sip:64.2.142.93;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length:  0

    -- PJSIP/outbound.vitelity.net-00000000 answered
    -- Executing [createcall at TestApp:1] Set("PJSIP/outbound.vitelity.net-00000000", "EXTIVR=") in new stack
    -- Executing [createcall at TestApp:2] AGI("PJSIP/outbound.vitelity.net-00000000", "agi:async") in new stack

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