[asterisk-users] PJSIP configuration question

Sun Dec 14 17:46:24 CST 2014

Trying this again after my first away from work in a couple weeks.

Running Asterisk 13.0.0
IP authentication with Vitelity

I can Originate with sip, but not pjsip.
Here is the sip settings and trace.

Action: Originate
ActionID: S8
Channel: SIP/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

sip.conf
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp

== Using SIP RTP CoS mark 5
Audio is at 18226
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.2.142.189:5060:
INVITE sip:8005555555 at outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183
Max-Forwards: 70
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de
To: <sip:8005555555 at outbound.vitelity.net>
Contact: <sip:1234 at 192.168.11.166:5060>
Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.0.0
Date: Sun, 21 Dec 2014 20:06:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1422632184 1422632184 IN IP4 192.168.11.166
s=Asterisk PBX 13.0.0
c=IN IP4 192.168.11.166
t=0 0
m=audio 18226 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called 8005555555 at outbound.vitelity.net

<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de
To: <sip:8005555555 at outbound.vitelity.net>
Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8005555555 at 64.2.142.189>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de
To: <sip:8005555555 at outbound.vitelity.net>;tag=as5458ca04
Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8005555555 at 64.2.142.189>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 21997 21997 IN IP4 64.2.142.189
s=session
c=IN IP4 64.2.142.189
t=0 0
m=audio 19282 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
sip_route_dump: route/path hop: <sip:8005555555 at 64.2.142.189>
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.189:19282
    -- SIP/outbound.vitelity.net-00000000 is making progress
       > 0x483cdb0 -- Probation passed - setting RTP source address to 64.2.142.189:19282

<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de
To: <sip:8005555555 at outbound.vitelity.net>;tag=as5458ca04
Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8005555555 at 64.2.142.189>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 21997 21998 IN IP4 64.2.142.189
s=session
c=IN IP4 64.2.142.189
t=0 0
m=audio 19282 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.189:19282
sip_route_dump: route/path hop: <sip:8005555555 at 64.2.142.189>
set_destination: Parsing <sip:8005555555 at 64.2.142.189> for address/port to send to
set_destination: set destination to 64.2.142.189:5060
Transmitting (no NAT) to 64.2.142.189:5060:
ACK sip:8005555555 at 64.2.142.189 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK6e0d8c45
Max-Forwards: 70
From: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de
To: <sip:8005555555 at outbound.vitelity.net>;tag=as5458ca04
Contact: <sip:1234 at 192.168.11.166:5060>
Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0

---
    -- SIP/outbound.vitelity.net-00000000 answered
    -- Executing [createcall at TestApp:1] Set("SIP/outbound.vitelity.net-00000000", "EXTIVR=") in new stack
    -- Executing [createcall at TestApp:2] AGI("SIP/outbound.vitelity.net-00000000", "agi:async") in new stack
       > 0x483cdb0 -- Probation passed - setting RTP source address to 64.2.142.189:19282

<--- SIP read from UDP:64.2.142.189:5060 --->
BYE sip:1234 at 192.168.11.166:5060 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.189:5060;branch=z9hG4bK521870f9;rport
From: <sip:8005555555 at outbound.vitelity.net>;tag=as5458ca04
To: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de
Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060
CSeq: 102 BYE
User-Agent: packetrino
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 64.2.142.189:5060 (no NAT)
Scheduling destruction of SIP dialog '59e9eff8339e32af271c23541298135d at 192.168.11.166:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 64.2.142.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.2.142.189:5060;branch=z9hG4bK521870f9;received=64.2.142.189;rport=5060
From: <sip:8005555555 at outbound.vitelity.net>;tag=as5458ca04
To: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de
Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060
CSeq: 102 BYE
Server: Asterisk PBX 13.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
  == Spawn extension (TestApp, createcall, 2) exited non-zero on 'SIP/outbound.vitelity.net-00000000'

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