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Dan Cropp

2014-Dec-14 23:50 UTC

[asterisk-users] PJSIP configuration question

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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Dan Cropp

2014-Dec-15 21:08 UTC

head link

[asterisk-users] PJSIP configuration question

Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.

Same problem is happening with both of them.

Could this be caused by PJPROJECT 2.3?

Anyone have any suggestions for what I can try?

My boss is giving me until tomorrow to get the PJSIP support working with
Vitelity.  Otherwise, he's told me to go back to using chan_sip and wait a
year or two for PJSIP to be in the field more.

Have a great day!

Dan
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George Joseph

2014-Dec-15 21:39 UTC

head link

[asterisk-users] PJSIP configuration question

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at amtelco.com>
wrote:>
> Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.
>
>
>
> Same problem is happening with both of them.
>
>
>
> Could this be caused by PJPROJECT 2.3?
>
>
>
> Anyone have any suggestions for what I can try?
>
>
>
> My boss is giving me until tomorrow to get the PJSIP support working with
> Vitelity.  Otherwise, he?s told me to go back to using chan_sip and wait a
> year or two for PJSIP to be in the field more.
>
>I have a Vitelity account I can try.  Re-post your pjsip config and I'll
try it now.


>
>
> Have a great day!
>
>
>
> Dan
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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