Open Source Sip: How to generate "180 Ringing" with ...

optotronic

10 posts since
Jun 16, 2008

Jun 18, 2008 9:58 AM

How to generate "180 Ringing" with ATLSIP

I'm using OSSPhone to test ATLSIP. It detects incoming calls, but it doesn't send out "180 Ringing" packets so caller doesn't get rings, and proxy eventually routes call to voicemail.

Is there a way to automatically or manually send the "180 Ringing" messages with ATLSIP?

Here is a log from OSSPhone:

----------------26:34:32.350----------------
Querying STUN server at stun01.sipphone.com.
This may take a while ...

----------------26:34:42.772----------------
STUN server "stun01.sipphone.com" replies Cone NAT, external IP 72.240.235.122
----------------26:34:47.820----------------

LISTENER STARTED *** OPAL 127.0.0.1:5060

----------------26:34:47.880----------------

LISTENER STARTED *** OPAL 192.168.0.20:5060 *** DEFAULT LISTENER ***

----------------26:34:47.936----------------

REGISTER sip:proxy01.sipphone.com SIP/2.0
From: recipient ;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:recipient@proxy01.sipphone.com
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport
CSeq: 2 REGISTER
Call-ID: c3ce95d9-d4fb-1810-9482-a33f609baf13@proxy01.sipphone.com
Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp>
User-Agent: OpenSIPStack v1.1.7-24
Expires: 3600
Max-Forwards: 70
Authorization: Digest username="recipient", realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1", uri="sip:proxy01.sipphone.com", response="4a78a8e0712b92b26d17ba786a89a4eb", opaque="", algorithm=MD5
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK
Content-Length: 0

----------------26:34:48.226----------------
<<< SIP/2.0 200 OK SRC: 198.65.166.131:5060:UDP enc=0 bytes=509
SIP/2.0 200 OK
From: recipient ;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:recipient@proxy01.sipphone.com;tag=21a483426c2cd5d9b85bffe6bba40a2e.90ab
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013
CSeq: 2 REGISTER
Call-ID: c3ce95d9-d4fb-1810-9482-a33f609baf13@proxy01.sipphone.com
Contact: <sip:recipient@72.240.235.122:22013;transport=udp>;expires=3600
Content-Length: 0

----------------26:34:58.666----------------
<<< INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=1103
INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" ;tag=as47bd3a4c
To:
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Contact:
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Date: Wed, 18 Jun 2008 13:47:42 GMT
User-Agent: Asterisk PBX
Max-Forwards: 16
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
RemoteIP: 66.54.140.46
Content-Type: application/sdp
Content-Length: 379

v=0
o=root 16325 16325 IN IP4 66.54.140.46
s=session
c=IN IP4 66.54.140.46
t=0 0
m=audio 14868 RTP/AVP 0 8 3 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------------26:34:58.684----------------

SIP/2.0 100 Trying DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=411

SIP/2.0 100 Trying
From: "FLINT MI" ;tag=as47bd3a4c
To:
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Content-Length: 0

----------------26:35:24.111----------------
<<< CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=332
CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" ;tag=as47bd3a4c
To:
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
CSeq: 102 CANCEL
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Content-Length: 0

----------------26:35:24.151----------------

SIP/2.0 200 OK DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=358

SIP/2.0 200 OK
From: "FLINT MI" ;tag=as47bd3a4c
To:
Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131
CSeq: 102 CANCEL
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Server: OpenSIPStack v1.1.7-24
Content-Length: 0

----------------26:35:24.163----------------

SIP/2.0 487 Request Cancelled DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=535

SIP/2.0 487 Request Cancelled
From: "FLINT MI" ;tag=as47bd3a4c
To: ;tag=a1f8ccd9d4fb1810991ba33f609baf13
Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Server: OpenSIPStack v1.1.7-24
Content-Length: 0

----------------26:35:24.263----------------
<<< ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=364
ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" ;tag=as47bd3a4c
To: ;tag=a1f8ccd9d4fb1810991ba33f609baf13
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
CSeq: 102 ACK
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Content-Length: 0

Recipient OSSPhone was built using latest code in CVS on 2008-06-13. I tested by calling in through an ipKall phone number. No rings were heard from the caller phone system. Eventually the call was routed to voicemail.

Finest regards,
Bill Root

Tags: atlsip, 180, ringing, ossphone

joegen

125 posts since
Apr 28, 2007

1. Jun 19, 2008 8:54 PM

in response to: optotronic

Re: How to generate "180 Ringing" with ATLSIP

You are right. I have created a ticket for this

http://www.assembla.com/spaces/opensbc/tickets/24

Unfortunately there is currently no way of doing this in the appliation layer. But it will be there soon enough. Thanks for bringing it up.

Joegen

optotronic wrote:
I'm using OSSPhone to test ATLSIP. It detects incoming calls, but it doesn't send out "180 Ringing" packets so caller doesn't get rings, and proxy eventually routes call to voicemail.

Is there a way to automatically or manually send the "180 Ringing" messages with ATLSIP?

Here is a log from OSSPhone:

LISTENER STARTED *** OPAL 127.0.0.1:5060