Re: [asterisk-users] Asterisk with PJSIP

2014-09-10 Thread エムディーシー太郎 
Thank you for your reply.

After setting pjsip set logger on,
the following message is displayed.

It seems that the 9002(SIP client) refuse INVITE message.
Are SIP methods too many?

Thanks,
MMEEGGAA


--- Transmitting SIP request (449 bytes) to UDP:192.168.177.180:16060 ---
OPTIONS sip:9001@192.168.177.180:16060 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1
From: sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df@192.168.177.190
;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: sip:9001@192.168.177.180
Contact: sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df@192.168.177.190:5060
Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c
CSeq: 44261 OPTIONS
Content-Length:  0


--- Received SIP response (333 bytes) from UDP:192.168.177.180:16060 ---
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1
From: sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df@192.168.177.190
;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: sip:9001@192.168.177.180;tag=EF1my
Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c
CSeq: 44261 OPTIONS


--- Transmitting SIP request (443 bytes) to UDP:192.168.177.191:5060 ---
OPTIONS sip:9002@192.168.177.191 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933
From: sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f@192.168.177.190
;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: sip:9002@192.168.177.191
Contact: sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f@192.168.177.190:5060
Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76
CSeq: 16803 OPTIONS
Content-Length:  0


--- Received SIP response (333 bytes) from UDP:192.168.177.191:5060 ---
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933
From: sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f@192.168.177.190
;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: sip:9002@192.168.177.191;tag=hSl7b
Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76
CSeq: 16803 OPTIONS


--- Received SIP request (1170 bytes) from UDP:192.168.177.180:16060 ---
INVITE sip:9002@192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
From: sip:9001@192.168.177.190;tag=~yIpJRFo9
To: sip:9002@192.168.177.190
CSeq: 20 INVITE
Call-ID: 2c1KLd1INo
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 633
User-Agent: Linphone/3.6.99 (belle-sip/1.2.4)
Contact: sip:9001@192.168.177.180:16060
;+sip.instance=urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2

v=0
o=9001 2189 3894 IN IP4 192.168.177.180
s=Talk
c=IN IP4 192.168.177.180
t=0 0
a=ice-pwd:3081373f4003
a=ice-ufrag:4c02
m=audio 17590 RTP/AVP 0 110 3 8 101
c=IN IP4 61.117.138.218
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:21548
a=candidate:1 1 UDP 2130706431 192.168.177.180 7078 typ host
a=candidate:1 2 UDP 2130706430 192.168.177.180 7079 typ host
a=candidate:2 1 UDP 1694498815 XXX.XXX.XXX.XXX 17590 typ srflx raddr
192.168.177.180 rport 7078
a=candidate:2 2 UDP 1694498814 XXX.XXX.XXX.XXX 21548 typ srflx raddr
192.168.177.180 rport 7079

--- Transmitting SIP response (431 bytes) to UDP:192.168.177.180:16060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.177.180:16060
;rport;received=192.168.177.180;branch=z9hG4bK.G4FYrVaiY
Call-ID: 2c1KLd1INo
From: sip:9001@192.168.177.190;tag=~yIpJRFo9
To: sip:9002@192.168.177.190;tag=z9hG4bK.G4FYrVaiY
CSeq: 20 INVITE
WWW-Authenticate: Digest
 
realm=asterisk,nonce=1410336707/cd97e01134333d7d5769e49872f750a4,opaque=58e109d10f49a371,algorithm=md5,qop=auth
Content-Length:  0


--- Received SIP request (373 bytes) from UDP:192.168.177.180:16060 ---
ACK sip:9002@192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
Call-ID: 2c1KLd1INo
From: sip:9001@192.168.177.190;tag=~yIpJRFo9
To: sip:9002@192.168.177.190;tag=z9hG4bK.G4FYrVaiY
Contact: sip:9001@192.168.177.180:16060
;+sip.instance=urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2
Max-Forwards: 70
CSeq: 20 ACK


--- Received SIP request (1428 bytes) from UDP:192.168.177.180:16060 ---
INVITE sip:9002@192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.00879CXMH;rport
From: sip:9001@192.168.177.190;tag=~yIpJRFo9
To: sip:9002@192.168.177.190
CSeq: 21 INVITE
Call-ID: 2c1KLd1INo
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 633
User-Agent: Linphone/3.6.99 (belle-sip/1.2.4)
Contact: sip:9001@192.168.177.180:16060
;+sip.instance=urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2
Authorization:  Digest realm=asterisk,
nonce=1410336707/cd97e01134333d7d5769e49872f750a4,
opaque=58e109d10f49a371, username=9001,  uri=sip:9002@192.168.177.190,
response=a822c66fe1c1d30492beeb08e6daaae5, cnonce=faaa92d5,
nc=0001, qop=auth

v=0

[asterisk-users] WebRTC meeting Norfolk, 15 October 2014

2014-09-10 Thread Daniel Pocock 



I'll be in Norfolk, VA for xTupleCon in October

On 15 October, there will be two events for WebRTC:


14:15 a talk about the xTuple WebRTC extension at xTupleCon
  - must register for xTupleCon to attend this

17:30 a technical / developer workshop at xTuple's offices
  - free, anybody welcome, even if not attending xTupleCon,
 RSVP through Eventbrite[1]


Please see my blog[2] for more comments about all of this and feel free
to email me in advance if you have questions about it or if you may like
to meet up there.



1.
http://www.eventbrite.com/e/browser-based-webrtc-telephony-for-web-apps-workshop-tickets-13002257101

2. http://danielpocock.com/xtuplecon-webrtc-talk-schedule-change

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[asterisk-users] Ast to Ast TLS trunk

2014-09-10 Thread Rizwan H Qureshi 
Hi Everyone,
How can I create a TLS based sip trunk between two asterisk servers. I have
been trying to do it but i dont know how to defined the client certificate
on the asterisk server. Has anyone tried this?

-- 
Best Ragards
Rizwan H Qureshi

V: +971 (0) 528272154
linkedin.com/in/rhqureshi
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[asterisk-users] SIP 380 Alternative Service with PJSIP

2014-09-10 Thread CDR 
I need to respond with 380 Alternative Service. Is there a way to do
this in PJSIP? Please note that I am not picking up the call. For
instance, the Transfer app closes the call if you did not answer it
first. There is a bug open about this. I want to stay with PJSIP, for
I found that it scales painlessly to 1000+ calls, basically, I have
not found an upper limit yet.
Thanks for your help.

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