Incoming Call is not receiving with SIPJS
365 views
Skip to first unread message
Anurag Mishra
Aug 21, 2015, 4:49:24 AM8/21/15
to 2600hz-dev
Hello All,
I am trying to integrating SIPJS with FreeSWITCH and got success to registered the device using SIPJS and also can make a call successfully but, when I tried to make a call from softphone to SIPJS, on that point I can't receive call from FreeSWITCH to SIPJS. Can anyone has idea regarding this, whats the problem behind this? I am not using secured web socket server, I am using ws://mydomain.com:5066.
I have also raised my point in SIPJS mailing list and they said the issue is with my SIP server.
So, can anyone help me in this regard, help will be appreciated.
Thanks,
Anurag Mishra
Darren Schreiber
Aug 21, 2015, 4:53:26 AM8/21/15
to 2600h...@googlegroups.com
Have you checked out our Websockets library/client?
Sent from the 2600hz mobile network
--
You received this message because you are subscribed to the Google Groups "2600hz-dev" group.
To unsubscribe from this group and stop receiving emails from it, send an email to 2600hz-dev+...@googlegroups.com.
For more options, visit https://groups.google.com/d/optout.
You received this message because you are subscribed to the Google Groups "2600hz-dev" group.
To unsubscribe from this group and stop receiving emails from it, send an email to 2600hz-dev+...@googlegroups.com.
For more options, visit https://groups.google.com/d/optout.
Anurag Mishra
Aug 21, 2015, 8:28:43 AM8/21/15
to 2600hz-dev, dschr...@2600hz.com
Hello Darren,
Thanks for your quick reply.
Yes, I was trying webphone and I got register the phone but, when I
tried to make a call then it doesn't send any request to FreeSWITCH.
The logs of console of Web is below:
Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event progress
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event accepted
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event rejected
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event failed
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event cancel
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event connecting
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event terminated
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event dtmf
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event invite
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | skipping event cancel - Event exists
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event refer
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event bye
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event hold
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event unhold
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event muted
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.clientcontext | adding event unmuted
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event userMediaRequest
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event userMedia
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event userMediaFailed
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event userMediaRequest
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event userMedia
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event userMediaFailed
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event iceGathering
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event iceCandidate
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event iceComplete
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event iceFailed
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event getDescription
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event setDescription
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event dataChannel
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | adding event addStream
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | new listener added to event userMediaRequest
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | new listener added to event userMedia
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | new listener added to event userMediaFailed
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | acquiring local media
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event progress
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event accepted
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event rejected
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event failed
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event connecting
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event cancel
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event refer
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event dtmf
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event muted
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event unmuted
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.inviteclientcontext | new listener added to event bye
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | emitting event userMediaRequest
sip.js:2680 Wed Aug 19 2015 00:11:33 GMT+0530 (IST) | sip.invitecontext.mediahandler | emitting event userMediaRequest
sip.js:2680 Wed Aug 19 2015 00:11:58 GMT+0530 (IST) | sip.transport | received WebSocket text message:
OPTIONS sip:l23v...@e19spsu5a138.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 196.123.**.**:5066;branch=z9hG4bKHveXZXaHvX81D
Route: <sip:l23v...@122.173.192.96:18465>;transport=ws
Max-Forwards: 70
From: <sip:mod_...@196.123.**.**:11000>;tag=96BFBNv7QDy8a
To: <sip:vi...@webdev.realm>
Call-ID: ce54dd4a-45d8-11e5-b42d-2bf86091ce42_eq79htjuk2io28g4piksom
CSeq: 79613859 OPTIONS
Contact: <sip:mod_...@196.123.**.**:11000>
User-Agent: 2600hz
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
sip.js:2680 Wed Aug 19 2015 00:11:58 GMT+0530 (IST) | sip.ua | emitting event newTransaction
sip.js:2680 Wed Aug 19 2015 00:11:58 GMT+0530 (IST) | sip.transaction.nist | adding event stateChanged
sip.js:2680 Wed Aug 19 2015 00:11:58 GMT+0530 (IST) | sip.transaction.nist | emitting event stateChanged
sip.js:2680 Wed Aug 19 2015 00:11:58 GMT+0530 (IST) | sip.transport | sending WebSocket message:
SIP/2.0 200 OK
Via: SIP/2.0/WS 196.123.**.**:5066;branch=z9hG4bKHveXZXaHvX81D
To: <sip:vi...@webdev.realm>;tag=t6volbmmgo
From: <sip:mod_...@196.123.**.**:11000>;tag=96BFBNv7QDy8a
Call-ID: ce54dd4a-45d8-11e5-b42d-2bf86091ce42_eq79htjuk2io28g4piksom
CSeq: 79613859 OPTIONS
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE
Accept: application/sdp,application/dtmf-relay
Supported: 100rel,outbound
Content-Length: 0
sip.js:2680 Wed Aug 19 2015 00:11:59 GMT+0530 (IST) | sip.transaction.nist | Timer J expired for non-INVITE server transaction z9hG4bKHveXZXaHvX81D
Thanks,
Anurag Mishra
James Aimonetti
Aug 21, 2015, 1:02:50 PM8/21/15
to 2600h...@googlegroups.com
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256
Curious why you're not going through Kamailio. I was under the
impression Kamailio was the entry point for webrtc calls?
Not to say FreeSWITCH can't do it, but the point of Kazoo was to use
Kamailio to distribute the calls among the available FreeSWITCH
servers. Seems a bit of wasted effort if you get calls working through
a FreeSWITCH server and that server goes away...
Le 08/21/2015 12:38 AM, Anurag Mishra a écrit :
> Hello All,
>
> I am trying to integrating SIPJS with FreeSWITCH and got success to
> registered the device using SIPJS and also can make a call
> successfully but, when I tried to make a call from softphone to
> SIPJS, on that point I can't receive call from FreeSWITCH to SIPJS.
> Can anyone has idea regarding this, whats the problem behind this?
> I am not using secured web socket server, I am using
> *ws://mydomain.com:5066*.
James Aimonetti
Lead Systems Architect / Impressionable Scallywag
"I thought I fixed that"
2600Hz | http://2600hz.com
sip:ja...@2600hz.com
tel:415.886.7905
irc:mc_ @ freenode
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2
Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
iQEcBAEBCAAGBQJV11miAAoJENTKa+JPXCVgNK4H+gIA2oqFPcY5Wc8vqqVrBsPw
j18cCgCs0rVTpg9Aqs3qFOeyDTw2DWdtRegs4DFjJuAihA2BAySLwE20u8Wx6vE2
pTdUi4NIIdbHiP4pejPAuMFedl0yjY9TQNLzR4Oos1QwDiA1BheG4ewxvOhRZf80
P+ozrXeu7CqTQfSOj8ryh+6Pa/a35pbd+/j8qxqogbZLut2K6LJtPceSevwALgzD
bkKaJGUzGVpEdyjDRuvzYcIpCrjExfYJMyriTdLBtG4euviZEKBTlQmDtSs0mLAo
Etf2KRHcLzA+8o4JymAoTr69UQVXGxJRmcXmYxEd9IxtSG3+1Zn828VODBSk4q4=
=cfn8
-----END PGP SIGNATURE-----
Hash: SHA256
Curious why you're not going through Kamailio. I was under the
impression Kamailio was the entry point for webrtc calls?
Not to say FreeSWITCH can't do it, but the point of Kazoo was to use
Kamailio to distribute the calls among the available FreeSWITCH
servers. Seems a bit of wasted effort if you get calls working through
a FreeSWITCH server and that server goes away...
Le 08/21/2015 12:38 AM, Anurag Mishra a écrit :
> Hello All,
>
> I am trying to integrating SIPJS with FreeSWITCH and got success to
> registered the device using SIPJS and also can make a call
> successfully but, when I tried to make a call from softphone to
> SIPJS, on that point I can't receive call from FreeSWITCH to SIPJS.
> Can anyone has idea regarding this, whats the problem behind this?
> I am not using secured web socket server, I am using
>
> I have also raised my point in SIPJS mailing list and they said the
> issue is with my SIP server.
>
> So, can anyone help me in this regard, help will be appreciated.
>
> Thanks, Anurag Mishra
>
- --
> I have also raised my point in SIPJS mailing list and they said the
> issue is with my SIP server.
>
> So, can anyone help me in this regard, help will be appreciated.
>
> Thanks, Anurag Mishra
>
James Aimonetti
Lead Systems Architect / Impressionable Scallywag
"I thought I fixed that"
2600Hz | http://2600hz.com
sip:ja...@2600hz.com
tel:415.886.7905
irc:mc_ @ freenode
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2
Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
iQEcBAEBCAAGBQJV11miAAoJENTKa+JPXCVgNK4H+gIA2oqFPcY5Wc8vqqVrBsPw
j18cCgCs0rVTpg9Aqs3qFOeyDTw2DWdtRegs4DFjJuAihA2BAySLwE20u8Wx6vE2
pTdUi4NIIdbHiP4pejPAuMFedl0yjY9TQNLzR4Oos1QwDiA1BheG4ewxvOhRZf80
P+ozrXeu7CqTQfSOj8ryh+6Pa/a35pbd+/j8qxqogbZLut2K6LJtPceSevwALgzD
bkKaJGUzGVpEdyjDRuvzYcIpCrjExfYJMyriTdLBtG4euviZEKBTlQmDtSs0mLAo
Etf2KRHcLzA+8o4JymAoTr69UQVXGxJRmcXmYxEd9IxtSG3+1Zn828VODBSk4q4=
=cfn8
-----END PGP SIGNATURE-----
Anurag Mishra
Aug 22, 2015, 2:34:22 PM8/22/15
to 2600h...@googlegroups.com
Hi James,
Actually, I have used SIPJS with kamailio and its working fine there is no issue, just I am configuring it with FreeSWITCH for testing. If you have any other suggestion, please share with me.
Thanks,
Anurag Mishra
Actually, I have used SIPJS with kamailio and its working fine there is no issue, just I am configuring it with FreeSWITCH for testing. If you have any other suggestion, please share with me.
Thanks,
Anurag Mishra
Reply all
Reply to author
Forward
0 new messages