Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Zoiper on Windows Phone


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
vitor.mazuco at gmail.com
Guest





PostPosted: Mon Feb 29, 2016 10:41 am    Post subject: [asterisk-users] Zoiper on Windows Phone Reply with quote

Hello everyone, I have some problems to enable push the Zoiper
Windows Phone in my Asterisk 11.


Below is the result of CLI

== Using SIP RTP CoS mark 5
-- Executing [1033@ramais:1] Answer("SIP/1030-00000201", "") in new stack
Quote:
0x7efc90024190 -- Probation passed - setting RTP source
address to 179.XX.XXX.XX:57741
[Feb 29 12:32:28] NOTICE[4348][C-000001ce]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '179.XX.XX.XX:57741'
Quote:
0x7efc90024190 -- Probation passed - setting RTP source
address to 179.XX.XX.XX:57741
-- Executing [1033@ramais:2] Set("SIP/1030-00000201", "location=")
in new stack
-- Executing [1033@ramais:3] Verbose("SIP/1030-00000201", "0,
getting push info ") in new stack
getting push info
-- Executing [1033@ramais:4] Set("SIP/1030-00000201",
"regx="X-PUSH-URI=([0-9a-zA-Z\.\:\/\_]+)"") in new stack
-- Executing [1033@ramais:5] Set("SIP/1030-00000201", "push=") in new stack
-- Executing [1033@ramais:6] System("SIP/1030-00000201",
"/usr/bin/push.sh ") in new stack
-- Executing [1033@ramais:7] Wait("SIP/1030-00000201", "1") in new stack
-- Executing [1033@ramais:8] Dial("SIP/1030-00000201", "SIP/1033")
in new stack
[Feb 29 12:32:29] WARNING[4348][C-000001ce]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/1030-00000201' status is 'CHANUNAVAIL'
asterisk*CLI>

I've created the file more push.sh qualification in the dialplan. But the
Windows Phone can not run on Asterisk.

Does anyone know another method for this?

Thanks in advanced.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services