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Hot Topic (More than 10 Replies) call get stuck (Read 12458 times)
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Re: call get stuck
Reply #15 - 02. Aug 2021 at 13:24
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That queue name is part of the SIP message (INVITE)? Can you send me an example if such INVITE message that contains the queue name?
  
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Re: call get stuck
Reply #16 - 03. Aug 2021 at 09:29
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-------------------------------------------
07:24:28,672: R: myserverip:5060 (UDP)
INVITE sip:199@mypublicip:56387 SIP/2.0
Via: SIP/2.0/UDP myserverip:5060;rport;branch=z9hG4bKPja58f6905-1bd9-4b63-a51c-7e9022eb50c2
From: "Test1999" <sip:1999@myserverip>;tag=7bd740f5-8ee6-45c5-8a31-263d1888fd84
To: <sip:199@mypublicip>
Contact: <sip:asterisk@myserverip:5060>
Call-ID: 676784a3-b80d-402d-a9e2-11988e6b16ed
CSeq: 30683 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Test1999" <sip:1999@myserverip>
Max-Forwards: 70
User-Agent: FPBX-15.0.17.43(16.19.0)
Content-Type: application/sdp
Content-Length:   343

v=0
o=- 1745614970 1745614970 IN IP4 myserverip
s=Asterisk
c=IN IP4 myserverip
t=0 0
m=audio 14194 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

07:24:28,672: Connect Indication: 45 00 01 00 02 82 1C 00 01 01 00 00 01 00 10 80 31 39 39 40 33 31 2E 33 2E 38 39 2E 32 35 30 15 00 80 31 39 39 39 40 38 33 2E 31 33 36 2E 32 35 32 2E 31 34 37 00 09 80 54 65 73 74 31 39 39 39 00 00 00 00 00 
07:24:28,672: Connect Indication
07:24:28,672:  CIP: 1 (speech)
07:24:28,672:  CalledPartyNumber: 199@mypublicip
07:24:28,672:  CallingPartyNumber: 1999@myserverip
07:24:28,672:  CallingPartySubaddress: Test1999
07:24:28,702: Alert Request: 12 00 01 00 01 80 08 00 01 01 00 00 05 00 00 00 00 00 
07:24:28,702: Alert Request
07:24:28,716: WaveOut thread created: 0x0778
07:24:28,716: Loading wave file succeeded: C:\Users\User\Desktop\phonerlite 2.94\RingIn.wav
07:24:28,716: Open render sound device: Default
07:24:28,716: WaveIn thread created: 0x0780
07:24:28,743: render sound device opened
-------------------------------------------
07:24:28,672: T: myserverip:5060 (UDP)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP myserverip:5060;rport=5060;branch=z9hG4bKPja58f6905-1bd9-4b63-a51c-7e9022eb50c2
From: "Test1999" <sip:1999@myserverip>;tag=7bd740f5-8ee6-45c5-8a31-263d1888fd84
To: <sip:199@mypublicip>
Call-ID: 676784a3-b80d-402d-a9e2-11988e6b16ed
CSeq: 30683 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: PhonerLite 2.94beta
Content-Length: 0


07:24:28,743: Alert Confirm: 0E 00 01 00 01 81 08 00 01 01 00 00 00 00 
07:24:28,743: Alert Confirm0x0000
-------------------------------------------
07:24:28,702: T: myserverip:5060 (UDP)
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP myserverip:5060;rport=5060;branch=z9hG4bKPja58f6905-1bd9-4b63-a51c-7e9022eb50c2
From: "Test1999" <sip:1999@myserverip>;tag=7bd740f5-8ee6-45c5-8a31-263d1888fd84
To: <sip:199@mypublicip>;tag=001e698199f2eb1197f52eb1b4b66b0d
Call-ID: 676784a3-b80d-402d-a9e2-11988e6b16ed
CSeq: 30683 INVITE
Contact: <sip:199@10.10.61.1:5060;gr=80003D7D-99F2-EB11-97EF-2EB1B4B66B0D>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Supported: 100rel, replaces, from-change, gruu
Server: PhonerLite 2.94beta
Content-Length: 0


07:24:29,932: Close render sound device: Default
07:24:29,936: render sound device closed
07:24:31,018: Loading wave file succeeded: C:\Users\User\Desktop\phonerlite 2.94\RingIn.wav
07:24:31,018: Open render sound device: Default
07:24:31,030: render sound device opened
-------------------------------------------
07:24:31,601: R: myserverip:5060 (UDP)
CANCEL sip:199@mypublicip:56387 SIP/2.0
Via: SIP/2.0/UDP myserverip:5060;rport;branch=z9hG4bKPja58f6905-1bd9-4b63-a51c-7e9022eb50c2
From: "Test1999" <sip:1999@myserverip>;tag=7bd740f5-8ee6-45c5-8a31-263d1888fd84
To: <sip:199@mypublicip>
Call-ID: 676784a3-b80d-402d-a9e2-11988e6b16ed
CSeq: 30683 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: FPBX-15.0.17.43(16.19.0)
Content-Length:  0


07:24:31,601: Disconnect Indicati
  

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noqueuename.png
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Re: call get stuck
Reply #17 - 03. Aug 2021 at 13:12
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Tanks! I uploaded a new beta version. Hopefully that works again.
  
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Re: call get stuck
Reply #18 - 03. Aug 2021 at 13:49
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great it works now, hope will not have issue with call getting stuck. Thanks
  
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Re: call get stuck
Reply #19 - 04. Aug 2021 at 10:39
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it happened more often now, not sure what is the problem its getting worse and worse. i could not hear the customer but customer can hear me as soon as i answered the call. check attachment
  

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Re: call get stuck
Reply #20 - 05. Aug 2021 at 11:29
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Does the application response to any input? If you change the sound device from "Default" to an explicit device - does that change anything? Do you see on the statistics page that you receive data?
  
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Re: call get stuck
Reply #21 - 06. Jan 2023 at 13:46
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hi, i started to use phonerlite again after 1 year but still having same issue, since than i have change asterisk version many things but same issue i event can not click disconnect button it doesnt work so i have to close application and open again, i have never had issue like this with other softphone. This is what i found on the debug file the moment that i couldn't hear the customer and this is happening everyday one or two times a day with different computer :

03:19:05,730: Close capture sound device: Default
03:19:05,738: capture sound device closed
  

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Re: call get stuck
Reply #22 - 09. Jan 2023 at 10:44
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Are you using the lastest version of PhonerLite? At the moment version 3.13 is actual.
  
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Re: call get stuck
Reply #23 - 09. Jan 2023 at 21:35
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Yes i am using version 3.12
  
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Re: call get stuck
Reply #24 - 10. Jan 2023 at 08:18
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Please update to version 3.13 and check again.
  
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