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Forums › VoIP › Sipura handsets remote BYE not hanging up handset.


37 posts

Geek


# 183705 27-Oct-2015 12:51
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Hi Guys

I have an issue where SPA9x and SPA5x models are receiving a BYE from my server and responding with an OK.

 

However the handset does not end the call on the handset.

 

 

 

The deployment uses a PBX call manager software which is compatible with the Talk package on these models.

 

Ending the call through the Call manager suite will end the call however the phone stays offhook and plays a disconnected tone for about 10secs then hangs up.

 

 

 

This happens when far end party hangups also so i suspect this is just a hardware feature.

I can't see any way to resolve this in the Admin manual or config.

 

 

 

Device is a SPA922 running the latest software.

 

The PBX is running Freeswitch with Sofia SIP with port 5080 instead of 5060/
Below is a Tcpdump of the call where this issue occurs.

I'm just wondering if anyone else has come across this and found a way to resolve it.

 

 

 

<<<<Call invite sent to handset>>>>

12:39:34.488419 IP [FREESWITCHIP].5080 > [HANDSETIP].5080: UDP, length 1230
E....S..@...dA..
...........INVITE sip:845@[HANDSETIP]:5080 SIP/2.0
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;rport;branch=z9hG4bKNXyS45H3tjt5K
Route: <sip:845@[HANDSETIP]:5080>
Max-Forwards: 69
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
To: <sip:845@[HANDSETIP]:5080>
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615323 INVITE
Contact: <sip:mod_sofia@[FREESWITCHIP]:5080>
Call-Info: <sip:[FREESWITCHIP]>
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130822T231319Z~dbfde499a4
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 255
X-FS-Support: update_display,send_info
P-Asserted-Identity: "TEST-USER" <sip:807@[FREESWITCHIP]>

v=0
o=FreeSWITCH 1445864742 1445864743 IN IP4 [FREESWITCHIP]
s=FreeSWITCH
c=IN IP4 [FREESWITCHIP]
t=0 0
m=audio 38032 RTP/AVP 0 3 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 42892 RTP/AVP 98
a=rtpmap:98 H264/90000

12:39:34.517454 IP [HANDSETIP].5080 > [FREESWITCHIP].5080: UDP, length 304
Eh.L......).
...dA.......8..SIP/2.0 100 Trying
To: <sip:845@[HANDSETIP]:5080>
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615323 INVITE
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;branch=z9hG4bKNXyS45H3tjt5K
Server: Linksys/SPA922-6.1.5(a)
Content-Length: 0

....
12:39:34.533438 IP [HANDSETIP].5080 > [FREESWITCHIP].5080: UDP, length 408
Eh........)q
...dA..........SIP/2.0 180 Ringing
To: <sip:845@[HANDSETIP]:5080>;tag=d3c8b583c5a00d0di0
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615323 INVITE
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;branch=z9hG4bKNXyS45H3tjt5K
Contact: "Test" <sip:845@[HANDSETIP]:5080>
Server: Linksys/SPA922-6.1.5(a)
Content-Length: 0
Allow-Events: hold,talk,conference

<<<<NOTIFY from PBX to answer the call using "Event: Talk">>>>

....
12:39:38.494290 IP [FREESWITCHIP].5080 > [HANDSETIP].5080: UDP, length 771
E....T..@...dA..
...........NOTIFY sip:845@[HANDSETIP]:5080 SIP/2.0
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;rport;branch=z9hG4bKp6Qj6026QUgrF
Max-Forwards: 70
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
To: <sip:845@[HANDSETIP]:5080>
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615324 NOTIFY
Contact: <sip:mod_sofia@[FREESWITCHIP]:5080>
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130822T231319Z~dbfde499a4
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Event: talk
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Subscription-State: active
Content-Length: 0


12:39:38.594592 IP [HANDSETIP].5080 > [FREESWITCHIP].5080: UDP, length 323
Eh._......).
...dA.......K..SIP/2.0 200 OK
To: <sip:845@[HANDSETIP]:5080>;tag=d3c8b583c5a00d0di0
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615324 NOTIFY
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;branch=z9hG4bKp6Qj6026QUgrF
Server: Linksys/SPA922-6.1.5(a)
Content-Length: 0

....
12:39:38.598692 IP [HANDSETIP].5080 > [FREESWITCHIP].5080: UDP, length 746
Eh........(.
...dA........M.SIP/2.0 200 OK
To: <sip:845@[HANDSETIP]:5080>;tag=d3c8b583c5a00d0di0
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615323 INVITE
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;branch=z9hG4bKNXyS45H3tjt5K
Contact: "Test" <sip:845@[HANDSETIP]:5080>
Server: Linksys/SPA922-6.1.5(a)
Content-Length: 226
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Allow-Events: hold,talk,conference
Supported: replaces
Content-Type: application/sdp

v=0
o=- 346677 346677 IN IP4 [HANDSETIP]
s=-
c=IN IP4 [HANDSETIP]
t=0 0
m=audio 42822 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 98
....
12:39:38.602434 IP [FREESWITCHIP].5080 > [HANDSETIP].5080: UDP, length 377
E....U..@..LdA..
..........IACK sip:845@[HANDSETIP]:5080 SIP/2.0
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;rport;branch=z9hG4bKQFHB8UKaN46aB
Max-Forwards: 70
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
To: <sip:845@[HANDSETIP]:5080>;tag=d3c8b583c5a00d0di0
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615323 ACK
Contact: <sip:mod_sofia@[FREESWITCHIP]:5080>
Content-Length: 0



<<<<BYE is sent from PBX>>>>

12:39:41.419623 IP [FREESWITCHIP].5080 > [HANDSETIP].5080: UDP, length 641
E....V..@..CdA..
..........QBYE sip:845@[HANDSETIP]:5080 SIP/2.0
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;rport;branch=z9hG4bKUKpeF8pr97ZNS
Max-Forwards: 70
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
To: <sip:845@[HANDSETIP]:5080>;tag=d3c8b583c5a00d0di0
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615325 BYE
Contact: <sip:mod_sofia@[FREESWITCHIP]:5080>
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130822T231319Z~dbfde499a4
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0


12:39:41.455958 IP [HANDSETIP].5080 > [FREESWITCHIP].5080: UDP, length 320
Eh.\.C....)/
...dA.......H.bSIP/2.0 200 OK
To: <sip:845@[HANDSETIP]:5080>;tag=d3c8b583c5a00d0di0
From: "TEST-USER" <sip:807@[FREESWITCHIP]>;tag=tK15UgDtD8taQ
Call-ID: a7677f28-f6dd-1233-d79b-005056b354af
CSeq: 82615325 BYE
Via: SIP/2.0/UDP [FREESWITCHIP]:5080;branch=z9hG4bKUKpeF8pr97ZNS
Server: Linksys/SPA922-6.1.5(a)
Content-Length: 0

<<<<SIP Call has ended, Phone is still "Offhook" and playing a Disconnected tone>>>>

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cisconz
1222 posts

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  # 1414611 27-Oct-2015 12:58
2 people support this post
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That is by design, same as if you don't hang up a POTS line




Hmmmm

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  # 1414648 27-Oct-2015 13:40
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I'm pretty sure the old 92X and 94X series won't hang up and will stay off hook. The new SPA5xx series phones will hang up.

I love phones with a full EHS such as the Snom's and Digium phones that have no manual hook switch as you then don't even need to hang the handset back up.

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