Hangup after call end not working on polycom phones

After I dial a key that calls 31817 the phone doesn’t hangup.

Here is my call stack with sip debug on . The line just holds open anyone have any ideas> is this something in polycom firmware?

-- Executing [31817@stations:1] NoOp("SIP/Daniel-Siemens-00000123", "loginprocess") in new stack
-- Executing [31817@stations:2] NoOp("SIP/Daniel-Siemens-00000123", "Get where the agents need to login to") in new stack
-- Executing [31817@stations:3] DumpChan("SIP/Daniel-Siemens-00000123", "") in new stack

Dumping Info For Channel: SIP/Daniel-Siemens-00000123:

Info:
Name= SIP/Daniel-Siemens-00000123
Type= SIP
UniqueID= 1559315609.4294
LinkedID= 1559315609.4294
CallerIDNum= 9xxxxxxxx
CallerIDName= Daniel Siemens
ConnectedLineIDNum= (N/A)
ConnectedLineIDName=(N/A)
DNIDDigits= 31817
RDNIS= (N/A)
Parkinglot= default
Language= en
State= Ring (4)
Rings= 0
NativeFormat= (g729)
WriteFormat= g729
ReadFormat= g729
RawWriteFormat= g729
RawReadFormat= g729
WriteTranscode= No
ReadTranscode= No
1stFileDescriptor= 43
Framesin= 0
Framesout= 0
TimetoHangup= 0
ElapsedTime= 0h0m0s
BridgeID= (Not bridged)
Context= stations
Extension= 31817
Priority= 3
CallGroup= 1
PickupGroup= 1
Application= DumpChan
Data= (Empty)
Blocking_in= (Not Blocking)

Variables:
SIPCALLID=b3a04b0f-89bbc0dc-a28f0ea5@10.119.220.166
SIPDOMAIN=multiservice.com
SIPURI=sip:Daniel-Siemens@10.119.220.166

-- Executing [31817@stations:4] Set("SIP/Daniel-Siemens-00000123", "MemberChanType=SIP") in new stack
-- Executing [31817@stations:5] Set("SIP/Daniel-Siemens-00000123", "MemberChannel=Daniel-Siemens") in new stack
-- Executing [31817@stations:6] UnpauseQueueMember("SIP/Daniel-Siemens-00000123", ",SIP/Daniel-Siemens") in new stack
-- Executing [31817@stations:7] NoOp("SIP/Daniel-Siemens-00000123", "UNPAUSED") in new stack
-- Executing [31817@stations:8] GotoIf("SIP/Daniel-Siemens-00000123", "UNPAUSED=UNPAUSED?good:bad") in new stack
-- Goto (stations,31817,9)
-- Executing [31817@stations:9] NoOp("SIP/Daniel-Siemens-00000123", "Good Test") in new stack
-- Executing [31817@stations:10] Playback("SIP/Daniel-Siemens-00000123", "agent/AgentLoggedIn") in new stack

Audio is at 17308
Adding codec g729 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 10.119.220.166:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.119.220.166;branch=z9hG4bK114864ba925CED2B;received=10.119.220.166
From: “ds-newstatck” sip:Daniel-Siemens@multiservice.com;tag=DA0CDAB3-7C8B87B0
To: sip:31817@multiservice.com;user=phone;tag=as64f8b1d6
Call-ID: b3a04b0f-89bbc0dc-a28f0ea5@10.119.220.166
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:31817@172.25.220.22:5060
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 736531786 736531786 IN IP4 172.25.220.22
s=Asterisk PBX 16.2.1
c=IN IP4 172.25.220.22
t=0 0
m=audio 17308 RTP/AVP 18 0 8 127
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=maxptime:150
a=sendrecv

<------------>

<— SIP read from UDP:10.119.220.166:5060 —>
ACK sip:31817@172.25.220.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.119.220.166;branch=z9hG4bKd8ac3086E3B49D07
From: “ds-newstatck” sip:Daniel-Siemens@multiservice.com;tag=DA0CDAB3-7C8B87B0
To: sip:31817@multiservice.com;user=phone;tag=as64f8b1d6
CSeq: 2 ACK
Call-ID: b3a04b0f-89bbc0dc-a28f0ea5@10.119.220.166
Contact: sip:Daniel-Siemens@10.119.220.166
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: Polycom/5.4.4.7720 PolycomVVX-VVX_310-UA/5.4.4.7720
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —
> 0x7fde880358e0 – Strict RTP switching to RTP target address 10.119.220.166:10052 as source
– <SIP/Daniel-Siemens-00000123> Playing ‘agent/AgentLoggedIn.slin’ (language ‘en’)
– Executing [31817@stations:11] Hangup(“SIP/Daniel-Siemens-00000123”, “”) in new stack
== Spawn extension (stations, 31817, 11) exited non-zero on ‘SIP/Daniel-Siemens-00000123’
Scheduling destruction of SIP dialog ‘b3a04b0f-89bbc0dc-a28f0ea5@10.119.220.166’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:Daniel-Siemens@10.119.220.166 for address/port to send to
set_destination: set destination to 10.119.220.166:5060
Reliably Transmitting (no NAT) to 10.119.220.166:5060:
BYE sip:Daniel-Siemens@10.119.220.166 SIP/2.0
Via: SIP/2.0/UDP 172.25.220.22:5060;branch=z9hG4bK2e885632
Max-Forwards: 70
From: sip:31817@multiservice.com;user=phone;tag=as64f8b1d6
To: “ds-newstatck” sip:Daniel-Siemens@multiservice.com;tag=DA0CDAB3-7C8B87B0
Call-ID: b3a04b0f-89bbc0dc-a28f0ea5@10.119.220.166
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.2.1
Proxy-Authorization: Digest username=“Daniel-Siemens”, realm=“asterisk”, algorithm=MD5, uri=“sip:multiservice.com”, nonce=“1c7e87d2”, response=“14b4b53500c071e882f7cf67b0967eec”
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:10.119.220.166:5060 —>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 172.25.220.22:5060;branch=z9hG4bK2e885632
From: sip:31817@multiservice.com;user=phone;tag=as64f8b1d6
To: “ds-newstatck” sip:Daniel-Siemens@multiservice.com;tag=DA0CDAB3-7C8B87B0
CSeq: 102 BYE
Call-ID: b3a04b0f-89bbc0dc-a28f0ea5@10.119.220.166
User-Agent: Polycom/5.4.4.7720 PolycomVVX-VVX_310-UA/5.4.4.7720
Accept-Language: en
Content-Length: 0

The phone has been told to hangup and the tags and call-ID all seem to match, so I think you need to look at this from the phone side, or look for any middleman that is manipulating the SIP requests.

I looked at adding a button. On Polycoms looks like I have to have Chu on the button config.

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