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Disconnect on hold D710 V8.7.5.28

Hello.

I have read the topic https://helpdesk.snom.com/support/discussions/topics/6000042744, but i still can't solve the problem so my phone has no video, so where "disable video codec offer" in my model.

My problem with the procedure for <hold> and respectively for <Transfer> in brief:

SNOM D710 for testing purpose using with Registrar – Telesis PX24Xr6 (Hibrid VoIP system).

When trying to use the function HOLD the call to be hold, it is not hold but released. In the tracer we may see that after the confirmation <200 ОК>  from the PABX,  the phone sends immediately <BYE>  thus releasing the connection. Apparently in the <200 ОК> of the PBX is missing a protocol element, needed to be received the confirmation from the phone and to be continued the establishment of the next connection.

I have attached the trace from the phone itself - SNOM_D710_BAD_Hold_to_PX24X.txt.  

txt

Best Answer

Dear Alexander,


I'm going to convert this post in a ticket so we can deeply analyse the issue.


Thanks,


Answer

Dear Alexander,


I'm going to convert this post in a ticket so we can deeply analyse the issue.


Thanks,

Dear Pietro,
No, the problem is still present.
I have sent the downloaded .bin file to the customer, they made upgrade of the firmware, but the issue is the same. I'll try to go to the site so to be able to perform some more tests. In addition to mentioned issue the customer says that "in some calls there is one way missing of voice". I suspect that this is the main problem and the "BYE" is a consequence of it.
I will keep you posted about the problem as soon as possible .
Best regards,
A.Kovachev.

 

Dear Aleksandar,


do you managed to solve this problem ?

I have send you via email latest firmware in order to test


Let us know how it works

Thank You for response.
I have tried  "disabling timer_support on the phone" and D710 no more adds Session-Expires: in SIP messages. But the problem is still present.
Furthermore with the settings: Timer Support (RFC4028):on  when D710 send <INVITE> with Session-Expires: 3600 and Telesis response with 180 Ringing and 200 OK both without Session-Expires:  the call is established normally.
So I attached trace from the phone itself without  Session-Expires:   -
txt

Akleksandar,


The reason for the BYE is the result of us setting the Session-Expires: 3600;refresher=uas header making the TELESIS responsible for the session timer. TELESIS ACKs without the Session-Expires header and then sends a BYE.


Does TELESIS support Session Timers? If so can you try enabling it? Alternatively, you can try disabling timer_support on the phone.


http://wiki.snom.com/Settings/timer_support


Sent to udp:192.168.100.62:5060 at Feb 6 15:42:28 (1074 bytes):

INVITE sip:1503@192.168.100.62:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.119:32768;branch=z9hG4bK-05ynk7ukxtsy;rport
From: "1503" <sip:1503@192.168.100.62>;tag=tbk2aafrlb
To: "1017" <sip:1017@192.168.100.62;user=phone>;tag=3aK7020K67E
Call-ID: 313438363338383534313133383332-ar49a6o3sjs7
CSeq: 2 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.28
Contact: <sip:1503@192.168.100.119:32768>;reg-id=1
X-Serialnumber: 0004137E4BA1
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 1457224829 1457224830 IN IP4 192.168.100.119
s=call
c=IN IP4 192.168.100.119
t=0 0
m=audio 62688 RTP/AVP 8 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
________________________________________
Received from udp:192.168.100.62:5060 at Feb 6 15:42:28 (747 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.119:32768;branch=z9hG4bK-05ynk7ukxtsy;received=192.168.100.119;rport=32768
From: "1503" <sip:1503@192.168.100.62>;tag=tbk2aafrlb
To: <sip:1017@192.168.100.62;user=phone>;tag=3aK7020K67E
Call-ID: 313438363338383534313133383332-ar49a6o3sjs7
CSeq: 2 INVITE
Contact: <sip:1503@192.168.100.62:5060>
User-Agent: TELESIS Xymphony PXr6 170119
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, REFER, REGISTER, SUBSCRIBE
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 209

v=0
o=- 3418483668 2 IN IP4 192.168.100.62
s=-
c=IN IP4 192.168.100.62
t=0 0
m=audio 30250 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=recvonly
a=ptime:90
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