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Makecall Javascipt function vs. Call forward no answer


gifti
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Following issue with snomONE 5.1.1:

 

- Call forward no answer (10sek) enabled for extension xy to number 123

- starting a new call over the website using javascript function "makecall" from extension xy to number 456

- after short ringing the connection to 456 is established for 10 seconds

- "SIP/2.0 302 Moved Temporarily" SIP Request to 123

- call to 456 is abandoned

- dood dood dood ... :wacko:

  snomONE                        snom370

1 : |T-5060-------INVITE------1752->|
2 : |<-5060-100 Trying/INVITE-1752-T|
3 : |<-5060-180 Ringing/INVIT-1752-T|
4 : |T-5060------PRACK-1752-------->|
5 : |<-52266-STUN_BIND_REQUE-49196-U|
6 : |<-52266--audio/PCMU(0)--49196-U|
7 : |<-5060----200 Ok/PRACK---1752-T|
8 : |<-5060---200 Ok/INVITE---1752-T|
9 : |T-5060----------ACK------1752->|
10: |U-52266--audio/PCMU(0)--49196->|
11: |T-5060---------INVITE----1752->|
12: |<-5060---200 Ok/INVITE---1752-T|
13: |T-5060-------PRACK-------1752->|
14: |<-5060-----SUBSCRIBE-----1752-T|
15: |T-5060--------SIP/2.0----1752->|
16: |<-5060------REGISTER-----1752-T|
17: |T-5060--------SIP/2.0----1752->|
18: |<-5060-302 Moved Tempora-1752-T|
19: |T-5060----------ACK------1752->|
20: |<-5060--------BYE--------1752-T|
21: |T-5060--------SIP/2.0----1752->|
22: |<-5060-----SUBSCRIBE-----1752-T|
23: |T-5060--------SIP/2.0----1752->|

When I try the same scenario with "Call forward no answer" disabled then the call is not abandoned.

 

regards

gifti

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There is a problem with the snom phones. The synchronization of the DND and redirection state between the PBX and the phones make the phone send a redirection, although this must not be done on the phone (this is done on the PBX). Yealink, Grandstream, Polycom work this way. There is a trouble ticket open with snom for a couple of moons and AFAIK it will be included in the next firmware release. If you want to test a beta firmware or get the release date, please contact snom for details.

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Its is only when you call "remote" over the website and "call forward no answer xx seconds" is enabled.

What is the difference to a normal "phone initiated" call?

The first INVITE starts from the PBX and shoud not allow refer, doesn't it?

INVITE sip:26@192.168.0.210:1752;transport=tcp;line=ya734foz SIP/2.0
Via: SIP/2.0/TCP 192.168.0.200:5060;branch=z9hG4bK-66360e544144f248b39928874c97d11a;rport
From: <sip:26@pbx.ggizef.lokal>;tag=1845693600
To: <sip:xxxxxxxxxx@pbx.ggizef.lokal>
Call-ID: 1b8dcdd5@pbx
CSeq: 14894 INVITE
Max-Forwards: 70
Contact: <sip:26@192.168.0.200:5060;transport=tcp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.1.1
Call-Info: <sip:26@pbx.ggizef.lokal>;answer-after=0
Content-Type: application/sdp
Content-Length: 324

But I'am not a SIP expert. It isn't a serious error.

Workaround -> disable CFNA before call remote.

I wait for the new release :rolleyes:.

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  • 5 months later...

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