Jump to content

Makecall Javascipt function vs. Call forward no answer


Recommended Posts

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.




Link to comment
Share on other sites

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.

Link to comment
Share on other sites

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@;transport=tcp;line=ya734foz SIP/2.0
Via: SIP/2.0/TCP;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@;transport=tcp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
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:.

Link to comment
Share on other sites

  • 5 months later...

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

  • Create New...