Solved Call Forwarding with Snom results in 480

[espritle]

Hi all,
thanks for the great forum here!
I'm right now struggling with CFU, CFB and CFNR with an Snom D725.
When A calls B, B answers with 302 Moved Temporarily to the 3cx v15.
But the 3cx sends out a 480 temporarily unavailable instead of an Invite in direction to C-Party.

Status-Line: SIP/2.0 302 Moved Temporarily
Message Header
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-524287-1---88c35149a0e97e42;rport=5060
From: <sip:[email protected]:5060;nf=e>;tag=79f55100
To: <sip:[email protected]>;tag=yulzvj4jtc
Call-ID: QimlVSk2Jn3u3rl5tqpToQ..
CSeq: 1 INVITE
User-Agent: snom725/8.7.5.8.11
Contact: <sip:[email protected];user=phone>
Diversion: <sip:[email protected]:55016;line=0equtlqe>;reason="unconditional"
Content-Length: 0

Status-Line: SIP/2.0 480 Temporarily Unavailable
Message Header
Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK-3DED-75D
Via: SIP/2.0/UDP 172.28.0.144:5060;branch=z9hG4bKg3Zqkv7ivmy8mqjiqxtlcfx7rsr7wre58
To: "Anruf 0049xxxxxxx"<sip:[email protected]:5060;user=phone>;tag=e9459a5d
From: <sip:[email protected];user=phone>;tag=C37
Call-ID: OA277F98F8498933984378552D9F0C
CSeq: 1780 INVITE
User-Agent: 3CXPhoneSystem 15.0.60903.0 (60903)
Warning: 499 3cxlinux.xxxxxxxxxx.de "Terminated"
Content-Length: 0

Thanks in advance
Klaus

 

[GiannosC_3CX]

Dear espritle,

As far I see on your above logs (SIP/2.0 302 Moved Temporarily), you did a call forwarding from the snom725 device with IP address 192.168.0.36 and extension number 32 to Contact: <sip:[email protected];user=phone>.

User-Agent: snom725/8.7.5.8.11
Contact: <sip:[email protected];user=phone>
Diversion: <sip:[email protected]:55016;line=0equtlqe>;reason="unconditional"

This is a normal log. Also at the second log with 480 Temporarily Unavailable, indicate the problem is front of the 3CX Phone system.
Please make sure that your gateway, voip provider, trunk configuration, max calls or your license etc.

 

[espritle]

Dear Giannos,
thanks a lot for Your reply.
The 3cx is not even trying to send out the Invite, so the gateway and the voip provider is not yet involved in this scenario.
I tried it out and it is possible to make two simultaneous calls. Also conference call with two outside phones is working well.

In the trunk settings, I can't find any relevant setting, which could influence this behavior.
For Reference, I copied the trunk settings below. (X means checked in checkbox, 0 means not checked)
Trunk Details
Enter name for Trunk
Generic SIP Trunk
Registrar/Server/Gateway Hostname or IP
192.168.0.5
5060
Outbound Proxy
192.168.0.5
5060
Number of SIM Calls
10

Authentication
Type of Authentication
Do not require - IP Based
Authentication ID (aka SIP User ID)
Authentication Password

0 3 Way Authentication
Routing of calls to Main Number
Main Trunk No
+49xxxxxxxxxxx
Destination for calls during office hours
33
Destination for calls outside office hours
33

0 Set up Specific Office Hours for this trunk
0 Play holiday prompt when it's a global holiday

===========================================================

Call options
X Allow inbound calls
X Allow outbound calls
X Disallow video calls

Advanced
X PBX Delivers Audio
0 Supports Re-Invite
0 Support Replaces
0 Put Public IP in SIP VIA Header
0 SRTP
Re-Register Timeout
0
Select which IP to use in 'Contact' (SIP) and 'Connection'(SDP) fields
Use Default Settings

Codec Priority
Add codecs Delete Move Up Move Down
G.711 U-law
G.711 A-law
GSM-FR

Thanks a lot
Klaus

 

[GiannosC_3CX]

Hi Klaus,

Please give us more information about your call flow. Also please update your Snom phone to supported min. firmware version.

 

[espritle]

Hi Giannos,
here are more details:
192.168.0.5 is the gateway, 192.168.0.8 is the 3cx
The snom phone works with different pbx in the same scenario ok. What is the minimum firmware version, 3cx needs for the D725?
Do You need the 100 Trying and ACK as well?

No. Time Source Destination Protocol Length Time to live Privacy Info
1 0.000 192.168.0.5 192.168.0.8 SIP/SDP 1193 64 none Request: INVITE sip:[email protected]:5060;user=phone |
2 0.153 192.168.0.8 192.168.0.5 SIP 415 64 Status: 100 Trying |
3 0.204 192.168.0.8 192.168.0.36 SIP/SDP 991 64 Request: INVITE sip:[email protected]:55016;line=0equtlqe |
4 0.224 192.168.0.36 192.168.0.8 SIP 415 64 Status: 100 Trying |
5 0.244 192.168.0.36 192.168.0.8 SIP 494 64 Status: 302 Moved Temporarily |
6 0.296 192.168.0.8 192.168.0.36 SIP 365 64 Request: ACK sip:[email protected]:55016;line=0equtlqe |
7 0.328 192.168.0.8 192.168.0.5 SIP 559 64 Status: 480 Temporarily Unavailable |
8 0.401 192.168.0.5 192.168.0.8 SIP 542 64 Request: ACK sip:[email protected]:5060;user=phone |


Request-Line: INVITE sip:[email protected]:5060;user=phone SIP/2.0
Message Header
Accept: application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Call-ID: OA277F98F8498933984378552D9F0C
Contact: <sip:[email protected];transport=udp;user=phone>
Content-Type: application/sdp
CSeq: 1780 INVITE
From: <sip:[email protected];user=phone>;tag=C37
Max-Forwards: 29
Min-SE: 900
P-Asserted-Identity: <sip:[email protected];user=phone>
Privacy: none
Record-Route: <sip:192.168.0.5;lr>,<sip:172.30.129.21:5064;lr>
Session-Expires: 1800;refresher=uac
Supported: timer
To: "Anruf 00498bbbbbbbb" <sip:[email protected]:5060;user=phone>
Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK-3DED-75D,SIP/2.0/UDP 172.28.0.144:5060;branch=z9hG4bKg3Zqkv7ivmy8mqjiqxtlcfx7rsr7wre58
Content-Length: 247
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): BroadWorks 81239 1 IN IP4 192.168.0.5
Session Name (s): -
Connection Information (c): IN IP4 192.168.0.5
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 11474 RTP/AVP 8 0 101
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): ptime:20
Media Attribute (a): bsoft: 1 image udptl t
===========================================================================================

Request-Line: INVITE sip:[email protected]:55016;line=0equtlqe SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-524287-1---88c35149a0e97e42;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: <sip:[email protected]:5060;nf=e>;tag=79f55100
Call-ID: QimlVSk2Jn3u3rl5tqpToQ..
CSeq: 1 INVITE
Alert-Info: <http://www.notused.invalidtld>;info=external
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 15.0.60903.0 (60903)
Content-Length: 333
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 3cxPS 12956586217570304 29545769220440065 IN IP4 192.168.0.8
Session Name (s): 3cxPS Audio call
Connection Information (c): IN IP4 192.168.0.8
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 7186 RTP/AVP 0 8 9 3 18 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:9 G722/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): sendre
===========================================================================================

Status-Line: SIP/2.0 302 Moved Temporarily
Message Header
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-524287-1---88c35149a0e97e42;rport=5060
From: <sip:[email protected]:5060;nf=e>;tag=79f55100
To: <sip:[email protected]>;tag=yulzvj4jtc
Call-ID: QimlVSk2Jn3u3rl5tqpToQ..
CSeq: 1 INVITE
User-Agent: snom725/8.7.5.8.11
Contact: <sip:[email protected];user=phone>
Diversion: <sip:[email protected]:55016;line=0equtlqe>;reason="unconditional"
Content-Length: 0
===========================================================================================

Status-Line: SIP/2.0 480 Temporarily Unavailable
Message Header
Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK-3DED-75D
Via: SIP/2.0/UDP 172.28.0.144:5060;branch=z9hG4bKg3Zqkv7ivmy8mqjiqxtlcfx7rsr7wre58
To: "Anruf 00498bbbbbbbb"<sip:[email protected]:5060;user=phone>;tag=e9459a5d
From: <sip:[email protected];user=phone>;tag=C37
Call-ID: OA277F98F8498933984378552D9F0C
CSeq: 1780 INVITE
User-Agent: 3CXPhoneSystem 15.0.60903.0 (60903)
Warning: 499 3cxlinux.xxxxxxxxxxx.de "Terminated"
Content-Length: 0


Thanks
Klaus

 

[GiannosC_3CX]

Dear Klaus,

I have sent you a p.m.

 

[GiannosC_3CX]

The issue was with outbound rules not matching and it is now solved.