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Call forwarding not working ( Vodafone Germany)

colcha
Level 1
Level 1

We have a Cisco Model ISR4321/K9 router with Cisco IOS XE Software Version 16.09.04
The problem occurs when we make a call forward to an external number.
The call made by User A to the destination User B (user B has activated a call forwarding to another external number User C)
User C answers the call, user C does not hear anything, User A does not hear anything either, but after 18 seconds the call closes automatically by itself.

User A -->+49xxxxxxx90 ...incoming call 

User B--> +49xxxxxxx55 --> call forward to  User C   

User C--> +49xxxxxxx30 .....Extern number ( external forwarding is configured on the router) 

colcha_0-1675086593820.png

 

 

 

 

 

-------

voice service voip
no notify redirect ip2ip
address-hiding
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw h323

------

voice register global
mode cme
source-address 172.20.XXX.XX port 5060
max-dn 10
auto-register
!
voice register dn 2
number +49xxxxxx55
call-forward b2bua unregistered +49xxxxxx30

-----------

Can someone help me to solve this problem

BR ,

Anibal 

 

3 Accepted Solutions

Accepted Solutions

b.winter
VIP
VIP

First of all: If you have CME and SIP Trunk on the same router, then you must use tenant configuration. Check out the cisco docs and also Cisco forums threads, there are a lot of how to configure tenants on the router.

Second: Without the full config and a debug of the call, nobody will be able to help you. So please provide both.

View solution in original post

colcha
Level 1
Level 1

Hi All , 

Thank you very much for your help, I solved the problem myself.

 

*Vodafone Provider*

Router Cisco ---> 

Router> enable

Router# configure terminal

Router(config)# voice service voip

Router(config-voi-serv)#notify redirect ip2ip

Router(config-voi-serv)#supplementary-service sip refer

Router(config-voi-serv)# supplementary-service sip moved-temporarily

Router(config-voi-serv)# exit

 

BR,

Anibal 

View solution in original post

Hi Roger ,

I have configured the external forwarding in the Cisco Router.

Example :

voice register global
mode cme
source-address 172.20.XXX.XX port 5060
max-dn 10
auto-register
!
voice register dn 2
number +49xxxxxxxx
call-forward b2bua unregistered +49xxxxxxxx

colcha_0-1675172141246.png

It is very true that I can do the forwarding from the PBX, but in my case I do it from the GW

BR, 

Anibal 

View solution in original post

20 Replies 20

b.winter
VIP
VIP

First of all: If you have CME and SIP Trunk on the same router, then you must use tenant configuration. Check out the cisco docs and also Cisco forums threads, there are a lot of how to configure tenants on the router.

Second: Without the full config and a debug of the call, nobody will be able to help you. So please provide both.

Hi winter,

debug of the call   = 

-----

000589: Jan 30 09:10:10.316: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+49xxxxxxx55@stapad.ngn.vodafone.de;user=phone SIP/2.0
Via: SIP/2.0/TCP 2.207.165.116:5060;branch=z9hxxxxx7b10.1
To: <sip:+49xxxxxxx55@hansx001.ngn.vodafone.de;user=phone>
From: <sip:+49xxxxxxx90@telekom.de;user=phone>;tag=SD5r6ec01-c5f923a9
Call-ID: SDxxxxxxx0
CSeq: 1 INVITE
Max-Forwards: 59
Contact: <sip:+49xxxxxxx90@2.207.165.116:5060;transport=tcp>
Date: Mon, 30 Jan 2023 10:10:10 GMT
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: resource-priority,timer,100rel
P-Asserted-Identity: <sip:+49xxxxxx90@telekom.de;user=phone>
P-Asserted-Identity: <tel:+49xxxxxxx90>
Session-Expires: 1810;refresher=uac
Min-SE: 600
Accept: application/sdp
P-Early-Media: supported
Content-Type: application/sdp
Content-Length: 172

v=0
o=- 0 0 IN IP4 2.207.165.116
s=IMSS
c=IN IP4 2.207.165.116
t=0 0
m=audio 29414 RTP/AVP 8 97
b=AS:80
a=rtpmap:97 telephone-event/8000
a=ptime:20
a=maxptime:30

000590: Jan 30 09:10:10.322: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:+49xxxxxx30@2.207.165.116:5060 SIP/2.0
Via: SIP/2.0/UDP 145.253.139.26:5060;branch=z9xxxxx2E
From: <sip:+49xxxxxxx90@145.253.139.26>;tag=12F7B813-1B9D
To: <sip:+49xxxxxxx30@2.207.165.116>
Call-ID: BBxxxxF@145.253.139.26
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 600
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1675069810
Contact: <sip:+49xxxxxxx90@145.253.139.26:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 58
P-Preferred-Identity: <sip:+49xxxxxxx65@145.253.139.26>
Session-Expires: 1810;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 226

v=0
o=- 7795 7427 IN IP4 145.253.139.26
s=SIP Call
c=IN IP4 145.253.139.26
t=0 0
m=audio 10050 RTP/AVP 8 101
c=IN IP4 145.253.139.26
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

000591: Jan 30 09:10:10.323: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 2.207.165.116:5060;branch=z9hG4bK67gvn120b8vhd8fe7b10.1
From: <sip:+49xxxxxxx90@telekom.de;user=phone>;tag=SD5r6ec01-c5f923a9
To: <sip:+49xxxxxxx55@hansx001.ngn.vodafone.de;user=phone>
Call-ID: SDxxxxxxE0
CSeq: 1 INVITE
Allow-Events: telephone-event
Content-Length: 0

000592: Jan 30 09:10:10.342: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 145.253.139.26:5060;branch=z9hG4bK1C81152E
From: <sip:+49xxxxxx90@145.253.139.26>;tag=12F7B813-1B9D
To: <sip:+49xxxxxxx030@2.207.165.116>
Call-ID: BBxxxxxF@145.253.139.26
CSeq: 101 INVITE
Timestamp: 1675069810
Content-Length: 0

000593: Jan 30 09:10:11.118: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 145.253.139.26:5060;branch=z9hG4bK1C81152E
From: <sip:+49xxxxxxx90@145.253.139.26>;tag=12Fxxxxx9D
To: <sip:+49xxxxxx30@2.207.165.116>;tag=SDviupa99-f7xxxxxx0000
Call-ID: BBxxxx2F@145.253.139.26
CSeq: 101 INVITE
Timestamp: 1675069810
Supported: resource-priority,timer
Require: timer
Session-Expires: 1810;refresher=uac
Accept: application/sdp
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Contact: <sip:+49xxxxxxxxx30@2.207.165.116:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 168

v=0
o=- 0 7859801 IN IP4 2.207.165.116
s=IMSS
c=IN IP4 2.207.165.116
t=0 0
m=audio 27718 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20

000594: Jan 30 09:10:11.121: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:+49xxxxxxxx30@2.207.165.116:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 145.253.139.26:5060;branch=z9hxxxxx6
From: <sip:+49xxxxxxx90@145.253.139.26>;tag=12F7B813-1B9D
To: <sip:+49xxxxxxxx30@2.207.165.116>;tag=SDviupa99-fxxxxx0-0000
Call-ID: BBxxxxxx2F@145.253.139.26
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

000595: Jan 30 09:10:11.124: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 2.207.165.116:5060;branch=z9xxxx10.1
From: <sip:+49xxxxxxxx90@telekom.de;user=phone>;tag=SDxxxxxa9
To: <sip:+49xxxxxxx55@hansx001.ngn.vodafone.de;user=phone>;tag=12F7BB33-CCC
Call-ID: SDxxxxx0E0
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
P-Preferred-Identity: <sip:+49xxxxxxxx65@145.253.139.26>
Contact: <sip:+49xxxxxxx30@145.253.139.26:5060;transport=tcp>
Supported: replaces
Supported: sdp-anat
Session-Expires: 1810;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 223

v=0
o=- 1353 4293 IN IP4 145.253.139.26
s=SIP Call
c=IN IP4 145.253.139.26
t=0 0
m=audio 10048 RTP/AVP 8 97
c=IN IP4 145.253.139.26
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20

000596: Jan 30 09:10:11.148: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+49xxxxxxxx30@145.253.139.26:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 2.207.165.116:5060;branch=z9hxxxx0.1
To: <sip:+49xxxxxxxx55@hansx001.ngn.vodafone.de;user=phone>;tag=12xxx
From: <sip:+49xxxxxxxx90@telekom.de;user=phone>;tag=SD5r6ec01-c5f923a9
Call-ID: SDxxxx
CSeq: 1 ACK
Max-Forwards: 59
Content-Length: 0

000597: Jan 30 09:10:11.892: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+49xxxxxxxx90@145.253.139.26:5060 SIP/2.0
Via: SIP/2.0/UDP 2.207.165.116:5060;branch=z9hxxxxj1.1
To: <sip:016xxxxxxxx90@145.253.139.26>;tag=12F7B813-1B9D
From: <sip:+49xxxxxxxxx30@2.207.165.116>;tag=SDxxxx-0000
Call-ID: BBxxxx53.139.26
CSeq: 102 INVITE
Max-Forwards: 69
Contact: <sip:+49xxxxxxxxx30@2.207.165.116:5060;transport=udp>
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: resource-priority,timer
Session-Expires: 1810;refresher=uas
Min-SE: 600
Accept: application/sdp
P-Early-Media: supported
Content-Type: application/sdp
Content-Length: 181

v=0
o=- 0 7859802 IN IP4 2.207.165.116
s=IMSS
c=IN IP4 2.207.165.116
t=0 0
a=X-UPSPEED
m=audio 27718 RTP/AVP 8 106
a=rtpmap:106 telephone-event/8000
a=sendrecv
a=ptime:20

000598: Jan 30 09:10:11.896: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:+49xxxxxxxxx90@2.207.165.116:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 145.253.139.26:5060;branch=z9xxxx
P-Preferred-Identity: <sip:+49xxxxxxx65@145.253.139.26>
From: <sip:+49xxxxxxxx55@hansx001.ngn.vodafone.de;user=phone>;tag=12xxx
To: <sip:+49xxxxxxx90@telekom.de;user=phone>;tag=SD5r6ec01-c5f923a9
Call-ID: SDxxx
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 600
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1675069811
Contact: <sip:+49xxxxxxxx55@145.253.139.26:5060;transport=tcp>
Expires: 180
Allow-Events: telephone-event
Session-Expires: 1810;refresher=uas
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 1353 4293 IN IP4 145.253.139.26
s=SIP Call
c=IN IP4 145.253.139.26
t=0 0
m=audio 10048 RTP/AVP 8 97
c=IN IP4 145.253.139.26
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20

000599: Jan 30 09:10:11.898: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.207.165.116:5060;branch=z9hxxxj1.1
From: <sip:+49xxxxxxxxxxx30@2.207.165.116>;tag=SDxxx
To: <sip:+49xxxxxxxxxx90@145.253.139.26>;tag=12F7B813-1B9D
Call-ID: BBxxx9.26
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

000600: Jan 30 09:10:11.934: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 145.253.139.26:5060;branch=z9xxx
From: <sip:+49xxxxxxxx55@hansx001.ngn.vodafone.de;user=phone>;tag=12xxx
To: <sip:+49xxxxxxxx90@telekom.de;user=phone>;tag=SD5r6ec01-c5f923a9
Call-ID: SDxxx
CSeq: 101 INVITE
Timestamp: 1675069811
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Session-Expires: 1810;refresher=uas
Accept: application/sdp
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Contact: <sip:+49xxxxxxxx90@2.207.165.116:5060;transport=tcp>
Content-Type: application/sdp
Content-Length: 172

v=0
o=- 0 0 IN IP4 2.207.165.116
s=IMSS
c=IN IP4 2.207.165.116
t=0 0
m=audio 29414 RTP/AVP 8 97
b=AS:80
a=rtpmap:97 telephone-event/8000
a=ptime:20
a=maxptime:30

000601: Jan 30 09:10:11.938: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.207.165.116:5060;branch=z9xxx.1
From: <sip:+49xxxxxxxx30@2.207.165.116>;tag=SDviupa99-f79b94f8-0014-0134-0000-0000
To: <sip:+49xxxxxxxx90@145.253.139.26>;tag=12Fxx
Call-ID: BBxxx
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
P-Preferred-Identity: <sip:+49xxxxx65@145.253.139.26>
Contact: <sip:+49xxxxxxxx90@145.253.139.26:5060>
Supported: replaces
Supported: sdp-anat
Session-Expires: 1810;refresher=uas
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 226

v=0
o=- 7795 7428 IN IP4 145.253.139.26
s=SIP Call
c=IN IP4 145.253.139.26
t=0 0
m=audio 10050 RTP/AVP 8 106
c=IN IP4 145.253.139.26
a=rtpmap:8 PCMA/8000
a=rtpmap:106 telephone-event/8000
a=fmtp:106 0-15
a=ptime:20

000602: Jan 30 09:10:11.940: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:+49xxxxxxx90@2.207.165.116:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 145.253.139.26:5060;branch=z9hxxx
From: <sip:+49xxxxxxxxx55@hansx001.ngn.vodafone.de;user=phone>;tag=12xx
To: <sip:+49xxxxxxxxx90@telekom.de;user=phone>;tag=SD5r6ec01-c5f923a9
Call-ID: SDxxx
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

000603: Jan 30 09:10:11.963: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+49xxxxxxxx90@145.253.139.26:5060 SIP/2.0
Via: SIP/2.0/UDP 2.207.165.116:5060;branch=z9hxxx
To: <sip:0xxxxxxxxxx90@145.253.139.26>;tag=12F7B813-1B9D
From: <sip:+49xxxxxxxx30@2.207.165.116>;tag=SDviupa99-fxxx
Call-ID: BBxxx
CSeq: 102 ACK
Max-Forwards: 69
Content-Length: 0

000604: Jan 30 09:10:30.022: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:+49xxxxxxx55@145.253.139.26:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 2.207.165.116:5060;branch=z9hxxx1
To: <sip:+49xxxxxxx55@hansx001.ngn.vodafone.de;user=phone>;tag=12xxx
From: <sip:+49xxxxxxx90@telekom.de;user=phone>;tag=SD5r6ec01-c5f923a9
Call-ID: SDxxx
CSeq: 2 BYE
Max-Forwards: 69
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Reason: Q.850 ;cause=16 ;text="2"
Content-Length: 0

000605: Jan 30 09:10:30.026: //11757/BB4562A0BE13/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 2.207.165.116:5060;branch=z9xxx
From: <sip:+49xxxxxxx90@telekom.de;user=phone>;tag=SDxxx
To: <sip:+49xxxxxxxxx55@hansx001.ngn.vodafone.de;user=phone>;tag=12F7BB33-CCC
Call-ID: SD5xxx
CSeq: 2 BYE
Reason: Q.850;cause=16
Content-Length: 0

000606: Jan 30 09:10:30.026: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:+49xxxxxxxxx30@2.207.165.116:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 145.253.139.26:5060;branch=z9xxx
From: <sip:+49xxxxxxxxxx090@145.253.139.26>;tag=12F7B813-1B9D
To: <sip:+49xxxxxxxxx30@2.207.165.116>;tag=SDxxx
Call-ID: BBxxx
Max-Forwards: 70
P-Preferred-Identity: <sip:+49xxxxx65@145.253.139.26>
Timestamp: 1675069830
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0

000607: Jan 30 09:10:30.049: //11758/BB45D80BBE18/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 145.253.139.26:5060;branch=z9xxx
From: <sip:+49xxxxxxxxx90@145.253.139.26>;tag=12F7B813-1B9D
To: <sip:+49xxxxxxxxx30@2.207.165.116>;tag=SDviupaxxx0
Call-ID: BB4xxx
Timestamp: 1675069830
CSeq: 102 BYE
Accept: application/sdp
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Contact: <sip:+49xxxxxxx30@2.207.165.116:5060;transport=udp>
Content-Length: 0

 

 

colcha
Level 1
Level 1

OSTAPB0421#show running-config | section voice register
voice register global
mode cme
source-address 172.XX.XXX.XX port 5060
max-dn 10
auto-register
!
voice register dn 2
number +49xxxxx55
call-forward b2bua unregistered +49xxxxxx30

 

 

colcha
Level 1
Level 1

For security reasons I can't upload the  config file complet . If you need any additional information from the config file, I can upload it.

---------------

voice service voip
no notify redirect ip2ip
address-hiding
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
h323
call service stop forced
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
min-se 120
registrar server expires max 600 min 60asserted-id ppi
conn-reuse
source filter
privacy-policy passthru
sip-profiles inbound

-----

BR ,

Anibal 

b.winter
VIP
VIP

Then its hard to help you. You are even stripping numbers and parameters in the debug ... What is the exact scenario? You wrote something about user A calls B, and B forwards to C (external).
But in the debug, it seems the user A is also an external. If yes, your problem is an old know problem.

What about my point 1?

Hi winter, 

Point 1

We only use the cisco router as a gateway that connects to the Vodafone provider, with a NIM-VA-B  " Vdsl2/adsloISDN" (NIM 2) Board , where the additional router has a 1MFT T1/E1 card that connects directly to a PBX.

But why do you have CME config on your router then?

because we monitor the states of the routers PRTG, we access via remote (VPN) for configuration, to back up the system configuration files, etc.
As I said before we do not have Cisco IP phones

 

colcha_0-1675086494436.png

I think this graph is better explained

Not going into detail about your setup or config, because this is not a good one. There are configuration templates on how to configure Cisco CUBE for Vodafone Germany.

But as already written, the problem of "no audio when external to external forward" is very known in Germany.
The problem is the SIP platform of Vodafone (and other providers like Telekom, ...).
They have the characteristica, that the platform only starts sending RTP packets, when it first receives RTP packets.
But in the your scenario the both call endpoints are the platform itself, so platform waits on itself, to start sending packets --> Therefore no audio.

Already explained here:
https://community.cisco.com/t5/unified-communications-infrastructure/cube-config-sip-trunk-to-deutsche-telekom/m-p/4447706/highlight/true#M167984

colcha
Level 1
Level 1

Hallo , 

As can be seen in the graph, the call forwarding is being carried out in the Cisco Router

colcha_0-1675083830167.png

Vodafone---GW2 (Router Cisco)  --- Pbx 

 

 

The call forwarding is done in the PBX, not in the gateway. If you don’t register any Cisco phones and or 3:rd party devices to the gateway you don’t need CME.



Response Signature


Hi Roger, 

Exactly

The cisco router can do external forwarding, as long as there is an external line provider in the configured router

BR, 

Anibal 

I'm not saying that the router can not handle the call, just that the call forwarding as such is not set in it, that would be done in the PBX. So your call path would be something along with this.

Calling party (A) - > PSTN -> GW -> PBX -> Called party (B) that has CFA set to C (call is actually not going to this entity, it is just a logical step in the PBX) -> PBX -> GW -> PSTN -> Called party (C)

As you can see the GW is just passing the call from one entity to another, it does not set the CFA, it acts on called number, either B or C, as received from the PSTN and/or PBX.



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