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CUBE SIP 403 Request From Unknown Source

uastac2011
Level 1
Level 1

Hi Guys,

 

Need some help. We are implementing SIP connection and I am new for this. I cannot make it work. I already did what the article or forums but still failed.

 

Here is the topology. Attached files are sip config and debug. 
I am hoping that you can help me. Thank you so much.

 

Toplogy.JPG

43 Replies 43

Gregory Brunn
Spotlight
Spotlight

403 ForbiddenThe server understood the request, but is refusing to fulfill it.Sometimes (but not always) this means the call has been rejected by the receiver.

 

Have you confirm with your ITSP they are allowing you yet?

 

Hi Gregory
Thank you for your reply. I will check on this our service provider and I'll update you.
Just want to confirm if this is correct. In cucm the sip trunk my destination address is 10.229.141.222.

Vaijanath Sonvane
VIP Alumni
VIP Alumni

Hi,

The CUBE router is receiving 403 Request From Unknown Source from your service provider. Please check with your Service Provider about what calling number they are expecting and in which format.

Capture.JPG

Also, please correct below configuration:

1. First remove bind interface commands from below configuration as you are binding the interface under dial-peers.

voice service voip
 sip
  bind control source-interface GigabitEthernet0/1
  bind media source-interface GigabitEthernet0/1

2. Under dial-peer 101 which is pointing to CUCM, change the bind interface to GigabitEthernet0/0 as this is your voice VLAN interface:

dial-peer voice 101 voip
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
!

3. Under dial-peer 100, add bind interface GigabitEthernet0/0.

dial-peer voice 100 voip
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
!

4. Under dial-peer 800, add bind interface GigabitEthernet0/2.

dial-peer voice 800 voip 
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
!

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

Hi Vaijanath
Thank you for your reply. I will check on this our service provider and will re config the cube.
Just want to confirm if this is correct. In cucm the sip trunk my destination address is 10.229.141.222.

In CUCM SIP Trunk, your destination IP Address needs to be 192.168.104.2.

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

but when i am using that ip add the status is no service. can i use lan ip given by the service provider? I also created voice vlan to that subnet (192.168.1.0) for testing.

Make sure that IP Address is reachable from your CUCM cluster. If you use different IP Address of different interface then make sure you bind correct interface under dial peers.

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

Incoming call is also not working :(
Our DID is 02891-8330 to 8339

 

Just want to check if this correct.

 

In my sip trunk

caller info.JPG

Route Pattern 

route pattern.JPGtransformation.JPG

IP Phone

external.JPGphone.JPG

dial-peer voice 800 voip
description ** Incoming From ITSP TO VG **
session protocol sipv2
session transport udp
incoming called-number .
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte

 

dial-peer voice 101 voip
description ** Outgoing From VG to CUCM **
destination-pattern 028918330
session protocol sipv2
session target ipv4:172.27.199.11
session transport udp
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte

 

 

thank you so much.

"Incoming call is also not working :("

Inbound calling will depend on the Trunk settings in "Inbound Calls" section.  For example inbound CSS of the trunk would either have to be able to "see" your partition Internal-PT using four significant digits, or would have to see a Translation Pattern to convert 02891833X to 833X

Hi Tony
PT and CSS are configured properly but when i show debug voice ccapi inout
there is no logs which means that our router is not receiving from the service provider?

What error message are you getting for inbound calls?

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

Hi Vaijanath

when i show debug voice ccapi inout there is no logs which means that our router is not receiving from the service provider?

Please try debug ccsip message and post the logs.
Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

Hi Vaijanath
here is the logs

Aug 16 09:39:32.793: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.1.5:5060 SIP/2.0
Via: SIP/2.0/UDP 172.27.199.11:5060;branch=z9hG4bK3fb4703578b4
From: <sip:172.27.199.11>;tag=1473428793
To: <sip:192.168.1.5>
Date: Fri, 16 Aug 2019 09:39:32 GMT
Call-ID: bec86780-d56179d4-3eed-bc71bac@172.27.199.11
User-Agent: Cisco-CUCM10.5
CSeq: 101 OPTIONS
Contact: <sip:172.27.199.11:5060>
Max-Forwards: 0
Content-Length: 0


Aug 16 09:39:32.797: //9752/9610430CA791/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.27.199.11:5060;branch=z9hG4bK3fb4703578b4
From: <sip:172.27.199.11>;tag=1473428793
To: <sip:192.168.1.5>;tag=275F5B08-1B02
Date: Fri, 16 Aug 2019 09:39:32 GMT
Call-ID: bec86780-d56179d4-3eed-bc71bac@172.27.199.11
Server: Cisco-SIPGateway/IOS-15.7.3.M4b
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
--More--  Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 375

v=0
o=CiscoSystemsSIP-GW-UserAgent 6061 6591 IN IP4 192.168.104.2
s=SIP Call
c=IN IP4 192.168.104.2
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 192.168.104.2
m=image 0 udptl t38
c=IN IP4 192.168.104.2
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
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