08-27-2020 01:09 AM
Hello everybody,
I'm trying to configure a Cisco 880 voice router to register SIP phones and serve as a gateway in a remoted office.
For now I've managed to get my two phones to register on the router, i can see them in the "show voice register pool all" command.
But when I try to make a call from one to another, nothing happens, I don't even hear the beeps.
In the "debug ccsip messages" I see only the INVITE from the first phone, until I cancel the call:
*Aug 27 07:28:37.936: //77/BD7523FB808D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6969@192.168.66.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.66.1:5060;branch=z9hG4bKC12F7
Remote-Party-ID: "6666 " <sip:6666@192.168.66.1>;party=calling;screen=no;privacy=off
From: "6666 " <sip:6666@192.168.66.1>;tag=A0A810-48A
To: <sip:6969@192.168.66.2>
Date: Thu, 27 Aug 2020 07:28:37 GMT
Call-ID: BD76F90C-E76D11EA-8093A823-2C00BD78@192.168.66.1
Supported: timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3178570747-3882684906-2156767267-0738246008
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M4b
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1598513317
Contact: <sip:6666@192.168.66.1:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-ID: 186f4b0a5f565e54aa40bd877054da83;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 293
v=0
o=CiscoSystemsSIP-GW-UserAgent 27 6336 IN IP4 192.168.66.1
s=SIP Call
c=IN IP4 192.168.66.1
t=0 0
m=audio 16438 RTP/AVP 8 0 101 19
c=IN IP4 192.168.66.1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
*Aug 27 07:28:41.968: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
.......
My config:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
rel1xx disable
registrar server expires max 1200 min 300
redirect contact order best-match
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice register global
mode cme
source-address **** port 5060
max-dn 4
max-pool 4
!
voice register dn 1
number 6666
no-reg
!
voice register dn 2
number 6969
no-reg
!
voice register pool 1
id mac ***
number 1 dn 1
dtmf-relay rtp-nte
voice-class codec 2
username 6666 password ***
!
voice register pool 2
id mac ***
number 1 dn 2
dtmf-relay rtp-nte
voice-class codec 2
username 6969 password ***
telephony-service
max-ephones 10
max-dn 10
ip source-address **** port 2000
max-conferences 4 gain -6
transfer-system full-consult
I've googled but found nothing specific. It's a simple setup and I can't understand what is wrong. Please, help me to solve this problem.
Thanks in advance
Solved! Go to Solution.
08-27-2020 07:08 PM
OK, I've found the solution. I don't know if it's a correct one though.
I've disabled "SIP Invite Restrict" option on the phones and now they are working without issues.
08-27-2020 01:52 AM
Try with adding this to you configuration.
sip-ua registrar ipv4:[A.A.A.A] expires 3600
Where A.A.A.A is the IP of your router.
08-27-2020 02:01 AM
Thank you for you reply!
I've added the registrar string, but nothing changed, unfortunately.
08-27-2020 03:08 AM
Remove username and password from pool. not required.
voice register pool 1
id mac ***
number 1 dn 1
dtmf-relay rtp-nte
voice-class codec 2
username 6666 password ***
08-27-2020 06:10 PM
Hi and thank you for your reply!
I've deleted username and password from pool. Nothing changed. I see only INVITE messages in the logs, the second phone does not respond.
08-27-2020 07:08 PM
OK, I've found the solution. I don't know if it's a correct one though.
I've disabled "SIP Invite Restrict" option on the phones and now they are working without issues.
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