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Replies

Can't make a call between SIP phones on 880 router

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

1 Accepted Solution

Accepted Solutions

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. 

View solution in original post

5 Replies 5

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.



Response Signature


Thank you for you reply! 

I've added the registrar string, but nothing changed, unfortunately. 

 

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 ***



Response Signature


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. 

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.