08-22-2015 12:13 PM - edited 03-17-2019 04:04 AM
Solved! Go to Solution.
08-28-2015 09:03 AM
Did you traslate the DID number to you Extension Number?
If not use:
num-exp 14159172008 "DN"
08-23-2015 10:13 PM
Share the output of debug ccsip messages (incoming call).
Which dial-peer you are expecting to match during incoming call?
Further you have not bind oubound dial-peer with any of the interface.
08-26-2015 05:16 PM
Sorry for late reply because I am testing with CME+Planet VIP-480 ougoing call problem.
While FreePBX+Planet VIP-480 outgoing call has no problem at all.
I'll post in another thread
1. Share the output of debug ccsip messages (incoming call).
Please see attached file
2. Which dial-peer you are expecting to match during incoming call?
dial-peer voice 10 voip
3. Further you have not bind oubound dial-peer with any of the interface.
With current config, I can call US or HK through my SIP trunk.
Please give an example where I should modify
thanks
08-26-2015 07:52 PM
The call is been rejected by the Router 2811. It could be the Toll Fraud Prevention feature actived. Just for testing use these command:
conf t
voice service voip
no ip address trusted authenticate
If still the same, maybe is a codec issue. Please attache a show run.
Regards
08-26-2015 08:34 PM
After no ip address trusted authenticate.
debug ccsip message output attached
sh run also attached.
When I call to my us number 14159170001.
I didn't hear my ext 801 which is 7912 SCCP firmware ringing.
Is that because my 7912 has SCCP firmware instead of SIP firmware
FYI
-I don't have PVDM card yet
thanks
08-28-2015 06:03 AM
Now, at least one issue has been solved.
The SIP Trunk is not registered. That's why the call is not answered.
Use this command:sh sip-ua register status
If same CODEC is used (ex. G711ulaw), there's not need of PVDM. However, is recomended to have at least a basic PVDM for feature invoking resources like MoH, Transfer, Conference and so on.
Regards
08-28-2015 06:14 AM
Thank you for your respond.
Here the command result
RTR2811a#sh sip-ua register status
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
08843 -1 2324 yes
1001 -1 174 no
801 20001 179 no
802 20002 174 no
803 20003 174 no
sip7228254 -1 177 no
--------------------- Registrar-Index 2 ---------------------
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
08843 -1 2325 yes
1001 -1 2325 yes
801 20001 2325 yes
802 20002 2325 yes
803 20003 2326 yes
sip7228254 -1 2325 yes
I am confuse why my extension trying to register to didlogic.net as shown below
.Aug 28 12:28:26.789: //73/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized. No cheating please
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK69AAF;rport=58677;received=139.0.190.24
From: <sip:801@sip.sg.didlogic.net>;tag=EFACC-1C45
To: <sip:801@sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e007
Call-ID: F807E66B-4CB411E5-800BB939-8A20C709
CSeq: 13 REGISTER
Content-Length: 0
.Aug 28 12:28:27.285: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip.sg.didlogic.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK6A1A91
From: <sip:802@sip.sg.didlogic.net>;tag=EFD3C-1B2B
To: <sip:802@sip.sg.didlogic.net>
Date: Fri, 28 Aug 2015 12:28:27 GMT
Call-ID: F8088293-4CB411E5-800EB939-8A20C709
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1440764907
CSeq: 12 REGISTER
Contact: <sip:802@10.0.10.206:5060>
Expires: 3600
Supported: path
Content-Length: 0
08-28-2015 06:33 AM
use these command:
registrar dns: "Serviceprovider URL" expires
3600
sip-server dns:
"Serviceprovider URL"
Please post those command making a call:
sh sip-ua con udp det
debug ccsip messages
08-28-2015 06:58 AM
I believe since I have 2 sip trunk. I need to input below for each provider
sip-ua
registrar dns: "Serviceprovider URL" expires 3600
sip-server dns:"Serviceprovider URL"
08-28-2015 07:26 AM
Registrar and sip-server already configured.
sip-server can only be listed one. If I type both provider, the last provider will win
sip-ua
credentials username 1001 password 7 055A545C751012 realm 10.0.10.87
credentials username 08843 password 7 01232617481C561123 realm sip.sg.didlogic.net
credentials username sip7228256 password 7 044A04030475786A254809033B2901 realm sip1.hoiio.com
authentication username 08843 password 7 15222B1F173D7B3123 realm sip.sg.didlogic.net
authentication username sip7228254 password 7 06170024471A3D3D29461E1F252123 realm sip1.hoiio.com
authentication username 1001 password 7 00554155500456 realm 10.0.10.87
registrar 1 dns:sip.sg.didlogic.net expires 3600
registrar 2 dns:sip1.hoiio.com expires 3600
registrar 3 ipv4:10.0.10.87 expires 3600
sip-server dns:sip.didlogic.net
08-28-2015 07:46 AM
Explain dialplan: Call from 85258010007 to 14159172008????
08-28-2015 07:51 AM
85258010007 is for example my client telp no
14159172008 is my DID number.
So I am expecting if my client call my number. It will be forwarded to my extension 801
08-28-2015 09:03 AM
Did you traslate the DID number to you Extension Number?
If not use:
num-exp 14159172008 "DN"
08-26-2015 09:27 PM
Hi Nawir,
In the attached debug, gateway is getting calls with called party number as 08001 but I don't see any inbound dial-peer which can match this number.
Please have a new dial-peer with incoming called-number . and apply the relevant translation rule/profile which can translate 08001 to 801.
Also include no ip address trusted authenticate under voice service voip.
Share the results after above modifications.
08-27-2015 05:08 AM
Hi Nawir,
Could you make it work?
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