CME can't recieve incoming call from SIP Trunk

[nawir]

PROBLEM:

-I want incoming call to my DID Number of siptrunk1 didlogic, will be forwarded to extension 801.

The same thing to siptrunk2 hoiio, will be forwarded to extension 801 as well

Currently no ringing tone when someone call my DID number to either siptrunk

Outgoing call no problem

Please help

thanks

DIAGRAM:

ISP -> (Dynamic IP) CLIENTROUTER (10.0.10.1) -> (10.0.10.206) 2811

CLIENTROUTER Firewall Rule

Forward ports 5060-5061 to 10.0.10.206 2811

Forward ports 9000-9100 to 10.0.10.206 2811

Forward ports 5090 to 10.0.10.206 2811

# sh run

version 15.1

boot system flash:c2800nm-adventerprisek9-mz.151-4.M10.bin

no ip domain lookup
ip domain name poc.com
ip name-server 8.8.8.8
ip name-server 8.8.4.4

!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
  registrar server
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
!
voice register global
mode cme
source-address 10.0.10.206 port 5060
max-dn 144
max-pool 35
tftp-path flash:
create profile sync 0037493227411278
voice-card 0

!

interface FastEthernet0/0
ip address 10.0.10.206 255.255.255.0
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 10.0.31.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto

!

dial-peer voice 1 voip
description Outgoing SIP DIDLogic
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target dns:sip.sg.didlogic.net
voice-class codec 1
dtmf-relay sip-notify
!
dial-peer voice 2 voip
description Outgoing SIP Hoiio
destination-pattern 852........
session protocol sipv2
session target dns:sip1.hoiio.com
voice-class codec 1
dtmf-relay sip-notify
!
dial-peer voice 10 voip
answer-address 801
destination-pattern 14159170001
voice-class codec 1
!
dial-peer voice 20 voip
answer-address 801
destination-pattern 85258010002
voice-class codec 1

!

sip-ua
credentials username 08001 password 7 password realm sip.sg.didlogic.net
credentials username sip7228002 password 7 password realm sip1.hoiio.com
authentication username 08001 password 7 password realm sip.sg.didlogic.net
authentication username sip7228002 password 7 password realm sip1.hoiio.com
timers connect 100
registrar 1 dns:sip.sg.didlogic.net expires 3600
registrar 2 dns:sip1.hoiio.com expires 3600

!

telephony-service
no auto-reg-ephone
max-ephones 35
max-dn 144
ip source-address 10.0.10.206 port 2000
max-redirect 5
cnf-file location flash:
load 7912 CP7912080004SCCP080108A.sbin
date-format dd-mm-yy
max-conferences 8 gain -6
web admin system name admin secret 5 $1$JGDo$afA54af/90u3S4gnt0sUP1
transfer-system full-consult
create cnf-files version-stamp 7960 Aug 23 2015 01:02:45
!
ephone-dn  1  dual-line
number 801
label Operator (x801)

!

ephone-dn  2  dual-line
number 802
label Sales (x802)
!
ephone-dn  3  dual-line
number 803
label CS (x803)
!
ephone  1
device-security-mode none
mac-address 000B.BE3B.9E9C
type 7912
button  1:1
!
ephone  2
device-security-mode none
mac-address 0012.0127.A14A
type 7912
button  1:2
!
ephone  3
device-security-mode none
mac-address E0DB.55A9.B3D5
type CIPC
button  1:3

[Vivek Batra]

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. 

[nawir]

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

[Leo Salcie Tejeda]

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

[nawir]

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

[Leo Salcie Tejeda]

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

[nawir]

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

[Leo Salcie Tejeda]

use these command:

sip-ua

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

[nawir]

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"

#show sip-ua connections udp detail
Total active connections      : 2
No. of send failures          : 0
No. of remote closures        : 0
No. of conn. failures         : 0
No. of inactive conn. ageouts : 4

---------Printing Detailed Connection Report---------
Note:
** Tuples with no matching socket entry
    - Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port>'
      to overcome this error condition
++ Tuples with mismatched address/port entry
    - Do 'clear sip <tcp[tls]/udp> conn t ipv4:<addr>:<port> id <connid>'
      to overcome this error condition

Remote-Agent:10.0.10.87, Connections-Count:1
  Remote-Port Conn-Id Conn-State  WriteQ-Size Local-Address
  =========== ======= =========== =========== ===========
         5060       2 Established           0  -

Remote-Agent:192.170.159.194, Connections-Count:1
  Remote-Port Conn-Id Conn-State  WriteQ-Size Local-Address
  =========== ======= =========== =========== ===========
         5060       3 Established           0  -


-------------- SIP Transport Layer Listen Sockets ---------------
  Conn-Id               Local-Address
===========    =============================
   0            [0.0.0.0]:5060

Call from 85258010007 to 14159172008

#
Aug 28 13:52:50.859: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:14159172008@10.0.10.206:5060 SIP/2.0
Record-Route: <sip:192.170.159.194;lr=on;ftag=as0a71f3aa;vst=AAAAAAEMCQEKcQUCBx4JeQkeHwIbHwEdATA2;nat=yes>
Record-Route: <sip:207.198.125.133;lr=on;ftag=as0a71f3aa;vsf=AAAAAAAAAAAAAAAAAAAAAAMJBQAADggAAAcHAAAEAzo1MDgw>
Via: SIP/2.0/UDP 192.170.159.194;branch=z9hG4bKbaa9.d1ea525a5dc6d35731516c3c0975a2c3.0
Via: SIP/2.0/UDP 207.198.125.133;rport=5060;branch=z9hG4bKbaa9.3a4df757193fc064f14366a130054145.0
Via: SIP/2.0/UDP 192.170.152.170:5080;received=192.170.152.170;branch=z9hG4bK46760265;rport=5080
Max-Forwards: 68
From:  <sip:85258010007@207.198.125.133>;tag=as0a71f3aa
To: 14159172008 <sip:14159172008@10.0.10.206>
Contact: <sip:85258010007@192.170.152.170:5080>
Call-ID: 48c13a49226a635421bef47c2bf80016@192.170.152.170
CSeq: 102 INVITE
Date: Fri, 28 Aug 2015 13:52:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-CID: 151029186_133145015@207.223.78.80
X-CallId:
Content-Type: application/sdp
Content-Length: 357
User-Agent: DIDLogic SBC

v=0
o=didlogic 1181670303 1181670303 IN IP4 192.170.152.170
s=DID Logic MGW
c=IN IP4 192.170.152.170
t=0 0
m=audio 12686 RTP/AVP 8 0 9 18 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Aug 28 13:52:50.887: //67/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.170.159.194;branch=z9hG4bKbaa9.d1ea525a5dc6d35731516c3c0975a2c3.0,SIP/2.0/UDP 207.198.125.133;rport=5060;branch=z9hG4bKbaa9.3a4df757193fc064f14366a130054145.0,SIP/2.0/UDP 192.170.152.170:5080;received=192.170.152.170;branch=z9hG4bK46760265;rport=5080
From: <sip:85258010007@207.198.125.133>;tag=as0a71f3aa
To: 14159172008 <sip:14159172008@10.0.10.206>
Date: Fri, 28 Aug 2015 13:52:50 GMT
Call-ID: 48c13a49226a635421bef47c2bf80016@192.170.152.170
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Aug 28 13:52:50.955: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:14159172008@sip.sg.didlogic.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK6210BF
Remote-Party-ID: <sip:85258010007@10.0.10.206>;party=calling;screen=no;privacy=off
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>
Date: Fri, 28 Aug 2015 13:52:50 GMT
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3896595059-1287786981-2151335822-3056050203
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1440769970
Contact: <sip:85258010007@10.0.10.206:5060>
Call-Info: <sip:10.0.10.206:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 67
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 7877 4204 IN IP4 10.0.10.206
s=SIP Call
c=IN IP4 10.0.10.206
t=0 0
m=audio 17742 RTP/AVP 8 0 18 19
c=IN IP4 10.0.10.206
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:19 CN/8000

Aug 28 13:52:51.031: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK6210BF;rport=60092;received=139.0.190.24
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.98e9
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.sg.didlogic.net", nonce="VeBo3FXgZ7BAgxiqnjoxKQBmmsEp6V/fYfBdCIA=", qop="auth"
Content-Length: 0


Aug 28 13:52:51.039: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:14159172008@sip.sg.didlogic.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK6210BF
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.98e9
Date: Fri, 28 Aug 2015 13:52:50 GMT
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


Aug 28 13:52:51.039: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:14159172008@sip.sg.didlogic.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK631EB1
Remote-Party-ID: <sip:85258010007@10.0.10.206>;party=calling;screen=no;privacy=off
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>
Date: Fri, 28 Aug 2015 13:52:51 GMT
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3896595059-1287786981-2151335822-3056050203
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1440769971
Contact: <sip:85258010007@10.0.10.206:5060>
Call-Info: <sip:10.0.10.206:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="08843",realm="sip.sg.didlogic.net",uri="sip:14159172008@sip.sg.didlogic.net:5060",response="8681576823e5d12a514d793b33047b19",nonce="VeBo3FXgZ7BAgxiqnjoxKQBmmsEp6V/fYfBdCIA=",cnonce="867F5F70",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 67
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 7877 4204 IN IP4 10.0.10.206
s=SIP Call
c=IN IP4 10.0.10.206
t=0 0
m=audio 17742 RTP/AVP 8 0 18 19
c=IN IP4 10.0.10.206
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:19 CN/8000

Aug 28 13:52:51.111: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK631EB1;rport=60092;received=139.0.190.24
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
CSeq: 102 INVITE
Content-Length: 0


Aug 28 13:52:59.527: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
Aug 28 13:53:10.151: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:14159172008@10.0.10.206:5060 SIP/2.0
Via: SIP/2.0/UDP 192.170.159.194;branch=z9hG4bKbaa9.d1ea525a5dc6d35731516c3c0975a2c3.0
Max-Forwards: 68
From:  <sip:85258010007@207.198.125.133>;tag=as0a71f3aa
To: 14159172008 <sip:14159172008@10.0.10.206>
Call-ID: 48c13a49226a635421bef47c2bf80016@192.170.152.170
CSeq: 102 CANCEL
Content-Length: 0


Aug 28 13:53:10.159: //67/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.170.159.194;branch=z9hG4bKbaa9.d1ea525a5dc6d35731516c3c0975a2c3.0
From: <sip:85258010007@207.198.125.133>;tag=as0a71f3aa
To: 14159172008 <sip:14159172008@10.0.10.206>
Date: Fri, 28 Aug 2015 13:53:10 GMT
Call-ID: 48c13a49226a635421bef47c2bf80016@192.170.152.170
CSeq: 102 CANCEL
Content-Length: 0


Aug 28 13:53:10.163: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:14159172008@sip.sg.didlogic.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK631EB1
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>
Date: Fri, 28 Aug 2015 13:52:51 GMT
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
CSeq: 102 CANCEL
Max-Forwards: 70
Timestamp: 1440769990
Reason: Q.850;cause=16
Content-Length: 0


Aug 28 13:53:10.163: //67/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.170.159.194;branch=z9hG4bKbaa9.d1ea525a5dc6d35731516c3c0975a2c3.0,SIP/2.0/UDP 207.198.125.133;rport=5060;branch=z9hG4bKbaa9.3a4df757193fc064f14366a130054145.0,SIP/2.0/UDP 192.170.152.170:5080;received=192.170.152.170;branch=z9hG4bK46760265;rport=5080
From: <sip:85258010007@207.198.125.133>;tag=as0a71f3aa
To: 14159172008 <sip:14159172008@10.0.10.206>;tag=CFC3C-13F8
Date: Fri, 28 Aug 2015 13:53:10 GMT
Call-ID: 48c13a49226a635421bef47c2bf80016@192.170.152.170
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0


Aug 28 13:53:10.199: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK631EB1;rport=60092;received=139.0.190.24
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-09b2
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
CSeq: 102 CANCEL
Content-Length: 0


Aug 28 13:53:10.203: //68/E8415273803A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.10.206:5060;rport=60092;received=139.0.190.24;branch=z9hG4bK631EB1
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>;tag=as6eac927c
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
CSeq: 102 INVITE
Server: DID Logic GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


Aug 28 13:53:10.207: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:14159172008@sip.sg.didlogic.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.206:5060;branch=z9hG4bK631EB1
From: <sip:85258010007@10.0.10.206>;tag=CB134-432
To: <sip:14159172008@sip.sg.didlogic.net>;tag=as6eac927c
Date: Fri, 28 Aug 2015 13:52:51 GMT
Call-ID: E844FBFB-4CC211E5-8040C78E-B627A01B@10.0.10.206
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0


Aug 28 13:53:10.211: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:14159172008@10.0.10.206:5060 SIP/2.0
Via: SIP/2.0/UDP 192.170.159.194;branch=z9hG4bKbaa9.d1ea525a5dc6d35731516c3c0975a2c3.0
Max-Forwards: 68
From:  <sip:85258010007@207.198.125.133>;tag=as0a71f3aa
To: 14159172008 <sip:14159172008@10.0.10.206>;tag=CFC3C-13F8
Call-ID: 48c13a49226a635421bef47c2bf80016@192.170.152.170
CSeq: 102 ACK
Content-Length: 0


RTR2811a#u all
All possible debugging has been turned off

[nawir]

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

[Leo Salcie Tejeda]

Explain dialplan: Call from 85258010007 to 14159172008???? 

[nawir]

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

[Leo Salcie Tejeda · Accepted solution]

Did you traslate the DID number to you Extension Number?

If not use:

num-exp 14159172008 "DN"

[Vivek Batra]

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.

[Vivek Batra]

Hi Nawir,

Could you make it work?