09-09-2024 08:52 PM
Hi community,
I'm trying to set up sip trunking from my ISP, but still have not able to received or get the call out of the router, this the context the ISP provide the sip server using the same fiber router where I get the internet, it is using the LAN port 1, the sip trunks do not required authentication due to this connection. here is the parameters they gave us.
The only thing I'm looking for it's for the sip trunking numbers to ring on extensión 103
Parameters |
no authentication |
IP: 10.65.65.19 |
MASK: 255.255.224.0 |
GTW: 10.65.64.1 |
Proxy: 10.187.128.3 |
routing: 10 digits |
routing coming in: last 4 digits |
Codec audio: G711ALAW Y G729 |
Port UDP 5060 |
Pilot: 614-388-8800 |
this is my current router configuration :
1. cisco 3945 configuration.
1.- The next to files have the command "debug ccsip messages" for incoming calls, I post it on word file because the app did not let me upload it in txt format.
2.- The command "debug voip dialpeer all" for outgoing calls.
I have been reading a lot of configurations, but still have not been able of received ir call out.
Thank you very much for your help in advance.
kotusha
Solved! Go to Solution.
09-11-2024 11:04 PM
09-11-2024 11:06 PM
The time clock on ip phones is 6 hours ahead from the time clock of the router. do u have any idea why this is happening?
Thank you
09-12-2024 01:53 AM - edited 09-12-2024 09:08 AM
Yes I have, it shows the wrong time since you haven't configured the time zone in you CME configuration. I don't know what TZ you're in, so you'll need to select the appropriate time-zone number to set under telephony-service.
telephony-service
time-zone <number>
time-format 24
time-zone ?
<1-56> select timezone name used by IP phones (offset in minutes)
1 Dateline Standard Time -720
2 Samoa Standard Time -660
3 Hawaiian Standard Time -600
4 Alaskan Standard/Daylight Time -540
5 Pacific Standard/Daylight Time -480
6 Mountain Standard/Daylight Time -420
7 US Mountain Standard Time -420
8 Central Standard/Daylight Time -360
9 Mexico Standard/Daylight Time -360
10 Canada Central Standard Time -360
11 SA Pacific Standard Time -300
12 Eastern Standard/Daylight Time -300
13 US Eastern Standard Time -300
14 Atlantic Standard/Daylight Time -240
15 SA Western Standard Time -240
16 Newfoundland Standard/Daylight Time -210
17 E. South America Standard/Daylight Time -180
18 SA Eastern Standard Time -180
19 Mid-Atlantic Standard/Daylight Time -120
20 Azores Standard/Daylight Time -60
21 GMT Standard/Daylight Time +0
22 Greenwich Standard Time +0
23 W. Europe Standard/Daylight Time +60
24 GTB Standard/Daylight Time +60
25 Egypt Standard/Daylight Time +120
26 E. Europe Standard/Daylight Time +120
27 Romance Standard/Daylight Time +120
28 Central Europe Standard/Daylight Time +120
29 South Africa Standard Time +120
30 Jerusalem Standard/Daylight Time +120
31 Saudi Arabia Standard Time +180
32 Russian Standard/Daylight Time +180
33 Iran Standard/Daylight Time +210
34 Caucasus Standard/Daylight Time +240
35 Arabian Standard Time +240
36 Afghanistan Standard Time +270
37 West Asia Standard Time +300
38 Ekaterinburg Standard Time +300
39 India Standard Time +330
40 Central Asia Standard Time +360
41 SE Asia Standard Time +420
42 China Standard/Daylight Time +480
43 Taipei Standard Time +480
44 Tokyo Standard Time +540
45 Cen. Australia Standard/Daylight Time +570
46 AUS Central Standard Time +570
47 E. Australia Standard Time +600
48 AUS Eastern Standard/Daylight Time +600
49 West Pacific Standard Time +600
50 Tasmania Standard/Daylight Time +600
51 Central Pacific Standard Time +660
52 Fiji Standard Time +720
53 New Zealand Standard/Daylight Time +720
54 Venezuela Standard Time -270
55 Pacific SA Daylight Time -180
56 Pacific SA Standard Time -240
09-12-2024 08:16 PM
Thank you very much, without you help it will have take forever to get it up and running, I going to paste the working configuration so we can help other. Thank you very much again, I will start working on a NME-CUE module to get voicemail and Auto attendant, if you have any information in how to get it working I will appreciated. Thank you so much one more time.
!
ip dhcp excluded-address 172.16.1.1 172.16.1.10
!
ip dhcp pool VOICE
network 172.16.1.0 255.255.255.0
default-router 172.16.1.1
option 150 ip 172.16.1.1
dns-server 8.8.8.8
!
!
!
no ip domain lookup
ip cef
no ipv6 cef
!
voice call carrier capacity active
!
voice service voip
ip address trusted list
ipv4 10.187.128.3 255.255.255.255
mode border-element license capacity 30
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/2
bind media source-interface GigabitEthernet0/2
header-passing
privacy-policy passthru
!
!
voice class uri PSTN sip
host ipv4:10.187.128.3
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
!
!
voice translation-rule 1
rule 1 /^9\(.*\)/ /\1/
!
voice translation-rule 2
rule 1 /.../ /6143888800/ **** Three digits any extention calls out thru the pilot number from ISP Telecomm
!
voice translation-rule 11
rule 1 /....$/ /103/ **** Last Four digits of incomming calls get transfer to Extention 103
!
!
voice translation-profile PSTN_INCOMING
translate called 11
!
voice translation-profile SIP
translate calling 2
translate called 1
!
!
interface GigabitEthernet0/0
ip address 192.168.1.19 255.255.252.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
ip address 172.16.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto
media-type rj45
!
interface GigabitEthernet0/2
ip address 10.65.65.19 255.255.224.0
duplex auto
speed auto
!
!
ip default-gateway 192.168.1.254
!
ip route 0.0.0.0 0.0.0.0 192.168.1.254
ip route 10.187.128.0 255.255.255.0 10.65.64.1
!
!
dial-peer voice 9000 voip
description ** INBOUND CALLS from telecoms **
translation-profile incoming PSTN_INCOMING
session protocol sipv2
incoming uri via PSTN
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 100 voip
description **** OUTBOUND CALLS to telecoms *****
translation-profile outgoing SIP
destination-pattern 9T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
sip-ua
no remote-party-id
retry invite 2
retry options 1
timers connect 100
sip-server ipv4:10.187.128.3
!
!
!
telephony-service
max-ephones 60
max-dn 60
ip source-address 172.16.1.1 port 2000
auto assign 1 to 10 type 7965
cnf-file location flash:
cnf-file perphone
user-locale ES load CME-locale-es_ES-Spanish-12.0.12.0.tar
network-locale ES
load 7937 apps37sccp.1-4-4-0
load 7965 SCCP45.9-2-1S
max-conferences 8 gain -6
web admin system name cisco password cisco
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1 dual-line
number 101
!
!
ephone-dn 2 dual-line
number 102
!
!
ephone-dn 3 dual-line
number 103
!
!
ephone 1
device-security-mode none
mac-address 0004.F2F7.20B8
type 7937
button 1:1
!
!
!
ephone 2
device-security-mode none
mac-address 0064.40B4.9D68
type 7965
button 1:2
!
!
!
ephone 3
device-security-mode none
mac-address 0026.0B5C.543E
type 7965
button 1:3
!
!
!
09-12-2024 09:46 PM
Glad to hear that you got it to work. One thing, why did you mark your own answer as the solution to your question instead of answer(s) from others who helped you solve your problems?
09-12-2024 11:43 PM
I did it know what accept has the solutions, I did accepted the last comment you gave me, and I add the information that worked for me, so in the future people can used it. thank you very much ones again. how can I rate you?
09-13-2024 03:19 AM
FYI You can change what answer you selected as the solution.
09-13-2024 10:32 PM
thank you very much.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide