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sip trunk problem (caller can't hear even ringing[ringback] tone)

Dear All

I have problem with configuring sip trunk on cube, my scenario is like below:

ITSP(SBC)>---------<(PBX) CUBE (sip trunk)>----------<ISSABEL>-----------IP SIP Phone

                                                  └-PRI[E1]>------------< Panasonic analog pbx

I can call from ip phone to panasonic and vice versa through my CUBE (CISCO2901) and everything is OK in two way

but when i call from outside to my ip phone (for example call 34567300 from 87654321) my sip phone ring and display caller id (87654321) correctly but caller can't hear anything even ringing (ringback) tone and when i pickup (off-hook) destination phone (ringing sip phone) , I can't hear anything and call disconnect after timeout (after 19 second) 

 

I test the sbc by removing CUBE and connect my issable server direct to ITSP modem and i can send and receive call normally with this scenario so i think problem is in my CUBE

 

I spoke to ITSP and they told me they didn't receive any sip message from my router (but i see this 100 trying,180 ringing and 200 OK in my debug ccsip message output so i think problem is SIP message doesn't deliver correctly to SBC

 

I even test without this lines but same result:

rel1xx disable

voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1

 

I have attached Debug output attached (debug voip ccapi inout & debug ccsip message)

 

My configuration on CUBE is:

 

my PBX ip is: 2.3.4.5 (ITSP gave this to me)
ITSP Gateway ip is: 2.3.4.6
ITSP SBC ip is: 5.6.7.8
my phone number is: 34567xxx
calling number for testing is: 87654321

 

CUBE config

!
version 15.7
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
card type e1 0 0
!
no aaa new-model
network-clock-participate wic 0
!
no ip domain lookup
ip cef
no ipv6 cef
!
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
voice-card 0
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711alaw
sip
rel1xx disable
min-se 300
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 ilbc
codec preference 5 g728
!
vxml logging-tag
hw-module pvdm 0/0
!
redundancy
!
controller E1 0/0/0
framing NO-CRC4
clock source internal
pri-group timeslots 1-31
!
controller E1 0/0/1
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
description Link to issable
ip address 192.168.0.1 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
description Link to SBC
ip address 2.3.4.5 255.255.255.252
duplex auto
speed auto
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-net5
isdn overlap-receiving T302 5000
isdn protocol-emulate network
isdn incoming-voice voice
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 192.168.0.254
ip route 5.6.7.8 255.255.255.255 2.3.4.6
ip ssh version 2
!
ipv6 ioam timestamp
!
control-plane
!
voice-port 0/0/0:15
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
dial-peer voice 1 voip
description INBOUND
destination-pattern .T
session protocol sipv2
session target ipv4:5.6.7.8:5060
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay h245-alphanumeric h245-signal sip-notify
no vad
!
dial-peer voice 2 pots
description PRI
destination-pattern 34567000
direct-inward-dial
port 0/0/0:15
forward-digits all
!
dial-peer voice 10 voip
description Issable
destination-pattern 3..
session protocol sipv2
session target ipv4:192.168.0.2:5060
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay h245-alphanumeric h245-signal sip-notify
no vad
!
dial-peer voice 20 voip
description Issable
destination-pattern 345672..
session protocol sipv2
session target ipv4:192.168.0.2:5060
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay h245-alphanumeric h245-signal sip-notify
no vad
!
dial-peer voice 50 pots
description PRI
destination-pattern 5..
direct-inward-dial
port 0/0/0:15
forward-digits all
!
gatekeeper
shutdown
!
line con 0
login local
line aux 0
!
scheduler allocate 20000 1000
!
end

 

4 Replies 4

I found problem!!!!

it was "connection-reuse"

I remove:voice-class sip bind control source-interface

and add:

sip-ua

connection-reuse

 

and problem solved

but now i have other problem when i call from PSTN to my router after call ended the PSTN side dose not receive end call tone!

How did you find the problem? You must have been reading some documentation and not just entering random commands. I would be interested to see how you solved this.

It was not just entering random commands but something like that!!

I search for same configs in web and in one of them see this command and i think it may be the problem because my router send INVITE msg to SBC (i think SBC reject that) and this command force router to use connection made by SBC

but i still have one problem:

when i call from PSTN to my router after call ended the PSTN side dose not receive end call tone!

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