11-25-2013 11:42 AM - edited 03-16-2019 08:34 PM
I have a SIP trunk from a provider terminated on my 3825 gateway which is connected via h323 to my CUCM 6.1 cluster.
I am not able to place outbound calls.
Any help would be much appreciated.
Thanks,-
Alexis
Solved! Go to Solution.
11-26-2013 04:38 PM
Alexis,
Now you need to speak to ITSP. We have tried a few things and they are still rejecting your call.
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
11-25-2013 11:45 AM
Hi Alexis.
Are you able to receive incoming calls?
Can you please post vg ios version and relevant config?
Thanks
Regards
Carlo
Sent from Cisco Technical Support iPhone App
11-25-2013 12:10 PM
We did not try incoming calls, configured an outbound dial-peer 900 for testing but its not working.
Below the config.
isdn switch-type primary-net5
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
voice call carrier capacity active
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
!
voice class codec 1
codec preference 1 g711ulaw bytes 160
codec preference 2 g729br8 bytes 20
codec preference 3 g729r8 bytes 20
!
voice translation-rule 1
rule 1 /^51..$/ /4355100/
rule 2 /^50..$/ /4355000/
rule 3 /^5...$/ /4355000/
rule 4 /^....$/ /4355000/
!
!
voice translation-profile 4to7
translate calling 1
!
!
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-31
!
controller E1 0/0/1
framing NO-CRC4
pri-group timeslots 1-31
!
controller E1 0/1/0
framing NO-CRC4
channel-group 0 timeslots 1-31
!
controller E1 0/1/1
!
ip ssh time-out 60
ip ssh authentication-retries 0
ip ssh version 2
!
!
!
!
interface GigabitEthernet0/0
no ip address
ip flow ingress
no ip mroute-cache
duplex auto
speed auto
media-type rj45
!
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address x.x.x.x 255.255.255.0
ip flow ingress
ip ospf message-digest-key 1 md5 7 132812260A23371A0D2F2D2A7402
ip ospf priority 0
h323-gateway voip bind srcaddr x.x.x.x
!
no ip http server
no ip http secure-server
!
!
ip access-list extended notelnet
deny tcp any any eq telnet
permit ip any any
!
voice-port 0/0/0:15
!
voice-port 0/0/1:15
!
!
!
sccp local GigabitEthernet0/0.30
sccp ccm 172.17.30.10 identifier 2 version 6.0
sccp ccm 172.17.30.11 identifier 1 version 6.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0.30
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 100 register XCODERENVGW01
associate profile 101 register CONFBRENVGW01
!
dspfarm profile 100 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 15
associate application SCCP
!
dspfarm profile 101 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
!
!
dial-peer voice 90 pots
translation-profile outgoing 4to7
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/0/0:15
!
dial-peer voice 5000 voip
preference 2
destination-pattern 5...
voice-class codec 1
session target ipv4:172.17.30.10
dtmf-relay h245-alphanumeric
!
dial-peer voice 5001 voip
preference 1
destination-pattern 5...
voice-class codec 1
session target ipv4:172.17.30.11
dtmf-relay h245-signal h245-alphanumeric
!
dial-peer voice 91 pots
translation-profile outgoing 4to7
shutdown
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/0/1:15
!
dial-peer voice 900 voip
destination-pattern 2T
voice-class codec 1
session protocol sipv2
session target ipv4:66.249.145.144
incoming called-number .
dtmf-relay rtp-nte
!
!
sip-ua
credentials username xxxx password 7 145F433219537C realm none
authentication username xxxx password 7 04135A3F1A761A
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:businesssolutions.digicelcuracao.net expires 3600
sip-server dns:sip.digicelcuracao.net
host-registrar
11-25-2013 02:22 PM
First of all I can see that your inbound (catch all) dial-peer is configured for sip protocol. While this is good for calls coming from your ITSP, this is not good for calls coming from cucm as these are H323 calls. Hence you will n eed two different inbound dial-peers..One to macth inbound calls to the gateway from cucm and the other for inbound calls from ITSP
dial-peer voice 900 voip
destination-pattern 2T
voice-class codec 1
session protocol sipv2
session target ipv4:66.249.145.144
incoming called-number XXXX, where XXX= the dialled number range from ITSP to you
dial-peer voice 901 voip
voice-class codec 1
incoming called-number 2T (assuming this is the called number to ITSP)
dtmf-relay h245-signal h245-alphanumeric
no vad
Once you have done that pls do another test call and send the ff:
1. debug voip ccapi inout
2.debug ccsip messages
3. debug h225 asn1
4.debug h245 asn2
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
11-26-2013 04:50 AM
Ok, I've made the changes as suggested (see config below). Find attached the debugs.
!
dial-peer voice 900 voip
destination-pattern 2T
voice-class codec 1
session protocol sipv2
session target ipv4:66.249.145.144
incoming called-number 7248...
!
dial-peer voice 901 voip
voice-class codec 1
incoming called-number 2T
dtmf-relay h245-signal h245-alphanumeric
no vad
!
11-26-2013 05:35 AM
Hi Alexis,
I traces the test call in with dialedDigits 25127172. The call was received by the gateway using H225 Setup via dial peer 901.
Then, the dial peer 900 was selected for sending the call to the provider.
The gateway sends an INVITE to the provider and the provider resposnds back with "403 Forbidden". We need to engage the provider to understand the reason for which they are disconnecting the call.
If they need any specific information in the INVITE message, we can make the necessary modification for them to accept the call.
1885471: Nov 26 08:42:41.087 CAR: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:25127172@66.249.145.144:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.30.5:5060;branch=z9hG4bKC4B21DA
From: "Alexis Sulbaran" <>>5155@businesssolutions.digicelcuracao.net>;tag=5AFCE04C-2549
To: <25127172>25127172>
Date: Tue, 26 Nov 2013 12:42:41 GMT
Call-ID: 12A891D7-55CF11E3-8D9ABF57-49F66684@172.17.30.5
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 16567355-1100431657-3422626819-2886800257
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1385469761
Contact: <5155>5155>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 212
v=0
o=CiscoSystemsSIP-GW-UserAgent 9242 6114 IN IP4 172.17.30.5
s=SIP Call
c=IN IP4 172.17.30.5
t=0 0
m=audio 16896 RTP/AVP 0 19
c=IN IP4 172.17.30.5
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
a=ptime:20
1885476: Nov 26 08:42:41.211 CAR: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.17.30.5:5060;branch=z9hG4bKC4B21DA
From: "Alexis Sulbaran" <>>5155@businesssolutions.digicelcuracao.net>;tag=5AFCE04C-2549
To: <25127172>;tag=aprqngfrt-3804ac30000a625127172>
Call-ID: 12A891D7-55CF11E3-8D9ABF57-49F66684@172.17.30.5
CSeq: 101 INVITE
Timestamp: 1385469761
HTH,
Jagpreet Singh Barmi
11-26-2013 06:37 AM
Just got of the phone with the provider. They are saying the invite "From" is incorrect. Its 5155@businesssolutions.digicelcuracao.net. It should be authenticationusername@businesssolutions.digicelcuracao.net.
The authentication username in the sip-ua.
And also the "To" is 25127172@businesssolutions.digicelcuracao.net and it should be 5127172@businesssolutions.digicelcuracao.net.
11-26-2013 08:16 AM
You can modify the "From and To" to match what your provider wants like this..using sip profiles
voice class sip-profiles 1
request INVITE sip-header From modify "<>" "
request INVITE sip-header To modify "<2>" "<>">2>
You then need to apply the profile to
dial-peer voice 900 voip
voice-class sip profile 1
Give this ago and test again..if you still have issues send us the logs again
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
11-26-2013 10:10 AM
Hi Alexis.
Just to add my 2 cents to the excellent suggestion from Aok (as usual +5P)
To modify from on invite to your SIP Provider, you can also add under sip-ua
calling-info sip-to-pstn number set
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
11-26-2013 10:27 AM
11-26-2013 10:58 AM
Hi Alexis.
Ok now from entry is correct but to not yet.
Try to modify voice class sip as follows:
voice class sip-profiles 1
no request INVITE sip-header To modify "<2>" "<>">2>
request INVITE sip-header To modify "<2>" "<>usinesssolutions.digicelcuracao.net>">2>
After changes plese post again a debug ccsip messages
Thanks
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
11-26-2013 11:15 AM
11-26-2013 11:37 AM
Hi Alexis.
As you can see, now from and to fields are as the provider requested.
From: "Alexis Sulbaran" <>MCAtcoUser01@businesssolutions.digicelcuracao.net>;tag=5C5C89F8-C57>
To: <>5127172@businesssolutions.digicelcuracao.net>>
But you are still receiving 403 Forbidden
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.17.30.5:5060;branch=z9hG4bKCE5130D
From: "Alexis Sulbaran" <>>MCAtcoUser01@businesssolutions.digicelcuracao.net>;tag=5C5C89F8-C57
To: <>>5127172@businesssolutions.digicelcuracao.net>;tag=aprqngfrt-i6qp1220000a6
Try to
add try to add these lines to your config
voice service voip
no ip address trusted authenticate
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
11-26-2013 11:57 AM
Dont have that option under voice service voip.
CX-REN-VGW01#conf t
Enter configuration commands, one per line. End with CNTL/Z.
CX-REN-VGW01(config)#voice ser
CX-REN-VGW01(config)#voice service voip
CX-REN-VGW01(conf-voi-serv)#no ip ?
% Unrecognized command
CX-REN-VGW01(conf-voi-serv)#no ip
11-26-2013 12:01 PM
Which ios version are you running on your vg?
Please let me know
Regards
Carlo
Sent from Cisco Technical Support iPhone App
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