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SIP Trunk, Cisco 3825 and CUCM 6.1

Alexis Sulbaran
Level 1
Level 1

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                  

1 Accepted Solution

Accepted Solutions

Alexis,

Now you need to speak to ITSP. We have tried a few things and they are still rejecting your call.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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View solution in original post

28 Replies 28

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

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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

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

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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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

!

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>

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>

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-3804ac30000a6

Call-ID: 12A891D7-55CF11E3-8D9ABF57-49F66684@172.17.30.5

CSeq: 101 INVITE

Timestamp: 1385469761

HTH,

Jagpreet Singh Barmi

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.

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 "<>" "" where XXXX = authentication username

request INVITE sip-header To modify "<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

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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

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"The more you help the more you learn"

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Ok, made the changes as suggested and applied the sip profile to the dial-peer 900. Still unable to call, attached the debug logs. Please let me know what else is missing, thanks for your excellent help.

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>" "<>"

request INVITE sip-header To modify "<2>" "<>usinesssolutions.digicelcuracao.net>"

After changes plese post again a debug ccsip messages

Thanks

HTH

Regards

Carlo

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"The more you help the more you learn"

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Ok, made the change to the sip profile. (To modify). Unable to call still, see attached logs.

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

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"The more you help the more you learn"

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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

Which ios version are you running on your vg?

Please let me know
Regards

Carlo

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