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SIP Trunk, Cisco 3825 and CUCM 6.1

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

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VIP Super Bronze

SIP Trunk, Cisco 3825 and CUCM 6.1

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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Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
28 REPLIES

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

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

Please rate all helpful posts "The more you help the more you learn"
New Member

SIP Trunk, Cisco 3825 and CUCM 6.1

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

VIP Super Bronze

SIP Trunk, Cisco 3825 and CUCM 6.1

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

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

!

Cisco Employee

SIP Trunk, Cisco 3825 and CUCM 6.1

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

New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

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.

VIP Super Bronze

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

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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Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

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"

Please rate all helpful posts "The more you help the more you learn"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

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.

SIP Trunk, Cisco 3825 and CUCM 6.1

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

Please rate all helpful posts

"The more you help the more you learn"

Please rate all helpful posts "The more you help the more you learn"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1


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

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

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"

Please rate all helpful posts "The more you help the more you learn"
New Member

SIP Trunk, Cisco 3825 and CUCM 6.1

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

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

Which ios version are you running on your vg?

Please let me know
Regards

Carlo

Sent from Cisco Technical Support iPhone App

Please rate all helpful posts "The more you help the more you learn"
New Member

SIP Trunk, Cisco 3825 and CUCM 6.1

CX-REN-VGW01#sh ver

Cisco IOS Software, 3800 Software (C3825-IPVOICEK9-M), Version 12.4(20)T1, RELEASE SOFTWARE (fc3)

Technical Support:

http://www.cisco.com/techsupport

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

Ok.
That's why you don't have that command ( introduced on 15.1)
Anyway try to check again with your provider why they are rejecting your call.

Let me know

Regards

Carlo

Sent from Cisco Technical Support iPhone App

Please rate all helpful posts "The more you help the more you learn"
VIP Super Bronze

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

Looking at the disconnect cause code of 57..This implies bearer capability not authorized. I suggest tha you offer your provider more than one codec...You are offering then only G711ulaw..I though you had voice class codec on your dial-peer. It looks like you have removed it and hard coded g711ulaw. I suggest you hard code g711alaw to your dial-peer and try again and also ensure that you include g711alaw in your voice class codec..

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

I did not remove the voice class codec. I added the codec g711alaw to the voice class as sugggested but still unable to place a call. See config below.

voice class codec 1

codec preference 1 g711ulaw bytes 160

codec preference 2 g711alaw bytes 160

codec preference 3 g729br8 bytes 20

codec preference 4 g729r8 bytes 20

!

dial-peer voice 901 voip

voice-class codec 1

incoming called-number 2T

dtmf-relay h245-signal h245-alphanumeric

no vad

!

dial-peer voice 900 voip

destination-pattern 2T

voice-class codec 1

voice-class sip profiles 1

session protocol sipv2

session target ipv4:66.249.145.144

incoming called-number 7248...

VIP Super Bronze

SIP Trunk, Cisco 3825 and CUCM 6.1

Ok, try this..

voice service voip

sip

early-offer forced

Then send me only debug ccsip messages

Please rate all useful posts

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

Ok, made the requested change. Find attached the debug ccsip messages.

VIP Super Bronze

SIP Trunk, Cisco 3825 and CUCM 6.1

OK, something is not quite right..The CUBE is not offering all the codecs to the ITSP...Please renove the voice class codec on the dial-peer to ITSP ( I believe its 900) and hard code codec g711alaw..

eg..

dial-peer voice 900 voip

no voice-class codec 1

codec g711alaw...

send me debug cccsip messages again

Please rate all useful posts

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

Changed the config as requested, I am unable to call and not getting any ccsip messages. I think my client is set to g729.

Also tried changing my client to g711 but still no ccsip messages.It looks like my client (softphone) is only talking g711ulaw.

VIP Super Bronze

SIP Trunk, Cisco 3825 and CUCM 6.1

You should check the region seeting between your softphone and the gateway..ensure it is set to 64Kbps...

Ok..please send the full logs now..debug h225 asn1 and debug h245 asn1 as well as debug voip ccapi inout

Please rate all useful posts

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

Here you go, dial-peer config below. Debug messages attached but not getting ccsip messages.

dial-peer voice 900 voip

destination-pattern 2T

voice-class sip profiles 1

session protocol sipv2

session target ipv4:66.249.145.144

incoming called-number 7248...

codec g711alaw

VIP Super Bronze

SIP Trunk, Cisco 3825 and CUCM 6.1

Can you check on CUCM if you have fast start configured? What codec is fast start set to use? Can you change this to G711alaw?

Please rate all useful posts

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP Trunk, Cisco 3825 and CUCM 6.1

Yes, it was set to fas start with g711ulaw, changed it g711alaw. Still unable to call but I am now getting ccsip messages (see attached).

VIP Super Bronze

SIP Trunk, Cisco 3825 and CUCM 6.1

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

SIP Trunk, Cisco 3825 and CUCM 6.1

Its working, it was an issue with the SIP account. The ITSP corrected the issue with the account.

Now, is it possible to configure multiple SIP accounts to the same ITSP as we want all our companies to dial out using their own SIP account.

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