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New Member

SIP trunk, CUBE and CUCM 8.5

I have a SIP trunk from a TSP terminated on the CUBE. - c2901 gateway connected via H323 to a CUCM 8.5 cluster. The signalling looks OK but I have no voice. The provider seems to support only G.729br8 codec and my phones (c6901 and c6921) seem to talk G.729r8 only so I need to do some transcoding. I'm in the UK btw.Now, I can do it on the CUBE and utilize dspfarm but that is I think only possible with SCCP. How do I register the gateway in CUCM, if it's already there as the H323 gateway?

Is there any way to do the transcoding on the CUCM? Any general advise would be much appreciated.

Cheers

Everyone's tags (7)
3 ACCEPTED SOLUTIONS

Accepted Solutions
VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Ok,

You still have not configured the dial-peer to CUCM to use SIP. But here is what I see..

Sent:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bK7kljtp2088b0cmgv34l0.1

From: ;tag=536c8c54

To: <5811>;tag=44F7878-C23

Date: Wed, 27 Jun 2012 09:38:06 GMT

Call-ID: ODE1Y2E4ZTlhZDg1NjM0MmRhZDE4YmNhNTRiMjhkN2M.

CSeq: 1 INVITE

Require: 100rel

RSeq: 650

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: "Michelle  Walters" <5811>;party=called;screen=no;privacy=off

Contact: <5811>

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 262


v=0

o=CiscoSystemsSIP-GW-UserAgent 5886 9981 IN IP4 146.191.201.41

s=SIP Call

c=IN IP4 146.191.201.41

t=0 0

m=audio 20748 RTP/AVP 18 96

c=IN IP4 146.191.201.41

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

You are sending G729br8 to your provider but they do not support annexb because in their invite we do not see annexb.

INVITE sip:5811@UniWestScot:5060 SIP/2.0

Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bK7kljtp2088b0cmgv34l0.1

Max-Forwards: 68

Contact:

To: <5811>

From: ;tag=536c8c54

Call-ID: ODE1Y2E4ZTlhZDg1NjM0MmRhZDE4YmNhNTRiMjhkN2M.

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, REFER, PRACK

Content-Type: application/sdp

Supported: 100rel

P-Asserted-Identity:

Content-Length: 278


v=0

o=Redwood_INX 347015 347015 IN IP4 146.191.243.4

s=Redwood Media Server

c=IN IP4 146.191.243.4

t=0 0

m=audio 50002 RTP/AVP 8 18 96

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:96 telephone-event/8000

a=sendrecv

a=silenceSupp:off

a=ecan:on

a=fmtp:96 0-15

So either configure your cube to transcode from g729br8 to g729r8 or send your provider g729r8

Here is the document to use to setup xcoding on your cube..

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a008092d6b3.shtml

Ensure you add g729br8 in your list of codecs

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VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Hi,

I have looked at the trace in detail and I have an idea why this is not working..

Can you please remove this from your dial-peers to cucm. Please ensure you remove from all dial-peers to cucm and sip provider.

progress_ind setup enable 3

progress_ind alert enable 8

Please do another test call and send me debug ccsip messages.

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

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

Re: SIP trunk, CUBE and CUCM 8.5

The reason why this is not working is because the called number is not 5811 from the sip provider..

Your dial-peer 1 is like this..

dial-peer voice 1 voip

  incoming called-number 5[8-9].. (this is not the incoming called number)

it should be this..

dial-peer voice 1 voip

incoming called-number 01387345[8-9]..

Then when this dial-peer is matched, the xlation will be applied.

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

SIP trunk, CUBE and CUCM 8.5

Hi,

I doubt if yourprovide ronly support G729br8. Most providers support a variety of codecs. You can turn on "debug ccsip messages" on your cube and see what codecs your provider is sending to you..

I will suggest you start with this before going down the xcoding route

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

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

SIP trunk, CUBE and CUCM 8.5

Have had a look already. The provider offers PCMA/8000 and G729/8000 codecs which seem to translate to g711a and g729br8 in cisco speak. And the flavour g729br8 is not liked by our phones.

New Member

SIP trunk, CUBE and CUCM 8.5

OK, I've progressed a bit - our phones should do g729br8 so no transcoding needed.

I still cannot get any voice. I have the right codec set up on the CUBE for the SIP trunk and for the CUCM.(g729br8)

I have the CUBE and the phones in regions set up for G.729.

One thing - my CUBE is just an IP host (with 1 ethernet interface) and I have no ip routing. It's on a remote site so I cannot easily turn it on - but I cannot think it would make a difference.

New Member

SIP trunk, CUBE and CUCM 8.5

So now I have only g729br8 codec on the CUBE gateway. I make a inbound call over the SIP trunk, phone rings, when I pick it up (no audio) I can check the Sender codec = G711u and RCVR codec =G711u on the phone. What's going on? The phone is in a different region than the CUBE and the Max Audio bit rate between the 2 regions is 8kbps -> G729. Why does the phone insist on G711u?

SIP trunk, CUBE and CUCM 8.5

Hi pijaracek,

Look like is not a CODEC negotiation problem because the call is established. Are you using Media Flow around or Passtrought?

Please remember to rate useful posts clicking on the stars below. Favor calificar todos las respuestas útiles dando click en las estrellas de mas abajo. ___________________________________________ LinkedIn Profile: do.linkedin.com/in/leosalcie
New Member

SIP trunk, CUBE and CUCM 8.5

No media flow,

I have only

fax protocol pass-through g711ulaw

I'm starting to suspect the config of the gateway in CUCM needs some change

SIP trunk, CUBE and CUCM 8.5

Please post your show run

Please remember to rate useful posts clicking on the stars below. Favor calificar todos las respuestas útiles dando click en las estrellas de mas abajo. ___________________________________________ LinkedIn Profile: do.linkedin.com/in/leosalcie
New Member

Re: SIP trunk, CUBE and CUCM 8.5

Here is the sh run on the CUBE:

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

I am not sure why you want help but you dont want to send over the things that can be used to help. I asked earlier for "debug ccsip messages". If you want us to help..please send the debug and your sh run.

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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, CUBE and CUCM 8.5

Sorry, I didn't get yoy asked for the debug. Here it is:

New Member

SIP trunk, CUBE and CUCM 8.5

I suspect now the SIP trunk and the CUBE configuration is OK.

For inbound calls over the SIP trunk I cannot hear the ringing tone although the phone is ringing and then if I redirect in CUCM using

Forward No Answer External

to Voicemail or another internal number, suddenly it works - the redirected call is ringing and there is audio after pick up.

Anything I should try to change in CUCM H323 gateway configuration?

It looks like this:

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Hi first of all I will advise to tidy up your configuration. This two dial-peers are not right.

dial-peer voice 2 voip

description *** INBOUND VOIP PEER ***

incoming called-number .

voice-class codec 1

dtmf-relay h245-alphanumeric

fax rate disable

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 1 voip

description *** INBOUND SIP TRUNK PEER ***

translation-profile incoming SIP-TRUNK-IN

session protocol sipv2

session target ipv4:XXX.XXX.243.4

incoming called-number .

voice-class codec 2

dtmf-relay rtp-nte

no vad

Both dial-peers are used for inbound calls from CUCM and from your SIP provider. Your sip provider sends a sip leg and one of these dial-peers is doing a h323 leg. I will advise that you either configure a specific dial-peer to match incoming calls from your sip provider and configure sip protocol on that dial-peer or you configure a single dial-peer with incoming called number. and session protocl sip. Then configure a sip trunk from your cucm to your cube.

So before we troubleshoot further, please do what I suggest.

The best scenario is to have an and to end sip solution like this..

CUCM-----SIP Trunk--->CUBE------>ITSP.

To do this, you need to configure a sip trunk on cucm to CUBE and then configure this interface like this..(make the IP on the destination siup trunk to be=ip of gig0/1)

voice service voip

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

After configuring this then configure the ff:

dial-peer voice 1 voip

description *** INBOUND DIAL-PEER ***

translation-profile incoming SIP-TRUNK-IN

session protocol sipv2

incoming called-number .

voice-class codec 2

dtmf-relay rtp-nte

no vad

no dial-peer voice 2 voip (remove this dial-peer).

Do a test call and send debug ccsip messages.

If you want to stick with the h323 scenario, your call flow will look like this..

cucm----h323----CUBE---sip--->ITSP

You need to configure the ff: in addition to what you have now

interface GigabitEthernet0/1

h323-gateway voip interface

dial-peer voice 1 voip

description *** INBOUND SIP TRUNK DIAL-PEER ***

translation-profile incoming SIP-TRUNK-IN

session protocol sipv2

incoming called-number 5[8-9]..

voice-class codec 2

dtmf-relay rtp-nte

no vad

dial-peer voice 2 voip

description *** INBOUND VOIP PEER ***

incoming called-number .

voice-class codec 1

dtmf-relay h245-alphanumeric

fax rate disable

ip qos dscp cs3 signaling

no vad

 

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New Member

Re: SIP trunk, CUBE and CUCM 8.5

Thanks for the reply.

I'd like to keep the h323 - sip scenario. So I changed the config as you suggested (I think in dial-peer voice 2 voip you missed session target clause). It still doesn't work. There is no dial tone for the inbound calls, although the phone rings and the call can even be successfully redirected and it rings once for the outbound calls and then the CUBE cancels the call and there is busy tone.

Here is the inbound call debug:

Re: SIP trunk, CUBE and CUCM 8.5

Need to check the config but the symptoms are like the call is matching default dial peer

Sent from Cisco Technical Support iPad App

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Hi,

First of I didnt miss the session target command, I omitted because you dont need. This is an incoming called number dial-peer. You dont not have a destination-pattern on this dial-peer so you dont need session target.

Secondly, I dont see the benefit of you keeping h323.  I helped resolved a problem similar to this on this forum by changing to SIP. So i strongly suggest you consider this..

Thirdly, I cdan see a disconnect cause of 47

Sent:

CANCEL sip:01413330661@146.191.243.4:6280 SIP/2.0

Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK217CC

From: "Michelle  Walters" <01387345811>;tag=40B9354-944

To: <01413330661>

Date: Wed, 27 Jun 2012 08:23:56 GMT

Call-ID: 45C8BC2C-BF6811E1-80BAA4DC-D76223A2@146.191.201.41

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1340785438

Reason: Q.850;cause=47

This is usually points to a codec issue.

We need to know what is happening on the call leg to cucm. Because you are using h323, you need to send debug voip ccapi inout.

I honestly advise changing to sip.

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

SIP trunk, CUBE and CUCM 8.5

OK, if I go SIP from cucm to CUBE, do I add the CUBE in cucm as gateway or as trunk? I'll give it a try. Thanks again. The reason why I started off with h323 is we have other gateways configured like this:

cucm -->h323 -->gateway --> PSTN

I understand it makes sense to keep it SIP all way through.

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Yes you need to add a sip trunk on cucm as I explained in the other post. Thats all. Then configure your sip binding on your gig0/1 interface as I also explained.

Once you have done this, do a test call and send the debug ccsip messages. With this we will see the interactuion between cucm and cube on the debug.

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

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, CUBE and CUCM 8.5

OK, I've made the changes. I also set up the newly created sip trunk in cucm to route outbound calls. Hrer are the traces:

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Ok,

You still have not configured the dial-peer to CUCM to use SIP. But here is what I see..

Sent:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bK7kljtp2088b0cmgv34l0.1

From: ;tag=536c8c54

To: <5811>;tag=44F7878-C23

Date: Wed, 27 Jun 2012 09:38:06 GMT

Call-ID: ODE1Y2E4ZTlhZDg1NjM0MmRhZDE4YmNhNTRiMjhkN2M.

CSeq: 1 INVITE

Require: 100rel

RSeq: 650

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: "Michelle  Walters" <5811>;party=called;screen=no;privacy=off

Contact: <5811>

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 262


v=0

o=CiscoSystemsSIP-GW-UserAgent 5886 9981 IN IP4 146.191.201.41

s=SIP Call

c=IN IP4 146.191.201.41

t=0 0

m=audio 20748 RTP/AVP 18 96

c=IN IP4 146.191.201.41

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

You are sending G729br8 to your provider but they do not support annexb because in their invite we do not see annexb.

INVITE sip:5811@UniWestScot:5060 SIP/2.0

Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bK7kljtp2088b0cmgv34l0.1

Max-Forwards: 68

Contact:

To: <5811>

From: ;tag=536c8c54

Call-ID: ODE1Y2E4ZTlhZDg1NjM0MmRhZDE4YmNhNTRiMjhkN2M.

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, REFER, PRACK

Content-Type: application/sdp

Supported: 100rel

P-Asserted-Identity:

Content-Length: 278


v=0

o=Redwood_INX 347015 347015 IN IP4 146.191.243.4

s=Redwood Media Server

c=IN IP4 146.191.243.4

t=0 0

m=audio 50002 RTP/AVP 8 18 96

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:96 telephone-event/8000

a=sendrecv

a=silenceSupp:off

a=ecan:on

a=fmtp:96 0-15

So either configure your cube to transcode from g729br8 to g729r8 or send your provider g729r8

Here is the document to use to setup xcoding on your cube..

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a008092d6b3.shtml

Ensure you add g729br8 in your list of codecs

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

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

SIP trunk, CUBE and CUCM 8.5

Hi there,

I've replaced the codec g729br8 with g729r8 - it didn't like it at all - our cube SENT SIP/2.0 488 Not Acceptable Media.

Is it possible the provider doesn't support either of the 2 codecs? I'm having difficulties to get a list from them.

Cheers

New Member

SIP trunk, CUBE and CUCM 8.5

What do you mean by

You still have not configured the dial-peer to CUCM to use SIP

Is it replacing voice-class h323 with session protocol sipv2 in the CUCM dial peers 5001, 5002, 5555?

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

configure

dial-peer voice 5002 voip

session protocol sipv2

do the same for all dial-peers to cucm.

do a test call and send "debug ccsip all"

remove the voice class h323 command..

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

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, CUBE and CUCM 8.5

This is debug ccsip /all for the inbound call:

New Member

SIP trunk, CUBE and CUCM 8.5

Hi

I got the outbound calls working.

The pronblem is a codec mismatch.

I added

g729 annexb-all

on the CUBE. However, that doesn't fix the inbound calls.

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Hi, I have looked at the trace again and I can see again the codec issue.

Here is a summary of trace analysis..

+++CUBE sends SDP to ITSP with g729br8+++

Sent:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bKg8plvd1008sg4n86i2p0.1

From: ;tag=5e3b5435

To: <5811>;tag=4D1597C-40E

Date: Wed, 27 Jun 2012 11:59:58 GMT

Call-ID: NmRhNmJlZGJkNjZhMmNiYTk1MmFmMjdlOGU0MzU1NDk.

CSeq: 1 INVITE

Require: 100rel

RSeq: 5815

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <5811>;party=called;screen=yes;privacy=off

Contact: <5811>

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 260


v=0

o=CiscoSystemsSIP-GW-UserAgent 3479 16 IN IP4 146.191.201.41

s=SIP Call

c=IN IP4 146.191.201.41

t=0 0

m=audio 26048 RTP/AVP 18 96

c=IN IP4 146.191.201.41

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

++++CUBE receives 200 ok from ITSP with G729br8+++So ITSP supports G729br8

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK35E20

From: <5811>;tag=4D1597C-40E

To: ;tag=5e3b5435

Call-ID: NmRhNmJlZGJkNjZhMmNiYTk1MmFmMjdlOGU0MzU1NDk.

CSeq: 101 INVITE

Timestamp: 1340798408

Contact:

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, REFER, PRACK

Content-Type: application/sdp

Content-Length: 258


v=0

o=Redwood_INX 432119 432238 IN IP4 146.191.243.4

s=Redwood Media Server

c=IN IP4 146.191.243.4

t=0 0

m=audio 50038 RTP/AVP 18 96

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=sendrecv

a=ptime:20

++++CUBE receives 200 ok from cucm with only g729r8 (no annex-b)+++

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK3021ED

From: ;tag=4D156B4-A23

To: <5811>;tag=107724~58ef7c6e-6d55-43ea-a324-3eb4beb02da5-75265600

Date: Wed, 27 Jun 2012 12:00:12 GMT

Call-ID: 72E7D3B6-BF8611E1-81BEA4DC-D76223A2@146.191.201.41

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence, kpml

Supported: replaces

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires:  1800;refresher=uas

Require:  timer

P-Asserted-Identity: "Michelle  Walters" <5811>

Remote-Party-ID: "Michelle  Walters" <5811>;party=called;screen=yes;privacy=off

Contact: <5811>

Content-Type: application/sdp

Content-Length: 182


v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 146.191.2.102

s=SIP Call

c=IN IP4 146.191.199.29

t=0 0

m=audio 16438 RTP/AVP 18

a=rtpmap:18 G729/8000

a=ptime:20

a=fmtp:18 annexb=no

+++CUBE rejects call and sends a BYE to CUCM with cause code of 65++++++++++++


Sent:

BYE sip:5811@146.191.2.102:5060 SIP/2.0

Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK32180E

From: ;tag=4D156B4-A23

To: <5811>;tag=107724~58ef7c6e-6d55-43ea-a324-3eb4beb02da5-75265600

Date: Wed, 27 Jun 2012 11:59:57 GMT

Call-ID: 72E7D3B6-BF8611E1-81BEA4DC-D76223A2@146.191.201.41

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1340798404

CSeq: 102 BYE

Reason: Q.850;cause=65

Now cause code of 65 is saying that CUBE does not support the codec call manager is trying to use.

Can you please configure your voice class codec to include g729r8 (I am not sure if the voice class codec you are using on dial-pee 5001, 500..is voice class codec 1 or 2. use the correct one.

eg

voice class codec 1 or voice class codec 2 (whichever o ne you are using on the dp to cucm

codec preference 1 g729r8

codec preference 2 g729br8

Please test again and send debug ccsip messages. Please attach debug in a text file dontg post here..cause the thread is getting too long

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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

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, CUBE and CUCM 8.5

Hi,

Thanks a lot for your help.

Stupid but I am unable to find a way how to atach a file to this discussion!

Anyway - the command

g729 annexb-all

did this:

- outbound calls work

- inbound calls still no audio, but codec g729r8 is now accepted by the ITSP.

Here is the inbound trace:

VIP Super Bronze

Re: SIP trunk, CUBE and CUCM 8.5

Ok, Your codec look okay now as I can see that g729r8 is used all the way from CUCM to your ITSP.

I can also see that your cube has 2 IP addresses..

Can you do a "sh voip rtp connection" when you do an inbound call. From your trace I can see the the ff IPs

10.0.48.98..what is this ip?

10.0.16.21..what is this IP?

146.191.243.4---------ITSP IP

146.191.201.41--------CUBE IP

146.191.251.40---------CUCM IP

146.191.199.29-----IP Phone IP

Your RTP connection should look like this..

146.191.201.41------------->146.191.199.29

146.191.201.41-------------->146.191.243.4

Can the CUBE reach the IP of the Phone 146.191.199.29? Can you ping this IP from the CUBE?

Can you please attach the updated config onf your CUBE. To attach click on the advanced editor on the right corner of the writing window. Then at the bottom you will see where to attach a file

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New Member

SIP trunk, CUBE and CUCM 8.5

Hi, yes the RTP connections are

VoIP RTP active connections :

No. CallId     dstCallId  LocalRTP RmtRTP     LocalIP                                RemoteIP

1     267        268        25010    50026    146.191.201.41                         146.191.243.4

2     268        267        31480    16496    146.191.201.41                         146.191.199.29

Found 2 active RTP connections

The 2 private IP addresses are coming from the ITSP. Our CUBE has 1 IP

hcpgw01#sh ip int brief

Interface                  IP-Address      OK? Method Status                Protocol

Embedded-Service-Engine0/0 unassigned      YES NVRAM  administratively down down

GigabitEthernet0/0         unassigned      YES NVRAM  administratively down down

GigabitEthernet0/1         146.191.201.41  YES NVRAM  up                    up

VIP Super Bronze

SIP trunk, CUBE and CUCM 8.5

Can the CUBE reach the IP of the Phone 146.191.199.29? Can you ping this IP from the CUBE?

Can you please attach the updated config onf your CUBE. To attach click on the advanced editor on the right corner of the writing window. Then at the bottom you will see where to attach a file

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

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