topic Re: SIP Trunk between CME and CME in Unified Communications Infrastructure

SIP Trunk between CME and CME

Vector5678 — Sun, 16 Jun 2019 15:56:15 GMT

I have two CME - 2911 which version 10.0 and version 12.0. I try to SIP trunk between two CME each other but I encountered some problems as follows:

- I can call from SCCP phone (registered CME-1 version 10.0) to SCCP phone (registered CME-2 version 12.0) and can call from SCCP phone (registered CME-2 version 12.0) to SCCP phone (registered CME-1 version 10.0)

- I can call from SIP phone, SCCP phone (registered CME-2 version 12.0) together

- I can't call from SIP phone, SCCP phone (registered CME-2 version 12.0) to SIP phone (registered CME-1 version 10) as well as call from SIP phone, SCCP phone(registered CME-1 version 10) to SIP phone and SCCP phone (registered CME-2 version 12.0).

Please help me this case.

Thanks.

Re: SIP Trunk between CME and CME

Maren Mahoney — Mon, 17 Jun 2019 16:11:59 GMT

Let me say first that you don't have a SIP trunk between these routers. The default VoIP protocol on a Cisco router is H.323, so that is the protocol used unless you add session protocol sipv2 to the dial-peers connecting the two systems.

Also, the default codec for VoIP calls on a Cisco router is G729. Without a transcoder, and with the SIP phones on HST (but not on TLJOC) set to use G711, that could cause call failure with the dial-peers all at the default of G729. So I suggest applying the voice-class codec 1 to the inter-site dial-peers on TLJOC, and also creating that voice class codec 1 on the HST router and applying that to the dial-peers on HST.

SCCP phones use G711 by default (but will support G729). I see the phones on HST have a G711 codec specified for the SIP phones, but the SIP phones on TLJOC are still at the default of G729. Is this by design? If not, I suggest making things consistent.

Give these suggestions a look and let us know how it goes.

Maren

Re: SIP Trunk between CME and CME

Vector5678 — Mon, 24 Jun 2019 07:25:16 GMT

Hi,

I already applied the command "Session Protocol Sipv2" on dial-peer from HST to TLJOC and from TLJOC to HST.

- After that I can't call from SIP phone (registered TLJOC) to SCCP phone (registered HST).

- when I call from SCCP phone (registered HST) to SIP phone (registered TLJOC). In LCD of SCCP phone displays connected but SIP phone doesn't ring out. Please support me.

Thanks.

Re: SIP Trunk between CME and CME

Maren Mahoney — Mon, 24 Jun 2019 10:34:07 GMT

You did the first of four things that I suggested you look at, the other three are codec-related:

Apply the voice-class codec 1 to the inter-site dial-peers on TLJOC
Create that same voice class codec 1 on the HST router and apply that to the inter-site dial-peers on HST
Set the SIP phones on TLJOC to use G711

Once that's done, please run debug ccsip messages and place the two non-working calls (TLJOC-SIP to HST-SCCP and HST-SCCP to TLJOC-SIP) and let us know what you find.

Maren

Re: SIP Trunk between CME and CME

Vaijanath Sonvane — Mon, 24 Jun 2019 14:38:05 GMT

Hi,

On HST_PABX router, the dial-peers are pointing to session target ipv4:192.168.20.3; but CME IP Address on TLJOC_PABX router is 192.168.34.1. I would suggest changing this session target to ipv4:192.168.34.1. Make sure the IP Address 192.168.34.1 is reachable from HST_PABX router.

Similarly, on TLJOC_PABX router, the dial-peers are pointing to session target ipv4:192.168.21.2. but CME IP Address on HST_PABX is 192.168.31.1.I would suggest changing this session target to ipv4:192.168.31.1. Make sure the IP Address 192.168.31.1 is reachable from TLJOC_PABX router.

Here is the configuration for your reference:

HST_PABX Router Configuration:

voice service voip
 no ip address trusted authenticate
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 1 g729r8
!
!
dial-peer voice 1000 voip
 description ** INBOUND CALLS FROM TLJOC_PABX **
 session protocol sipv2
 incoming called-number .
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay cisco-rtp rtp-nte
 no vad
!
dial-peer voice 1001 voip
 description ** OUTBOUND CALLS TO TLJOC_PABX **
 destination-pattern 4..$
 session protocol sipv2
 session target ipv4:192.168.34.1
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay cisco-rtp rtp-nte
 no vad
!

TLJOC_PABX Router Configuration:

voice service voip
 no ip address trusted authenticate
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 1 g729r8
!
!
dial-peer voice 4000 voip
 description ** INBOUND CALLS FROM HST_PABX **
 session protocol sipv2
 incoming called-number .
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay cisco-rtp rtp-nte
 no vad
!
dial-peer voice 4001 voip
 description ** OUTBOUND CALLS TO HST_PABX **
 destination-pattern [12]..$
 session protocol sipv2
 session target ipv4:192.168.31.1
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay cisco-rtp rtp-nte
 no vad
!

Re: SIP Trunk between CME and CME

Vector5678 — Tue, 25 Jun 2019 09:44:52 GMT

hi,

Thank for your support.

After I revised my configuration follow your advise. I have a new problem:

- when I call from SCCP phone (registered HST) to SIP phone (reigistered TLJOC) or I call from SIP phone (registered TLJOC) to SCCP phone (registered HST) or I call from SCCP phone (registered TLJOC) to SIP phone (registered HST), SIP phone or SCCP will be ringged out. But, all phones registered to HST-PABX will be unregistered.

I can't know that what's happen?

Can you help me?

Thanks.

Re: SIP Trunk between CME and CME

Vaijanath Sonvane — Tue, 25 Jun 2019 12:10:22 GMT

So, are you saying that the call are working between HST and TLJOC and the new problem is all phones registered to HST are unregistered.

Re: SIP Trunk between CME and CME

Vector5678 — Wed, 26 Jun 2019 04:56:23 GMT

hi,

Yes, correct. All telephones will be unregistered when the call are working between HST and TLJOC and request timed out when ping to IP address of PABX-HST.

Re: SIP Trunk between CME and CME

Vaijanath Sonvane — Wed, 26 Jun 2019 05:15:16 GMT

Just to reconfirm again, are the calls between below phones working?

SCCP Phone (CME-1 v10.0) to SCCP Phone (CME-2 v12.0)
SCCP Phone (CME-1 v10.0) to SIP Phone (CME-2 v12.0)
SIP Phone (CME-1 v10.0) to SCCP Phone (CME-2 v12.0)
SIP Phone (CME-1 v10.0) to SIP Phone (CME-2 v12.0)

Similarly;

SCCP Phone (CME-2 v12.0) to SCCP Phone (CME-1 v10.0)
SCCP Phone (CME-2 v12.0) to SIP Phone (CME-1 v10.0)
SIP Phone (CME-2 v12.0) to SCCP Phone (CME-1 v10.0)
SIP Phone (CME-2 v12.0) to SIP Phone (CME-1 v10.0)

Are all the phones at HST are un-registering when you are on call with TLJOC Phones? From which device/location are you trying to ping HST-PABX Router? Are you sure you do not have network connectivity or routing issue?

Re: SIP Trunk between CME and CME

Vector5678 — Wed, 26 Jun 2019 07:22:37 GMT

hi,

Yes, I confirm that my issue occur when:

- SCCP Phone (CME-1 v10.0) to SCCP Phone (CME-2 v12.0)
- SCCP Phone (CME-1 v10.0) to SIP Phone (CME-2 v12.0)
- SIP Phone (CME-1 v10.0) to SCCP Phone (CME-2 v12.0)
- SIP Phone (CME-1 v10.0) to SIP Phone (CME-2 v12.0)
Similarly;

- SCCP Phone (CME-2 v12.0) to SCCP Phone (CME-1 v10.0)
- SCCP Phone (CME-2 v12.0) to SIP Phone (CME-1 v10.0)
- SIP Phone (CME-2 v12.0) to SCCP Phone (CME-1 v10.0)
- SIP Phone (CME-2 v12.0) to SIP Phone (CME-1 v10.0)
I upgraded TLJOC-PABX version 10.0 to version 12.0, but can not resolve my issue.

Are all the phones at HST are un-registering when you are on call with TLJOC Phones?

=> yes, correct.

From which device/location are you trying to ping HST-PABX Router?

=> I ping from Router HST (please refer to Block diagram in attached)

Are you sure you do not have network connectivity or routing issue?

=> I sure that do not have network connectivity or routing issue.

Re: SIP Trunk between CME and CME

Vaijanath Sonvane — Wed, 26 Jun 2019 12:27:15 GMT

I guess you accepted this as solution by mistake. Please attach your block diagram and updated configuration. Please clarify what is not working. Also, before you test your calls, start below debugs on both CME routers and test calls for non working scenario:
debug ccsip message
debug voip ccapi inout
debug ephone register
debug voice register errors
debug voice register events