Configuration for SIP Trunking between Avaya IP Office and Cisco CallManager Express 7.1

[femi.agboade]

Hi,

I have the task of allowing calls route between an Avaya IP Office and a CCME running on a 3825 router. I have both devices on the same subnet, but I have been unsuccessful so far in getting calling routed between them.

I would appreciate any pointers in getting this done as I would rather start from scratch.

All comments and suggestions are very welcomed.

Regards,

Femi

[acampbell]

Hi,

I am not sure about the Avaya side

May be this document will help

http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns829/785594.pdf

HTH

Alex

[William Bell]

I haven't seen an integration guide specific to Avaya IP Office and CME but here is an avaya technote on SIP trunks between IP Office and CUCM.

https://devconnect.avaya.com/public/flink.do?f=/public/download/interop/CUCM8IPO61SIPtrk.pdf

Maybe you can use the above in conjunction with Alex's link (+5 A.) to get where you need to be.

HTH.

Regards,

Bill

[femi.agboade]

Hello William and Alex,

Thank you both for the URLs. I am presently looking into it to try and meet a common for my own scenario, considering that both URLs are applicable only to a part of my setup.

I'll give it a go and revert as soon as I make progress or not

Regards,

Femi

[femi.agboade]

Hello again William and Alex,

I have taken time to study all the materials on those URLs and even plenty more but i dont see any that directly addresses my scenario. Below is the config i have on the CME router now:

!

voice-card 0

dspfarm

dsp services dspfarm

!

voice-card 2

dspfarm

dsp services dspfarm

!

voice service voip

no notify redirect ip2ip

qsig decode

redirect ip2ip

sip

registrar server

g729 annexb-all

voice class codec 1

codec preference 4 g729br8

!

!

dial-peer voice 1 voip

description TO LAGOS

destination-pattern 1....

voice-class sip g729 annexb-all

session protocol sipv2

session target ipv4:10.65.0.5

session transport tcp

incoming called-number 17...

dtmf-relay rtp-nte

!

dial-peer voice 2 voip

description TO WARRI

destination-pattern 2....

voice-class sip g729 annexb-all

session protocol sipv2

session target ipv4:10.65.0.5

session transport tcp

incoming called-number 17...

dtmf-relay rtp-nte

!

dial-peer voice 3 voip

description TO PORTHARCOURT

destination-pattern 3....

voice-class sip g729 annexb-all

session protocol sipv2

session target ipv4:10.65.0.5

session transport tcp

incoming called-number 17...

dtmf-relay rtp-nte

!

dial-peer voice 4 voip

description TO KADUNA

destination-pattern 4....

voice-class sip g729 annexb-all

session protocol sipv2

session target ipv4:10.65.0.5

session transport tcp

incoming called-number 17...

dtmf-relay rtp-nte

!

dial-peer voice 5 voip

description TO MOSIMI/ORE

destination-pattern 5....

voice-class sip g729 annexb-all

session protocol sipv2

session target ipv4:10.65.0.5

session transport tcp

incoming called-number 17...

dtmf-relay rtp-nte

!

dial-peer voice 7 voip

description TO BENIN

destination-pattern 7....

voice-class sip g729 annexb-all

session protocol sipv2

session target ipv4:10.65.0.5

session transport tcp

incoming called-number 17...

dtmf-relay rtp-nte

!

dial-peer voice 8 voip

description TO ABUJA

destination-pattern 8....

voice-class sip g729 annexb-all

session protocol sipv2

session target ipv4:10.65.0.5

session transport tcp

incoming called-number 17...

dtmf-relay rtp-nte

!

So the issue i have now is getting the Avaya IP Office setup for same.

Regards,

Femi

[acampbell]

Hi,

I think you should add these few lines to enable internetworking

!

voice service voip

allow-connections sip to sip

allow-connections sip to h323

allow-connections h323 to sip

sip

bind control source-interface

bind media source-interface

!

HTH

alex

[femi.agboade]

Hi Alex,

Thanks for the feedback.

What interface is referred to in these command lines:

bind control source-interface

bind media source-interface

!

I have 3 interfaces on the router, an interface that is connected to the LAN which services LAN users to the CME, while I also have another interface that connects to a point-to-point link that has the Avaya IPO on the other end of that link. SO which of these interfaces would it be?

Regards,

Femi

[Aaron Harrison]

Hi

It doesn't generally matter which, as long as it's a reliable interface.

It does have to match the config on other devices however - so if you use Fa0/0 as the bind address for example, ensure that any device configured to send calls to this kit uses the IP of fa0/0 or it may not recognise packets coming from this router as being from this router.

Aaron

[femi.agboade]

Hi Aaron,

Not sure I follow.

Assuming Fa0/0 is my LAN interface where the Cisco IP phones are connecting through to get to the CME.

Fa0/1 is the interface that can reach the Avaya IPO.

On my SIP config, I have a dial peer that points to the Avaya IPO which shares the same ip subnet as Fa0/1.

From your comment, does this imply that the bind address is the Avaya IPO's address which is reachable via the Fa0/1? If so, then I guess the interface to quote here will be the Fa0/1 interface?

Secondly, how do you ensure that a device uses the IP of Fa0/1?

Thanks.

Regards,

Femi

[Aaron Harrison]

Hi

The bind commands mean that the router will set it's source address to the bind address. So if you put fa0/1, then when a call is initiated to the Avaya the source address in the packets at the application level (I.e. in the SIP Invite etc) will be the address of fa 0/1.

Typically on an internal deployment both the LAN and WAN interface (yoru Fa 0/0 and Fa0/1) would both be able to reach the avaya - if this is the case it's up to you which you use. If only one interface can reach it, then use that one.

Either way - when you refer to the CME gateway by IP address in the Avaya config, refer to the address you have bound to. If not, the Avaya may see the inbound invite as from an 'untrusted' or 'uknown' source, and reject it.

Aaron

[femi.agboade]

Hi Aaron,

Thanks for that explantion!

I'll take a look at this in the morning and revert later on.

Cheers.

Femi

[femi.agboade]

Hi all,

I have applied the configs but still no luck with the calls. Is there a way I can check that the SIP Trunk is actually established and also monitor calls going through or attempting to go through that trunk?

I also included these command lines:

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

dial-peer voice 1 voip

voice-class codec 1

dial-peer voice 2 voip

voice-class codec 1

!

The guys configuring the Avaya IPO also mentioned that they had g711alaw configured on the box, does it affect my configuration in anyway such that I may need to change codec parameters or preferences?

Regards,

Femi

[Aaron Harrison]

Hi

It's not so much a 'trunk' really, as a loose idea of a connection - so there's no status exchange. When you make a call you can run a debug to see the SIP messages being exchanged...

try:

debug ccsip messages

debug ccapi inout

Expect a lot of output so log it to a file from putty, over an SSH/telnet session rather than console. If this router is in production you might want to turn off logging to the console (no logging console) to save CPU and risk of problems...

Once the debugs are running just make a test call and see what messages are exchanged...

Aaron

[femi.agboade]

Hi Aaron,

I had a tough time going through the messages as it was in hundreds. I'll keep at it and revert. One question I have though, I just realized I didn't configure "sip-ua". Am I supposed to configure this or not?

Regards,

Femi

[femi.agboade]

Hi there,

So I have made a few changes and there seems to be some improvements. I changed from SIP to H323 on the CME and I am now able to call extensions on the Avaya IPO system. The bad news is when the call gets through and someone picks up, neither of us can hear each other. The call then disconnects automatically after 10 secs. Below is a snippet of my config now:

!

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service h450.2

no supplementary-service h450.3

supplementary-service h450.12

fax protocol cisco

!

!

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

!

!

interface FastEthernet0/0/0

ip address 10.65.0.9 255.255.0.0

duplex auto

speed auto

h323-gateway voip interface

h323-gateway voip id CO ipaddr 10.65.0.5 1719

h323-gateway voip h323-id CO

h323-gateway voip tech-prefix 1#

h323-gateway voip bind srcaddr 10.65.0.9

!

!

dial-peer voice 1 voip

description TO LAGOS

destination-pattern 1....

session target ipv4:10.65.0.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 2 voip

description TO WARRI

destination-pattern 2....

session target ipv4:10.65.0.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 3 voip

description TO PORTHARCOURT

destination-pattern 3....

session target ipv4:10.65.0.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 4 voip

description TO KADUNA

destination-pattern 4....

session target ipv4:10.65.0.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 5 voip

description TO MOSIMI/ORE

destination-pattern 5....

session target ipv4:10.65.0.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 7 voip

description TO BENIN

destination-pattern 7....

session target ipv4:10.65.0.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 8 voip

description TO ABUJA

destination-pattern 8....

session target ipv4:10.65.0.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

!

gateway

timer receive-rtp 1200

!

Any suggestions?

Regards,

Femi