11-11-2011 03:08 AM - edited 03-16-2019 08:00 AM
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
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11-29-2011 03:20 AM
Hi
This could be one of a few things... it's possible it's a routing problem or perhaps a firewall or other security restriction...
For example, whilst the two systems (CME router, Avaya) can reach each other, the actual phones might not be able to reach each other. Can you ping a phone in one system directly from the voice VLAN of the other?
Are there any security devices/ACLs in between that migth need you to permit traffic direct from the Cisco phones to the Avaya phones/endpoint?
11-11-2011 03:23 AM
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
11-11-2011 04:59 AM
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
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11-11-2011 09:19 PM
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
11-14-2011 05:56 AM
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
11-14-2011 06:21 AM
11-14-2011 02:03 PM
Hi Alex,
Thanks for the feedback.
What interface is referred to in these command lines:
!
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
11-14-2011 02:08 PM
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
11-14-2011 03:29 PM
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
11-14-2011 03:36 PM
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
11-14-2011 03:56 PM
Hi Aaron,
Thanks for that explantion!
I'll take a look at this in the morning and revert later on.
Cheers.
Femi
11-15-2011 10:41 AM
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
11-15-2011 12:26 PM
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
11-16-2011 12:41 PM
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
11-29-2011 03:11 AM
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
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