Greeting to all:
It is a long time since my last post. I was preparing for my CCIE R&S Written exam and I have finally passed it. Now it is time to get my hand dirty with R&S lab preparation.
Anyway... we have finally won this project and I will be taking care of implementation. After discussing with customer, we did some onsite testing with these two PABXes. The equipment used for proof-of-concept purpose only & includes Cisco 2821 and 2811 series router with VIC-2MFT-703 cards and Cisco 12.4(20)T and 12.4(3h) SP Services feature sets.
Let 2821 be Router-A and 2811 be Router-B and Nortel 11C has 12xx series of extensions and 61c has 2XXX, 3XXX, 4XXX, 5XXX, 6XXX, 7XXX series of phone extensions. Another detail that I want to point out the 61C has Class of Restrictions applied and PSTN trunk termination. Since calls will be traversing WAN, we need quality of services (QoS) and off-course the codec used is G.729. I calculated the bandwidth requirement with Cisco TAC VoIP Bandwidth calculator located at:
You might need you CCO login for this website.
The customer is running EIGRP but right now I am not concerned with routing and QoS details, but he did stated that the total number of call should be under 512kbps.
The respective configuration are as follows:
2821_Router-A:
!! Connected with 11C PABX
network-clock-participate wic 3
hostname 2821_Router-A
!
voice class codec 1
codec preference 1 g729r8
!
controller E1 0/1/0
channel-group 0 timeslots 1-31
!
controller E1 0/3/0
pri-group timeslots 1-31
!
interface Serial0/1/0:0
ip address 192.168.1.1 255.255.255.0
!
interface Serial0/3/0:15
no ip address
no logging event link-status
isdn switch-type primary-qsig
isdn incoming-voice voice
no cdp enable
!
dial-peer voice 1200 pots
description ** to 11c extensions **
destination-pattern 12..
port 0/3/0:15
forward-digits all
!
dial-peer voice 2000 voip
description ** to 61c extensions **
destination-pattern ....
voice-class codec 1
session target ipv4:192.168.1.2
!
end
2811_Router-B:
!! Connected with 61C PABX
network-clock-participate wic 3
hostname 2811_Router-B
!
voice class codec 1
codec preference 1 g729r8
!
controller E1 0/1/0
channel-group 0 timeslots 1-31
!
controller E1 0/3/0
pri-group timeslots 1-31
!
interface Serial0/1/0:0
ip address 192.168.1.2 255.255.255.0
!
interface Serial0/3/0:15
no ip address
no logging event link-status
isdn switch-type primary-qsig
isdn incoming-voice voice
no cdp enable
!
dial-peer voice 2000 pots
description ** to 61c extensions **
destination-pattern ....
port 0/3/0:15
forward-digits all
!
dial-peer voice 1200 voip
description ** to 11c extensions **
destination-pattern 12..
voice-class codec 1
session target ipv4:192.168.1.2
!
end
FINDINGS:
Issue Number 1:
After configuring the PRI, I was continuously getting channel not available error message with Q931 debugs. Actually PABX was not configured with full PRI channels. So I asked the PABX admin and he fixed it.
Issue Number-2:
Although call were working fine but the 11c user could not hear the voice of 61c PBX users, that is it was a ONE-WAY-AUDIO-ISSUE. So what to do... I looked at the routing and verified that it is two way (and it was) so what could be the problem. May be it is the DELAY. I confirmed with ISP and doing some ping with 256 and 512 payload sized and found it under 80ms. Now it is time to do some google. I found a very good document at Cisco website. The ur l is :
Reading through the document I find out it was a Cut Through Audio problem. So the command: voice rtp send-recv on Router-B solved the problem.
Issue Number 3:
After connecting with voice gateway there is some delay in dialing extensions. This delay is eliminated as soon the the pound (#) key is pressed. Also if both PABXes are connected directly with E1, this post-dial-delay is eliminated. I am still scratching my head and hopefully this will be resolved.
Sometimes there are other thing that you must do to keep the customer happy and I call this Farmaeshi Program (in URDU) and most of the time the sales guy commits such things. So he asked if the calling name can be displayed. I was thinking that isdn supp-service name calling should solve the problem but it did not. So I have read some docs to make it happen. One thing that did came to my mind if the PABXes were sending the Calling Name and after confirming they were not. I will definitely post the solution once this issue is resolved as well.
I am looking forward for comments or suggestions on this post.
