[cisco-voip] ISDN T1 voice calls hardware validation
PF
pucknether at foril.com
Wed May 9 10:29:16 EDT 2018
Hi again
We are preparing the hardware request
I am not sure, should we get NIM-8MFT-T1/E1 modules or NIM-8CE1T1-PRI modules ?
the clear-channel vs channellized channel is not clear for me
thanks
Pat
----- Original Message -----
From: Andrew Dorsett
To: PF
Cc: Ryan Huff ; cisco-voip at puck.nether.net
Sent: Friday, May 04, 2018 3:39 PM
Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation
Pat -
Also let’s look at the hypothetical ASR you’re wanting versus the ISR.
The ASR1006 supports 12 half height SPAs in 6RU of space.
To keep this simple and not get into codec complexity, let’s assume that the DSP SPA is matched 1:1 with the 8xT1 SPA.
That gives us 6 8xT1 SPAs providing a total of 48 T1 ports and 1,104 PRI channels in the 6RU.
The ISR4451 supports 5 8xT1 NIMs providing a total of 40 T1 ports and 920 channels in 2RU.
Three ISR4451 boxes would give us a total of 120 T1 ports versus the 48 in a single ASR1006 for the same rack density.
Hopefully you see the reason why the ASR just isn’t the right platform for voice T1 termination. Now SBC is a totally different story and you can chock it full of DSP SPAs and make a huge transcoder which the ISR can’t do because it doesn’t have enough DSPs.
Andrew
On Fri, May 4, 2018 at 2:47 PM PF <pucknether at foril.com> wrote:
ok
I guess we will have to look for another product like the ISR4k
the ASR1006 was very interresting because it was full redundant
and we could put several 8xT1 card in it to have greater density
the ISR4k seems to be limited to 3 cards and is not redundant.
Pat
----- Original Message -----
From: Andrew Dorsett
To: Ryan Huff
Cc: PF ; cisco-voip at puck.nether.net
Sent: Friday, May 04, 2018 12:45 PM
Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation
I replied privately but the SPA-DSP was built for SBC transcoding using CUBE between SIP or H323 trunks. Unless something has changed recently it cannot be linked to the T1 SPA to terminate PRI Voice.
To terminate voice from a PRI you will need to use one of the ISR products and the relevant T1 module with DSPs. If you’re using the latest it would be an ISR4k with the NIM first mentioned and make sure the DSPs are onboard the NIM.
Andrew
On Fri, May 4, 2018 at 12:32 PM Ryan Huff <ryanhuff at outlook.com> wrote:
Not sure if DSP is onboard the SPA like the NIM; I’d have to look that one up ... you might try dsp services dspfarm under the voice-card configuration parameter. I’d also disable cdp on the serial interface.
Sent from my iPhone
On May 4, 2018, at 12:18, PF <pucknether at foril.com> wrote:
Hi
I have tried several configurations
Here is the relevent par of the actual config
version 15.3
boot system flash bootflash:asr1000rp2-adventerprisek9.03.10.07.S.153-3.S7-ext.bin
!
card type t1 0 2
!
multilink bundle-name authenticated
isdn switch-type primary-5ess
!
voice-card 0/0
!
voice service pots
supported-language FR
!
voice service voip
clid network-provided
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
signaling forward unconditional
sip
bind control source-interface GigabitEthernet1/2/0
bind media source-interface GigabitEthernet1/2/0
ds0-num
header-passing
privacy id
outbound-proxy ipv4:172.16.x.y
!
voice class codec 1
codec preference 1 g711ulaw
!
controller T1 0/2/0
framing esf
clock source internal
linecode b8zs
cablelength long 0db
pri-group timeslots 1-24
!
interface Service-Engine0/0/0
!
interface Serial0/2/0:23
encapsulation hdlc
isdn switch-type primary-5ess
isdn negotiate-bchan
!
map-class dialer DOVtest
dialer voice-call
!
dspfarm profile 1 transcode universal
rsvp
shutdown
!
dial-peer voice 1 voip
destination-pattern *.
signaling forward unconditional
session protocol sipv2
session target sip-server
voice-class codec 1
!
dial-peer voice 2 pots
destination-pattern xxxxxxx
incoming called-number xxxxxxx
direct-inward-dial
!
!
sip-ua
credentials username asr1006 password 7 xxxxxxxx realm yyyyy
retry invite 3
retry bye 3
retry cancel 3
timers trying 1000
timers register 100
sip-server ipv4:172.16.x.y
!
!
Pat
----- Original Message -----
From: Ryan Huff
To: Patrick Fortin
Cc: cisco-voip at puck.nether.net
Sent: Friday, May 04, 2018 11:43 AM
Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation
I assume you have the card type specified and controller interface configured with timeslots?
Can you send the running-config and code version?
Sent from my iPhone
On May 4, 2018, at 11:35, Patrick Fortin <pfortin at royaume.com> wrote:
Hi
I get this error
**ERROR**: call_incoming: Received a call id 0x2F with a bad bearercap from xxxxxxxxxx on b channel 1
seems like the dsp are not associated with the t1 card
in the config we don't have acces to the following :
isdn incoming-voice voice
which should go on the Serial interface.
we also don't have the "port" command that should go in the dial-peer voice section
any ideas ?
Thanks
Pat
----- Original Message -----
From: Ryan Huff
To: PF
Cc: cisco-voip at puck.nether.net
Sent: Friday, May 04, 2018 11:11 AM
Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation
That correct! I misread! Yeah the shared port adapter T1 should work.
Sent from my iPhone
On May 4, 2018, at 10:36, PF <pucknether at foril.com> wrote:
Hi
Thanks for your help
But there are no NIM slot in the ASR1006 chassis
NIM is for ASR1001-X I think
Pat
----- Original Message -----
From: Ryan Huff
To: PF
Cc: cisco-voip at puck.nether.net
Sent: Friday, May 04, 2018 9:56 AM
Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation
For isdn voice you’ll need a NIM-8MFT-T1/E1
Sent from my iPhone
On May 4, 2018, at 09:31, PF <pucknether at foril.com> wrote:
Hi
Can someone help us validate if we can use this hardware to receive voice calls from a isdn T1 (23B+D) and send them in SIP to a softswitch and vice-versa
ASR1006
SPA-8XCHT1/E1
SPA-DSP
ASR1000-RP2
ASR1000-ESP40
in short can it be used to build a voip gateway like an audiocode mediant or a patton smartnode
Thanks
Pat
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