[cisco-voip] ISDN T1 voice calls hardware validation

[Andrew Dorsett]

Pat -
I will agree that the ASR1k offers redundant RPs but by the time you
configure it the way you want, you'd probably find dual ISR4k's to be the
same cost.  Just a guess there as I haven't ran the numbers but I know the
ASR comes at quite the premium over the ISR.

The ISR4k offers an 8xT1 NIM and the ISR-4451 supports 5 NIMs (using SM-X
NIM adapter) which gives you 40 T1 ports and 1200 E1 channels or 920 T1
channels in a single 2RU box.

For the ASR to do voice decoding it would have to carry the TDM clock
signal from the T1 SPA to the DSP SPA which doesn't happen.  On the old
ISRG2's and older we carried the TDM clock across the backplane which meant
you had a single clock domain inside the entire box.  The problem with that
was you could encounter line slips if one of your circuits was not coming
from the CO or from even the same provider in the same CO.  As a result,
the new ISR4k does not carry the TDM clock outside of the NIM module and so
the DSP resources all reside locally to the T1 cards themselves.  This
means that you have to keep the same providers/CO on the same NIM but you
can bring other providers/COs into other NIMs and have separate clocking
domains.  This is huge for teleco redundancy across your PRIs.

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 <vtadorsett at gmail.com>
> *To:* Ryan Huff <ryanhuff at outlook.com>
> *Cc:* PF <pucknether at foril.com> ; 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 <ryanhuff at outlook.com>
>> *To:* Patrick Fortin <pfortin at royaume.com>
>> *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 <ryanhuff at outlook.com>
>> *To:* PF <pucknether at foril.com>
>> *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 <ryanhuff at outlook.com>
>> *To:* PF <pucknether at foril.com>
>> *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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