[cisco-voip] ISDN T1 voice calls hardware validation

Andrew Dorsett vtadorsett at gmail.com
Wed May 9 10:33:46 EDT 2018


You want the MFT which is the Multi-Function Trunk card.

Make sure you pick it under the voice card section of CCW and then pick the
PVDM4-256.

Andrew


On Wed, May 9, 2018 at 10:29 AM PF <pucknether at foril.com> wrote:

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