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 ; [email protected] 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 <[email protected]> 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 ; [email protected] 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 <[email protected]> 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 <[email protected]> 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: [email protected] 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 <[email protected]> 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: [email protected] 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 <[email protected]> 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: [email protected] 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 <[email protected]> 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 _______________________________________________ cisco-voip mailing list [email protected] https://puck.nether.net/mailman/listinfo/cisco-voip _______________________________________________ cisco-voip mailing list [email protected] https://puck.nether.net/mailman/listinfo/cisco-voip
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