<div dir="auto">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. </div><div dir="auto"><br></div><div dir="auto">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. </div><div dir="auto"><br></div><div dir="auto">Andrew</div><div dir="auto"><br></div><div><br><div class="gmail_quote"><div>On Fri, May 4, 2018 at 12:32 PM Ryan Huff <<a href="mailto:ryanhuff@outlook.com">ryanhuff@outlook.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="auto">
Not sure if DSP is onboard the SPA like the NIM; I’d have to look that one up ... you might try <span style="background-color:rgba(255,255,255,0)"><i>dsp services dspfarm
</i>under the voice-card configuration parameter. I’d also disable cdp on the serial interface.</span>
<div><br>
</div>
<div>
<div id="m_8553762886255870531AppleMailSignature">Sent from my iPhone</div></div></div><div dir="auto"><div>
<div><br>
On May 4, 2018, at 12:18, PF <<a href="mailto:pucknether@foril.com" target="_blank">pucknether@foril.com</a>> wrote:<br>
<br>
</div>
<blockquote type="cite">
<div>
<div><font size="2" face="Arial">Hi</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">I have tried several configurations</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">Here is the relevent par of the actual config</font></div>
<div><font size="2" face="Arial"><br>
version 15.3<br>
boot system flash bootflash:asr1000rp2-adventerprisek9.03.10.07.S.153-3.S7-ext.bin<br>
!<br>
card type t1 0 2<br>
!<br>
multilink bundle-name authenticated<br>
isdn switch-type primary-5ess<br>
!<br>
voice-card 0/0<br>
!<br>
voice service pots<br>
supported-language FR<br>
!<br>
voice service voip<br>
clid network-provided<br>
allow-connections h323 to sip<br>
allow-connections sip to h323<br>
allow-connections sip to sip<br>
signaling forward unconditional<br>
sip<br>
bind control source-interface GigabitEthernet1/2/0<br>
bind media source-interface GigabitEthernet1/2/0<br>
ds0-num<br>
header-passing<br>
privacy id<br>
outbound-proxy ipv4:172.16.x.y<br>
!<br>
voice class codec 1<br>
codec preference 1 g711ulaw<br>
!<br>
controller T1 0/2/0<br>
framing esf<br>
clock source internal<br>
linecode b8zs<br>
cablelength long 0db<br>
pri-group timeslots 1-24<br>
!<br>
interface Service-Engine0/0/0<br>
!<br>
interface Serial0/2/0:23<br>
encapsulation hdlc<br>
isdn switch-type primary-5ess<br>
isdn negotiate-bchan<br>
!<br>
map-class dialer DOVtest<br>
dialer voice-call<br>
!<br>
dspfarm profile 1 transcode universal <br>
rsvp<br>
shutdown<br>
!<br>
dial-peer voice 1 voip<br>
destination-pattern *.<br>
signaling forward unconditional<br>
session protocol sipv2<br>
session target sip-server<br>
voice-class codec 1 <br>
!<br>
dial-peer voice 2 pots<br>
destination-pattern xxxxxxx<br>
incoming called-number xxxxxxx<br>
direct-inward-dial<br>
!<br>
!<br>
sip-ua <br>
credentials username asr1006 password 7 xxxxxxxx realm yyyyy<br>
retry invite 3<br>
retry bye 3<br>
retry cancel 3<br>
timers trying 1000<br>
timers register 100<br>
sip-server ipv4:172.16.x.y<br>
!<br>
!</font></div>
<font size="2" face="Arial">
<div>Pat</div>
</font>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial"></font> </div>
<blockquote style="BORDER-LEFT:#000000 2px solid;PADDING-LEFT:5px;PADDING-RIGHT:0px;MARGIN-LEFT:5px;MARGIN-RIGHT:0px">
<div style="FONT:10pt arial">----- Original Message ----- </div>
<div style="FONT:10pt arial;BACKGROUND:#e4e4e4"><b>From:</b>
<a title="ryanhuff@outlook.com" href="mailto:ryanhuff@outlook.com" target="_blank">Ryan Huff</a> </div>
<div style="FONT:10pt arial"><b>To:</b> <a title="pfortin@royaume.com" href="mailto:pfortin@royaume.com" target="_blank">
Patrick Fortin</a> </div>
<div style="FONT:10pt arial"><b>Cc:</b> <a title="cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net" target="_blank">
cisco-voip@puck.nether.net</a> </div>
<div style="FONT:10pt arial"><b>Sent:</b> Friday, May 04, 2018 11:43 AM</div>
<div style="FONT:10pt arial"><b>Subject:</b> Re: [cisco-voip] ISDN T1 voice calls hardware validation</div>
<div><br>
</div>
I assume you have the card type specified and controller interface configured with timeslots?
<div><br>
</div>
<div>Can you send the running-config and code version?<br>
<br>
<div id="m_8553762886255870531AppleMailSignature">Sent from my iPhone</div>
<div><br>
On May 4, 2018, at 11:35, Patrick Fortin <<a href="mailto:pfortin@royaume.com" target="_blank">pfortin@royaume.com</a>> wrote:<br>
<br>
</div>
<blockquote type="cite">
<div>
<div><font size="2" face="Arial">Hi</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">I get this error</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial"> **ERROR**: call_incoming: Received a call id 0x2F with a bad bearercap from xxxxxxxxxx on b channel 1</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">seems like the dsp are not associated with the t1 card</font></div>
<div><font size="2" face="Arial"><br>
in the config we don't have acces to the following :</font></div>
<font size="2" face="Arial"></font>
<div><font size="2" face="Arial">
<p>isdn incoming-voice voice<br>
which should go on the Serial interface. <br>
<br>
we also don't have the "port" command that should go in the dial-peer voice section</p>
</font></div>
<div><font size="2" face="Arial">any ideas ?</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">Thanks</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">Pat</font></div>
<blockquote style="BORDER-LEFT:#000000 2px solid;PADDING-LEFT:5px;PADDING-RIGHT:0px;MARGIN-LEFT:5px;MARGIN-RIGHT:0px">
<div style="FONT:10pt arial">----- Original Message ----- </div>
<div style="FONT:10pt arial;BACKGROUND:#e4e4e4"><b>From:</b>
<a title="ryanhuff@outlook.com" href="mailto:ryanhuff@outlook.com" target="_blank">Ryan Huff</a> </div>
<div style="FONT:10pt arial"><b>To:</b> <a title="pucknether@foril.com" href="mailto:pucknether@foril.com" target="_blank">
PF</a> </div>
<div style="FONT:10pt arial"><b>Cc:</b> <a title="cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net" target="_blank">
cisco-voip@puck.nether.net</a> </div>
<div style="FONT:10pt arial"><b>Sent:</b> Friday, May 04, 2018 11:11 AM</div>
<div style="FONT:10pt arial"><b>Subject:</b> Re: [cisco-voip] ISDN T1 voice calls hardware validation</div>
<div><br>
</div>
That correct! I misread! Yeah the shared port adapter T1 should work.
<div><br>
<br>
<div id="m_8553762886255870531AppleMailSignature">Sent from my iPhone</div>
<div><br>
On May 4, 2018, at 10:36, PF <<a href="mailto:pucknether@foril.com" target="_blank">pucknether@foril.com</a>> wrote:<br>
<br>
</div>
<blockquote type="cite">
<div>
<div><font size="2" face="Arial">Hi</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">Thanks for your help</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">But there are no NIM slot in the ASR1006 chassis</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">NIM is for ASR1001-X I think</font></div>
<div><font size="2" face="Arial"></font> </div>
<div><font size="2" face="Arial">Pat</font></div>
<div><font size="2" face="Arial"></font> </div>
<blockquote style="BORDER-LEFT:#000000 2px solid;PADDING-LEFT:5px;PADDING-RIGHT:0px;MARGIN-LEFT:5px;MARGIN-RIGHT:0px">
<div style="FONT:10pt arial">----- Original Message ----- </div>
<div style="FONT:10pt arial;BACKGROUND:#e4e4e4"><b>From:</b>
<a title="ryanhuff@outlook.com" href="mailto:ryanhuff@outlook.com" target="_blank">Ryan Huff</a> </div>
<div style="FONT:10pt arial"><b>To:</b> <a title="pucknether@foril.com" href="mailto:pucknether@foril.com" target="_blank">
PF</a> </div>
<div style="FONT:10pt arial"><b>Cc:</b> <a title="cisco-voip@puck.nether.net" href="mailto:cisco-voip@puck.nether.net" target="_blank">
cisco-voip@puck.nether.net</a> </div>
<div style="FONT:10pt arial"><b>Sent:</b> Friday, May 04, 2018 9:56 AM</div>
<div style="FONT:10pt arial"><b>Subject:</b> Re: [cisco-voip] ISDN T1 voice calls hardware validation</div>
<div><br>
</div>
For isdn voice you’ll need a NIM-8MFT-T1/E1<br>
<br>
<div id="m_8553762886255870531AppleMailSignature">Sent from my iPhone</div>
<div><br>
On May 4, 2018, at 09:31, PF <<a href="mailto:pucknether@foril.com" target="_blank">pucknether@foril.com</a>> wrote:<br>
<br>
</div>
<blockquote type="cite">
<div>
<div><font size="2" face="Arial">
<div>Hi</div>
<div>
<p>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</p>
<p>ASR1006<br>
SPA-8XCHT1/E1<br>
SPA-DSP<br>
ASR1000-RP2<br>
ASR1000-ESP40</p>
<p>in short can it be used to build a voip gateway like an audiocode mediant or a patton smartnode</p>
<p>Thanks</p>
<p>Pat</p>
</div>
</font></div>
</div>
</blockquote>
<blockquote type="cite">
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