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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>
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On May 4, 2018, at 12:18, PF <<a href="mailto:pucknether@foril.com">pucknether@foril.com</a>> wrote:<br>
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<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>
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<div><font size="2" face="Arial"></font> </div>
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<div style="FONT: 10pt arial">----- Original Message ----- </div>
<div style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><b>From:</b>
<a title="ryanhuff@outlook.com" href="mailto:ryanhuff@outlook.com">Ryan Huff</a> </div>
<div style="FONT: 10pt arial"><b>To:</b> <a title="pfortin@royaume.com" href="mailto:pfortin@royaume.com">
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">
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>
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I assume you have the card type specified and controller interface configured with timeslots?
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<div>Can you send the running-config and code version?<br>
<br>
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<div><br>
On May 4, 2018, at 11:35, Patrick Fortin <<a href="mailto:pfortin@royaume.com">pfortin@royaume.com</a>> wrote:<br>
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<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>
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<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>
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<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" dir="ltr">
<div style="FONT: 10pt arial">----- Original Message ----- </div>
<div style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><b>From:</b>
<a title="ryanhuff@outlook.com" href="mailto:ryanhuff@outlook.com">Ryan Huff</a> </div>
<div style="FONT: 10pt arial"><b>To:</b> <a title="pucknether@foril.com" href="mailto:pucknether@foril.com">
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">
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.
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<div><br>
On May 4, 2018, at 10:36, PF <<a href="mailto:pucknether@foril.com">pucknether@foril.com</a>> wrote:<br>
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<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" dir="ltr">
<div style="FONT: 10pt arial">----- Original Message ----- </div>
<div style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><b>From:</b>
<a title="ryanhuff@outlook.com" href="mailto:ryanhuff@outlook.com">Ryan Huff</a> </div>
<div style="FONT: 10pt arial"><b>To:</b> <a title="pucknether@foril.com" href="mailto:pucknether@foril.com">
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">
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="AppleMailSignature">Sent from my iPhone</div>
<div><br>
On May 4, 2018, at 09:31, PF <<a href="mailto:pucknether@foril.com">pucknether@foril.com</a>> wrote:<br>
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<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>
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