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<div class="moz-text-flowed"
style="font-family: -moz-fixed; font-size: 15px;" lang="x-western">Hi...
we're using asterisk to send calls over SIP to a cisco with two PRIs.
<br>
There's really two questions here, one about how to get the cisco to
properly
<br>
authenticate over SIP, and the second about callerid.
The second is much more<br>
critical...<br>
<br>
1. in sip.conf I have a section labeled [a.b.c.d] (the cisco's IP
address) but
<br>
asterisk doesn't end up using that for the cisco's communications.
Instead it uses
<br>
the [general] section. I know this because I had to add a usable
context to [general]
<br>
for it to work. How do I make the cisco at a.b.c.d use a particular
part of sip.conf
<br>
instead of general?
(or how do I make chan_sip use [a.b.c.d] instead of [general])<br>
<br>
2. No caller ID shows up on remote side of PRI. through Cisco
<br>
<br>
The inbound calls (PSTN->PRI->cisco->sip->asterisk) work
fine, and display CLID great.
<br>
<br>
The outbound calls (sip
phone->asterisk->sip->cisco->PRI->PSTN) have an ID set
in *<br>
(I used NoOp to check it, as well as an explicit SetCallerID)
<br>
and the ID shows on the Cisco console debug messages but NONE shows up
on the PSTN side.
<br>
<br>
We've checked the switch type (NI2) with the carrier, and they say
their settings are good.
<br>
<br>
We then moved the PRI to the Zaptel PRI card... and sure enough
outbound caller ID works,
<br>
so we know the carrier wasn't lying to us.
<br>
<br>
That tells me it's a cisco configuration problem. I have been unable
to find what's wrong<br>
with the cisco config. Any help would be appreciated.<br>
<br>
Ehud
<br>
<a class="moz-txt-link-abbreviated" href="mailto:gavron@wetwork.net">gavron@wetwork.net</a><br>
<br>
Here's the console output, my comments in parens.<br>
<br>
Jul 28 14:47:47 MST: %ISDN-6-CONNECT: Interface Serial1/1:0 is now
connected to 520XXXXXXX unknown
<br>
Jul 28 14:48:13 MST: %ISDN-6-CONNECT: Interface Serial1/0:0 is now
connected to YYYYYYY unknown
<br>
(XXXXXXXis the correct callING phone number. YYYYYYY is the correct
callED phone number. When YYYY
<br>
gets the call there is no callerid for the call.)
<br>
<br>
(and relevant parts of the config)<br>
<br>
controller T1 1/0
<br>
framing esf
<br>
linecode b8zs
<br>
cablelength long 0db
<br>
pri-group timeslots 1-24
<br>
!
<br>
controller T1 1/1
<br>
framing esf
<br>
linecode b8zs
<br>
cablelength long 0db
<br>
pri-group timeslots 1-24
<br>
<br>
interface Serial1/0:23
<br>
no ip address
<br>
isdn switch-type primary-ni
<br>
isdn incoming-voice voice
<br>
isdn bchan-number-order ascending
<br>
no cdp enable
<br>
!
<br>
interface Serial1/1:23
<br>
no ip address
<br>
isdn switch-type primary-ni
<br>
isdn incoming-voice voice
<br>
isdn bchan-number-order ascending
<br>
no cdp enable
<br>
voice-port 1/0:23
<br>
echo-cancel coverage 32
<br>
!
<br>
voice-port 1/1:23
<br>
echo-cancel coverage 32
<br>
!
<br>
!
<br>
!
<br>
dial-peer voice 1 voip
<br>
application session
<br>
destination-pattern 8T
<br>
session protocol sipv2
<br>
session target ipv4:ip.address.goes.here
<br>
session transport udp
<br>
dtmf-relay rtp-nte
<br>
codec g711ulaw
<br>
no vad
<br>
!
<br>
dial-peer voice 2 pots
<br>
incoming called-number 520.......
<br>
direct-inward-dial
<br>
!
<br>
dial-peer voice 3 pots
<br>
destination-pattern T
<br>
port 1/0:23
<br>
!
<br>
dial-peer voice 4 voip
<br>
application session
<br>
destination-pattern 520.......
<br>
session protocol sipv2
<br>
session target ipv4:ip.address.goes.here
<br>
dtmf-relay rtp-nte
<br>
codec g711ulaw
<br>
no vad
<br>
<br>
!
<br>
dial-peer voice 6 pots
<br>
destination-pattern T
<br>
port 1/0:23
<br>
!
<br>
<br>
<br>
<br>
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