CS1000E Rls. 7.5 SIP to * 1.8

I’m having trouble calling the Asterisk box. My network is completely isolated from the outside world, meaning no security concerns. I’m OK with letting the CS1000 call anywhere and anyone the Asterisk box can reach. Right now I’m trying to use the Asterisk box as a conference bridge.

Call Scenario: DN 2011 calls extension 6001 on the Asterisk box. The phone making the call hears dead air for about 10 - 12 seconds then reorder.

Here is some SIP debug output:

<--- SIP read from UDP: --->
INVITE sip:6001;phone-context=cdp.udp@xyz.com;user=phone SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bKe39181583f147aad1468c982-a6dc4212.1
Via: SIP/2.0/UDP;branch=z9hG4bK-b24e-2b883b8-42851ea4;received=
Record-Route: <sip:;transport=udp;lr>
From: <sip:2011;phone-context=cdp.udp@xyz.com;user=phone>;tag=9bfcd28-114be421-13c4-55013-b24e-3c26ba7-b24e
To: <sip:6001;phone-context=cdp.udp@xyz.com;user=phone>
Call-ID: a8bc36a8-114be421-13c4-55013-b24e-4c06c871-b24e
Contact: <sip:2011;phone-context=cdp.udp@xyz.com:5060;maddr=;transport=udp;user=phone>
Max-forwards: 69
Supported: 100rel,x-nortel-sipvc,replaces
User-agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.50.17
P-asserted-identity: <sip:2011;phone-context=cdp.udp@xyz.com;user=phone>
Privacy: none
History-info: <sip:6001;phone-context=cdp.udp@xyz.com;user=phone>;index=1
Content-Type: multipart/mixed ;boundary=unique-boundary-1
Alert-Info: <cid:internal@xyz.com>
Content-Length: 1132

Content-Type: application/sdp

o=- 9 1 IN IP4
c=IN IP4
t=0 0
m=audio 5438 RTP/AVP 0 8 18 4 101 111
c=IN IP4
a=tcap:1 RTP/SAVP
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XIeUgoutjfaWvcTZfTVVGw2u2lNrZGPF2jigzj+P|2^31|1126864636:4
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:XIeUgoutjfaWvcTZfTVVGw2u2lNrZGPF2jigzj+P|2^31
a=pcfg:1 t=1
a=rtpmap:4 G723/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000

Content-Type: application/x-nt-mcdn-frag-hex;version=ssLinux-7.50.17;base=x2611
Content-Disposition: signal;handling=optional

Content-Type: application/x-nt-epid-frag-hex;version=ssLinux-7.50.17;base=x2611
Content-Disposition: signal;handling=optional

--- (19 headers 40 lines) ---
Sending to (NAT)
Using INVITE request as basis request - a8bc36a8-114be421-13c4-55013-b24e-4c06c871-b24e
No matching peer for '2011;phone-context=cdp.udp' from ''
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Found RTP audio format 111
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Found unknown media description format X-nt-inforeq for ID 111

<--- Reliably Transmitting (NAT) to --->
SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP;branch=z9hG4bKe39181583f147aad1468c982-a6dc4212.1;received=;rport=35919 Via: SIP/2.0/UDP;branch=z9hG4bK-b24e-2b883b8-42851ea4;received= From: <sip:2011;phone-context=cdp.udp@xyz.com;user=phone>;tag=9bfcd28-114be421-13c4-55013-b24e-3c26ba7-b24e To: <sip:6001;phone-context=cdp.udp@xyz.com;user=phone>;tag=as2e651215 Call-ID: a8bc36a8-114be421-13c4-55013-b24e-4c06c871-b24e CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r349968 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0  
Scheduling destruction of SIP dialog 'a8bc36a8-114be421-13c4-55013-b24e-4c06c871-b24e' in 32000 ms (Method: INVITE)

Would one of you agree that the problem I’m experiencing is identified in this line:

No matching peer for '2011;phone-context=cdp.udp' from ''
Does this mean my system doesn’t recognize the CS1000 as a peer, or is it specifically saying it has a problem with DN 2011? Right or wrong, I’d be very greatful for guidance on how I can let any call from any DN on the CS1000 be considered a peer or user, or whatever it takes to get calls into Asterisk.

I’m not interested in calling the CS1000 from the Asterisk box right now.

I’ve tried setting up all the default incoming call privileges I’ve found referenced at various sites, but none seem to work.

I’m certainly an Asterisk novice, though been in the voice business for over a decade.

The INVITE appears to have a malformed sip URL (; in the user field).

Hey all,

Just putting my two cents in;

It is looking for a peer (I.E. CS1K) and the phone-context coming from the CS1K is normal as its trying to past through the CS1K’s NRS which tags on the cdp.udp;

Are you using NRS or Session Manager?

Also, are you able to do calls from Asterisk into CS1K?


Asterisk is trying to match the whole user name against the sip.conf section heading. I am still fairly sure that the user field is not one where parameters are allowed. They are allowed after the domain and, in some contexts, after the final >. However I don’t have time to double check the RFC for the exact productions.

; may actually be a valid character in the user field, in which case it needs to be in the sip.conf section name, if you want to do a user type match.

Of course, getting ; into sip.conf, as anything except a comment delimiter, may be impossible! You could try , or less likely the, the % escape.

Even though you claim to be in a secure environment, I would still disable allowguest.

One other question. Why are you trying to do a user, rather than an IP, match?