[Asterisk-Users] Cisco 2620 as PRI gateway

Tim Reimers tim.reimers at asheville.k12.nc.us
Mon Feb 13 12:10:23 MST 2006


Here's a debug---

Longish, but I'm not sure what info in this might be useful to anyone--

I have a zyxel SIP phone configured as ext '6351' on Asterisk--
I can successfully call the SIP phone from an Sjphone client on my PC
and talk between the two-
so the SIP phone is in fact registered with * correctly...

The Cisco router has matching for 63[5-9]x configured- 

Here's a debug from the router:

ACS-GW#
*May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid
type/plan 0x0
0x1 may be overriden; sw-type 13
*May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid
type/plan 0x0
0x0 may be overriden; sw-type 13
*May 26 13:02:42: ISDN Se1/1:23 Q931: Applying typeplan for sw-type 0xD
is 0x4 0
x1, Called num 3506351
*May 26 13:02:42: ISDN Se1/1:23 Q931: TX -> SETUP pd = 8  callref =
0x7A54
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18387
                Preferred, Channel 7
        Calling Party Number i = 0x2181, '8283506180'
                Plan:ISDN, Type:National
        Called Party Number i = 0xC1, '3506351'
                Plan:ISDN, Type:Subscriber(local)
*May 26 13:02:42: ISDN Se1/1:23 Q931: RX <- CALL_PROC pd = 8  callref =
0xFA54
        Channel ID i = 0xA98387
                Exclusive, Channel 7
*May 26 13:02:43: ISDN Se1/0:23 Q931: RX <- SETUP pd = 8  callref =
0x0014
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98383
                Exclusive, Channel 3
        Calling Party Number i = 0x2181, '8283506180'
                Plan:ISDN, Type:National
        Called Party Number i = 0x80, '6351'
                Plan:Unknown, Type:Unknown
*May 26 13:02:43: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6351 at 10.10.1.28:5060 SIP/2.0
Via: SIP/2.0/UDP  10.12.1.252:5060;branch=z9hG4bK12A7
From: <sip:8283506180 at 10.12.1.252>;tag=3FB70415-7B4
To: <sip:6351 at 10.10.1.28>
Date: Sun, 26 May 2002 18:02:43 gmt
Call-ID: 9C227F6C-700911D6-803FEF66-8E03C25C at 10.12.1.252
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 2619306202-1879642582-3189047311-607696096
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY
, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID:
<sip:8283506180 at 10.12.1.252>;party=calling;screen=yes;privacy=o
ff
Timestamp: 1022436163
Contact: <sip:8283506180 at 10.12.1.252:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 211

v=0
o=CiscoSystemsSIP-GW-UserAgent 7318 2409 IN IP4 10.12.1.252
s=SIP Call
c=IN IP4 10.12.1.252
t=0 0
m=audio 16396 RTP/AVP 18
c=IN IP4 10.12.1.252
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20

*May 26 13:02:43: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP  10.12.1.252:5060;branch=z9hG4bK12A7
From: <sip:8283506180 at 10.12.1.252>;tag=3FB70415-7B4
To: <sip:6351 at 10.10.1.28>;tag=as21b1fb71
Call-ID: 9C227F6C-700911D6-803FEF66-8E03C25C at 10.12.1.252
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:6351 at 10.10.1.28>
Content-Length: 0



*May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> CALL_PROC pd = 8  callref =
0x8014
        Channel ID i = 0xA18383
                Preferred, Channel 3
ACS-GW#
*May 26 13:02:43: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:6351 at 10.10.1.28:5060 SIP/2.0
Via: SIP/2.0/UDP  10.12.1.252:5060;branch=z9hG4bK12A7
From: <sip:8283506180 at 10.12.1.252>;tag=3FB70415-7B4
To: <sip:6351 at 10.10.1.28>;tag=as21b1fb71
Date: Sun, 26 May 2002 18:02:43 gmt
Call-ID: 9C227F6C-700911D6-803FEF66-8E03C25C at 10.12.1.252
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0



*May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> DISCONNECT pd = 8  callref =
0x8014
        Cause i = 0x8081 - Unallocated/unassigned number
ACS-GW#
*May 26 13:02:43: ISDN Se1/1:23 Q931: RX <- PROGRESS pd = 8  callref =
0xFA54
        Cause i = 0x8281 - Unallocated/unassigned number
        Progress Ind i = 0x8288 - In-band info or appropriate now
available
*May 26 13:02:43: ISDN Se1/0:23 Q931: RX <- RELEASE pd = 8  callref =
0x0014
*May 26 13:02:43: ISDN Se1/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref
= 0x801
4
ACS-GW#
*May 26 13:02:49: ISDN Se1/1:23 Q931: TX -> DISCONNECT pd = 8  callref =
0x7A54
        Cause i = 0x8090 - Normal call clearing
*May 26 13:02:49: ISDN Se1/1:23 Q931: RX <- RELEASE pd = 8  callref =
0xFA54
*May 26 13:02:49: ISDN Se1/1:23 Q931: TX -> RELEASE_COMP pd = 8  callref
= 0x7A5
4
ACS-GW#
*May 26 13:03:07: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.12.1.252 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.28:5060;branch=z9hG4bK3741dad8
From: "Unknown" <sip:Unknown at 10.10.1.28>;tag=as6479f479
To: <sip:10.12.1.252>
Contact: <sip:Unknown at 10.10.1.28>
Call-ID: 6668302302bd600a6cdd9b830991189e at 10.10.1.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Mon, 13 Feb 2006 18:58:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0



*May 26 13:03:07: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.28:5060;branch=z9hG4bK3741dad8
From: "Unknown" <sip:Unknown at 10.10.1.28>;tag=as6479f479
To: <sip:10.12.1.252>;tag=3FB76385-1200
Date: Sun, 26 May 2002 18:03:07 gmt
Call-ID: 6668302302bd600a6cdd9b830991189e at 10.10.1.28
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY
, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 163
Content-Type:
ACS-GW# application/sdp

v=0
o=CiscoSystemsSIP-GW-UserAgent 777 2006 IN IP4 10.12.1.252
s=SIP Call
c=IN IP4 10.12.1.252
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 10.12.1.252


Asterisk shows nothing in '-r -vvvv' mode..
Anywhere I can trace to see if * is actually taking the call at all?

Here's this--- it LOOKS like the router is known to the * server....

acsasterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Mask
Port     Status    
pots                       10.12.1.252                 255.255.255.255
5060     OK (47 ms)
callmgr                    10.12.1.11                  255.255.255.255
5060     Unmonitored
acs-gw-rtr                 10.12.1.252                 255.255.255.255
5060     Unmonitored
6399/6399                  (Unspecified)    D          255.255.255.255
0        Unmonitored
6352/6352                  10.10.1.183      D          255.255.255.255
5060     Unmonitored
6351/6351                  10.10.1.101      D          255.255.255.255
5060     Unmonitored
6 sip peers [6 online , 0 offline]
acsasterisk*CLI> 




 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of kurt x
Sent: Friday, February 10, 2006 10:51 AM
To: Asterisk
Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway

debug ccsip message

Kurt
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