thr3ads.net - asterisk users - [Asterisk-Users] Cisco 3620 PRI and Asterisk [Sep 2004]

If this information is useful, please help other people find it:
Share via:

Jesse Tyler

2004-Sep-29 10:32 UTC

[Asterisk-Users] Cisco 3620 PRI and Asterisk

Hi All:

I have a Cisco3620 with a proper T1/PRI card installed with asterisk 
running on the same LAN. Since I have lit up the line, I can dial out 
and make calls to regular lands lines. However when a call comes back 
in it rings the destination phone once and disconnects.

Here is an error from my router
15:40:45: ISDN Se1/0:23 SERROR: L3_GetUser_NLCB: EVENT 0X45 No NLCB 2
15:40:45: ISDN Se1/0:23 **ERROR**: Ux_BadMsg: Invalid Message for call 
state 9, call id 0x253, call ref 0x83DF, event 0x62
15:40:45: ISDN Se1/0:23 SERROR: CCPRI_Go: call id 0x254 event 0x57 No 
ccb Source->HOST
15:40:45: ISDN  **ERROR**: Module-l3_sdl_u  Function-U19_BadMsg  
Error-Bad message received.
15:40:45: ISDN Se1/0:23 SERROR: CCPRI_Go: call id 0x253 event 0x57 No 
ccb Source->HOSTConnection closed by foreign host.


Here is some data from a SNIFF on port 5060
3648.406191 192.168.10.1 -> 192.168.10.2 SIP Status: 200 OK
3659.554288 192.168.10.2 -> 192.168.10.1 SIP Request: OPTIONS 
sip:192.168.10.1
3659.573166 192.168.10.1 -> 192.168.10.2 SIP/SDP Status: 200 OK, with 
session description
3684.730069 192.168.10.1 -> 192.168.10.2 SIP/SDP Request: INVITE 
sip:[my_hidden_phone_number]@192.168.10.2:5060, with session 
description
3684.730479 192.168.10.2 -> 192.168.10.1 SIP Status: 100 Trying
3684.732364 192.168.10.2 -> 192.168.10.1 SIP Status: 180 Ringing
3685.077268 192.168.10.1 -> 192.168.10.2 SIP Request: CANCEL 
sip:[my_hidden_phone_number]@192.168.10.2:5060
3685.077617 192.168.10.2 -> 192.168.10.1 SIP Status: 200 OK


Asterisk
     -- Executing Goto("SIP/192.168.10.1-0819f7d8", "350|1")
in new stack
     -- Goto (default,350,1)
     -- Executing Dial("SIP/192.168.10.1-0819f7d8",
"SIP/350|20|tr") in
new stack
     -- Called 350
   == Spawn extension (default, 350, 1) exited non-zero on 
'SIP/192.168.10.1-0819f7d8'

350 is my extension on Asterisk
192.168.10.1 is the router with the PRI installed and running
192.168.10.2 is the asterisk box


Anyone with any ideas please contact me.

Thanks to all in advance,


Jesse Tyler

Astrit

2004-Sep-29 10:43 UTC

head link

[Asterisk-Users] MGCP for SA Webstar Cable Modems

Hi all,
 Does anybody have a sample mgcp.conf that can be used to connect a
Scientific Atlanta eMTA Cable Modem or any eMTA cable modem with Asterisk ?
Or, does anybody can tell me the meaning of most command used in mgcp.conf
file ?

Thanks in advance 
Astrit Morina
DOCSIS
IPKO Net

Benjamin on Asterisk Mailing Lists

2004-Sep-29 10:47 UTC

head link

[Asterisk-Users] Cisco 3620 PRI and Asterisk

On Wed, 29 Sep 2004 11:32:44 -0600, Jesse Tyler <jtyler@goarctic.com>
wrote:> Here is an error from my router
> 15:40:45: ISDN Se1/0:23 SERROR: L3_GetUser_NLCB: EVENT 0X45 No NLCB 2
> 15:40:45: ISDN Se1/0:23 **ERROR**: Ux_BadMsg: Invalid Message for call
> state 9, call id 0x253, call ref 0x83DF, event 0x62
> 15:40:45: ISDN Se1/0:23 SERROR: CCPRI_Go: call id 0x254 event 0x57 No
> ccb Source->HOST
> 15:40:45: ISDN  **ERROR**: Module-l3_sdl_u  Function-U19_BadMsg
> Error-Bad message received.
> 15:40:45: ISDN Se1/0:23 SERROR: CCPRI_Go: call id 0x253 event 0x57 No
> ccb Source->HOSTConnection closed by foreign host.
> 
> Here is some data from a SNIFF on port 5060
> 3648.406191 192.168.10.1 -> 192.168.10.2 SIP Status: 200 OK
> 3659.554288 192.168.10.2 -> 192.168.10.1 SIP Request: OPTIONS
> sip:192.168.10.1
> 3659.573166 192.168.10.1 -> 192.168.10.2 SIP/SDP Status: 200 OK, with
> session description
> 3684.730069 192.168.10.1 -> 192.168.10.2 SIP/SDP Request: INVITE
> sip:[my_hidden_phone_number]@192.168.10.2:5060, with session
> description
> 3684.730479 192.168.10.2 -> 192.168.10.1 SIP Status: 100 Trying
> 3684.732364 192.168.10.2 -> 192.168.10.1 SIP Status: 180 Ringing
> 3685.077268 192.168.10.1 -> 192.168.10.2 SIP Request: CANCEL
> sip:[my_hidden_phone_number]@192.168.10.2:5060
> 3685.077617 192.168.10.2 -> 192.168.10.1 SIP Status: 200 OK
> 
> Asterisk
>      -- Executing Goto("SIP/192.168.10.1-0819f7d8",
"350|1") in new stack
>      -- Goto (default,350,1)
>      -- Executing Dial("SIP/192.168.10.1-0819f7d8",
"SIP/350|20|tr") in
> new stack
>      -- Called 350
>    == Spawn extension (default, 350, 1) exited non-zero on
> 'SIP/192.168.10.1-0819f7d8'
> 
> 350 is my extension on Asterisk
> 192.168.10.1 is the router with the PRI installed and running
> 192.168.10.2 is the asterisk box
Apparently, the Cisco is getting something back from Asterisk which it
thinks is a bad message and as a result of that it sends a CANCEL
request back to Asterisk.

Although I am not sure what that bad message might be, judging by the
fact that the last message Asterisk sent out to the Cisco shortly
before was the RINGING message, I take a guess that there might be
something funny with that. Following that thought, I would try the
dial command without the "r" and also without "t" and
without "tr" and
see if that makes any difference.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.

Deon Rodden

2004-Sep-29 11:05 UTC

head link

[Asterisk-Users] Cisco 3620 PRI and Asterisk

Is your carrier sending you the numbers in 7 digit format or 10 digit 
format?

What does your dial-peer statements look like in your routers config? We 
have a similar setup on a Cisco 3640

Here's a couple of examples of our setup:

dial-peer voice 28 voip
 description WU Sales Temp
 destination-pattern 4559593
 session protocol sipv2
 session target ipv4:216.242.94.6
 session transport udp
 codec g711ulaw
 no vad

or

dial-peer voice 85 voip
 description 569-3000 through 569-3039
 destination-pattern 56930[0-3][0-9]
 session protocol sipv2
 session target sip-server
 session transport udp
 codec g711ulaw
 no vad


Jesse Tyler wrote:
> Hi All:
>
> I have a Cisco3620 with a proper T1/PRI card installed with asterisk 
> running on the same LAN. Since I have lit up the line, I can dial out 
> and make calls to regular lands lines. However when a call comes back 
> in it rings the destination phone once and disconnects.
>
> Here is an error from my router
> 15:40:45: ISDN Se1/0:23 SERROR: L3_GetUser_NLCB: EVENT 0X45 No NLCB 2
> 15:40:45: ISDN Se1/0:23 **ERROR**: Ux_BadMsg: Invalid Message for call 
> state 9, call id 0x253, call ref 0x83DF, event 0x62
> 15:40:45: ISDN Se1/0:23 SERROR: CCPRI_Go: call id 0x254 event 0x57 No 
> ccb Source->HOST
> 15:40:45: ISDN  **ERROR**: Module-l3_sdl_u  Function-U19_BadMsg  
> Error-Bad message received.
> 15:40:45: ISDN Se1/0:23 SERROR: CCPRI_Go: call id 0x253 event 0x57 No 
> ccb Source->HOSTConnection closed by foreign host.
>
>
> Here is some data from a SNIFF on port 5060
> 3648.406191 192.168.10.1 -> 192.168.10.2 SIP Status: 200 OK
> 3659.554288 192.168.10.2 -> 192.168.10.1 SIP Request: OPTIONS 
> sip:192.168.10.1
> 3659.573166 192.168.10.1 -> 192.168.10.2 SIP/SDP Status: 200 OK, with 
> session description
> 3684.730069 192.168.10.1 -> 192.168.10.2 SIP/SDP Request: INVITE 
> sip:[my_hidden_phone_number]@192.168.10.2:5060, with session description
> 3684.730479 192.168.10.2 -> 192.168.10.1 SIP Status: 100 Trying
> 3684.732364 192.168.10.2 -> 192.168.10.1 SIP Status: 180 Ringing
> 3685.077268 192.168.10.1 -> 192.168.10.2 SIP Request: CANCEL 
> sip:[my_hidden_phone_number]@192.168.10.2:5060
> 3685.077617 192.168.10.2 -> 192.168.10.1 SIP Status: 200 OK
>
>
> Asterisk
>     -- Executing Goto("SIP/192.168.10.1-0819f7d8",
"350|1") in new stack
>     -- Goto (default,350,1)
>     -- Executing Dial("SIP/192.168.10.1-0819f7d8",
"SIP/350|20|tr") in
> new stack
>     -- Called 350
>   == Spawn extension (default, 350, 1) exited non-zero on 
> 'SIP/192.168.10.1-0819f7d8'
>
> 350 is my extension on Asterisk
> 192.168.10.1 is the router with the PRI installed and running
> 192.168.10.2 is the asterisk box
>
>
> Anyone with any ideas please contact me.
>
> Thanks to all in advance,
>
>
> Jesse Tyler
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> .
>

Michael Bielicki

2004-Sep-29 11:28 UTC

head link

[Asterisk-Users] MGCP for SA Webstar Cable Modems

A friend of mine who is testing this for his cable operator has done a
page about * and cable. just go here ====>
http://asterisk.urtho.net/tiki-index.php

On Wed, 29 Sep 2004 19:43:27 +0200, Astrit <morina@ipko.net>
wrote:> 
> Hi all,
>  Does anybody have a sample mgcp.conf that can be used to connect a
> Scientific Atlanta eMTA Cable Modem or any eMTA cable modem with Asterisk ?
> Or, does anybody can tell me the meaning of most command used in mgcp.conf
> file ?
> 
> Thanks in advance
> Astrit Morina
> DOCSIS
> IPKO Net
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
Michael Bielicki

Apparently Analagous Threads

Search for more seemingly similar threads

asterisk users - Sep 2004 - Cisco 3620 PRI and Asterisk

[Asterisk-Users] Cisco 3620 PRI and Asterisk

[Asterisk-Users] MGCP for SA Webstar Cable Modems

[Asterisk-Users] Cisco 3620 PRI and Asterisk

[Asterisk-Users] Cisco 3620 PRI and Asterisk

[Asterisk-Users] MGCP for SA Webstar Cable Modems

Apparently Analagous Threads