[Asterisk-Users] Cisco 3620 PRI and Asterisk

[Deon Rodden]

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
>
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