[Asterisk-Users] Cisco 3620 PRI and Asterisk

[Jesse Tyler]

Deon:

I am using a * for matching numbers on the router. I would like all 
calls to go straight to Asterisk-PBX. Do you think this method is 
acceptable???

Thanks,

Jesse  Tyler

On 29-Sep-04, at 12:05 PM, Deon Rodden wrote:

> 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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