[Asterisk-Users] Cisco 3620 PRI and Asterisk
Jesse Tyler
jtyler at goarctic.com
Wed Sep 29 14:03:16 MST 2004
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
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
>> .
>>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>
More information about the asterisk-users
mailing list