[asterisk-users] Fwd: Asterisk With Cisco Voice Router

Timothy Smith timotsmith at gmail.com
Sat May 16 09:22:22 CDT 2009


David,

Thanks a lot for your input.  I will enable DSP farming. Like some
other techies, I just wanted to see it work before i consider others
things.

I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.

It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.

Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.

; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten => _X.,1,Read(NUM,beep,4,2,3)
exten => _X.,n,Dial(SIP/${NUM})

; This is all i get when i call and the call hangs up!

cs-intranet*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [730732888 at default:1]
Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new stack
    -- Accepting a maximum of 4 digits.
  == Using SIP RTP CoS mark 5
    -- Executing [730732888 at default:1]
Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new stack
    -- Accepting a maximum of 4 digits.
cs-intranet*CLI>

Thanks alot for your assistance.

On Sat, May 16, 2009 at 4:02 PM, David Backeberg <dbackeberg at gmail.com> wrote:
> On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at gmail.com> wrote:
>> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
>> and also a dialpeer to forward on the router to forward calls to my
>> asterisk. It works properly but the problem is there is NO AUDIO! I
>> have tried to change codec but no sucess!
>> Has anyone had the above set up working successfully?
>
> Yes.
>
> You have been caught by a not-very-well-documented issue with setting
> up voice routing on the 3845, and probably other similar Cisco gear.
> And I'm not sure how you've done your test.
> This is the closest I've ever seen to a document that explains your problem:
> http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml
>
> Did you have a SIP phone on one side of asterisk and a POTS phone on
> the outside of the 3845?
>
> If you did, and you could talk on both at the same time, I think you
> would discover in fact that you do have some audio, in fact, one-way
> audio to be precise. But I don't remember for sure, because it's been
> a while since I've done this to myself.
>
> At any rate, your problem is you have dial-peers to get voice packets
> out from the 3845 to Cisco, but no dial-peers to get the packets from
> SIP back to a physical circuit on the 3845. Think about this. What
> should happen to a call inbound from asterisk, to the 3845? Should it
> go out an E1 to the outside phones world? If so, you need to build a
> dial-peer that does that. Until you do, you won't be getting two-way
> audio.
>
> you need another rule something like:
> dial-peer voice 790792888 pots
> map this back to the proper E1 circuit
>
> A secondary problem could also be with the way you're managing your
> DSPs. I don't know how many physical DSPs you have in your router, but
> usually it's a GOOD thing to enable DSP farming.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
May 16 14:18:28.640: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x0CAB
        Cause i = 0x80D1 - Invalid call reference value
May 16 14:18:28.640: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A No NLCB 2
May 16 14:18:28.640: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8CAB callid 0x0
May 16 14:18:28.660: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8  callref = 0x0CAC
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
        Calling Party Number i = 0x2183, '730730199'
                Plan:ISDN, Type:National
        Called Party Number i = 0xA1, '730732888'
                Plan:ISDN, Type:National
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_INCOMING
May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x3407, Guid = 42D2FC70B0D1
May 16 14:18:28.664: fb_get_reject_cause_code: ERROR cause_code NULL

May 16 14:18:28.668: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD
May 16 14:18:28.668: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8CAC
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
May 16 14:18:28.676: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd: false ret
May 16 14:18:28.676: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD
May 16 14:18:28.676: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8  callref = 0x8CAC
May 16 14:18:28.688: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8  callref = 0x0CAC
May 16 14:18:28.688: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_PROGRESS
May 16 14:18:28.688: %ISDN-6-CONNECT: Interface Serial0/0/1:12 is now connected to 730730199 N/A
May 16 14:18:28.788: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x0CAC
        Cause i = 0x82AF - Resource unavailable, unspecified
May 16 14:18:28.788: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_DISC
May 16 14:18:28.788: %ISDN-6-CONNECT: Interface Serial0/0/1:12 is now connected to 730730199 N/A
May 16 14:18:28.788: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8  callref = 0x8CAC
May 16 14:18:28.792: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8  callref = 0x0000
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
        Restart Indicator i = 0x80
May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_DISC
May 16 14:18:28.792: ISDN Se0/0/1:15 SERROR: CCPRI_Go: call id 0x3407 event 0x57 No ccb Source->HOST
May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_CLEARED
May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x0 calltype 6 CHAN_STATUS
OULKLAVG2#
May 16 14:18:28.792: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8  callref = 0x8000
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
        Restart Indicator i = 0x80
May 16 14:18:28.800: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x0CAC
        Cause i = 0x80D1 - Invalid call reference value
May 16 14:18:28.800: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A No NLCB 2
May 16 14:18:28.800: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8CAC callid 0x0
OULKLAVG2#


More information about the asterisk-users mailing list