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Ivan Vershigora

2005-Nov-03 06:26 UTC

[Asterisk-Users] call from asterisk to SIP cisco 5300

i dial on my phone to to 80912222222
and convert it on asterisk to #00#70912222222
But Cisco says 404

============cisco peer============!
dial-peer voice 22 pots
 huntstop
 preference 5
 destination-pattern #00#......\*
 translate-outgoing calling 1
 direct-inward-dial
 port 0:D
 prefix 810
!
===============================
============peer in sip.conf=========[krdvox]
context=from-sip
type=peer
host=123.123.123.123
canreinvite=yes
dtmfmode=inband       
===============================
============extensions.conf=========exten =>
_.,1,SetCallerID("8612730000" <8612731107>[|a])
exten => _.,2,Dial(SIP/#00#7${EXTEN:1}@krdvox,60)
exten => _.,3,Congestion
===============================
============Asterisk says==========-- Executing Dial("SIP/201-2966",
"SIP/#00#70912222222@krdvox|60") in
new stack
    -- Called #00#70912222222@krdvox
    -- Got SIP response 404 "Not Found" back from XXX.XXX.XXX.XXX
    -- SIP/krdvox-3910 is circuit-busy
  == Everyone is busy/congested at this time
==============================
======CISCO debug ccsip ==========Nov  3 16:10:03.516: Received:
INVITE sip:#00#70912222222@123.123.123.123 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6697eb34
From: "8612730000" <sip:8612730000@1.1.1.1>;tag=as74db268c
To: <sip:#00#70957555655@123.123.123.123>
Contact: <sip:8612730000@1.1.1.1>
Call-ID: 3ac14bc91f81edb732cc3681388b811d@123.123.123.123
CSeq: 102 INVITE
User-Agent: CSCO/6
Date: Thu, 03 Nov 2005 13:10:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 235

.....

Nov  3 16:10:03.524: MatchNextPeer: Peer 999 matched
Nov  3 16:10:03.524: Using Voice Class Codec, tag=1

.....

Disconnect Cause (SIP)   : 404

==============================Nov  3 16:10:03.524: MatchNextPeer: Peer 999
matched

Peer 999- wrong one !!!!!!!
why he cant find dial-peer voice 22


????????????????????

Leandro Tenorio

2005-Nov-03 07:24 UTC

head link

[Asterisk-Users] call from asterisk to SIP cisco 5300

Probably by preference and peer type matching, try setting a new VoIP peer
for inbound calls from asterisk 

LTenorio
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> Ivan Vershigora
> Sent: Thursday, November 03, 2005 10:27 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] call from asterisk to SIP cisco 5300
> 
> 
> i dial on my phone to to 80912222222
> and convert it on asterisk to #00#70912222222 But Cisco says 404
> 
> ============cisco peer============> !
> dial-peer voice 22 pots
>  huntstop
>  preference 5
>  destination-pattern #00#......\*
>  translate-outgoing calling 1
>  direct-inward-dial
>  port 0:D
>  prefix 810
> !
> ===============================> 
> ============peer in sip.conf=========> [krdvox]
> context=from-sip
> type=peer
> host=123.123.123.123
> canreinvite=yes
> dtmfmode=inband       
> ===============================> 
> ============extensions.conf=========> exten =>
_.,1,SetCallerID("8612730000" <8612731107>[|a])
> exten => _.,2,Dial(SIP/#00#7${EXTEN:1}@krdvox,60)
> exten => _.,3,Congestion
> ===============================> 
> ============Asterisk says==========> -- Executing
Dial("SIP/201-2966",
> "SIP/#00#70912222222@krdvox|60") in new stack
>     -- Called #00#70912222222@krdvox
>     -- Got SIP response 404 "Not Found" back from XXX.XXX.XXX.XXX
>     -- SIP/krdvox-3910 is circuit-busy
>   == Everyone is busy/congested at this time 
> ==============================> 
> ======CISCO debug ccsip ==========> Nov  3 16:10:03.516: Received:
> INVITE sip:#00#70912222222@123.123.123.123 SIP/2.0
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6697eb34
> From: "8612730000" <sip:8612730000@1.1.1.1>;tag=as74db268c
> To: <sip:#00#70957555655@123.123.123.123>
> Contact: <sip:8612730000@1.1.1.1>
> Call-ID: 3ac14bc91f81edb732cc3681388b811d@123.123.123.123
> CSeq: 102 INVITE
> User-Agent: CSCO/6
> Date: Thu, 03 Nov 2005 13:10:06 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 235
> 
> .....
> 
> Nov  3 16:10:03.524: MatchNextPeer: Peer 999 matched Nov  3 
> 16:10:03.524: Using Voice Class Codec, tag=1
> 
> .....
> 
> Disconnect Cause (SIP)   : 404
> 
> ==============================> Nov  3 16:10:03.524: MatchNextPeer: Peer
999 matched
> 
> Peer 999- wrong one !!!!!!!
> why he cant find dial-peer voice 22
> 
> 
> ????????????????????
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> 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
>

Jason Brashear

2005-Nov-06 09:12 UTC

head link

[Asterisk-Users] asterisks talking to asterisks

I have a request. I have a server in Texas
And one in NJ.
Is it possible for the system in Texas to log into the system in NJ so that 
Extensions can call each other?
-J

Ivan Vershigora

2005-Nov-07 08:50 UTC

head link

[Asterisk-Users] call from asterisk to SIP cisco 5300

sorry, i didnt write i have voip  peer

so i have sloved thy problem, nubder like

#00#70912222222
*00*70912222222
*777
doesnt work
Cisco says
dpMatchPeersMoreArg: Match Dest. pattern; called ()

and when i tries to dial *777*777
it says

dpMatchPeersMoreArg: Match Dest. pattern; called (777)

But I cant understand why CISCO cant understand this "MAGIC" # and *
:)
>I think you should set dial-peer voice 21 voip with incoming called number
>#00#......\* too, this catch this call and the dial peer 22 send it.
>
>Adam
>
>Cytowanie Ivan Vershigora <noodlez@kubtelecom.ru>:
>
>>i dial on my phone to to 80912222222
>>and convert it on asterisk to #00#70912222222
>>But Cisco says 404
>>
>>============cisco peer============>>!
>>dial-peer voice 22 pots
>> huntstop
>> preference 5
>> destination-pattern #00#......\*
>> translate-outgoing calling 1
>> direct-inward-dial
>> port 0:D
>> prefix 810
>>!
>>===============================>>
>>============peer in sip.conf=========>>[krdvox]
>>context=from-sip
>>type=peer
>>host=123.123.123.123
>>canreinvite=yes
>>dtmfmode=inband
>>===============================>>
>>============extensions.conf=========>>exten =>
_.,1,SetCallerID("8612730000" <8612731107>[|a])
>>exten => _.,2,Dial(SIP/#00#7${EXTEN:1}@krdvox,60)
>>exten => _.,3,Congestion
>>===============================>>
>>============Asterisk says==========>>-- Executing
Dial("SIP/201-2966", "SIP/#00#70912222222@krdvox|60") in
>>new stack
>>    -- Called #00#70912222222@krdvox
>>    -- Got SIP response 404 "Not Found" back from
XXX.XXX.XXX.XXX
>>    -- SIP/krdvox-3910 is circuit-busy
>>  == Everyone is busy/congested at this time
>>==============================>>
>>======CISCO debug ccsip ==========>>Nov  3 16:10:03.516: Received:
>>INVITE sip:#00#70912222222@123.123.123.123 SIP/2.0
>>Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6697eb34
>>From: "8612730000"
<sip:8612730000@1.1.1.1>;tag=as74db268c
>>To: <sip:#00#70957555655@123.123.123.123>
>>Contact: <sip:8612730000@1.1.1.1>
>>Call-ID: 3ac14bc91f81edb732cc3681388b811d@123.123.123.123
>>CSeq: 102 INVITE
>>User-Agent: CSCO/6
>>Date: Thu, 03 Nov 2005 13:10:06 GMT
>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>>Content-Type: application/sdp
>>Content-Length: 235
>>
>>.....
>>
>>Nov  3 16:10:03.524: MatchNextPeer: Peer 999 matched
>>Nov  3 16:10:03.524: Using Voice Class Codec, tag=1
>>
>>.....
>>
>>Disconnect Cause (SIP)   : 404
>>
>>==============================>>Nov  3 16:10:03.524:
MatchNextPeer: Peer 999 matched
>>
>>Peer 999- wrong one !!!!!!!
>>why he cant find dial-peer voice 22
>>
>>
>>????????????????????
>>_______________________________________________
>>--Bandwidth and Colocation sponsored by Easynews.com --
>>
>>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
>>
>
>
>Pozdrawiam,
>Adam Rybak
>
>

BJ Weschke

2005-Nov-07 09:17 UTC

head link

[Asterisk-Users] asterisks talking to asterisks

Yes. Most certainly. Take a look at IAX (Inter Asterisk eXchange)
protocol to enable this functionality for you with minimal impact on
your firewall/NAT setups.

On 11/6/05, Jason Brashear <jason@austinfx.com>
wrote:> I have a request. I have a server in Texas
> And one in NJ.
> Is it possible for the system in Texas to log into the system in NJ so that
> Extensions can call each other?
> -J
>
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>

Ivan Vershigora

2005-Nov-21 07:20 UTC

head link

[Asterisk-Users] sip show users

asterisk1*CLI> sip show users
Username	Secret	Accountcode	Def.Context	ACL	NAT
205		test	from-internal   No   No
204		test	from-internal   No   No
203		test	from-internal   No   No
202		020	from-internal   No   No
201		test	from-internal   No   No


how can I get this information in my asterisk Macros in extesions.conf

something like sip(204(Def.Context))=from-internal

??

asterisk users - Nov 2005 - call from asterisk to SIP cisco 5300

[Asterisk-Users] call from asterisk to SIP cisco 5300

[Asterisk-Users] call from asterisk to SIP cisco 5300

[Asterisk-Users] asterisks talking to asterisks

[Asterisk-Users] call from asterisk to SIP cisco 5300

[Asterisk-Users] asterisks talking to asterisks

[Asterisk-Users] sip show users