Paweł Staszewski
2005-Apr-21 07:11 UTC
Odp: Re: [Asterisk-Users] capi problem with dialout
Hello
I live in poland and :)
local numbers are: 752xxxx (7 digits)
zone prefix: 32
country prefix: 48
And i must add that i am behind a local PBX (Alcatel 4200E)
Configured isdn port with msn 7523071
Why dial in is working but dial-out not ... ??
And: I can dial-in from outside.... some debug from capi :
-- CONNECT_IND ID=001 #0x0e29 LEN=0045
Controller/PLCI/NCCI = 0x101
CIPValue = 0x10
CalledPartyNumber = <81>153
CallingPartyNumber = <09 80>172
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = <80 90 a3>
LLC = default
HLC = <91 81>
AdditionalInfo
BChannelinformation = <00 00>
Keypadfacility = default
Useruserdata = <04>
Facilitydataarray = default
== CONNECT_IND (PLCI=0x101,DID=153,CID=172,CIP=0x10,CONTROLLER=0x1)
-- started pbx on channel (callgroup=0)!
-- INFO_IND ID=001 #0x0e2a LEN=0016
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x7e
InfoElement = <04>
-- INFO_IND ID=001 #0x0e2b LEN=0019
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x70
InfoElement = <81>153
-- INFO_IND ID=001 #0x0e2c LEN=0016
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x18
InfoElement = <89>
-- ALERT_CONF ID=001 #0x0e29 LEN=0014
Controller/PLCI/NCCI = 0x101
Info = 0x0
== Starting CAPI[contr1/153]/6 at from-isdn,153,1 failed so falling back to
exten 's'
-- Executing SetLanguage(CAPI[contr1/153]/6, en) in new stack
-- Executing Dial(CAPI[contr1/153]/6, SIP/478) in new stack
We're at 195.205.186.7 port 10786
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:478@10.0.230.14:5060 SIP/2.0
Via: SIP/2.0/UDP 195.205.186.7:5060;branch=z9hG4bK4541e422
From: 172 <sip:172@195.205.186.7>;tag=as24721ef0
To: <sip:478@10.0.230.14:5060>
Contact: <sip:172@195.205.186.7>
Call-ID: 6238a0627e65947105ac1c004ecbb7a4@195.205.186.7
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 21 Apr 2005 14:03:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 10839 10839 IN IP4 195.205.186.7
s=session
c=IN IP4 195.205.186.7
t=0 0
m=audio 10786 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 10.0.230.14:5060
-- Called 478
Sip read:
SIP/2.0 100 Trying
To: <sip:478@10.0.230.14:5060>
From: 172<sip:172@195.205.186.7>;tag=as24721ef0
Via: SIP/2.0/UDP
195.205.186.7:5060;branch=z9hG4bK4541e422;received=195.205.186.7
Call-ID: 6238a0627e65947105ac1c004ecbb7a4@195.205.186.7
CSeq: 102 INVITE
Contact: <sip:10.0.230.14:5060>
User-Agent: Firefly
Content-Length: 0
9 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
To: <sip:478@10.0.230.14:5060>;tag=c84d4d07
From: 172<sip:172@195.205.186.7>;tag=as24721ef0
Via: SIP/2.0/UDP
195.205.186.7:5060;branch=z9hG4bK4541e422;received=195.205.186.7
Call-ID: 6238a0627e65947105ac1c004ecbb7a4@195.205.186.7
CSeq: 102 INVITE
Contact: <sip:10.0.230.14:5060>
User-Agent: Firefly
Content-Length: 0
9 headers, 0 lines
-- SIP/478-2750 is ringing
-- INFO_IND ID=001 #0x0e2d LEN=0017
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x8
InfoElement = <81 90>
-- DISCONNECT_IND ID=001 #0x0e2e LEN=0014
Controller/PLCI/NCCI = 0x101
Reason = 0x3490
== DISCONNECT_IND PLCI=0x101 REASON=0x3490
Reliably Transmitting:
CANCEL sip:478@10.0.230.14:5060 SIP/2.0
Via: SIP/2.0/UDP 195.205.186.7:5060;branch=z9hG4bK4541e422
From: 172 <sip:172@195.205.186.7>;tag=as24721ef0
To: <sip:478@10.0.230.14:5060>
Contact: <sip:172@195.205.186.7>
Call-ID: 6238a0627e65947105ac1c004ecbb7a4@195.205.186.7
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 10.0.230.14:5060
Scheduling destruction of call
'6238a0627e65947105ac1c004ecbb7a4@195.205.186.7' in 15000 ms
== Spawn extension (from-isdn, s, 2) exited non-zero on
'CAPI[contr1/153]/6'
Sip read:
SIP/2.0 200 OK
To: <sip:478@10.0.230.14:5060>;tag=c84d4d07
From: 172 <sip:172@195.205.186.7>;tag=as24721ef0
Via: SIP/2.0/UDP
195.205.186.7:5060;branch=z9hG4bK4541e422;received=195.205.186.7
Call-ID: 6238a0627e65947105ac1c004ecbb7a4@195.205.186.7
CSeq: 102 CANCEL
Contact: <sip:10.0.230.14:5060>
User-Agent: Firefly
Content-Length: 0
9 headers, 0 lines
Destroying call 'ba7cb64ac679144b@Z3J1Ynk.'
Destroying call '6238a0627e65947105ac1c004ecbb7a4@195.205.186.7'
I can talk with sip client but sip client can't dial-out using isdn line
(sip-cli -> isdn)
Best Regards
Pawe? Staszewski
ART-COM
+48327522333
+480609183038
>>>asterisk@ropeguru.com 04/21/05 2:57 pm >>>
<SNIP>
>> == DISCONNECT_IND PLCI=0x101 REASON=0x3481
>> == No one is available to answer at this time
>>
>
>How changing from CAPI to a zaphfc card will correct
>this error I don't
>know, and problably neither does the person who
>suggested it.
>
>REASON 0x3481 is Unallocated (unassigned) number. >Wrong number.
>
>--
>Dave Cotton <dcotton@linuxautrement.com>
>
Just as a shot in the dark, but does the telco maybe
require 10 digit dialing for ISDN??
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050421/8076708e/attachment.htm
Peer Oliver Schmidt
2005-Apr-21 08:24 UTC
Odp: Re: [Asterisk-Users] capi problem with dialout
Pawe? Staszewski wrote:> > Hello > > I live in poland and :) > local numbers are: 752xxxx (7 digits) > zone prefix: 32 > country prefix: 48 > > And i must add that i am behind a local PBX (Alcatel 4200E) > Configured isdn port with msn 7523071 > > Why dial in is working but dial-out not ... ??maybe your local PBX requires a 0 in front for an outside line? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA