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