I got it !!
host=192.168.0.151
port=5060
type=friend
nat=yes
qualify=yes
fromdomain=192.168.0.151
insecure=invite,port
dtmfmode=auto
disallow=all
allow=alaw&g729 -----<<<<<-----here! make a tention at the order! G729 is
not allowed !
i reorder it get work!!
thks a lot,all !
On 26 March 2010 13:44, Alyed <al...@vivoxie.com> wrote:
> it doesn't seems to be a problem of communication between A and B
>
>
> > -- Executing [...@macro-dialout-trunk:19]
> Dial("SIP/192.168.0.151-088e7938",
> "ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack
> > == Everyone is busy/congested at this time (1:0/0/1)
>
> That's says it's more a problem with your Zap channels than your SIP
> connection.
>
> First try playing a sound in B when receiving the call, that way you can be
> sure the connection is ok. If that one works then move to PSTN.
>
> Alyed
>
>
> 2010/3/25 Aaron chen <evane1...@gmail.com>
>
>> i have a prablom here,
>>
>> i want to send a call from A to B use sip trunk ,
>>
>> the call can sended B,but not work to PSTN.
>>
>> the message from B server. help pls,what's rong?
>>
>>
>>
>>>
>>> <--- SIP read from 192.168.0.176:5060 --->
>>> INVITE sip:15921256...@192.168.0.151
>>> <sip%3a15921256...@192.168.0.151>SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
>>> From: "50005" <sip:50...@192.168.0.151 <sip%3a50...@192.168.0.151>
>>> >;tag=as72a55960
>>> To: <sip:15921256...@192.168.0.151 <sip%3a15921256...@192.168.0.151>>
>>> Contact: <sip:50...@192.168.0.176 <sip%3a50...@192.168.0.176>>
>>> Call-ID: 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX
>>> Max-Forwards: 70
>>> Date: Fri, 26 Mar 2010 02:12:07 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>> Supported: replaces
>>> Content-Type: application/sdp
>>> Content-Length: 380
>>> v=0
>>> o=root 15081 15081 IN IP4 192.168.0.176
>>> s=session
>>> c=IN IP4 192.168.0.176
>>> t=0 0
>>> m=audio 12726 RTP/AVP 0 18 8 3 4 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:18 G729/8000
>>> a=fmtp:18 annexb=no
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:4 G723/8000
>>> a=fmtp:4 annexa=no
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>> <------------->
>>> --- (14 headers 18 lines) ---
>>> Sending to 192.168.0.176 : 5060 (NAT)
>>> Using INVITE request as basis request -
>>> 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>>> Found peer 's1'
>>> Found RTP audio format 0
>>> Found RTP audio format 18
>>> Found RTP audio format 8
>>> Found RTP audio format 3
>>> Found RTP audio format 4
>>> Found RTP audio format 101
>>> Peer audio RTP is at port 192.168.0.176:12726
>>> Found audio description format PCMU for ID 0
>>> Found audio description format G729 for ID 18
>>> Found audio description format PCMA for ID 8
>>> Found audio description format GSM for ID 3
>>> Found audio description format G723 for ID 4
>>> Found audio description format telephone-event for ID 101
>>> Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10f
>>> (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10f
>>> (g723|gsm|ulaw|alaw|g729)
>>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
>>> (telephone-event), combined - 0x1 (telephone-event)
>>> Peer audio RTP is at port 192.168.0.176:12726
>>> Looking for 15921256331 in from-internal (domain 192.168.0.151)
>>> list_route: hop: <sip:50...@192.168.0.176 <sip%3a50...@192.168.0.176>>
>>> gd-branch*CLI>
>>> <--- Transmitting (NAT) to 192.168.0.176:5060 --->
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 192.168.0.176:5060
>>> ;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060
>>> From: "50005" <sip:50...@192.168.0.151 <sip%3a50...@192.168.0.151>
>>> >;tag=as72a55960
>>> To: <sip:15921256...@192.168.0.151 <sip%3a15921256...@192.168.0.151>>
>>> Call-ID: 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Supported: replaces
>>> Contact: <sip:15921256...@192.168.0.151<sip%3a15921256...@192.168.0.151>
>>> >
>>> Content-Length: 0
>>>
>>> <------------>
>>> -- Executing [15921256...@from-internal:1]
>>> Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none") in new stack
>>> -- Executing [15921256...@from-internal:2]
>>> Macro("SIP/192.168.0.151-088e7938", "user-callerid|SKIPTTL|") in new stack
>>> -- Executing [...@macro-user-callerid:1]
>>> Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005") in new stack
>>> -- Executing [...@macro-user-callerid:2]
>>> GotoIf("SIP/192.168.0.151-088e7938", "0?report") in new stack
>>> -- Executing [...@macro-user-callerid:3]
>>> ExecIf("SIP/192.168.0.151-088e7938", "1|Set|REALCALLERIDNUM=50005") in new
>>> stack
>>> -- Executing [...@macro-user-callerid:4]
>>> Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in new stack
>>> -- Executing [...@macro-user-callerid:5]
>>> Set("SIP/192.168.0.151-088e7938", "AMPUSERCIDNAME=") in new stack
>>> -- Executing [...@macro-user-callerid:6]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?report") in new stack
>>> -- Goto (macro-user-callerid,s,10)
>>> -- Executing [...@macro-user-callerid:10]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?continue") in new stack
>>> -- Goto (macro-user-callerid,s,19)
>>> -- Executing [...@macro-user-callerid:19]
>>> NoOp("SIP/192.168.0.151-088e7938", "Using CallerID "50005" <50005>") in new
>>> stack
>>> -- Executing [15921256...@from-internal:3]
>>> Set("SIP/192.168.0.151-088e7938", "_NODEST=") in new stack
>>> -- Executing [15921256...@from-internal:4]
>>> Macro("SIP/192.168.0.151-088e7938", "record-enable||OUT|") in new stack
>>> -- Executing [...@macro-record-enable:1]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?check") in new stack
>>> -- Goto (macro-record-enable,s,4)
>>> -- Executing [...@macro-record-enable:4]
>>> AGI("SIP/192.168.0.151-088e7938",
>>> "recordingcheck|20100326-101436|1269569676.20") in new stack
>>> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
>>> recordingcheck|20100326-101436|1269569676.20: No AMPUSER db entry for .
>>> Not recording
>>> -- AGI Script recordingcheck completed, returning 0
>>> -- Executing [...@macro-record-enable:5]
>>> MacroExit("SIP/192.168.0.151-088e7938", "") in new stack
>>> -- Executing [15921256...@from-internal:5]
>>> Macro("SIP/192.168.0.151-088e7938", "dialout-trunk|1|15921256331||") in new
>>> stack
>>> -- Executing [...@macro-dialout-trunk:1]
>>> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK=1") in new stack
>>> -- Executing [...@macro-dialout-trunk:2]
>>> GosubIf("SIP/192.168.0.151-088e7938", "0?sub-pincheck|s|1") in new stack
>>> -- Executing [...@macro-dialout-trunk:3]
>>> GotoIf("SIP/192.168.0.151-088e7938", "0?disabletrunk|1") in new stack
>>> -- Executing [...@macro-dialout-trunk:4]
>>> Set("SIP/192.168.0.151-088e7938", "DIAL_NUMBER=15921256331") in new stack
>>> -- Executing [...@macro-dialout-trunk:5]
>>> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
>>> -- Executing [...@macro-dialout-trunk:6]
>>> Set("SIP/192.168.0.151-088e7938", "OUTBOUND_GROUP=OUT_1") in new stack
>>> -- Executing [...@macro-dialout-trunk:7]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new stack
>>> -- Goto (macro-dialout-trunk,s,9)
>>> -- Executing [...@macro-dialout-trunk:9]
>>> GotoIf("SIP/192.168.0.151-088e7938", "0?skipoutcid") in new stack
>>> -- Executing [...@macro-dialout-trunk:10]
>>> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Tt") in new stack
>>> -- Executing [...@macro-dialout-trunk:11]
>>> Macro("SIP/192.168.0.151-088e7938", "outbound-callerid|1") in new stack
>>> -- Executing [...@macro-outbound-callerid:1]
>>> ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|") in new stack
>>> -- Executing [...@macro-outbound-callerid:2]
>>> ExecIf("SIP/192.168.0.151-088e7938", "0|Set|REALCALLERIDNUM=50005") in new
>>> stack
>>> -- Executing [...@macro-outbound-callerid:3]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?normcid") in new stack
>>> -- Goto (macro-outbound-callerid,s,6)
>>> -- Executing [...@macro-outbound-callerid:6]
>>> Set("SIP/192.168.0.151-088e7938", "USEROUTCID=") in new stack
>>> -- Executing [...@macro-outbound-callerid:7]
>>> Set("SIP/192.168.0.151-088e7938", "EMERGENCYCID=") in new stack
>>> -- Executing [...@macro-outbound-callerid:8]
>>> Set("SIP/192.168.0.151-088e7938", "TRUNKOUTCID=64858162") in new stack
>>> -- Executing [...@macro-outbound-callerid:9]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?trunkcid") in new stack
>>> -- Goto (macro-outbound-callerid,s,12)
>>> -- Executing [...@macro-outbound-callerid:12]
>>> ExecIf("SIP/192.168.0.151-088e7938", "1|Set|CALLERID(all)=64858162") in new
>>> stack
>>> -- Executing [...@macro-outbound-callerid:13]
>>> ExecIf("SIP/192.168.0.151-088e7938", "0|Set|CALLERID(all)=") in new stack
>>> -- Executing [...@macro-outbound-callerid:14]
>>> ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|prohib_passed_screen")
>>> in new stack
>>> -- Executing [...@macro-dialout-trunk:12]
>>> ExecIf("SIP/192.168.0.151-088e7938", "0|AGI|fixlocalprefix") in new stack
>>> -- Executing [...@macro-dialout-trunk:13]
>>> Set("SIP/192.168.0.151-088e7938", "OUTNUM=15921256331") in new stack
>>> -- Executing [...@macro-dialout-trunk:14]
>>> Set("SIP/192.168.0.151-088e7938", "custom=ZAP/g0") in new stack
>>> -- Executing [...@macro-dialout-trunk:15]
>>> ExecIf("SIP/192.168.0.151-088e7938",
>>> "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack
>>> -- Executing [...@macro-dialout-trunk:16]
>>> Macro("SIP/192.168.0.151-088e7938", "dialout-trunk-predial-hook|") in new
>>> stack
>>> -- Executing [...@macro-dialout-trunk-predial-hook:1]
>>> MacroExit("SIP/192.168.0.151-088e7938", "") in new stack
>>> -- Executing [...@macro-dialout-trunk:17]
>>> GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1") in new stack
>>> -- Executing [...@macro-dialout-trunk:18]
>>> GotoIf("SIP/192.168.0.151-088e7938", "0?customtrunk") in new stack
>>> -- Executing [...@macro-dialout-trunk:19]
>>> Dial("SIP/192.168.0.151-088e7938",
>>> "ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack
>>> == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Executing [...@macro-dialout-trunk:20]
>>> Goto("SIP/192.168.0.151-088e7938", "s-CHANUNAVAIL|1") in new stack
>>> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
>>> -- Executing [s-chanunav...@macro-dialout-trunk:1]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?noreport") in new stack
>>> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
>>> -- Executing [s-chanunav...@macro-dialout-trunk:3]
>>> NoOp("SIP/192.168.0.151-088e7938", "TRUNK Dial failed due to CHANUNAVAIL
>>> (hangupcause: 58) - failing through to other trunks") in new stack
>>> -- Executing [15921256...@from-internal:6]
>>> Macro("SIP/192.168.0.151-088e7938", "outisbusy|") in new stack
>>> -- Executing [...@macro-outisbusy:1]
>>> Playback("SIP/192.168.0.151-088e7938", "all-circuits-busy-now|noanswer") in
>>> new stack
>>> -- Executing [...@macro-outisbusy:2]
>>> Playback("SIP/192.168.0.151-088e7938", "pls-try-call-later|noanswer") in new
>>> stack
>>> -- Executing [...@macro-outisbusy:3]
>>> Macro("SIP/192.168.0.151-088e7938", "hangupcall") in new stack
>>> -- Executing [...@macro-hangupcall:1]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?skiprg") in new stack
>>> -- Goto (macro-hangupcall,s,4)
>>> -- Executing [...@macro-hangupcall:4]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?skipblkvm") in new stack
>>> -- Goto (macro-hangupcall,s,7)
>>> -- Executing [...@macro-hangupcall:7]
>>> GotoIf("SIP/192.168.0.151-088e7938", "1?theend") in new stack
>>> -- Goto (macro-hangupcall,s,9)
>>> -- Executing [...@macro-hangupcall:9]
>>> Hangup("SIP/192.168.0.151-088e7938", "") in new stack
>>> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
>>> 'SIP/192.168.0.151-088e7938' in macro 'hangupcall'
>>> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
>>> 'SIP/192.168.0.151-088e7938' in macro 'outisbusy'
>>> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
>>> 'SIP/192.168.0.151-088e7938'
>>> Scheduling destruction of SIP dialog
>>> '28272ebb12ee6e4c1f06fca651456...@192.168.0.151'<%2728272ebb12ee6e4c1f06fca651456...@192.168.0.151%27>in
>>> 6400 ms (Method: INVITE)
>>> gd-branch*CLI>
>>> <--- Reliably Transmitting (NAT) to 192.168.0.176:5060 --->
>>> SIP/2.0 488 Not Acceptable Here
>>> Via: SIP/2.0/UDP 192.168.0.176:5060
>>> ;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060
>>> From: "50005" <sip:50...@192.168.0.151 <sip%3a50...@192.168.0.151>
>>> >;tag=as72a55960
>>> To: <sip:15921256...@192.168.0.151 <sip%3a15921256...@192.168.0.151>
>>> >;tag=as12db2697
>>> Call-ID: 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Supported: replaces
>>> Content-Length: 0
>>>
>>> <------------>
>>> gd-branch*CLI>
>>> <--- SIP read from 192.168.0.176:5060 --->
>>> ACK sip:15921256...@192.168.0.151 <sip%3a15921256...@192.168.0.151>SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
>>> From: "50005" <sip:50...@192.168.0.151 <sip%3a50...@192.168.0.151>
>>> >;tag=as72a55960
>>> To: <sip:15921256...@192.168.0.151 <sip%3a15921256...@192.168.0.151>
>>> >;tag=as12db2697
>>> Contact: <sip:50...@192.168.0.176 <sip%3a50...@192.168.0.176>>
>>> Call-ID: 28272ebb12ee6e4c1f06fca651456...@192.168.0.151
>>> CSeq: 102 ACK
>>> User-Agent: Asterisk PBX
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> <------------->
>>> --- (10 headers 0 lines) ---
>>> sip no debug
>>> SIP Debugging Disabled
>>>
>>
>>
>> Best regards!
>>
>> Aaron Chen
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
祝您愉快！！
Aaron Chen
陈江涛
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
