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Aaron chen

2010-Mar-26 02:37 UTC

[asterisk-users] send a call from A to B use sip trunk prablem

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:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at
192.168.0.151>
> >;tag=as72a55960
> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>>
> Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at
192.168.0.176>>
> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 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 -
> 28272ebb12ee6e4c1f06fca651456469 at 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:50005 at 192.168.0.176 <sip%3A50005 at
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:50005 at 192.168.0.151 <sip%3A50005 at
192.168.0.151>
> >;tag=as72a55960
> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>>
> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>>
> Content-Length: 0
>
> <------------>
>     -- Executing [15921256331 at from-internal:1]
> Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none") in
new stack
>     -- Executing [15921256331 at from-internal:2]
> Macro("SIP/192.168.0.151-088e7938",
"user-callerid|SKIPTTL|") in new stack
>     -- Executing [s at macro-user-callerid:1]
> Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005") in
new stack
>     -- Executing [s at macro-user-callerid:2]
> GotoIf("SIP/192.168.0.151-088e7938", "0?report") in new
stack
>     -- Executing [s at macro-user-callerid:3]
> ExecIf("SIP/192.168.0.151-088e7938",
"1|Set|REALCALLERIDNUM=50005") in new
> stack
>     -- Executing [s at macro-user-callerid:4]
> Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in new
stack
>     -- Executing [s at macro-user-callerid:5]
> Set("SIP/192.168.0.151-088e7938", "AMPUSERCIDNAME=") in
new stack
>     -- Executing [s at macro-user-callerid:6]
> GotoIf("SIP/192.168.0.151-088e7938", "1?report") in new
stack
>     -- Goto (macro-user-callerid,s,10)
>     -- Executing [s at macro-user-callerid:10]
> GotoIf("SIP/192.168.0.151-088e7938", "1?continue") in
new stack
>     -- Goto (macro-user-callerid,s,19)
>     -- Executing [s at macro-user-callerid:19]
> NoOp("SIP/192.168.0.151-088e7938", "Using CallerID
"50005" <50005>") in new
> stack
>     -- Executing [15921256331 at from-internal:3]
> Set("SIP/192.168.0.151-088e7938", "_NODEST=") in new
stack
>     -- Executing [15921256331 at from-internal:4]
> Macro("SIP/192.168.0.151-088e7938",
"record-enable||OUT|") in new stack
>     -- Executing [s at macro-record-enable:1]
> GotoIf("SIP/192.168.0.151-088e7938", "1?check") in new
stack
>     -- Goto (macro-record-enable,s,4)
>     -- Executing [s at 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 [s at macro-record-enable:5]
> MacroExit("SIP/192.168.0.151-088e7938", "") in new
stack
>     -- Executing [15921256331 at from-internal:5]
> Macro("SIP/192.168.0.151-088e7938",
"dialout-trunk|1|15921256331||") in new
> stack
>     -- Executing [s at macro-dialout-trunk:1]
> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK=1") in
new stack
>     -- Executing [s at macro-dialout-trunk:2]
> GosubIf("SIP/192.168.0.151-088e7938",
"0?sub-pincheck|s|1") in new stack
>     -- Executing [s at macro-dialout-trunk:3]
> GotoIf("SIP/192.168.0.151-088e7938",
"0?disabletrunk|1") in new stack
>     -- Executing [s at macro-dialout-trunk:4]
> Set("SIP/192.168.0.151-088e7938",
"DIAL_NUMBER=15921256331") in new stack
>     -- Executing [s at macro-dialout-trunk:5]
> Set("SIP/192.168.0.151-088e7938",
"DIAL_TRUNK_OPTIONS=Ttr") in new stack
>     -- Executing [s at macro-dialout-trunk:6]
> Set("SIP/192.168.0.151-088e7938",
"OUTBOUND_GROUP=OUT_1") in new stack
>     -- Executing [s at macro-dialout-trunk:7]
> GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new
stack
>     -- Goto (macro-dialout-trunk,s,9)
>     -- Executing [s at macro-dialout-trunk:9]
> GotoIf("SIP/192.168.0.151-088e7938", "0?skipoutcid") in
new stack
>     -- Executing [s at macro-dialout-trunk:10]
> Set("SIP/192.168.0.151-088e7938",
"DIAL_TRUNK_OPTIONS=Tt") in new stack
>     -- Executing [s at macro-dialout-trunk:11]
> Macro("SIP/192.168.0.151-088e7938",
"outbound-callerid|1") in new stack
>     -- Executing [s at macro-outbound-callerid:1]
> ExecIf("SIP/192.168.0.151-088e7938",
"0|SetCallerPres|") in new stack
>     -- Executing [s at macro-outbound-callerid:2]
> ExecIf("SIP/192.168.0.151-088e7938",
"0|Set|REALCALLERIDNUM=50005") in new
> stack
>     -- Executing [s at macro-outbound-callerid:3]
> GotoIf("SIP/192.168.0.151-088e7938", "1?normcid") in
new stack
>     -- Goto (macro-outbound-callerid,s,6)
>     -- Executing [s at macro-outbound-callerid:6]
> Set("SIP/192.168.0.151-088e7938", "USEROUTCID=") in new
stack
>     -- Executing [s at macro-outbound-callerid:7]
> Set("SIP/192.168.0.151-088e7938", "EMERGENCYCID=") in
new stack
>     -- Executing [s at macro-outbound-callerid:8]
> Set("SIP/192.168.0.151-088e7938",
"TRUNKOUTCID=64858162") in new stack
>     -- Executing [s at macro-outbound-callerid:9]
> GotoIf("SIP/192.168.0.151-088e7938", "1?trunkcid") in
new stack
>     -- Goto (macro-outbound-callerid,s,12)
>     -- Executing [s at macro-outbound-callerid:12]
> ExecIf("SIP/192.168.0.151-088e7938",
"1|Set|CALLERID(all)=64858162") in new
> stack
>     -- Executing [s at macro-outbound-callerid:13]
> ExecIf("SIP/192.168.0.151-088e7938",
"0|Set|CALLERID(all)=") in new stack
>     -- Executing [s at macro-outbound-callerid:14]
> ExecIf("SIP/192.168.0.151-088e7938",
"0|SetCallerPres|prohib_passed_screen")
> in new stack
>     -- Executing [s at macro-dialout-trunk:12]
> ExecIf("SIP/192.168.0.151-088e7938",
"0|AGI|fixlocalprefix") in new stack
>     -- Executing [s at macro-dialout-trunk:13]
> Set("SIP/192.168.0.151-088e7938", "OUTNUM=15921256331")
in new stack
>     -- Executing [s at macro-dialout-trunk:14]
> Set("SIP/192.168.0.151-088e7938", "custom=ZAP/g0") in
new stack
>     -- Executing [s at macro-dialout-trunk:15]
> ExecIf("SIP/192.168.0.151-088e7938",
> "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack
>     -- Executing [s at macro-dialout-trunk:16]
> Macro("SIP/192.168.0.151-088e7938",
"dialout-trunk-predial-hook|") in new
> stack
>     -- Executing [s at macro-dialout-trunk-predial-hook:1]
> MacroExit("SIP/192.168.0.151-088e7938", "") in new
stack
>     -- Executing [s at macro-dialout-trunk:17]
> GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1") in
new stack
>     -- Executing [s at macro-dialout-trunk:18]
> GotoIf("SIP/192.168.0.151-088e7938", "0?customtrunk")
in new stack
>     -- Executing [s at 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 [s at 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-CHANUNAVAIL at 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-CHANUNAVAIL at 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 [15921256331 at from-internal:6]
> Macro("SIP/192.168.0.151-088e7938", "outisbusy|") in
new stack
>     -- Executing [s at macro-outisbusy:1]
> Playback("SIP/192.168.0.151-088e7938",
"all-circuits-busy-now|noanswer") in
> new stack
>     -- Executing [s at macro-outisbusy:2]
> Playback("SIP/192.168.0.151-088e7938",
"pls-try-call-later|noanswer") in new
> stack
>     -- Executing [s at macro-outisbusy:3]
Macro("SIP/192.168.0.151-088e7938",
> "hangupcall") in new stack
>     -- Executing [s at macro-hangupcall:1]
> GotoIf("SIP/192.168.0.151-088e7938", "1?skiprg") in new
stack
>     -- Goto (macro-hangupcall,s,4)
>     -- Executing [s at macro-hangupcall:4]
> GotoIf("SIP/192.168.0.151-088e7938", "1?skipblkvm") in
new stack
>     -- Goto (macro-hangupcall,s,7)
>     -- Executing [s at macro-hangupcall:7]
> GotoIf("SIP/192.168.0.151-088e7938", "1?theend") in new
stack
>     -- Goto (macro-hangupcall,s,9)
>     -- Executing [s at 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
> '28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151' 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:50005 at 192.168.0.151 <sip%3A50005 at
192.168.0.151>
> >;tag=as72a55960
> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>
> >;tag=as12db2697
> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 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:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at
192.168.0.151>
> >;tag=as72a55960
> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>
> >;tag=as12db2697
> Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at
192.168.0.176>>
> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 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
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Alyed

2010-Mar-26 05:44 UTC

head link

[asterisk-users] send a call from A to B use sip trunk prablem

it doesn't seems to be a problem of communication between A and B
>    -- Executing [s at 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 <evane1890 at 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:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>SIP/2.0
>> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
>> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005
at 192.168.0.151>
>> >;tag=as72a55960
>> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>>
>> Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at
192.168.0.176>>
>> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 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 -
>> 28272ebb12ee6e4c1f06fca651456469 at 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:50005 at 192.168.0.176 <sip%3A50005 at
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:50005 at 192.168.0.151 <sip%3A50005
at 192.168.0.151>
>> >;tag=as72a55960
>> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>>
>> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Contact: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>
>> >
>> Content-Length: 0
>>
>> <------------>
>>     -- Executing [15921256331 at from-internal:1]
>> Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none")
in new stack
>>     -- Executing [15921256331 at from-internal:2]
>> Macro("SIP/192.168.0.151-088e7938",
"user-callerid|SKIPTTL|") in new stack
>>     -- Executing [s at macro-user-callerid:1]
>> Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005")
in new stack
>>     -- Executing [s at macro-user-callerid:2]
>> GotoIf("SIP/192.168.0.151-088e7938", "0?report") in
new stack
>>     -- Executing [s at macro-user-callerid:3]
>> ExecIf("SIP/192.168.0.151-088e7938",
"1|Set|REALCALLERIDNUM=50005") in new
>> stack
>>     -- Executing [s at macro-user-callerid:4]
>> Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in
new stack
>>     -- Executing [s at macro-user-callerid:5]
>> Set("SIP/192.168.0.151-088e7938",
"AMPUSERCIDNAME=") in new stack
>>     -- Executing [s at macro-user-callerid:6]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?report") in
new stack
>>     -- Goto (macro-user-callerid,s,10)
>>     -- Executing [s at macro-user-callerid:10]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?continue")
in new stack
>>     -- Goto (macro-user-callerid,s,19)
>>     -- Executing [s at macro-user-callerid:19]
>> NoOp("SIP/192.168.0.151-088e7938", "Using CallerID
"50005" <50005>") in new
>> stack
>>     -- Executing [15921256331 at from-internal:3]
>> Set("SIP/192.168.0.151-088e7938", "_NODEST=") in
new stack
>>     -- Executing [15921256331 at from-internal:4]
>> Macro("SIP/192.168.0.151-088e7938",
"record-enable||OUT|") in new stack
>>     -- Executing [s at macro-record-enable:1]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?check") in
new stack
>>     -- Goto (macro-record-enable,s,4)
>>     -- Executing [s at 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 [s at macro-record-enable:5]
>> MacroExit("SIP/192.168.0.151-088e7938", "") in new
stack
>>     -- Executing [15921256331 at from-internal:5]
>> Macro("SIP/192.168.0.151-088e7938",
"dialout-trunk|1|15921256331||") in new
>> stack
>>     -- Executing [s at macro-dialout-trunk:1]
>> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK=1")
in new stack
>>     -- Executing [s at macro-dialout-trunk:2]
>> GosubIf("SIP/192.168.0.151-088e7938",
"0?sub-pincheck|s|1") in new stack
>>     -- Executing [s at macro-dialout-trunk:3]
>> GotoIf("SIP/192.168.0.151-088e7938",
"0?disabletrunk|1") in new stack
>>     -- Executing [s at macro-dialout-trunk:4]
>> Set("SIP/192.168.0.151-088e7938",
"DIAL_NUMBER=15921256331") in new stack
>>     -- Executing [s at macro-dialout-trunk:5]
>> Set("SIP/192.168.0.151-088e7938",
"DIAL_TRUNK_OPTIONS=Ttr") in new stack
>>     -- Executing [s at macro-dialout-trunk:6]
>> Set("SIP/192.168.0.151-088e7938",
"OUTBOUND_GROUP=OUT_1") in new stack
>>     -- Executing [s at macro-dialout-trunk:7]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in
new stack
>>     -- Goto (macro-dialout-trunk,s,9)
>>     -- Executing [s at macro-dialout-trunk:9]
>> GotoIf("SIP/192.168.0.151-088e7938",
"0?skipoutcid") in new stack
>>     -- Executing [s at macro-dialout-trunk:10]
>> Set("SIP/192.168.0.151-088e7938",
"DIAL_TRUNK_OPTIONS=Tt") in new stack
>>     -- Executing [s at macro-dialout-trunk:11]
>> Macro("SIP/192.168.0.151-088e7938",
"outbound-callerid|1") in new stack
>>     -- Executing [s at macro-outbound-callerid:1]
>> ExecIf("SIP/192.168.0.151-088e7938",
"0|SetCallerPres|") in new stack
>>     -- Executing [s at macro-outbound-callerid:2]
>> ExecIf("SIP/192.168.0.151-088e7938",
"0|Set|REALCALLERIDNUM=50005") in new
>> stack
>>     -- Executing [s at macro-outbound-callerid:3]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?normcid")
in new stack
>>     -- Goto (macro-outbound-callerid,s,6)
>>     -- Executing [s at macro-outbound-callerid:6]
>> Set("SIP/192.168.0.151-088e7938", "USEROUTCID=") in
new stack
>>     -- Executing [s at macro-outbound-callerid:7]
>> Set("SIP/192.168.0.151-088e7938", "EMERGENCYCID=")
in new stack
>>     -- Executing [s at macro-outbound-callerid:8]
>> Set("SIP/192.168.0.151-088e7938",
"TRUNKOUTCID=64858162") in new stack
>>     -- Executing [s at macro-outbound-callerid:9]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?trunkcid")
in new stack
>>     -- Goto (macro-outbound-callerid,s,12)
>>     -- Executing [s at macro-outbound-callerid:12]
>> ExecIf("SIP/192.168.0.151-088e7938",
"1|Set|CALLERID(all)=64858162") in new
>> stack
>>     -- Executing [s at macro-outbound-callerid:13]
>> ExecIf("SIP/192.168.0.151-088e7938",
"0|Set|CALLERID(all)=") in new stack
>>     -- Executing [s at macro-outbound-callerid:14]
>> ExecIf("SIP/192.168.0.151-088e7938",
"0|SetCallerPres|prohib_passed_screen")
>> in new stack
>>     -- Executing [s at macro-dialout-trunk:12]
>> ExecIf("SIP/192.168.0.151-088e7938",
"0|AGI|fixlocalprefix") in new stack
>>     -- Executing [s at macro-dialout-trunk:13]
>> Set("SIP/192.168.0.151-088e7938",
"OUTNUM=15921256331") in new stack
>>     -- Executing [s at macro-dialout-trunk:14]
>> Set("SIP/192.168.0.151-088e7938", "custom=ZAP/g0")
in new stack
>>     -- Executing [s at macro-dialout-trunk:15]
>> ExecIf("SIP/192.168.0.151-088e7938",
>> "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack
>>     -- Executing [s at macro-dialout-trunk:16]
>> Macro("SIP/192.168.0.151-088e7938",
"dialout-trunk-predial-hook|") in new
>> stack
>>     -- Executing [s at macro-dialout-trunk-predial-hook:1]
>> MacroExit("SIP/192.168.0.151-088e7938", "") in new
stack
>>     -- Executing [s at macro-dialout-trunk:17]
>> GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1")
in new stack
>>     -- Executing [s at macro-dialout-trunk:18]
>> GotoIf("SIP/192.168.0.151-088e7938",
"0?customtrunk") in new stack
>>     -- Executing [s at 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 [s at 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-CHANUNAVAIL at 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-CHANUNAVAIL at 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 [15921256331 at from-internal:6]
>> Macro("SIP/192.168.0.151-088e7938", "outisbusy|")
in new stack
>>     -- Executing [s at macro-outisbusy:1]
>> Playback("SIP/192.168.0.151-088e7938",
"all-circuits-busy-now|noanswer") in
>> new stack
>>     -- Executing [s at macro-outisbusy:2]
>> Playback("SIP/192.168.0.151-088e7938",
"pls-try-call-later|noanswer") in new
>> stack
>>     -- Executing [s at macro-outisbusy:3]
>> Macro("SIP/192.168.0.151-088e7938", "hangupcall")
in new stack
>>     -- Executing [s at macro-hangupcall:1]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?skiprg") in
new stack
>>     -- Goto (macro-hangupcall,s,4)
>>     -- Executing [s at macro-hangupcall:4]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?skipblkvm")
in new stack
>>     -- Goto (macro-hangupcall,s,7)
>>     -- Executing [s at macro-hangupcall:7]
>> GotoIf("SIP/192.168.0.151-088e7938", "1?theend") in
new stack
>>     -- Goto (macro-hangupcall,s,9)
>>     -- Executing [s at 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
>> '28272ebb12ee6e4c1f06fca651456469 at
192.168.0.151'<%2728272ebb12ee6e4c1f06fca651456469 at
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:50005 at 192.168.0.151 <sip%3A50005
at 192.168.0.151>
>> >;tag=as72a55960
>> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>
>> >;tag=as12db2697
>> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 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:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>SIP/2.0
>> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
>> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005
at 192.168.0.151>
>> >;tag=as72a55960
>> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at
192.168.0.151>
>> >;tag=as12db2697
>> Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at
192.168.0.176>>
>> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 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
>
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