Check your trunk @pstn-out there's something reaching that server
192.168.1.4?
2014-04-09 12:06 GMT-05:00 Luis Eduardo Cortes <luedcor...@gmail.com>:
> Hello:
>
> I have this situation: I can make calls internally, I can make inbound
> calls but I can't make outbound calls.
>
> Thanks in advance.
>
>
>
> These are my devices:
> * asterisk 11.8.1 = 192.168.1.22
> * sipphone grandstream gxp2160 = 192.168.1.5
> * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4
> port 1 (FXS) connected to an analog phone
> port 3 (FXO) connected to the PSTN
>
> These are my sip.conf and extensions.conf files:
>
> sip.conf
> --------
> [general]
> context = incoming-call
> allowguest = no
> srvlookup = no
> udpbindaddr = 0.0.0.0
> tcpenable = no
> qualify = yes
> language = es
>
> [office](!)
> type = friend
> context = internal-call
> host = dynamic
> nat = force_rport,comedia
> dtmfmode = auto
> disallow = all
> allow = g722
> allow = alaw
> allow = ulaw
>
> [telefono](office)
> description = grandstream gxp2160
> secret = telefono
>
> [celular](office)
> description = samsung gt-s7562
> secret = celular
>
> [fxs](office)
> description = fxs port1
> secret = fxs
>
> [pstn](!)
> nat = no
> canreinvite = no
> dtmfmode = auto
> disallow = all
> allow = g722
> allow = alaw
> allow = ulaw
>
> [pstn-in](pstn)
> description = pstn-in port3
> type = user
> host = dynamic
> secret = pstn-in
> context = incoming-call
>
> [pstn-out](pstn)
> description = pstn-out port3
> type = peer
> host = 192.168.1.4
>
> extensions.conf
> ---------------
> [incoming-call]
> exten => _24872006,1,Answer()
> same => n,Dial(SIP/telefono)
> same => n,Hangup()
>
> [outgoing-call]
> exten => _X.,1,Dial(SIP/${EXTEN}@pstn-out)
>
> [internal-call]
> exten => 101,1,Dial(SIP/telefono)
> exten => 102,1,Dial(SIP/celular)
> exten => 103,1,Dial(SIP/fxs)
> exten => 104,1,Answer()
> same => n,Playback(tt-weasels)
> same => n,Hangup()
> include => outgoing-call
>
> This is the result of "sip show peers"
> --------------------------------------
> Name/username Host Dyn Forcerport Comedia ACL Port
> Status Description
> celular/celular 192.168.1.21 D Yes Yes
> 47747 OK (6 ms) samsung gt-s7562
> fxs/fxs 192.168.1.4 D Yes Yes 5060
> OK (27 ms) fxs port1
> pstn-out 192.168.1.4 No No 5060
> OK (25 ms) pstn-out port3
> telefono/telefono 192.168.1.5 D Yes Yes 1555
> OK (3 ms) grandstream gxp2160
> 4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0
> offline]
>
> This is the result of "sip show users"
> --------------------------------------
> Username Secret Accountcode Def.Context ACL Forcerport
> celular celular internal-call No Yes
> pstn-in pstn-in incoming-call No No
> fxs fxs internal-call No Yes
> telefono telefono internal-call No Yes
> debian-asterisk*CLI>
>
> This is the result of "sip set debug on" when I try to make an outbound
> call:
>
> ----------------------------------------------------------------------------
> <--- SIP read from UDP:192.168.1.5:1555 --->
> INVITE sip:22222222@192.168.1.22 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
> From: <sip:telefono@192.168.1.22>;tag=1524540678
> To: <sip:22222222@192.168.1.22>
> Call-ID: 667168938-155...@bjc.bgi.B.F
> CSeq: 30 INVITE
> Contact: <sip:telefono@192.168.1.5:1555>
> X-Grandstream-PBX: true
> Max-Forwards: 70
> User-Agent: Grandstream GXP2160 1.0.0.17
> Privacy: none
> P-Preferred-Identity: <sip:telefono@192.168.1.22>
> Supported: replaces, path, timer
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
> REFER, UPDATE, MESSAGE
> Content-Type: application/sdp
> Accept: application/sdp, application/dtmf-relay
> Content-Length: 335
>
> v=0
> o=telefono 8000 8000 IN IP4 192.168.1.5
> s=SIP Call
> c=IN IP4 192.168.1.5
> t=0 0
> m=audio 5004 RTP/AVP 0 8 18 9 2 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:9 G722/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> <------------->
> --- (17 headers 16 lines) ---
> Sending to 192.168.1.5:1555 (no NAT)
> Sending to 192.168.1.5:1555 (no NAT)
> Using INVITE request as basis request - 667168938-155...@bjc.bgi.B.F
> Found peer 'telefono' for 'telefono' from 192.168.1.5:1555
>
> <--- Reliably Transmitting (NAT) to 192.168.1.5:1555 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 192.168.1.5:1555;branch=z9hG4bK2009427179;received=192.168.1.5;rport=1555
> From: <sip:telefono@192.168.1.22>;tag=1524540678
> To: <sip:22222222@192.168.1.22>;tag=as50d1512e
> Call-ID: 667168938-155...@bjc.bgi.B.F
> CSeq: 30 INVITE
> Server: Asterisk PBX 11.8.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1032f9e6"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '667168938-155...@bjc.bgi.B.F' in
> 6400 ms (Method: INVITE)
>
> <--- SIP read from UDP:192.168.1.5:1555 --->
> ACK sip:22222222@192.168.1.22 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
> From: <sip:telefono@192.168.1.22>;tag=1524540678
> To: <sip:22222222@192.168.1.22>;tag=as50d1512e
> Call-ID: 667168938-155...@bjc.bgi.B.F
> CSeq: 30 ACK
> Content-Length: 0
>
> <------------->
> --- (7 headers 0 lines) ---
>
> <--- SIP read from UDP:192.168.1.5:1555 --->
> INVITE sip:22222222@192.168.1.22 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport
> From: <sip:telefono@192.168.1.22>;tag=1524540678
> To: <sip:22222222@192.168.1.22>
> Call-ID: 667168938-155...@bjc.bgi.B.F
> CSeq: 31 INVITE
> Contact: <sip:telefono@192.168.1.5:1555>
> Authorization: Digest username="telefono", realm="asterisk",
> nonce="1032f9e6", uri="sip:22222222@192.168.1.22",
> response="491072c64fd264bd28d0ac088a738dc3", algorithm=MD5
> X-Grandstream-PBX: true
> Max-Forwards: 70
> User-Agent: Grandstream GXP2160 1.0.0.17
> Privacy: none
> P-Preferred-Identity: <sip:telefono@192.168.1.22>
> Supported: replaces, path, timer
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
> REFER, UPDATE, MESSAGE
> Content-Type: application/sdp
> Accept: application/sdp, application/dtmf-relay
> Content-Length: 335
>
> v=0
> o=telefono 8000 8000 IN IP4 192.168.1.5
> s=SIP Call
> c=IN IP4 192.168.1.5
> t=0 0
> m=audio 5004 RTP/AVP 0 8 18 9 2 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:9 G722/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> <------------->
> --- (18 headers 16 lines) ---
> Sending to 192.168.1.5:1555 (NAT)
> Using INVITE request as basis request - 667168938-155...@bjc.bgi.B.F
> Found peer 'telefono' for 'telefono' from 192.168.1.5:1555
> == Using SIP RTP CoS mark 5
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 9
> Found RTP audio format 2
> Found RTP audio format 101
> Found audio description format PCMU for ID 0
> Found audio description format PCMA for ID 8
> Found audio description format G729 for ID 18
> Found audio description format G722 for ID 9
> Found audio description format G726-32 for ID 2
> Found audio description format telephone-event for ID 101
> Capabilities: us - (ulaw|alaw|g722), peer -
> audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing),
> combined - (ulaw|alaw|g722)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
> (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 192.168.1.5:5004
> Looking for 22222222 in internal-call (domain 192.168.1.22)
> list_route: hop: <sip:telefono@192.168.1.5:1555>
>
> <--- Transmitting (NAT) to 192.168.1.5:1555 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.1.5:1555;branch=z9hG4bK415263616;received=192.168.1.5;rport=1555
> From: <sip:telefono@192.168.1.22>;tag=1524540678
> To: <sip:22222222@192.168.1.22>
> Call-ID: 667168938-155...@bjc.bgi.B.F
> CSeq: 31 INVITE
> Server: Asterisk PBX 11.8.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:22222222@192.168.1.22:5060>
> Content-Length: 0
>
>
> <------------>
> -- Executing [22222222@internal-call:1]
> Dial("SIP/telefono-00000004", "SIP/22222222@pstn-out") in new stack
> == Using SIP RTP CoS mark 5
> Audio is at 29272
> Adding codec 100012 (g722) to SDP
> Adding codec 100004 (alaw) to SDP
> Adding codec 100003 (ulaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 192.168.1.4:5060:
> INVITE sip:22222222@192.168.1.4 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.22:5060;branch=z9hG4bK3f81cf2e
> Max-Forwards: 70
> From: <sip:telefono@192.168.1.22>;tag=as7cd8ea4c
> To: <sip:22222222@192.168.1.4>
> Contact: <sip:telefono@192.168.1.22:5060>
> Call-ID: 323866b71557eac419f667ee37ee16ae@192.168.1.22:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 11.8.1
> Date: Wed, 09 Apr 2014 15:00:11 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 281
>
> v=0
> o=root 268828888 268828888 IN IP4 192.168.1.22
> s=Asterisk PBX 11.8.1
> c=IN IP4 192.168.1.22
> t=0 0
> m=audio 29272 RTP/AVP 9 8 0 101
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---
> -- Called SIP/22222222@pstn-out
>
> <--- SIP read from UDP:192.168.1.4:5060 --->
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 192.168.1.22:5060;branch=z9hG4bK3f81cf2e
> From: <sip:telefono@192.168.1.22>;tag=as7cd8ea4c
> To: <sip:22222222@192.168.1.4>;tag=1c1296932060
> Call-ID: 323866b71557eac419f667ee37ee16ae@192.168.1.22:5060
> CSeq: 102 INVITE
> Allow:
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
> Server: MP-114 FXS_FXO/v.6.60A.041.005
> Reason: Q.850 ;cause=3 ;text="local"
> Content-Length: 0
>
> <------------->
> --- (10 headers 0 lines) ---
> Transmitting (no NAT) to 192.168.1.4:5060:
> ACK sip:22222222@192.168.1.4 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.22:5060;branch=z9hG4bK3f81cf2e
> Max-Forwards: 70
> From: <sip:telefono@192.168.1.22>;tag=as7cd8ea4c
> To: <sip:22222222@192.168.1.4>;tag=1c1296932060
> Contact: <sip:telefono@192.168.1.22:5060>
> Call-ID: 323866b71557eac419f667ee37ee16ae@192.168.1.22:5060
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 11.8.1
> Content-Length: 0
>
>
> ---
> Scheduling destruction of SIP dialog
> '323866b71557eac419f667ee37ee16ae@192.168.1.22:5060' in 6400 ms
> (Method: INVITE)
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Auto fallthrough, channel 'SIP/telefono-00000004' status is
> 'CHANUNAVAIL'
>
> <--- Reliably Transmitting (NAT) to 192.168.1.5:1555 --->
> SIP/2.0 503 Service Unavailable
> Via: SIP/2.0/UDP
> 192.168.1.5:1555;branch=z9hG4bK415263616;received=192.168.1.5;rport=1555
> From: <sip:telefono@192.168.1.22>;tag=1524540678
> To: <sip:22222222@192.168.1.22>;tag=as4caf91d6
> Call-ID: 667168938-155...@bjc.bgi.B.F
> CSeq: 31 INVITE
> Server: Asterisk PBX 11.8.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> X-Asterisk-HangupCause: Unallocated (unassigned) number
> X-Asterisk-HangupCauseCode: 1
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from UDP:192.168.1.5:1555 --->
> ACK sip:22222222@192.168.1.22 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport
> From: <sip:telefono@192.168.1.22>;tag=1524540678
> To: <sip:22222222@192.168.1.22>;tag=as4caf91d6
> Call-ID: 667168938-155...@bjc.bgi.B.F
> CSeq: 31 ACK
> Content-Length: 0
>
> <------------->
> --- (7 headers 0 lines) ---
> Really destroying SIP dialog '667168938-155...@bjc.bgi.B.F' Method: ACK
> debian-asterisk*CLI> sip set debug off
> SIP Debugging Disabled
> debian-asterisk*CLI>
>
>
>
>
>
> --
> Usuario Linux Registrado # 342019
> --> http://linuxcounter.net/ <--
> skype --> luedcortes
> gtalk --> luedcor...@gmail.com
> msn --> luedcor...@gmail.com
>
> --
> _____________________________________________________________________
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>
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>
--
Saludos
Gustavo Ch. Apaza
Core Consulting Group
CEL. 993783686
--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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