Luis Eduardo Cortes
2014-Apr-09 17:06 UTC
[asterisk-users] I can't make outbound calls (status is 'CHANUNAVAIL')
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 at 192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
From: <sip:telefono at 192.168.1.22>;tag=1524540678
To: <sip:22222222 at 192.168.1.22>
Call-ID: 667168938-1555-4 at BJC.BGI.B.F
CSeq: 30 INVITE
Contact: <sip:telefono at 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 at 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 - 667168938-1555-4 at 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 at 192.168.1.22>;tag=1524540678
To: <sip:22222222 at 192.168.1.22>;tag=as50d1512e
Call-ID: 667168938-1555-4 at 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 '667168938-1555-4 at BJC.BGI.B.F'
in
6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.5:1555 --->
ACK sip:22222222 at 192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
From: <sip:telefono at 192.168.1.22>;tag=1524540678
To: <sip:22222222 at 192.168.1.22>;tag=as50d1512e
Call-ID: 667168938-1555-4 at 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 at 192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport
From: <sip:telefono at 192.168.1.22>;tag=1524540678
To: <sip:22222222 at 192.168.1.22>
Call-ID: 667168938-1555-4 at BJC.BGI.B.F
CSeq: 31 INVITE
Contact: <sip:telefono at 192.168.1.5:1555>
Authorization: Digest username="telefono", realm="asterisk",
nonce="1032f9e6", uri="sip:22222222 at 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 at 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 - 667168938-1555-4 at 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 at 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 at 192.168.1.22>;tag=1524540678
To: <sip:22222222 at 192.168.1.22>
Call-ID: 667168938-1555-4 at 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 at 192.168.1.22:5060>
Content-Length: 0
<------------>
-- Executing [22222222 at internal-call:1]
Dial("SIP/telefono-00000004", "SIP/22222222 at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c
To: <sip:22222222 at 192.168.1.4>
Contact: <sip:telefono at 192.168.1.22:5060>
Call-ID: 323866b71557eac419f667ee37ee16ae at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c
To: <sip:22222222 at 192.168.1.4>;tag=1c1296932060
Call-ID: 323866b71557eac419f667ee37ee16ae at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c
To: <sip:22222222 at 192.168.1.4>;tag=1c1296932060
Contact: <sip:telefono at 192.168.1.22:5060>
Call-ID: 323866b71557eac419f667ee37ee16ae at 192.168.1.22:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0
---
Scheduling destruction of SIP dialog
'323866b71557eac419f667ee37ee16ae at 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 at 192.168.1.22>;tag=1524540678
To: <sip:22222222 at 192.168.1.22>;tag=as4caf91d6
Call-ID: 667168938-1555-4 at 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 at 192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport
From: <sip:telefono at 192.168.1.22>;tag=1524540678
To: <sip:22222222 at 192.168.1.22>;tag=as4caf91d6
Call-ID: 667168938-1555-4 at BJC.BGI.B.F
CSeq: 31 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '667168938-1555-4 at BJC.BGI.B.F' Method:
ACK
debian-asterisk*CLI> sip set debug off
SIP Debugging Disabled
debian-asterisk*CLI>
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Gustavo Ch. Apaza
2014-Apr-09 17:57 UTC
[asterisk-users] I can't make outbound calls (status is 'CHANUNAVAIL')
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 <luedcortes at 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 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22> > Call-ID: 667168938-1555-4 at BJC.BGI.B.F > CSeq: 30 INVITE > Contact: <sip:telefono at 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 at 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 - 667168938-1555-4 at 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 at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as50d1512e > Call-ID: 667168938-1555-4 at 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 '667168938-1555-4 at BJC.BGI.B.F' in > 6400 ms (Method: INVITE) > > <--- SIP read from UDP:192.168.1.5:1555 ---> > ACK sip:22222222 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as50d1512e > Call-ID: 667168938-1555-4 at 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 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22> > Call-ID: 667168938-1555-4 at BJC.BGI.B.F > CSeq: 31 INVITE > Contact: <sip:telefono at 192.168.1.5:1555> > Authorization: Digest username="telefono", realm="asterisk", > nonce="1032f9e6", uri="sip:22222222 at 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 at 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 - 667168938-1555-4 at 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 at 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 at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22> > Call-ID: 667168938-1555-4 at 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 at 192.168.1.22:5060> > Content-Length: 0 > > > <------------> > -- Executing [22222222 at internal-call:1] > Dial("SIP/telefono-00000004", "SIP/22222222 at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c > To: <sip:22222222 at 192.168.1.4> > Contact: <sip:telefono at 192.168.1.22:5060> > Call-ID: 323866b71557eac419f667ee37ee16ae at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c > To: <sip:22222222 at 192.168.1.4>;tag=1c1296932060 > Call-ID: 323866b71557eac419f667ee37ee16ae at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c > To: <sip:22222222 at 192.168.1.4>;tag=1c1296932060 > Contact: <sip:telefono at 192.168.1.22:5060> > Call-ID: 323866b71557eac419f667ee37ee16ae at 192.168.1.22:5060 > CSeq: 102 ACK > User-Agent: Asterisk PBX 11.8.1 > Content-Length: 0 > > > --- > Scheduling destruction of SIP dialog > '323866b71557eac419f667ee37ee16ae at 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 at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as4caf91d6 > Call-ID: 667168938-1555-4 at 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 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as4caf91d6 > Call-ID: 667168938-1555-4 at BJC.BGI.B.F > CSeq: 31 ACK > Content-Length: 0 > > <-------------> > --- (7 headers 0 lines) --- > Really destroying SIP dialog '667168938-1555-4 at 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 --> luedcortes at gmail.com > msn --> luedcortes at gmail.com > > -- > _____________________________________________________________________ > -- 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 >-- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140409/bf06df74/attachment.html>
Gustavo Chalco
2014-Apr-09 17:58 UTC
[asterisk-users] I can't make outbound calls (status is 'CHANUNAVAIL')
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 <luedcortes at 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 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22> > Call-ID: 667168938-1555-4 at BJC.BGI.B.F > CSeq: 30 INVITE > Contact: <sip:telefono at 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 at 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 - 667168938-1555-4 at 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 at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as50d1512e > Call-ID: 667168938-1555-4 at 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 '667168938-1555-4 at BJC.BGI.B.F' in > 6400 ms (Method: INVITE) > > <--- SIP read from UDP:192.168.1.5:1555 ---> > ACK sip:22222222 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as50d1512e > Call-ID: 667168938-1555-4 at 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 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22> > Call-ID: 667168938-1555-4 at BJC.BGI.B.F > CSeq: 31 INVITE > Contact: <sip:telefono at 192.168.1.5:1555> > Authorization: Digest username="telefono", realm="asterisk", > nonce="1032f9e6", uri="sip:22222222 at 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 at 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 - 667168938-1555-4 at 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 at 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 at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22> > Call-ID: 667168938-1555-4 at 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 at 192.168.1.22:5060> > Content-Length: 0 > > > <------------> > -- Executing [22222222 at internal-call:1] > Dial("SIP/telefono-00000004", "SIP/22222222 at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c > To: <sip:22222222 at 192.168.1.4> > Contact: <sip:telefono at 192.168.1.22:5060> > Call-ID: 323866b71557eac419f667ee37ee16ae at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c > To: <sip:22222222 at 192.168.1.4>;tag=1c1296932060 > Call-ID: 323866b71557eac419f667ee37ee16ae at 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 at 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 at 192.168.1.22>;tag=as7cd8ea4c > To: <sip:22222222 at 192.168.1.4>;tag=1c1296932060 > Contact: <sip:telefono at 192.168.1.22:5060> > Call-ID: 323866b71557eac419f667ee37ee16ae at 192.168.1.22:5060 > CSeq: 102 ACK > User-Agent: Asterisk PBX 11.8.1 > Content-Length: 0 > > > --- > Scheduling destruction of SIP dialog > '323866b71557eac419f667ee37ee16ae at 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 at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as4caf91d6 > Call-ID: 667168938-1555-4 at 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 at 192.168.1.22 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport > From: <sip:telefono at 192.168.1.22>;tag=1524540678 > To: <sip:22222222 at 192.168.1.22>;tag=as4caf91d6 > Call-ID: 667168938-1555-4 at BJC.BGI.B.F > CSeq: 31 ACK > Content-Length: 0 > > <-------------> > --- (7 headers 0 lines) --- > Really destroying SIP dialog '667168938-1555-4 at 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 --> luedcortes at gmail.com > msn --> luedcortes at gmail.com > > -- > _____________________________________________________________________ > -- 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 >-- Saludos Gustavo Ch. Apaza Core Consulting Group CEL. 993783686 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140409/a0ff1b3f/attachment.html>