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[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@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: 667168938-1555-4@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 - 667168938-1555-4@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: 667168938-1555-4@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@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: 667168938-1555-4@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: 667168938-1555-4@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 - 667168938-1555-4@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: 667168938-1555-4@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: 667168938-1555-4@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: 667168938-1555-4@BJC.BGI.B.F
CSeq: 31 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '667168938-1555-4@BJC.BGI.B.F' Method: ACK
debian-asterisk*CLI> sip set debug off
SIP Debugging Disabled
debian-asterisk*CLI>





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[gustavo at ccg.com.pe]

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@gmail.com (luedcortes@gmail.com)>:

[gcaclaroempresas at gm...]

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@gmail.com (luedcortes@gmail.com)>: