Thanks Sam. Please see below CLI log:
/[root@localhost ~]# asterisk -rvvvv
Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for detail
s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
Connected to Asterisk 1.6.2.7 currently running on localhost (pid = 11533)
Verbosity is at least 4
== Spawn extension (avaya-internal, 15707088788, 2) exited non-zero
on 'OOH323
/(null)-b7798910'
-- Executing [s@avaya-internal:1] Answer("OOH323/(null)-b7798910",
"") in ne
w stack
-- Executing [s@avaya-internal:2]
BackGround("OOH323/(null)-b7798910", "pls-
entr-num-uwish2-call") in new stack
-- <OOH323/(null)-b7798910> Playing 'pls-entr-num-uwish2-call.gsm'
(language
'en')
== CDR updated on OOH323/(null)-b7798910
-- Executing [15707088788@avaya-internal:1]
Authenticate("OOH323/(null)-b779
8910", "/etc/asterisk/passcode.txt,a") in new stack
-- <OOH323/(null)-b7798910> Playing 'agent-pass.ulaw' (language 'en')
-- <OOH323/(null)-b7798910> Playing 'auth-thankyou.ulaw' (language
'en')
-- Executing [15707088788@avaya-internal:2]
Dial("OOH323/(null)-b7798910", "
SIP/15707088788@cordia") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 15707088788@cordia
-- SIP/cordia-00000017 answered OOH323/(null)-b7798910
== Spawn extension (avaya-internal, 15707088788, 2) exited non-zero
on 'OOH323
/(null)-b7798910'
-- Executing [s@avaya-internal:1] Answer("OOH323/(null)-0a389388",
"") in ne
w stack
-- Executing [s@avaya-internal:2]
BackGround("OOH323/(null)-0a389388", "pls-
entr-num-uwish2-call") in new stack
-- <OOH323/(null)-0a389388> Playing 'pls-entr-num-uwish2-call.gsm'
(language
'en')
-- Executing [s@avaya-internal:3]
WaitExten("OOH323/(null)-0a389388", "") in
new stack
== CDR updated on OOH323/(null)-0a389388
-- Executing [18772281023@avaya-internal:1]
Authenticate("OOH323/(null)-0a38
9388", "/etc/asterisk/passcode.txt,a") in new stack
-- <OOH323/(null)-0a389388> Playing 'agent-pass.ulaw' (language 'en')
-- <OOH323/(null)-0a389388> Playing 'auth-thankyou.ulaw' (language
'en')
-- Executing [18772281023@avaya-internal:2]
Dial("OOH323/(null)-0a389388", "
SIP/18772281023@cordia") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 18772281023@cordia
-- SIP/cordia-00000018 is making progress passing it to
OOH323/(null)-0a3893
88
-- SIP/cordia-00000018 is ringing
-- SIP/cordia-00000018 is making progress passing it to
OOH323/(null)-0a3893
88
-- SIP/cordia-00000018 answered OOH323/(null)-0a389388
== Spawn extension (avaya-internal, 18772281023, 2) exited non-zero
on 'OOH323
/(null)-0a389388'
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
localhost*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@localhost ~]# nano /etc/asterisk/extensions_custom.conf
[root@localhost ~]# asterisk -rvvvv
Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
Connected to Asterisk 1.6.2.7 currently running on localhost (pid = 11533)
Verbosity is at least 4
-- Remote UNIX connection
localhost*CLI> sip set debug peer cordia
SIP Debugging Enabled for IP: 66.148.120.167:5060
localhost*CLI> core set verbose 0
Verbosity is now OFF
Audio is at 192.168.254.15 port 19144
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 66.148.120.167:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK1e0698f8;rport
Max-Forwards: 70
From: "10.1.129.247" <sip:[email protected]>;tag=as4f38e456
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 28 Sep 2011 02:43:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 2082067001 2082067001 IN IP4 192.168.254.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 192.168.254.15
t=0 0
m=audio 19144 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
localhost*CLI>
<--- SIP read from UDP:66.148.120.167:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport
From: "10.1.129.247" <sip:[email protected]>;tag=as4f38e456
Call-ID: [email protected]
To: <sip:[email protected]>;tag=28091911111014129494936357
Contact: <sip:66.148.120.167:5060;transport=udp>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
localhost*CLI>
<--- SIP read from UDP:66.148.120.167:5060 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport
From: "10.1.129.247" <sip:[email protected]>;tag=as4f38e456
Call-ID: [email protected]
To: <sip:[email protected]>;tag=28091911111014129494936357
Contact: <sip:66.148.120.167:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 225
v=0
o=VoipSwitch 6356 7356 IN IP4 66.148.120.167
s=VoipSIP
i=Audio Session
c=IN IP4 66.148.120.167
t=0 0
m=audio 6356 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (9 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
(telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 66.148.120.167:6356
list_route: hop: <sip:66.148.120.167:5060;transport=udp>
set_destination: Parsing <sip:66.148.120.167:5060;transport=udp> for
address/port to send to
set_destination: set destination to 66.148.120.167, port 5060
Transmitting (no NAT) to 66.148.120.167:5060:
ACK sip:66.148.120.167:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK28aa746f;rport
Max-Forwards: 70
From: "10.1.129.247" <sip:[email protected]>;tag=as4f38e456
To: <sip:[email protected]>;tag=28091911111014129494936357
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7
Content-Length: 0
---
localhost*CLI>/
Regards,
Malvin
On 9/28/2011 1:46 PM, Sam Govind wrote:
I see a couple of conflicting extensions as well as something I assume
copy-paste malfunction. Please paste the CLI logs of the call.
On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito
<[email protected] <mailto:[email protected]>>
wrote:
Thanks All. Here is my config:
*On my Firewall NAT:*
/I allowed the following ports: 4569,5004-5082, 10000-20000/
*
On Asterisk Box:*
Here is the extensions.conf:
/[general]
static=yes
autofallthrough=yes
[avaya-internal]
exten => s,1,Answer()
exten => s,2,background(pls-entr-num-uwish2-call)
exten => s,3,WaitExten()
exten => s,4,Dial(SIP/${EXTEN})
exten => s,5,Dial(H323/${EXTEN})
exten => s,6,PlayBack(vm-nobodyavail)
exten => s,7,HangUp()
exten => 1000,1,Dial(SIP/1000)
exten => 1000,1,Answer()
exten => 1000,2,PlayBack(vm-goodbye)
exten => 1000,3,HangUp()
#Extension for recording
exten => 9000,1,Answer()
exten => 9000,2,Background(pm-to-record-phrase)
exten => 9000,3,Hangup()
#exten => 9000,3,Wait(2)
exten => 9000,4,Record(alt_recording%d:ulaw)
exten => 9000,5,Wait(2)
exten => 9000,6,Playback(${RECORDED_FILE})
exten => 9000,7,Wait(2)
exten => 9000,8,Hangup
exten => _XXXX,1,Dial(SIP/${EXTEN}@Avaya)
exten => _11XX,1,Dial(H323/${EXTEN}@Avaya)
exten => _XXXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten => _XXXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia)
exten => _XXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten => _XXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia)/
Regards,
Malvin
On 9/26/2011 9:56 PM, Ruben Rögels wrote:
Am26.09.2011 13 <tel:26.09.2011%2013>:18, schrieb Malvin Rito:
Hi list,
My call does not pass through on the first dial, I have to redial again
the number for the call to pass through. I'm not sure if the problem is
on my voip proovider or my asterisk server setup. Any thoughts pls?
Regards,
Malvin
Hi,
could be a NAT related issue.
Please be more specific about your setup.
best regards,
Ruben
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