[asterisk-users] Call does not pass through
Malvin Rito
mrito at mail.altcladding.com.ph
Wed Sep 28 00:59:00 CDT 2011
Thanks Sam. Please see below CLI log:
/[root at localhost ~]# asterisk -rvvvv
Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
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 at avaya-internal:1] Answer("OOH323/(null)-b7798910",
"") in ne
w stack
-- Executing [s at 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 at 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 at avaya-internal:2]
Dial("OOH323/(null)-b7798910", "
SIP/15707088788 at cordia") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 15707088788 at cordia
-- SIP/cordia-00000017 answered OOH323/(null)-b7798910
== Spawn extension (avaya-internal, 15707088788, 2) exited non-zero
on 'OOH323
/(null)-b7798910'
-- Executing [s at avaya-internal:1] Answer("OOH323/(null)-0a389388",
"") in ne
w stack
-- Executing [s at 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 at avaya-internal:3]
WaitExten("OOH323/(null)-0a389388", "") in
new stack
== CDR updated on OOH323/(null)-0a389388
-- Executing [18772281023 at 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 at avaya-internal:2]
Dial("OOH323/(null)-0a389388", "
SIP/18772281023 at cordia") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 18772281023 at 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 at localhost ~]# nano /etc/asterisk/extensions_custom.conf
[root at localhost ~]# asterisk -rvvvv
Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
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:15707088788 at 66.148.120.167 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:1105 at 192.168.254.15>;tag=as4f38e456
To: <sip:15707088788 at 66.148.120.167>
Contact: <sip:1105 at 192.168.254.15>
Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
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:1105 at 192.168.254.15>;tag=as4f38e456
Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
To: <sip:15707088788 at 66.148.120.167>;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:1105 at 192.168.254.15>;tag=as4f38e456
Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
To: <sip:15707088788 at 66.148.120.167>;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:1105 at 192.168.254.15>;tag=as4f38e456
To: <sip:15707088788 at 66.148.120.167>;tag=28091911111014129494936357
Contact: <sip:1105 at 192.168.254.15>
Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
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
> <mrito at mail.altcladding.com.ph <mailto:mrito at mail.altcladding.com.ph>>
> 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
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided byhttp://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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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- 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
>
>
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