[asterisk-users] Use Polycom FX with Asterisk

2011-11-16 Thread Malvin Rito 

Hi List,
I have a Polycom FX video unit and I'm thinking maybe I can integrate it 
on our Asterisk Server to be able to do teleconference and video as well 
via Polycom FX.


I already have oh323 configured on my Asterisk box and I just no idea on 
how to let them work.Any help please?


Regards,
Malvin
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Re: [asterisk-users] Call does not pass through

2011-09-28 Thread Malvin Rito 
 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@192.168.254.15;tag=as4f38e456
Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15
To: sip:15707088788@66.148.120.167;tag=2809191014129494936357
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@192.168.254.15;tag=as4f38e456
Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15
To: sip:15707088788@66.148.120.167;tag=2809191014129494936357
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@192.168.254.15;tag=as4f38e456
To: sip:15707088788@66.148.120.167;tag=2809191014129494936357
Contact: sip:1105@192.168.254.15
Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@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 
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph 
wrote:


Thanks All. Here is my config:

*On my Firewall NAT:*

/I allowed the following ports: 4569,5004-5082, 1-2/
*
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 = _,1,Dial(SIP/${EXTEN}@Avaya)
exten = _11XX,1,Dial(H323/${EXTEN}@Avaya)

exten = _X,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _X,2,Dial(SIP/${EXTEN}@cordia)

exten = _,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _,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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Re: [asterisk-users] Call does not pass through

2011-09-27 Thread Malvin Rito 

Thanks All. Here is my config:

*On my Firewall NAT:*

/I allowed the following ports: 4569,5004-5082, 1-2/
*
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 = _,1,Dial(SIP/${EXTEN}@Avaya)
exten = _11XX,1,Dial(H323/${EXTEN}@Avaya)

exten = _X,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _X,2,Dial(SIP/${EXTEN}@cordia)

exten = _,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _,2,Dial(SIP/${EXTEN}@cordia)/



Regards,
Malvin

On 9/26/2011 9:56 PM, Ruben Rögels wrote:

Am 26.09.2011 13:18, 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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[asterisk-users] Call does not pass through

2011-09-26 Thread 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
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Re: [asterisk-users] Add PinCode on my dialplan

2011-09-22 Thread Malvin Rito 

Hi,

I tried Authenticate where pass codes are stored on the file pass.conf 
and it works.


exten = _,1,Authenticate(/etc/asterisk/pass.conf)

Since I have my CDR, I want to have a report wherein I can check which 
pass code did the call. How can I achieve it?
Using authenticate through file does not replace ACCOUNT_CODE field with 
the pass code entered, it only show *ast_h323 *under the Account_Code field.


Regards,
Malvin

On 9/21/2011 1:01 PM, Sam Govind wrote:

See core show application autheTAB
If passwords are already the same as those of voicemail.conf go for 
application VMAuthenticate() - DIA generates a dial-tone which I don't 
think is suitable for dialling out from users(insiders)


  -= Info about application 'Authenticate' =-

[Synopsis]
Authenticate a user

[Description]
This application asks the caller to enter a given password in order to 
continue

dialplan execution.
If the password begins with the '/' character,  it is interpreted as a 
file
which contains a list of valid passwords, listed 1 password per line 
in the

file.
When using a database key, the value associated with the key can be 
anything.

Users have three attempts to authenticate before the channel is hung
up.

[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])

[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line 
in the

file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.

[See Also]
VMAuthenticate(), DISA()


On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito 
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph 
wrote:


Thanks. ?If I want to use unique PIN for every user that dials out
how can I implement it using Authenticate app?

Regards,
Malvin


On 9/21/2011 12:42 PM, Sam Govind wrote:

DISA and DB based Auth could be an overkill.

Kyle showed the very simplistic dial plan if Dial-out pin is
common for the whole system.
See application
*Authenticate(password[,options[,maxdigits[,prompt]]] *and if
Voicemail PIN are required to be used use application
*MAuthenticate([mailbox][@context][,options] *

Regards,

- Sammy

On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org
mailto:k...@mocker.org wrote:

Something like this should work:

exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)

exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)

exten = _011.,n,Hangup
exten = _011.,n,Playback(invalid)
exten = _011.,n,Hangup

Could be cleaned up (the GotoIf isn't very descriptive about
where it's going), but it's a starting point.


On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:


Hi List,
I currently have a asterisk server running used for
dialing-out for IDD but I want to Put a pincode wherein only
users with the right pin code will be allowed to dial IDD.
Any sample dialplan you can suggest pls?

Thanks,
Malvin
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[asterisk-users] Add PinCode on my dialplan

2011-09-20 Thread Malvin Rito 
Hi List,
I currently have a asterisk server running used for dialing-out for IDD but I 
want to Put a pincode wherein only users with the right pin code will be 
allowed to dial IDD. Any sample dialplan you can suggest pls?

Thanks,
Malvin--
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Re: [asterisk-users] Add PinCode on my dialplan

2011-09-20 Thread Malvin Rito 
Thanks. ?If I want to use unique PIN for every user that dials out how 
can I implement it using Authenticate app?


Regards,
Malvin

On 9/21/2011 12:42 PM, Sam Govind wrote:

DISA and DB based Auth could be an overkill.

Kyle showed the very simplistic dial plan if Dial-out pin is common 
for the whole system.
See application *Authenticate(password[,options[,maxdigits[,prompt]]] 
*and if Voicemail PIN are required to be used use application 
*MAuthenticate([mailbox][@context][,options] *


Regards,

- Sammy

On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org 
mailto:k...@mocker.org wrote:


Something like this should work:

exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)

exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)

exten = _011.,n,Hangup
exten = _011.,n,Playback(invalid)
exten = _011.,n,Hangup

Could be cleaned up (the GotoIf isn't very descriptive about where
it's going), but it's a starting point.


On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:


Hi List,
I currently have a asterisk server running used for dialing-out
for IDD but I want to Put a pincode wherein only users with the
right pin code will be allowed to dial IDD. Any sample dialplan
you can suggest pls?

Thanks,
Malvin
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[asterisk-users] Avaya Asterisk FreePBX Integration Problem

2011-07-28 Thread Malvin Rito 

Hi,

I'm currently testing my FreePbx Box to work with our Avaya PBX to allow 
dialing outgoing international call and FreePBX extensions to avaya PBX 
Extensions calling.
Unfortunately no luck to do it successfully. Any help would be much be 
appreciated, here is the sample codes I already tried:


On FreePBX GUI:
1. I created a custom Trunk called AvayaPBXTrunk with custom dial string 
OOH323/$OUTNUM$/Avaya
2. Created an Outbound route called InternationalCall and select 
AvayaPBXTrunk on the trunk sequence.

3. Created an Extension 1000 with dial extension OOH323/$OUTNUM$@Avaya

On Asterisk CLI:
1. Edit ooh323.conf with the following codes:
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=127.0.0.1
port=1720
callerID=Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
callerid=asterisk
context=default
disallow=all
allow=ulaw

[Avaya]
type=friend
context=from-internal
host=X.X.X.X 'IP Address of our Avaya PBX
port=1720
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=inband
rtptimeout=60
e164=50

2. Edit sip_custom.conf with the following code:

[general]
context=from-internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=127.0.0.1
srvlookup=yes
canreinvite=no

Below also the log result during the call:
-- Executing [s@default:1] Playback(OOH323/(null)-b78bd818, 
vm-goodbye) in new stack

--Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b78bd818, hangupcall) 
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b78bd818, 
1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b78bd818, 
1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b78bd818, 
1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b78bd818, ) 
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'OOH323/(null)-b78bd818' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on 
'OOH323/(null)-b78bd818'
-- Executing [h@default:1] Macro(OOH323/(null)-b78bd818, 
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b78bd818, 
1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b78bd818, 
1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b78bd818, 
1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b78bd818, ) 
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'OOH323/(null)-b78bd818' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on 
'OOH323/(null)-b78bd818'


Regards,
Malvin

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Re: [asterisk-users] My Asterisk Box was hacked

2011-07-21 Thread Malvin Rito 

Thanks. Any link for me to check for the procedure to implement those?

Regards,
Malvin

On 7/21/2011 1:59 PM, Захаров Антон wrote:

Hello!

First of all, you should disable unused VoIP protocols. Than remove 
all guest accounts from used protocols, disable guest unauth access.
Always use strong passwords for accounts, for users on your system. 
Passwords shouldn't be eq username. Move port binds on LAN network for 
all active services as much as you can (i.e. SHH should be on WAN too 
I think).
Use iptables for blocking password bruteforce. Try to install fail2ban 
with jails for asterisk, ssh, HTTP and other public services. Then you 
can try to install PSAD (port scan autodetect) to prevent attacks.

And never use default context in asterisk for word calls directions.
And you should always keep your software up to date. There much more 
security issues than you think.


Good Luck!

On 21.07.2011 09:29, Malvin Rito wrote:

Hi List,

My asterisk box was hacked! Can anyone help on how do I secure my 
asterisk box, currently my box is installed with 2 NIC. 1st NIC is 
for LAN access and 2nd NIC has a public IP which is registered to our 
VoIP Provider.


As I remember I already tried putting our Box on NAT but 
unfortunately due to some issue like call is dropped after 30 seconds 
and sometimes voice are not heard. Then we disable again the NAT.


Your advise will be much appreciated. Thanks in advance.

Regards,
Malvin

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[asterisk-users] My Asterisk Box was hacked

2011-07-20 Thread Malvin Rito 

Hi List,

My asterisk box was hacked! Can anyone help on how do I secure my 
asterisk box, currently my box is installed with 2 NIC. 1st NIC is for 
LAN access and 2nd NIC has a public IP which is registered to our VoIP 
Provider.


As I remember I already tried putting our Box on NAT but unfortunately 
due to some issue like call is dropped after 30 seconds and sometimes 
voice are not heard. Then we disable again the NAT.


Your advise will be much appreciated. Thanks in advance.

Regards,
Malvin

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Re: [asterisk-users] Problem on Dialling-out

2011-07-13 Thread Malvin Rito 

Bruce,

Thanks. I already figured out the problem. It seems that a firewall issue.

Regards,
Malvin

On 7/13/2011 12:30 PM, Bruce B wrote:

Your trunk shows busy:

*/  -- Called CordiaVoIP/639285010430
   -- SIP/CordiaVoIP-0015 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)/*

Try this in the CLI (asterisk -r):
*core set verbose 0*
*sip set debug peer CordiaVoIP*

And then make a call and read why the SIP trunk is failing.

-Bruce


On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito 
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph 
wrote:


Hi List,

I have a Asterisk + FreePbx Server setup with around 10 SIP
extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any
number call is being dropped with the following message on
asterisk log:

 == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
   -- Called CordiaVoIP/639285010430
   -- SIP/CordiaVoIP-0015 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
   -- Executing [s@macro-dialout-trunk:20]
NoOp(SIP/1001-0014, Dial failed for some reason with
DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack
   -- Executing [s@macro-dialout-trunk:21]
Goto(SIP/1001-0014, s-CONGESTION,1) in new stack
   -- Goto (macro-dialout-trunk,s-CONGESTION,1)
   -- Executing [s-CONGESTION@macro-dialout-trunk:1]
Set(SIP/1001-0014, RC=0) in new stack
   -- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto(SIP/1001-0014, 0,1) in new stack
   -- Goto (macro-dialout-trunk,0,1)
   -- Executing [0@macro-dialout-trunk:1]
Goto(SIP/1001-0014, continue,1) in new stack
   -- Goto (macro-dialout-trunk,continue,1)
   -- Executing [continue@macro-dialout-trunk:1]
GotoIf(SIP/1001-0014, 1?noreport) in new stack
   -- Goto (macro-dialout-trunk,continue,3)
   -- Executing [continue@macro-dialout-trunk:3]
NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION
HANGUPCAUSE: 0 - failing through to other trunks) in new stack
   -- Executing [continue@macro-dialout-trunk:4]
Set(SIP/1001-0014, CALLERID(number)=1001) in new stack
   -- Executing [639285010430@from-internal:8]
Macro(SIP/1001-0014, outisbusy,) in new stack
   -- Executing [s@macro-outisbusy:1]
Progress(SIP/1001-0014, ) in new stack
   -- Executing [s@macro-outisbusy:2]
Playback(SIP/1001-0014, all-circuits-busy-now,noanswer) in
new stack
   -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm'
(language 'en')
   -- Executing [s@macro-outisbusy:3]
Playback(SIP/1001-0014, pls-try-call-later,noanswer) in
new stack
   -- SIP/1001-0014 Playing 'pls-try-call-later.gsm'
(language 'en')
   -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014,
hangupcall) in new stack
   -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014,
1?skiprg) in new stack
   -- Goto (macro-hangupcall,s,4)
   -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014,
1?skipblkvm) in new stack
   -- Goto (macro-hangupcall,s,7)
   -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014,
1?theend) in new stack
   -- Goto (macro-hangupcall,s,9)
   -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014,
) in new stack
 == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/1001-0014' in macro 'hangupcall'
 == Spawn extension (macro-outisbusy, s, 4) exited non-zero on
'SIP/1001-0014' in macro 'outisbusy'
 == Spawn extension (from-internal, 639285010430, 8) exited
non-zero on 'SIP/1001-0014'
   -- Executing [h@from-internal:1] Macro(SIP/1001-0014,
hangupcall) in new stack
   -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014,
1?skiprg) in new stack
   -- Goto (macro-hangupcall,s,4)
   -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014,
1?skipblkvm) in new stack
   -- Goto (macro-hangupcall,s,7)
   -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014,
1?theend) in new stack
   -- Goto (macro-hangupcall,s,9)
   -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014,
) in new stack
 == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/1001-0014' in macro 'hangupcall'
 == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/1001-0014'
localhost*CLI


Can someone assist me please. Thanks in advance.

Regards,
Malvin



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[asterisk-users] Connect Avaya to Asterisk PBX

2011-07-13 Thread Malvin Rito 

Hi List,

I have another issue on allowing outgoing calls to PSTN on Asterisk via 
Avaya Phones, I hope that anyone could help me fix this issue:


*When I dial through Avaya phone i just here a good bye message reply 
from asterisk server. And here is the log:*


 == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling 
back to exten 's'
  == Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so 
falling back to context 'default'
-- Executing [s@default:1] Playback(OOH323/(null)-b7db8aa0, 
vm-goodbye) in new stack

-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0, 
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] 
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] 
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
  == Spawn extension (default, s, 2) exited non-zero on 
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0, 
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] 
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] 
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
  == Spawn extension (default, h, 1) exited non-zero on 
'OOH323/(null)-b7db8aa0'


*Here is also the content of my extensions_custom.conf:*
[general]
static=yes
autofallthrough=yes

[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()

exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten  = _XX,1,Dial(H323/${EXTEN}@Avaya)

*Here is also the content of my ooh323.conf:*
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal

[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband

*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no

[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes


Thanks  regards,
Malvin
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Re: [asterisk-users] Connect Avaya to Asterisk PBX

2011-07-13 Thread Malvin Rito 

How do I write it on my code?

On 7/13/2011 4:04 PM, Warren Selby wrote:

Looks like you need an 's' exten in your [internal] context.

Thanks,
--Warren Selby, dCAP

On Jul 13, 2011, at 2:02 AM, Malvin Rito 
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph 
wrote:



Hi List,

I have another issue on allowing outgoing calls to PSTN on Asterisk 
via Avaya Phones, I hope that anyone could help me fix this issue:


*When I dial through Avaya phone i just here a good bye message 
reply from asterisk server. And here is the log:*


 == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling 
back to exten 's'
  == Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so 
falling back to context 'default'
-- Executing [s@default:1] Playback(OOH323/(null)-b7db8aa0, 
vm-goodbye) in new stack

-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0, 
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] 
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] 
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
  == Spawn extension (default, s, 2) exited non-zero on 
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0, 
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] 
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] 
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] 
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
  == Spawn extension (default, h, 1) exited non-zero on 
'OOH323/(null)-b7db8aa0'


*Here is also the content of my extensions_custom.conf:*
[general]
static=yes
autofallthrough=yes

[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()

exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten  = _XX,1,Dial(H323/${EXTEN}@Avaya)

*Here is also the content of my ooh323.conf:*
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal

[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband

*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no

[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes


Thanks  regards,
Malvin
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Re: [asterisk-users] Connect Avaya to Asterisk PBX

2011-07-13 Thread Malvin Rito 
Thanks. I want to dial-out to PSTN using Asterisk Server via Avaya Phone 
using Cordia VoIP Service provider. How can I achieve it using the same 
code below?


Regards,
Malvin

On 7/13/2011 4:59 PM, DHAVAL INDRODIYA wrote:

you can edit dial-plan by adding following lines to your code

[internal]

exten = s,1,Dial(SIP/1000)
exten = s,2,HangUp()


exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()

exten = _,1,Dial(H323/${EXTEN}@
Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten  = _XX,1,Dial(H323/${EXTEN}@Avaya)


On Wed, Jul 13, 2011 at 1:35 PM, Malvin Rito 
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph 
wrote:


How do I write it on my code?


On 7/13/2011 4:04 PM, Warren Selby wrote:

Looks like you need an 's' exten in your [internal] context.

Thanks,
--Warren Selby, dCAP

On Jul 13, 2011, at 2:02 AM, Malvin Rito
mr...@mail.altcladding.com.ph
mailto:mr...@mail.altcladding.com.ph wrote:


Hi List,

I have another issue on allowing outgoing calls to PSTN on
Asterisk via Avaya Phones, I hope that anyone could help me fix
this issue:

*When I dial through Avaya phone i just here a good bye
message reply from asterisk server. And here is the log:*

 == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so
falling back to exten 's'
  == Starting OOH323/(null)-b7db8aa0 at internal,s,1 still
failed so falling back to context 'default'
-- Executing [s@default:1]
Playback(OOH323/(null)-b7db8aa0, vm-goodbye) in new stack
-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw'
(language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0,
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
  == Spawn extension (default, s, 2) exited non-zero on
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0,
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
  == Spawn extension (default, h, 1) exited non-zero on
'OOH323/(null)-b7db8aa0'

*Here is also the content of my extensions_custom.conf:*
[general]
static=yes
autofallthrough=yes

[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()

exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten  = _XX,1,Dial(H323/${EXTEN}@Avaya)

*Here is also the content of my ooh323.conf:*
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal

[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband

*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no

[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes


Thanks  regards,
Malvin
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[asterisk-users] Problem on Dialling-out

2011-07-12 Thread Malvin Rito 

Hi List,

I have a Asterisk + FreePbx Server setup with around 10 SIP extensions 
and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is 
being dropped with the following message on asterisk log:


 == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called CordiaVoIP/639285010430
-- SIP/CordiaVoIP-0015 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp(SIP/1001-0014, 
Dial failed for some reason with DIALSTATUS = CONGESTION and 
HANGUPCAUSE = 0) in new stack
-- Executing [s@macro-dialout-trunk:21] Goto(SIP/1001-0014, 
s-CONGESTION,1) in new stack

-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] 
Set(SIP/1001-0014, RC=0) in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] 
Goto(SIP/1001-0014, 0,1) in new stack

-- Goto (macro-dialout-trunk,0,1)
-- Executing [0@macro-dialout-trunk:1] Goto(SIP/1001-0014, 
continue,1) in new stack

-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] 
GotoIf(SIP/1001-0014, 1?noreport) in new stack

-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] 
NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION 
HANGUPCAUSE: 0 - failing through to other trunks) in new stack
-- Executing [continue@macro-dialout-trunk:4] 
Set(SIP/1001-0014, CALLERID(number)=1001) in new stack
-- Executing [639285010430@from-internal:8] 
Macro(SIP/1001-0014, outisbusy,) in new stack
-- Executing [s@macro-outisbusy:1] Progress(SIP/1001-0014, 
) in new stack
-- Executing [s@macro-outisbusy:2] Playback(SIP/1001-0014, 
all-circuits-busy-now,noanswer) in new stack
-- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm' 
(language 'en')
-- Executing [s@macro-outisbusy:3] Playback(SIP/1001-0014, 
pls-try-call-later,noanswer) in new stack

-- SIP/1001-0014 Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014, 
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 
1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 
1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 
1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) 
in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'SIP/1001-0014' in macro 'hangupcall'
  == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 
'SIP/1001-0014' in macro 'outisbusy'
  == Spawn extension (from-internal, 639285010430, 8) exited non-zero 
on 'SIP/1001-0014'
-- Executing [h@from-internal:1] Macro(SIP/1001-0014, 
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 
1?skiprg) in new stack

-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 
1?skipblkvm) in new stack

-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 
1?theend) in new stack

-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) 
in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
'SIP/1001-0014' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 
'SIP/1001-0014'

localhost*CLI


Can someone assist me please. Thanks in advance.

Regards,
Malvin



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Re: [asterisk-users] Problem on Dialling-out

2011-07-12 Thread Malvin Rito 
:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #6 (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: Cordia sip:Unknown@172.16.9.15;tag=as2267fdcc
To: sip:639285010...@lasip1.cordiaip.net
Contact: sip:Unknown@172.16.9.15
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Scheduling destruction of SIP dialog 
'12d2279238e5851572c30cad11bb9492@172.16.9.15' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog 
'12d2279238e5851572c30cad11bb9492@172.16.9.15' in 32000 ms (Method: INVITE)
Really destroying SIP dialog 
'12d2279238e5851572c30cad11bb9492@172.16.9.15' Method: INVITE

localhost*CLI



On 7/13/2011 12:30 PM, Bruce B wrote:

Your trunk shows busy:

*/  -- Called CordiaVoIP/639285010430
   -- SIP/CordiaVoIP-0015 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)/*

Try this in the CLI (asterisk -r):
*core set verbose 0*
*sip set debug peer CordiaVoIP*

And then make a call and read why the SIP trunk is failing.

-Bruce


On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito 
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph 
wrote:


Hi List,

I have a Asterisk + FreePbx Server setup with around 10 SIP
extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any
number call is being dropped with the following message on
asterisk log:

 == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
   -- Called CordiaVoIP/639285010430
   -- SIP/CordiaVoIP-0015 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
   -- Executing [s@macro-dialout-trunk:20]
NoOp(SIP/1001-0014, Dial failed for some reason with
DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack
   -- Executing [s@macro-dialout-trunk:21]
Goto(SIP/1001-0014, s-CONGESTION,1) in new stack
   -- Goto (macro-dialout-trunk,s-CONGESTION,1)
   -- Executing [s-CONGESTION@macro-dialout-trunk:1]
Set(SIP/1001-0014, RC=0) in new stack
   -- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto(SIP/1001-0014, 0,1) in new stack
   -- Goto (macro-dialout-trunk,0,1)
   -- Executing [0@macro-dialout-trunk:1]
Goto(SIP/1001-0014, continue,1) in new stack
   -- Goto (macro-dialout-trunk,continue,1)
   -- Executing [continue@macro-dialout-trunk:1]
GotoIf(SIP/1001-0014, 1?noreport) in new stack
   -- Goto (macro-dialout-trunk,continue,3)
   -- Executing [continue@macro-dialout-trunk:3]
NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION
HANGUPCAUSE: 0 - failing through to other trunks) in new stack
   -- Executing [continue@macro-dialout-trunk:4]
Set(SIP/1001-0014, CALLERID(number)=1001) in new stack
   -- Executing [639285010430@from-internal:8]
Macro(SIP/1001-0014, outisbusy,) in new stack
   -- Executing [s@macro-outisbusy:1]
Progress(SIP/1001-0014, ) in new stack
   -- Executing [s@macro-outisbusy:2]
Playback(SIP/1001-0014, all-circuits-busy-now,noanswer) in
new stack
   -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm'
(language 'en')
   -- Executing [s@macro-outisbusy:3]
Playback(SIP/1001-0014, pls-try-call-later,noanswer) in
new stack
   -- SIP/1001-0014 Playing 'pls-try-call-later.gsm'
(language 'en')
   -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014,
hangupcall) in new stack
   -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014,
1?skiprg) in new stack
   -- Goto (macro-hangupcall,s,4)
   -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014,
1?skipblkvm) in new stack
   -- Goto (macro-hangupcall,s,7)
   -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014,
1?theend) in new stack
   -- Goto (macro-hangupcall,s,9)
   -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014,
) in new stack
 == Spawn extension (macro-hangupcall

[asterisk-users] Trunk grouping

2011-02-17 Thread Malvin Rito 
Hi List,

 

Were upgrading our network switches and need to create multiple VLAN groups,
but since our Squid Proxy (Transparent Proxy) Server should be accessible to
all VLAN groups we need to setup a trunk grouping inside our Squid Proxy
Box. Is anyone has a documentation or code on how to implement trunk
grouping?

 

Your thoughts will be highly appreciated.

 

Regards,

Malvin

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[asterisk-users] Receive Call from unknown user

2010-10-12 Thread Malvin Rito 
Hello List,

I have noticed for the past few weeks that someone from an unknown IP is
trying to make a call to my Asterisk box, below is the sample content of the
log file. Sometimes the calls are being made every seconds.

Is my system being hack by someone?

Oct 12 09:41:47] VERBOSE[3114] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing
[99541442073479...@from-sip-external:1] NoOp(SIP/113.105.153.251-0265,
Received incoming SIP connection from unknown peer to 9954144207347)
in new stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing
[99541442073479...@from-sip-external:2] Set(SIP/113.105.153.251-0265,
DID=9954144207347) in new stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing
[99541442073479...@from-sip-external:3] Goto(SIP/113.105.153.251-0265,
s,1) in new stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Goto (from-sip-external,s,1)
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing [...@from-sip-external:1]
GotoIf(SIP/113.105.153.251-0265, 0?checklang:noanonymous) in new
stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Goto (from-sip-external,s,5)
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing [...@from-sip-external:5]
Set(SIP/113.105.153.251-0265, TIMEOUT(absolute)=15) in new stack
[Oct 12 09:41:47] VERBOSE[21627] func_timeout.c: Channel will hangup at
2010-10-12 09:42:02.042 PHT.
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing [...@from-sip-external:6]
Answer(SIP/113.105.153.251-0265, ) in new stack

Please Advise.

Malvin


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[asterisk-users] Unable to open pseudo device

2010-07-13 Thread Malvin Rito 
Hi List,

I'm new to asterisk and currently running the newest of version. I'm
encountering the error below when I dial my meetme conference #:
WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo device

I already tried googling this issue and found some procedure but still no
luck on fixing it. My server does not have any digium hardware and I'm
trying this via ztdummy.

Please advise,

Malvin


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Re: [asterisk-users] Unable to open pseudo device

2010-07-13 Thread Malvin Rito 
Thanks for the reply. There is no folder dahdi under /dev folder. I cannot
also find /udev.d on /etc folder.

Under /dev folder I only see /dev/zap/pseudo.

Regards,
Malvin

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Tuesday, July 13, 2010 9:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to open pseudo device

On Tuesday 13 July 2010 06:35:45 Malvin Rito wrote:
 Hi List,

 I'm new to asterisk and currently running the newest of version. I'm
 encountering the error below when I dial my meetme conference #:
 WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo
 device

 I already tried googling this issue and found some procedure but still no
 luck on fixing it. My server does not have any digium hardware and I'm
 trying this via ztdummy.

It's likely an issue of permissions.  Check the permissions of
/dev/dahdi/pseudo versus the user your Asterisk daemon runs at.  If
necessary,
change your permission script in /etc/udev.d/ to match the ownership of the
pseudo device to the user running Asterisk.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] Use asterisk as a backend PBX

2010-07-12 Thread Malvin Rito 
Thanks Justin. I'm using a hardware based PBX, can you tell me how I can
deal this or a procedure on setting it up.

 

From: Justin Case [mailto:nogoodnameswereavaila...@gmail.com] 
Sent: Monday, July 12, 2010 2:16 PM
To: mr...@mail.altcladding.com.ph; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] Use asterisk as a backend PBX

 

Sure. If you write the dial plan correctly and your legacy PBX supports it.

On Mon, Jul 12, 2010 at 7:31 AM, Malvin Rito mr...@mail.altcladding.com.ph
wrote:

Hi List,

 

We're planning to use Asterisk as our backend PBX for our legacy PBX
where-in received calls from legacy PBX can be transferred to Asterisk PBX
extension, is this possible? 

 

Regards,
Malvin

 

 


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[asterisk-users] Use asterisk as a backend PBX

2010-07-11 Thread Malvin Rito 
Hi List,

 

We're planning to use Asterisk as our backend PBX for our legacy PBX
where-in received calls from legacy PBX can be transferred to Asterisk PBX
extension, is this possible? 

 

Regards,
Malvin

 

 

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