Re: [asterisk-users] Music On Hold

[Cyprus VoIP]

Hi,

I deleted all the default files and put one that I know that works on 
another Asterisk, but since then, I recompiled Asterisk and the default 
files were added.

In order to test moh, I created a context for it:

[default]
exten => 888,1,Goto(moh,s,1)
[moh]
exten => s,1,Answer
exten => s,2,MusicOnHold()

When we dial 888, we hear the music and this appears in the console:
 -- Executing [...@default:1] Goto("SIP/24-08650e80", "moh,s,1") in 
new stack
 -- Goto (moh,s,1)
 -- Executing [...@moh:1] Answer("SIP/24-08650e80", "") in new stack
 -- Executing [...@moh:2] MusicOnHold("SIP/24-08650e80", "") in new stack
 -- Started music on hold, class 'default', on SIP/24-08650e80
 -- Stopped music on hold on SIP/24-08650e80
   == Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80'


But, when I just put a call on hold, nothing is played and nothing 
appears in the console.

I have no idea why this happens and what to do about it. Any suggestions?

Thanks.


 Original Message  
Subject: Re: [asterisk-users] Music On Hold
From: John A. Sullivan III 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Date: Wednesday, 30 September, 2009 15:27:28

> On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
> 
>>  > You see the wav files but do you see the files encoded for the codecs 
>> you are using?
>> There's only one wav file there. No encoded files, but on asterisk 1.2 
>> we have, it's the same file and it works.
> 
> Hmm . . only one wav file.  We had several.  As I recall now, we
> actually installed 1.6.1.1 and upgraded.  1.6.1.1 had the old hold
> music.  1.6.1.6 has the new hold music.  But I believe there are several
> files.  Is that wav file valid, i.e., if you copy it to a system with a
> sound card and play it, does it play? Could it have been corrupted in
> copying or have incorrect permissions? - John

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[asterisk-users] Free version of softswitch with billing and routing released

[Mindaugas Kezys]

Hello,

 

We are happy to announce that FREE version of MOR 8 - our advanced
Softswitch with billing and Routing is released.

 

It comes as ISO image which installs everything from scratch.

 

FREE edition has all functionality just limited to 10 simultaneous calls.

 

We hope it will be useful for starters and makes life easier for many
people.

 

Link to get it:
http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/

 

More info about software: http://www.voip-info.org/wiki/view/MOR

 

 

Regards,

Mindaugas Kezys

http://www.kolmisoft.com

VoIP Billing and Routing Solutions

 

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[asterisk-users] srtp issue

[Szasz Szabolcs]

Hi,

I have set up an asterisk with TLS and SRTP support. The SRTP is working
with Phonerlite softphone. I have problem with the SRTP, when I make calls
on Audiocodes gateway . I got the folloowing messages on asterisk:

[Oct  2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto
life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80
inline:SL+jOTOj8J1jTFgC+ETx5ORfFEWB5kxk5Ysr0XcI|2^31
[Oct  2 10:59:48] NOTICE[24868]: sdp_crypto.c:242 sdp_crypto_process: SRTP
crypto offer not acceptable
[Oct  2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto
life time unsupported: crypto:2 AES_CM_128_HMAC_SHA1_32
inline:TyBSx7QAdczhqkuh+/eK2dWEH3c9sq7qa8r9FycS|2^31
[Oct  2 10:59:48] NOTICE[24868]: sdp_crypto.c:242 sdp_crypto_process: SRTP
crypto offer not acceptable
[Oct  2 10:59:48] WARNING[24868]: chan_sip.c:7939 process_sdp: Can't provide
secure audio requested in SDP offer

What means this?

By debugging sip messages:

<--- SIP read from TLS:UA_IP_ADDRESS:60415 --->
INVITE sips:2...@ast_ip_address;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781732149;alias
Max-Forwards: 70
From: "201" ;tag=1c781729204
To: 
Call-ID: 781728720312000192...@192.168.105.199
CSeq: 1 INVITE
Contact: 
Supported:
em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 528

v=0
o=AudiocodesGW 781713142 781713021 IN IP4 192.168.105.199
s=Phone-Call
c=IN IP4 192.168.105.199
t=0 0
m=audio 6000 RTP/SAVP 0 8 18 4 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:EFG/GFJBnNMdfJ2/hBCyJmgdPS6MNkuOscQEJR3E|2^31
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:IdPMZ5yfypyQPt2q0HPYnVojTSWj1el7cOB6LOEq|2^31


<->
--- (14 headers 19 lines) ---
Using INVITE request as basis request -
781728720312000192...@192.168.105.199
Found peer '201' for '201' from UA_IP_ADDRESS:60415
sbc06*CLI>
<--- Reliably Transmitting (NAT) to UA_IP_ADDRESS:60415 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.105.199:5051
;branch=z9hG4bKac781732149;alias;received=UA_IP_ADDRESS
From: "201" ;tag=1c781729204
To: ;tag=as1bf72d42
Call-ID: 781728720312000192...@192.168.105.199
CSeq: 1 INVITE
Server: Asterisk PBX SVN-group-srtp-r183146-/trunk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="526064ea"
Content-Length: 0


<>
Scheduling destruction of SIP dialog '781728720312000192...@192.168.105.199'
in 32000 ms (Method: INVITE)
sbc06*CLI>
<--- SIP read from TLS:UA_IP_ADDRESS:60415 --->
ACK sips:2...@ast_ip_address;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781732149;alias
Max-Forwards: 70
From: "201" ;tag=1c781729204
To: ;tag=as1bf72d42
Call-ID: 781728720312000192...@192.168.105.199
CSeq: 1 ACK
Contact: 
Supported: em,timer,replaces,path,early-session,resource-priority
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003
Content-Length: 0


<->
--- (12 headers 0 lines) ---
sbc06*CLI>
<--- SIP read from TLS:UA_IP_ADDRESS:60415 --->
INVITE sips:2...@ast_ip_address;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.105.199:5051;branch=z9hG4bKac781931225;alias
Max-Forwards: 70
From: "201" ;tag=1c781729204
To: 
Call-ID: 781728720312000192...@192.168.105.199
CSeq: 2 INVITE
Authorization: Digest
username="201",realm="asterisk",nonce="526064ea",uri="sips:2...@ast_ip_address
",algorithm=MD5,response="64f012c1334a4eb355f256c2569c61f6"
Contact: 
Supported:
em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 528

v=0
o=AudiocodesGW 781713142 781713021 IN IP4 192.168.105.199
s=Phone-Call
c=IN IP4 192.168.105.199
t=0 0
m=audio 6000 RTP/SAVP 0 8 18 4 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:EFG/GFJBnNMdfJ2/hBCyJmgdPS6MNkuOscQEJR3E|2^31
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:IdPMZ5yfypyQPt2q0HPYnVojTSWj1el7cOB6LOEq|2^31


<->
--- (15 headers 19 lines) ---
Sending to UA_IP_ADDRESS : 60415 (NAT)
Using INVITE request as basis request -
781728720312000192...@192.168.105.199
Found peer '201' for '201' from 

Re: [asterisk-users] Bringing people into a conference

[Ioan Indreias]

Hello Harley,

Please find the directions I've used in order to get this work on our
Asterisk machine.

The "flow"
===
A conference user (A) decide to invite somebody else (B) into the
conference. Pressing 0 from his dialpad A will hear a dial tone and he
have to enter the destination number followed by #. In case B agree to
enter into conference A will have to press 9 otherwise A should press
7. In both cases A will return into the conference.

Step 1. Add the following lines in [applicationmap] section from  features.conf

conf-inv   => 9,caller,Macro,conf-inv
conf-noinv => 7,caller,Macro,conf-noinv

After issuing reload/restart, when you issue "feature show" from CLI
you have to see:

Dynamic Feature           Default Current
---           --- ---
conf-noinv                no def  7
conf-inv                  no def  9

Step 2. When add a person into the conference use option "X" for MeetMe
This will allow user to exit the conference by entering a valid single
digit extension. If you want to go to a special context you have to
set MEETME_EXIT_CONTEXT otherwise the same context will be used.
Let's assume that MeetMe call is done on extension s, priority
"cont_3" and we do not change the MEETME_EXIT_CONTEXT
Also, let's assume that the conference number is set into
__Room_number variable (please notice __ in front of the variable).

Thus we have the folowing line in our dialplan:
exten => s,n(cont_3),MeetMe(${Room_number},X)  <- here you could add
other options as well

Step 3. Let's say the user will press 0 -> they will jump to extension
0, step 1 in the MEETME_EXIT_CONTEXT

Use the following lines:
exten => 0,1,Wait(1)
exten => 0,n,Read(Get_dest,dial,,i)
exten => 0,n,Set(DYNAMIC_FEATURES=conf-inv#conf-noinv)
exten => 0,n,Dial(Local/${get_de...@from-internal,,g)   <- here you
have to use the context you use for the internal users. Also, notice
the "g" option - it is very important
exten => 0,n,Set(DYNAMIC_FEATURES=)
exten => 0,n,Goto(s,cont_3)    <- here you have to jump to the "line"
where you add the person to the conference

Step 4. Put in your dialplan the following lines for the Dynamic features:

[macro-conf-inv]
exten => s,1,ChannelRedirect(${BRIDGEPEER},conference-redirect,${Room_number},1)

[macro-conf-noinv]
exten => s,1,SoftHangup(${BRIDGEPEER})

This setup should work and maybe some twicks should be done in order
to work in your system.

I'll be happy to help you and you could contact me off list in case
you think it is more easy.

Best regards,
Ioan (Nini) Indreias
www.modulo.ro

On Fri, Oct 2, 2009 at 1:08 AM, Harley Holcombe
 wrote:
>
> The extension does exist, as the other caller is redirected to the room.  
> Here's the relevant lines in extensions.conf:
>
> [dynamic-nway]
> exten => _XXX,1,Answer
>
> I've been trying to get this to work on and off for a while now, and it's 
> time to get serious.  If someone would like to get paid for getting this to 
> work please contact me off-list (I also have a Google Wave invite if you're 
> interested).  The solution (and the steps we take) will of course be posted 
> back here and you will also have my eternal gratitude.
>
> Thanks,
>   Harley
>
>
>
> From:
> Matt Riddell 
> To:
> asterisk-users@lists.digium.com
> Date: 23/09/2009 04:00 PM
> Subject:
> Re: [asterisk-users] Bringing people into a conference
> Sent by:
> asterisk-users-boun...@lists.digium.com
> 
>
>
> On 23/09/09 5:07 PM, Harley Holcombe wrote:
> > 1. Internal person A calls person B
> > 2. Person A presses *0, he is given a dial tone and person B is taken to
> > a conference room
> > 3. Person A calls person C and they can talk, and then person A presses **.
> > 4. Person C is brought to the conference room, but person A is
> > disconnected.
>
> Is there an extension:
>
> dynamic-nway,282,1
>
> Oh, and please refrain from using HTML emails to lists.
>
> --
> Cheers,
>
> Matt Riddell
> Director
> ___
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Re: [asterisk-users] Music On Hold

[Ioan Indreias]

Hello Cyprus,

What is the output of "moh files show" CLI command ?

Best regards,
Ioan (Nini) Indreias
www.modulo.ro

On Fri, Oct 2, 2009 at 11:46 AM, Cyprus VoIP  wrote:
> Hi,
>
> I deleted all the default files and put one that I know that works on
> another Asterisk, but since then, I recompiled Asterisk and the default
> files were added.
>
> In order to test moh, I created a context for it:
>
> [default]
> exten => 888,1,Goto(moh,s,1)
> [moh]
> exten => s,1,Answer
> exten => s,2,MusicOnHold()
>
> When we dial 888, we hear the music and this appears in the console:
>     -- Executing [...@default:1] Goto("SIP/24-08650e80", "moh,s,1") in
> new stack
>     -- Goto (moh,s,1)
>     -- Executing [...@moh:1] Answer("SIP/24-08650e80", "") in new stack
>     -- Executing [...@moh:2] MusicOnHold("SIP/24-08650e80", "") in new stack
>     -- Started music on hold, class 'default', on SIP/24-08650e80
>     -- Stopped music on hold on SIP/24-08650e80
>   == Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80'
>
>
> But, when I just put a call on hold, nothing is played and nothing
> appears in the console.
>
> I have no idea why this happens and what to do about it. Any suggestions?
>
> Thanks.
>
>
>  Original Message  
> Subject: Re: [asterisk-users] Music On Hold
> From: John A. Sullivan III 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Wednesday, 30 September, 2009 15:27:28
>
>> On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
>> 
>>>  > You see the wav files but do you see the files encoded for the codecs
>>> you are using?
>>> There's only one wav file there. No encoded files, but on asterisk 1.2
>>> we have, it's the same file and it works.
>> 
>> Hmm . . only one wav file.  We had several.  As I recall now, we
>> actually installed 1.6.1.1 and upgraded.  1.6.1.1 had the old hold
>> music.  1.6.1.6 has the new hold music.  But I believe there are several
>> files.  Is that wav file valid, i.e., if you copy it to a system with a
>> sound card and play it, does it play? Could it have been corrupted in
>> copying or have incorrect permissions? - John
>
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Re: [asterisk-users] Music On Hold

[Cyprus VoIP]

 >
 > What is the output of "moh files show" CLI command ?
 >


pbx*CLI> moh show files
Class: default
 File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
 File: /var/lib/asterisk/moh/macroform-the_simplicity
 File: /var/lib/asterisk/moh/macroform-robot_dity
 File: /var/lib/asterisk/moh/macroform-cold_day
 File: /var/lib/asterisk/moh/reno_project-system
 File: /var/lib/asterisk/moh/music_100
 File: /var/lib/asterisk/moh/CHANGES-asterisk-moh-opsound-2
 File: /var/lib/asterisk/moh/CREDITS-asterisk-moh-opsound-2
 File: /var/lib/asterisk/moh/LICENSE-asterisk-moh-opsound-2

One more thing I tried is to add the m option in the dial command, to 
check what happens, when the call is initially originated:

exten => ,n,Dial(SIP/21&SIP/22&SIP/23&SIP/24&SIP/25,,m(default))

This is what the console shows me:

 -- Executing [1...@default:6] Dial("SIP/IN-PROXY-08563908", 
"SIP/21&SIP/22&SIP/23&SIP/24&SIP/25,,m(default)") in new stack
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 21
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 22
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 23
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 24
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 25
 -- Started music on hold, class 'default', on SIP/IN-PROXY-08563908
 -- SIP/23-b7c2d928 is ringing
 -- SIP/24-b7a7b3a8 is ringing
 -- SIP/22-b7c35cd8 is ringing
 -- SIP/21-b7c3a508 is ringing
 -- SIP/25-b7a83350 is ringing
 -- Stopped music on hold on SIP/IN-PROXY-08563908
   == Spawn extension (default, 99935709, 6) exited non-zero on 
'SIP/IN-PROXY-08563908'


The music is played instead of the RBT, but during the conversation, 
when put on hold, I only get silence and I don't get any reference in 
the console to the fact that the call has been put on hold.

Thanks.

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[asterisk-users] Problem with inbound calls - asterisk 1.6.1.6

[Carlo Dimaggio]

Hi all,

I have a new installation with asterisk 1.6.1.6 but I'm unable to  
receive calls from a SIP trunk:

[Oct  2 14:30:09] NOTICE[21554]: chan_sip.c:18523  
handle_request_invite: Call from 'user001' to extension 'user001'  
rejected because extension not found.

Are there any changes from 1.6.0 to 1.6.1 (or there is a bug)?
Below my simple configuration:

sip.conf

register => user001:pass...@sip.clio.it/user001

[user001-sip-in]
context=default
defaultuser=user001
fromuser=001
fromdomain=sip.xxx.it
type=user
insecure=port,invite
secret=pass001
qualify=yes
port=5060
nat=no
host=sip.xxx.it
canreinvite=no

---

extensions.conf

[default]
exten => user001,1,Noop(Inbound call)


Thanks and regards
Carlo Dimaggio

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[asterisk-users] Creating a clear channel on zaptel

[Zeeshan Zakaria]

Hi,

Is it possible to create a clear zaptel channel which doesn't require to be
picked up? The requirement of my client is to open a clear channel to a
recorder which starts recording certain message. Currently the channel which
is created by zaptel requires the other end to answer the call, and the
recorded can't answer, so the channel get hung up after a certain number of
rings.

Zaptel hardware is a Rhino R4T1 card with fxo_ks signalling, esf framing and
b8zs encoding. zapata.conf settings are as follows:

[channels]
group = 0,11
context = zaptel
signalling = fxo_ks
channel => 1-24
group =
context = default

Any help would be highly appreciated.

-- 
Zeeshan A Zakaria
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Re: [asterisk-users] MeetMe Hints

[Paul Dugas]

bump...   Anybody using MeetMe hints for busy-lamps on phones?  Anyone
seeing this issue?

On Mon, Sep 28, 2009 at 11:08 AM, Paul Dugas wrote:

> I suspect the issue I'm having is more specific to the MeetMe app but
> that's just a guess.  app_meetme.c (in 1.6.1.6) calls
>
>  ast_devstate_changed(AST_DEVICE_INUSE, "meetme:%s", conf->confno)
>
> when the first caller enters a conference and
>
>  ast_devstate_changed(AST_DEVICE_NOT_INUSE, "meetme:%s", conf->confno)
>
> when the last caller leaves.  Running "core show hints" shows the
> meetme:600 as Status:Unavailable most of the time after a conference
> ends.  Occasionally, it'll show Status:Idle.  I guess I'm wondering
> where the hint could be getting changed to Unavailable since it
> doesn't look like the MeetMe app ever sets them this way.
>
> I wonder if it's a result of my using an extension pattern.  AFAIK,
> the 6...@dialtone hint is created from the _...@dialtone pattern when
> the first station subscribes.  I wonder if these polycom stations are
> unsubscribing or timing out thus removing the pattern-generated hint.
> I'll try a statically configured hint to see if that's the issue.
>
> I also wonder if there's any logic elsewhere that would ever set the
> initial hint status for a meetme.  With a brief look at app_meetme.c,
> I don't see where it updates the device state except on join/leave.
>
> P
>
> On Mon, Sep 28, 2009 at 9:24 AM, Danny Nicholas  wrote:
> > The routine ast_extension_states in main/pbx.c is set up (in 1.4.26.1) to
> > return these values:
> >
> >   switch (devstate) {
> >case AST_DEVICE_ONHOLD:
> >return AST_EXTENSION_ONHOLD;
> >case AST_DEVICE_BUSY:
> >return AST_EXTENSION_BUSY;
> >case AST_DEVICE_UNAVAILABLE:
> >case AST_DEVICE_UNKNOWN:
> >case AST_DEVICE_INVALID:
> >return AST_EXTENSION_UNAVAILABLE;
> >case AST_DEVICE_RINGINUSE:
> >return (AST_EXTENSION_INUSE | AST_EXTENSION_RINGING);
> >case AST_DEVICE_RINGING:
> >return AST_EXTENSION_RINGING;
> >case AST_DEVICE_INUSE:
> >return AST_EXTENSION_INUSE;
> >case AST_DEVICE_NOT_INUSE:
> >return AST_EXTENSION_NOT_INUSE;
> >case AST_DEVICE_TOTAL: /* not a device state, included for
> > completeness */
> >break;
> >
> > These come back "in English" as
> > static const struct cfextension_states {
> >int extension_state;
> >const char * const text;
> > } extension_states[] = {
> >{ AST_EXTENSION_NOT_INUSE, "Idle" },
> >{ AST_EXTENSION_INUSE, "InUse" },
> >{ AST_EXTENSION_BUSY,  "Busy" },
> >{ AST_EXTENSION_UNAVAILABLE,   "Unavailable" },
> >{ AST_EXTENSION_RINGING,   "Ringing" },
> >{ AST_EXTENSION_INUSE | AST_EXTENSION_RINGING, "InUse&Ringing" },
> >{ AST_EXTENSION_ONHOLD,"Hold" },
> >{ AST_EXTENSION_INUSE | AST_EXTENSION_ONHOLD,  "InUse&Hold" }
> >
> > So a line Is Unavailable on 3 conditions, but only Idle on one; unless
> you
> > tweak to make UNAVAILABLE equivalent to NOT_INUSE.  I don't know the
> > ramifications if any of the tweak.
> >
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Dugas
> > Sent: Sunday, September 27, 2009 5:31 PM
> > To: Asterisk Users Mailing List
> > Subject: [asterisk-users] MeetMe Hints
> >
> > I've got hints setup for my MeetMe conferences like so:
> >
> >exten => _60X,hint,MeetMe:${EXTEN}
> >
> > and they show up in "core show hints" like so
> >
> >6...@dialtone: MeetMe:600State:Unavailable
> >   Watchers  1
> >_...@dialtone: MeetMe:${EXTEN}   State:Unavailable
> >Watchers  0
> >
> > I'm wondering why they're Unavailable instead of Idle.  They go to
> > "State:InUse" when active but usually return to Unavailable when the
> > conference ends.  Occasionally they end up in InUse but not
> > consistently.
> >
> > Anybody know why?
> >
> > Paul
> > --
> > Paul Dugas -- Computer Engineer -- Dugas Enterprises, LLC
> > 522 Black Canyon Park, Canton GA 30114 USA
> > p...@dugasenterprises.com -- +1.404.932.1355
> >
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[asterisk-users] How to call extensions and add them to a conference room

[Zeeshan Zakaria]

Greetings,

I have created simple conferencing solution before using meetme application,
but this times its a little tricky.

My client needs a functionality to call multiple extensions to join a
conference room. Extensions will ring like in a ring group, and on pick up,
user will be either automatically added to the conference room, or maybe
I'll program them to enter 9 to accept and 8 to reject the invitation.

I need a starting point to start writing this dialplan. Any advise on what
applications and commands I can use to accomplish this would be highly
appreciated.

Thanks,

-- 
Zeeshan A Zakaria
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Re: [asterisk-users] How to call extensions and add them to aconference room

[Danny Nicholas]

This one is pretty simple.  You just need to set up two contexts.  Context 1
will answer and connect to your room

[answerandgo]

-  exten => s,1,answer

-  exten => s,2,meetme(100,options)

-  exten => s,3,playback(vm-goodbye)

-  exten => s,4,hangup

 

context 2 will answer and play a message, then do IVR function

[askandgo]

-  exten => s,1,answer

-  exten => s,2,background(greeting)

-  exten => s,3,waitexten(10,m)

-  exten => s,4,playback(vm-goodbye)

-  exten => s,5,hangup

-  exten => 9,1,meetme(100,options)

-  exten => 9,2.goto(askandgo,s,4)

-  exten => 8,1,playback(rejected)

-  exten => 8,2,goto(askandgo,s,4)

 

Then do an AMI or call file to call your extension/number and start the
appropriate context.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Friday, October 02, 2009 7:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to call extensions and add them to aconference
room

 

Greetings,

I have created simple conferencing solution before using meetme application,
but this times its a little tricky.

My client needs a functionality to call multiple extensions to join a
conference room. Extensions will ring like in a ring group, and on pick up,
user will be either automatically added to the conference room, or maybe
I'll program them to enter 9 to accept and 8 to reject the invitation.

I need a starting point to start writing this dialplan. Any advise on what
applications and commands I can use to accomplish this would be highly
appreciated.

Thanks,

-- 
Zeeshan A Zakaria

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Re: [asterisk-users] Music On Hold

[Danny Nicholas]

What does your musiconhold.conf look like?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP
Sent: Friday, October 02, 2009 3:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music On Hold

Hi,

I deleted all the default files and put one that I know that works on 
another Asterisk, but since then, I recompiled Asterisk and the default 
files were added.

In order to test moh, I created a context for it:

[default]
exten => 888,1,Goto(moh,s,1)
[moh]
exten => s,1,Answer
exten => s,2,MusicOnHold()

When we dial 888, we hear the music and this appears in the console:
 -- Executing [...@default:1] Goto("SIP/24-08650e80", "moh,s,1") in 
new stack
 -- Goto (moh,s,1)
 -- Executing [...@moh:1] Answer("SIP/24-08650e80", "") in new stack
 -- Executing [...@moh:2] MusicOnHold("SIP/24-08650e80", "") in new stack
 -- Started music on hold, class 'default', on SIP/24-08650e80
 -- Stopped music on hold on SIP/24-08650e80
   == Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80'


But, when I just put a call on hold, nothing is played and nothing 
appears in the console.

I have no idea why this happens and what to do about it. Any suggestions?

Thanks.


 Original Message  
Subject: Re: [asterisk-users] Music On Hold
From: John A. Sullivan III 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Date: Wednesday, 30 September, 2009 15:27:28

> On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
> 
>>  > You see the wav files but do you see the files encoded for the codecs 
>> you are using?
>> There's only one wav file there. No encoded files, but on asterisk 1.2 
>> we have, it's the same file and it works.
> 
> Hmm . . only one wav file.  We had several.  As I recall now, we
> actually installed 1.6.1.1 and upgraded.  1.6.1.1 had the old hold
> music.  1.6.1.6 has the new hold music.  But I believe there are several
> files.  Is that wav file valid, i.e., if you copy it to a system with a
> sound card and play it, does it play? Could it have been corrupted in
> copying or have incorrect permissions? - John

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[asterisk-users] Followme

[Anahi Ludueña]

Hi everybody,
What I need to do is to run a context where I'll pass some phones (for example: 
3 numbers). 
I need to make something like a followme, if the first phone is not answered, 
I'll call the second one, and so on. 
That dial plan is not the problem, my problem is when I execute the AMI, I'm 
using the Originate. It needs a channel as an argument, so the context can be 
executed; but what channel should I pass there? (the phone numbers are in the 
Variable argument)
Thanks in advance.






Anahi Ludueña
 

  
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Re: [asterisk-users] Followme

[Danny Nicholas]

Local/1 will run the context without tying up resources.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 8:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Followme

 

Hi everybody,
What I need to do is to run a context where I'll pass some phones (for
example: 3 numbers). 
I need to make something like a followme, if the first phone is not
answered, I'll call the second one, and so on. 
That dial plan is not the problem, my problem is when I execute the AMI, I'm
using the Originate. It needs a channel as an argument, so the context can
be executed; but what channel should I pass there? (the phone numbers are in
the Variable argument)
Thanks in advance.




  _  

Anahi Ludueña

 





  _  

Comparte tus fotos con tus amigos. Más fácil con Windows Live
 

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Re: [asterisk-users] How to call extensions and add them to aconference room

[Zeeshan Zakaria]

Thanks Danny, seems like you have saved my day. I'll try it right now. It
looks easier than I thought.

Zeeshan

On Fri, Oct 2, 2009 at 9:05 AM, Danny Nicholas  wrote:

>  This one is pretty simple.  You just need to set up two contexts.
> Context 1 will answer and connect to your room
>
> [answerandgo]
>
> -  exten => s,1,answer
>
> -  exten => s,2,meetme(100,options)
>
> -  exten => s,3,playback(vm-goodbye)
>
> -  exten => s,4,hangup
>
>
>
> context 2 will answer and play a message, then do IVR function
>
> [askandgo]
>
> -  exten => s,1,answer
>
> -  exten => s,2,background(greeting)
>
> -  exten => s,3,waitexten(10,m)
>
> -  exten => s,4,playback(vm-goodbye)
>
> -  exten => s,5,hangup
>
> -  exten => 9,1,meetme(100,options)
>
> -  exten => 9,2.goto(askandgo,s,4)
>
> -  exten => 8,1,playback(rejected)
>
> -  exten => 8,2,goto(askandgo,s,4)
>
>
>
> Then do an AMI or call file to call your extension/number and start the
> appropriate context.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
> *Sent:* Friday, October 02, 2009 7:53 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] How to call extensions and add them to
> aconference room
>
>
>
> Greetings,
>
> I have created simple conferencing solution before using meetme
> application, but this times its a little tricky.
>
> My client needs a functionality to call multiple extensions to join a
> conference room. Extensions will ring like in a ring group, and on pick up,
> user will be either automatically added to the conference room, or maybe
> I'll program them to enter 9 to accept and 8 to reject the invitation.
>
> I need a starting point to start writing this dialplan. Any advise on what
> applications and commands I can use to accomplish this would be highly
> appreciated.
>
> Thanks,
>
> --
> Zeeshan A Zakaria
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Zeeshan A Zakaria
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Re: [asterisk-users] "got stuck at 150 calls, above that not working in stress test"

[das sandesh]

These calls are from asterisk. I am using sipp to generate the calls and I
increased the limits using the command line interface and used ulimit -n
1 and as well as changed in /etc/security/limits.conf. I dint find any
errors on the console...

Thanks
Sandesh

On Thu, Oct 1, 2009 at 4:29 PM, Matt Riddell  wrote:

> On 2/10/09 12:41 AM, das sandesh wrote:
> > Hi Matt,
> >
> > When I get can more that 150 calls, i get a busy signal (Congestion) for
> > the calls above 150 - says "your call cannot be completed now", its
> > allowing only 150 callsIs there any thing related to field
> > descriptors from linux point of view that I need to increase inorder to
> > increase the call capacity.
>
> Is that coming from Asterisk?
>
> It seems strange that Asterisk would reject the call unless you have
> settings in asterisk.conf to do this. You've said you've already
> increased the file descriptor limits - did you do this in the console
> you were using to subsequently run Asterisk from?
>
> Do you get any errors in the Asterisk console?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
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Re: [asterisk-users] "got stuck at 150 calls, above that not working in stress test"

[Matt Riddell]

On 3/10/09 2:40 AM, das sandesh wrote:
> These calls are from asterisk. I am using sipp to generate the calls and
> I increased the limits using the command line interface and used ulimit
> -n 1 and as well as changed in /etc/security/limits.conf. I dint
> find any errors on the console...

How are you spacing the calls out?  I.E. how many calls per second?

-- 
Cheers,

Matt Riddell
Director
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Re: [asterisk-users] Creating a clear channel on zaptel

[Tzafrir Cohen]

On Fri, Oct 02, 2009 at 08:34:54AM -0400, Zeeshan Zakaria wrote:
> Hi,
> 
> Is it possible to create a clear zaptel channel which doesn't require to be
> picked up? The requirement of my client is to open a clear channel to a
> recorder which starts recording certain message. Currently the channel which
> is created by zaptel requires the other end to answer the call, and the
> recorded can't answer, so the channel get hung up after a certain number of
> rings.

What do you mean by "recorder"? How does it connect? What does it
record?

> 
> Zaptel hardware is a Rhino R4T1 card with fxo_ks signalling, esf framing and
> b8zs encoding. zapata.conf settings are as follows:
> 
> [channels]
> group = 0,11
> context = zaptel
> signalling = fxo_ks
> channel => 1-24
> group =
> context = default
> 
> Any help would be highly appreciated.
> 
> -- 
> Zeeshan A Zakaria

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-- 
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icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] One side SIP goes dead on length conversation

[David Cook]

Has anyone seen something like this before. Randomly, on longish calls, the
local side of the call audio goes dead. Meaning remote caller can hear us
but we cannot hear the remote person?

Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009 x86_64
x86_64 x86_64 GNU/Linux

Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.

WANPIPE Release: 3.4.1

Wanpipe Config:

Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud
rate |

wanpipe1| N/A  | A101/1D/A102/2D/4/4D/8| 169 | 4   | 1|
N/A | 0 |

Wanrouter Status:

Device name | Protocol | Station | Status|

wanpipe1| AFT TE1  | N/A | Connected |

 

Local sets are all Aastra 9143's.

- dbc.

 

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Re: [asterisk-users] Followme

[Anahi Ludueña]

Thanks Danny,
It seems I'm doing something wrong.
Let forget the followme, I have this context:

[new-context]
exten => 1,1,Answer
exten => 1,2,Dial(sip/1000)
exten => 1,3,Playback(sorrynoanswer)
exten => 1,4,Hangup 

Now, I execute the Originate with these parameters:
Channel: Local/1
Context: new-context
Priority: 1

But it gives this error:

Response: Error
Message: Originate failed

Do you know if there is something wrong?

Thanks again.





From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 08:25:58 -0500
Subject: Re: [asterisk-users] Followme



















Local/1 will run the context without tying
up resources.

 









From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Friday, October 02, 2009
8:20 AM

To:
asterisk-users@lists.digium.com

Subject: [asterisk-users] Followme



 

Hi everybody,

What I need to do is to run a context where I'll pass some phones (for example:
3 numbers). 

I need to make something like a followme, if the first phone is not answered,
I'll call the second one, and so on. 

That dial plan is not the problem, my problem is when I execute the AMI, I'm
using the Originate. It needs a channel as an argument, so the context can be
executed; but what channel should I pass there? (the phone numbers are in the
Variable argument)

Thanks in advance.















Anahi
Ludueña

 















Comparte tus fotos con tus amigos. Más fácil con Windows Live

  
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Re: [asterisk-users] "got stuck at 150 calls, above that not working in stress test"

[das sandesh]

I am using the command:
./sipp -sn uac -d 200 -s-l
200

Its 10 calls per second and 200 concurrent calls, similarly I used 2 ssh
sessions each sending 100 concurrent calls. But this was limiting to only
150 calls.

Thanks
Sandesh

On Fri, Oct 2, 2009 at 9:23 AM, Matt Riddell  wrote:

> On 3/10/09 2:40 AM, das sandesh wrote:
> > These calls are from asterisk. I am using sipp to generate the calls and
> > I increased the limits using the command line interface and used ulimit
> > -n 1 and as well as changed in /etc/security/limits.conf. I dint
> > find any errors on the console...
>
> How are you spacing the calls out?  I.E. how many calls per second?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
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>
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[asterisk-users] Sending a DTMF remotely with PlayDTMF problem.

[Pablo Bernasconi]

Hello,

I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.

isb177*CLI> features show
Builtin Feature   Default Current
---   --- ---
Pickup*8  *8
Blind Transfer#   #8
Attended Transfer *2
One Touch Monitor *1
Disconnect Call   *   **
Park Call
One Touch MixMonitor

Dynamic Feature   Default Current
---   --- ---
(none)

Call parking

Parking extension   :  70
Parking context :  parkedcalls
Parked call extensions:  71-78


My script is:

#!/usr/bin/php -q



The script output is:

Array
(
[0] => Asterisk Call Manager/1.1
[1] => Response: Success
[2] => Message: Authentication accepted
[3] =>
[4] => Response: Success
[5] => Message: DTMF successfully queued
[6] =>
[7] => Response: Success
[8] => Message: DTMF successfully queued
[9] =>
[10] => Response: Goodbye
[11] => Message: Thanks for all the fish.
[12] =>
[13] =>
)


When I run the script I can hear the two digit (only the audio) but nothing
happens, the Transfer menu doesnt start. The Cli shows:


[Oct  2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'login'
  == Manager 'admin' logged on from 127.0.0.1
[Oct  2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'PlayDTMF'
[Oct  2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread
-1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread
-1217414256 in procedure ast_waitfor_nandfds
[Oct  2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided
for write to channel 'SIP/1000-0a292360'
[Oct  2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'PlayDTMF'
[Oct  2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread
-1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread
-1217414256 in procedure ast_waitfor_nandfds
[Oct  2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided
for write to channel 'SIP/1000-0a292360'
[Oct  2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'Logoff'
  == Manager 'admin' logged off from 127.0.0.1


BUT, if I press #8 in the softphone, I can hear the two digit and
inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli
output in this case is:

[Oct  2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of
60 bytes
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#),
at 192.168.0.148
[Oct  2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#'
received on SIP/1000-0a292360
[Oct  2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin
passthrough '#' on SIP/1000-0a292360
[Oct  2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF
begin on channel (SIP/1000-0a292360)
[Oct  2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge
stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#),
at 192.168.0.148
[Oct  2 11:09:20] DTMF[29533]: channel.c:2768 __ast_read: DTMF end '#'
received on SIP/1000-0a292360, duration 80 ms
[Oct  2 11:09:20] DTMF[29533]: channel.c:2808 __ast_read: DTMF end accepted
with begin '#' on SIP/1000-0a292360
[Oct  2 11:09:20] DTMF[29533]: channel.c:2824 __ast_read: DTMF end
passthrough '#' on SIP/1000-0a292360
[Oct  2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF
end on channel (SIP/1000-0a292360)
[Oct  2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge
stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408
[Oct  2 11:09:20] DEBUG[29533]: features.c:1836 ast_feature_interpret:
Feature interpret: chan=SIP/1000-0a292360, peer=SIP/1001-0a026408, code=#,
sense=1, features=2, dynamic=#
[Oct  2 11:09:20] DEBUG[29533]: features.c:2496 ast_bridge_call: Set time
limit to 2000
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
0008 (len = 4)
[Oct  2 11:09:21] DEBUG[29533]: rtp.c:806 send_dtmf: S

Re: [asterisk-users] Followme

[Danny Nicholas]

Change the 1’s to s.  The 1 assumes that you pressed 1 from an IVR/DTMF
selection.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 9:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Followme

 

Thanks Danny,
It seems I'm doing something wrong.
Let forget the followme, I have this context:

[new-context]
exten => 1,1,Answer
exten => 1,2,Dial(sip/1000)
exten => 1,3,Playback(sorrynoanswer)
exten => 1,4,Hangup 

Now, I execute the Originate with these parameters:
Channel: Local/1
Context: new-context
Priority: 1

But it gives this error:

Response: Error
Message: Originate failed

Do you know if there is something wrong?

Thanks again.






  _  

From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 08:25:58 -0500
Subject: Re: [asterisk-users] Followme

Local/1 will run the context without tying up resources.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 8:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Followme

 

Hi everybody,
What I need to do is to run a context where I'll pass some phones (for
example: 3 numbers). 
I need to make something like a followme, if the first phone is not
answered, I'll call the second one, and so on. 
That dial plan is not the problem, my problem is when I execute the AMI, I'm
using the Originate. It needs a channel as an argument, so the context can
be executed; but what channel should I pass there? (the phone numbers are in
the Variable argument)
Thanks in advance.



  _  

Anahi Ludueña

 

 

  _  

Comparte tus fotos con tus amigos. Más fácil con Windows Live
 

 

  _  

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Re: [asterisk-users] Followme

[Anahi Ludueña]

Thanks, anyway the result is the same...


Response: Error

Message: Originate failed




Anahi Ludueña
 



From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 10:01:05 -0500
Subject: Re: [asterisk-users] Followme



















Change the 1’s to s.  The 1 assumes
that you pressed 1 from an IVR/DTMF selection.

 









From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Friday, October 02, 2009
9:53 AM

To:
asterisk-users@lists.digium.com

Subject: Re: [asterisk-users]
Followme



 

Thanks Danny,

It seems I'm doing something wrong.

Let forget the followme, I have this context:



[new-context]

exten => 1,1,Answer

exten => 1,2,Dial(sip/1000)

exten => 1,3,Playback(sorrynoanswer)

exten => 1,4,Hangup 



Now, I execute the Originate with these parameters:

Channel: Local/1

Context: new-context

Priority: 1



But it gives this error:



Response: Error

Message: Originate failed



Do you know if there is something wrong?



Thanks again.

















From: da...@debsinc.com

To: asterisk-users@lists.digium.com

Date: Fri, 2 Oct 2009 08:25:58 -0500

Subject: Re: [asterisk-users] Followme



Local/1 will run the context without tying
up resources.

 









From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Friday, October 02, 2009
8:20 AM

To:
asterisk-users@lists.digium.com

Subject: [asterisk-users] Followme



 

Hi everybody,

What I need to do is to run a context where I'll pass some phones (for example:
3 numbers). 

I need to make something like a followme, if the first phone is not answered,
I'll call the second one, and so on. 

That dial plan is not the problem, my problem is when I execute the AMI, I'm
using the Originate. It needs a channel as an argument, so the context can be
executed; but what channel should I pass there? (the phone numbers are in the
Variable argument)

Thanks in advance.













Anahi
Ludueña

 



 







Comparte tus fotos con tus amigos. Más fácil con Windows Live



 







Diferentes formas de estar en contacto con amigos y
familiares. Descúbrelas. Descúbrelas.

  
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Followme

[Danny Nicholas]

The 1 (1,1; 1,2; 1,3) needs to be s (s,1; s,2; s,3).  It works for me with
that change

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 9:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Followme

 

Thanks Danny,
It seems I'm doing something wrong.
Let forget the followme, I have this context:

[new-context]
exten => 1,1,Answer
exten => 1,2,Dial(sip/1000)
exten => 1,3,Playback(sorrynoanswer)
exten => 1,4,Hangup 

Now, I execute the Originate with these parameters:
Channel: Local/1
Context: new-context
Priority: 1

But it gives this error:

Response: Error
Message: Originate failed

Do you know if there is something wrong?

Thanks again.






  _  

From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 08:25:58 -0500
Subject: Re: [asterisk-users] Followme

Local/1 will run the context without tying up resources.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 8:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Followme

 

Hi everybody,
What I need to do is to run a context where I'll pass some phones (for
example: 3 numbers). 
I need to make something like a followme, if the first phone is not
answered, I'll call the second one, and so on. 
That dial plan is not the problem, my problem is when I execute the AMI, I'm
using the Originate. It needs a channel as an argument, so the context can
be executed; but what channel should I pass there? (the phone numbers are in
the Variable argument)
Thanks in advance.



  _  

Anahi Ludueña

 

 

  _  

Comparte tus fotos con tus amigos. Más fácil con Windows Live
 

 

  _  

Diferentes formas de estar en contacto con amigos y familiares. Descúbrelas.
Descúbrelas.  

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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Followme

[Anahi Ludueña]

Maybe there is another problem.
I changed the context like you said.
Where is the local channel configured? or is it implicit?
Sorry but I'm newbie with Asterisk...





Anahi Ludueña
 



From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 10:15:45 -0500
Subject: Re: [asterisk-users] Followme



















The 1 (1,1; 1,2; 1,3) needs to be s (s,1;
s,2; s,3).  It works for me with that change

 









From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Friday, October 02, 2009
9:53 AM

To:
asterisk-users@lists.digium.com

Subject: Re: [asterisk-users]
Followme



 

Thanks Danny,

It seems I'm doing something wrong.

Let forget the followme, I have this context:



[new-context]

exten => 1,1,Answer

exten => 1,2,Dial(sip/1000)

exten => 1,3,Playback(sorrynoanswer)

exten => 1,4,Hangup 



Now, I execute the Originate with these parameters:

Channel: Local/1

Context: new-context

Priority: 1



But it gives this error:



Response: Error

Message: Originate failed



Do you know if there is something wrong?



Thanks again.

















From: da...@debsinc.com

To: asterisk-users@lists.digium.com

Date: Fri, 2 Oct 2009 08:25:58 -0500

Subject: Re: [asterisk-users] Followme



Local/1 will run the context without tying
up resources.

 









From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Friday, October 02, 2009
8:20 AM

To:
asterisk-users@lists.digium.com

Subject: [asterisk-users] Followme



 

Hi everybody,

What I need to do is to run a context where I'll pass some phones (for example:
3 numbers). 

I need to make something like a followme, if the first phone is not answered,
I'll call the second one, and so on. 

That dial plan is not the problem, my problem is when I execute the AMI, I'm
using the Originate. It needs a channel as an argument, so the context can be
executed; but what channel should I pass there? (the phone numbers are in the
Variable argument)

Thanks in advance.













Anahi
Ludueña

 



 







Comparte tus fotos con tus amigos. Más fácil con Windows Live



 







Diferentes formas de estar en contacto con amigos y
familiares. Descúbrelas. Descúbrelas.

  
_
¿Quieres ver los mejores videos de MSN? Enciende Messenger TV
http://messengertv.msn.com/mkt/es-es/default.htm___
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Re: [asterisk-users] Followme

[Jim Dickenson]

The Originate packet is used to connect two legs of a call. This  
packet does not have all the needed parts:


Channel: Local/1
Context: new-context
Priority: 1


I do something like this:

Action: Originate
Channel: Local/get_i...@cfmc_cdi_private
Exten: do_noop
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=GetInfo
ActionID: GetInfo
Async: true

This dials a local channel and connects the call with another local  
channel.


--
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Oct 2, 2009, at 8:15 AM, Danny Nicholas wrote:

The 1 (1,1; 1,2; 1,3) needs to be s (s,1; s,2; s,3).  It works for  
me with that change


From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- 
boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Friday, October 02, 2009 9:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Followme

Thanks Danny,
It seems I'm doing something wrong.
Let forget the followme, I have this context:

[new-context]
exten => 1,1,Answer
exten => 1,2,Dial(sip/1000)
exten => 1,3,Playback(sorrynoanswer)
exten => 1,4,Hangup

Now, I execute the Originate with these parameters:
Channel: Local/1
Context: new-context
Priority: 1

But it gives this error:

Response: Error
Message: Originate failed

Do you know if there is something wrong?

Thanks again.





From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 08:25:58 -0500
Subject: Re: [asterisk-users] Followme

Local/1 will run the context without tying up resources.

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- 
boun...@lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Friday, October 02, 2009 8:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Followme

Hi everybody,
What I need to do is to run a context where I'll pass some phones  
(for example: 3 numbers).
I need to make something like a followme, if the first phone is not  
answered, I'll call the second one, and so on.
That dial plan is not the problem, my problem is when I execute the  
AMI, I'm using the Originate. It needs a channel as an argument, so  
the context can be executed; but what channel should I pass there?  
(the phone numbers are in the Variable argument)

Thanks in advance.


Anahi Ludueña



Comparte tus fotos con tus amigos. Más fácil con Windows Live

Diferentes formas de estar en contacto con amigos y familiares.  
Descúbrelas. Descúbrelas.

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  http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Followme

[Danny Nicholas]

Apologies – Local/1 did not work.  Local/170 (170 is a valid SIP extension)
did.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 10:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Followme

 

Thanks, anyway the result is the same...

Response: Error
Message: Originate failed

  _  

Anahi Ludueña

 






  _  

From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 10:01:05 -0500
Subject: Re: [asterisk-users] Followme

Change the 1’s to s.  The 1 assumes that you pressed 1 from an IVR/DTMF
selection.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 9:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Followme

 

Thanks Danny,
It seems I'm doing something wrong.
Let forget the followme, I have this context:

[new-context]
exten => 1,1,Answer
exten => 1,2,Dial(sip/1000)
exten => 1,3,Playback(sorrynoanswer)
exten => 1,4,Hangup 

Now, I execute the Originate with these parameters:
Channel: Local/1
Context: new-context
Priority: 1

But it gives this error:

Response: Error
Message: Originate failed

Do you know if there is something wrong?

Thanks again.





  _  

From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 2 Oct 2009 08:25:58 -0500
Subject: Re: [asterisk-users] Followme

Local/1 will run the context without tying up resources.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 02, 2009 8:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Followme

 

Hi everybody,
What I need to do is to run a context where I'll pass some phones (for
example: 3 numbers). 
I need to make something like a followme, if the first phone is not
answered, I'll call the second one, and so on. 
That dial plan is not the problem, my problem is when I execute the AMI, I'm
using the Originate. It needs a channel as an argument, so the context can
be executed; but what channel should I pass there? (the phone numbers are in
the Variable argument)
Thanks in advance.

  _  

Anahi Ludueña

 

 

  _  

Comparte tus fotos con tus amigos. Más fácil con Windows Live
 

 

  _  

Diferentes formas de estar en contacto con amigos y familiares. Descúbrelas.
Descúbrelas.  

 

  _  

Disfruta antes que nadie del nuevo Windows Live Messenger
 

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Re: [asterisk-users] One side SIP goes dead on length conversation

[Steven Stromer]

I'm under the impression that this sometimes happens when a firewall  
decides that the port you've opened no longer needs to be so. Are you  
using sip_nat? Do you have a firewall between the asterisk host and  
public? How are your VoIP related firewall rules configured?




Has anyone seen something like this before. Randomly, on longish  
calls, the local side of the call audio goes dead. Meaning remote  
caller can hear us but we cannot hear the remote person?


Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009  
x86_64 x86_64 x86_64 GNU/Linux


Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.

WANPIPE Release: 3.4.1

Wanpipe Config:

Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |  
Baud rate |


wanpipe1| N/A  | A101/1D/A102/2D/4/4D/8| 169 | 4   |  
1| N/A | 0 |


Wanrouter Status:

Device name | Protocol | Station | Status|

wanpipe1| AFT TE1  | N/A | Connected |



Local sets are all Aastra 9143’s.

- dbc.


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Re: [asterisk-users] What are the reasons for VoIP echo?

[Steve Underwood]

On 10/02/2009 08:36 AM, Martin wrote:
> if a user calling you hears echo of himself then it's the fault of
> your sip device/sip phone.
> The manufacturer must be using a cheap or an open source echo canceller ...
>
> try getting a different sip device made by some 'normal' company like
> polycom or linksys/cisco
>
You might like to know that a number of people have used the open source 
OSLEC canceller to replace the rather broken one Linksys put in their ATAs.
> Martin
>
> On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham  wrote:
>
>> I have an Asterisk 1.4.2 system that has been installed for about 3
>> months now in our home.  We converted all of our phones to SIP phones,
>> and use two different trunk providers (BroadVoice for incoming&
>> FlowRoute for outgoing).
>>
>> Most of the time its working flawlessly.  But about 1/3rd of the calls
>> that come into us complain of an echo and what is best described as
>> latency issues.  Its not consistent though.  I was on the phone with an
>> insurance company yesterday for about 1 hour and the call was perfect (I
>> originated the call which used Flowroute for the SIP provider).
>>
>> What seems to be a pattern here is cell phones.  When we receive a call
>> from a cell phone, or from certain people on certain phone systems, they
>> consistently complain of echo in the call.  Its far less regular when we
>> originate the call, which suggested to me that the problem might be with
>> Broadvoice.  But I'm now hearing that us calling back the party doesn't
>> always solve the problem either.
>>
>> We upgraded our Internet feed (we're on a cable Internet through our
>> cable company, with 12mb/s down, 1.5mb/s up) and that seems to have
>> helped but not solved this problem.  From what I can see, its some form
>> of latency issue.  We use IPCop as a firewall for our Internet access,
>> but have turned off any IDS on it so that its running fast.  I can play
>> online computer games through the network with no issues at all, so I
>> don't think its slowing down the traffic and if it was I'd expect this
>> problem to be occurring consistently on all calls.
>>
>> Are there any tweaks that I can do with Asterisk to increase the network
>> performance to reduce these issues?  Have others who have experienced
>> this been able to identify the issues to external VoIP SIP providers
>> only, or does our system have something to do with all of this?  At the
>> time of the calls coming in, IPCop is telling me that we don't have more
>> than 100K/s of bandwidth in use, and according to the network bandwidth
>> graphs there, even with 2 people on the phone at the same time, the
>> bandwidth never seems to exceed 300K/s, so I think we have plenty of
>> headroom for this.  I checked with our cable provider for issues with
>> modem latency, and they couldn't detect anything.  Again, I'm not
>> experiencing any lag issues with computer games, particularly those that
>> are heavy in interactivity, so I don't think that is the reason.
>>
>> Any suggestions as to what could be tweaked would be greatly appreciated.
>>
>>  
Steve


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[asterisk-users] Asterisk + Skype deployment

[Alan Lord (News)]

Just FYI Really, nothing to do with me...

http://www.thevarguy.com/2009/10/01/systems-integrator-dials-skype-for-asterisk/



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Re: [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed

[Steve Underwood]

On 10/02/2009 01:44 AM, Martin wrote:
> anyone can just grab the PEF framer datasheet and tweak the driver though...
> last I checked there's a whole section devoted to high impedance in
> the datasheet
>
> Martin
>
The hardware needs to have been built in a particular way, with 
particular versions of the FALC devices, for high impedance mode to be 
possible. Are the Digium cards built in the right way?
> On Thu, Oct 1, 2009 at 9:56 AM, Kevin P. Fleming  wrote:
>
>> Moises Silva wrote:
>>
>>  
>>> May be Martin can help with that, I don't know how to setup Digium
>>> boards in high impedance mode. It seems the feature may not be exported
>>> via configuration files yet, so changes to the driver may be needed?
>>>
>> That is correct, none of our drivers currently expose a method to put
>> the framer interface into high-impedance mode.
>>
>>  
Steve


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Re: [asterisk-users] What are the reasons for VoIP echo?

[Steve Underwood]

On 10/02/2009 09:18 AM, Martin wrote:
> Are you saying there are half duplex phones out there  with half
> duplex speakerphones ?
>
Practically all analogue speakerphones are half duplex. Only a small 
number of analogue phones ever implemented a proper echo canceller based 
speakerphone - usually ones which included the necessary DSP power for 
other purposes, like answering machine functions.
> All analog phones are full duplex ...
>
> Anyways the echo can be created by the analog phone even when it's
> connected to the
> sip ata or even the sip phone ... then you usually have acoustic echo
> which goes from speaker
> to microphone of the handset ... that should be cancelled by the sip
> phone/device... or someone out there will
> hear echo
>
> Martin
>
> On Thu, Oct 1, 2009 at 7:57 PM, John A. Sullivan III
>   wrote:
>
>> I'm quite new to all this but I was under the impression that most
>> electrically induced echo was at the physical interface to the PSTN.  If
>> one is using SIP trunking, I would think this would point to a carrier
>> issue.
>>
>> We also hit an interesting problem with echo today but I don't think
>> this is the issue Myles is having.  We installed fairly high end phones
>> with full duplex speakerphones.  Callers are having a bad problem with
>> echo when the users use the speakerphone.  Because it is full duplex
>> rather than half, if the speakerphone volume and speakerphone mike
>> volume are turned up, the callers are indeed hearing themselves by
>> virtue of the higher quality full duplex!
>>
>> On Thu, 2009-10-01 at 19:36 -0500, Martin wrote:
>>  
>>> if a user calling you hears echo of himself then it's the fault of
>>> your sip device/sip phone.
>>> The manufacturer must be using a cheap or an open source echo canceller ...
>>>
>>> try getting a different sip device made by some 'normal' company like
>>> polycom or linksys/cisco
>>>
>>> Martin
>>>
>>> On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham  wrote:
>>>
 I have an Asterisk 1.4.2 system that has been installed for about 3
 months now in our home.  We converted all of our phones to SIP phones,
 and use two different trunk providers (BroadVoice for incoming&
 FlowRoute for outgoing).

 Most of the time its working flawlessly.  But about 1/3rd of the calls
 that come into us complain of an echo and what is best described as
 latency issues.  Its not consistent though.  I was on the phone with an
 insurance company yesterday for about 1 hour and the call was perfect (I
 originated the call which used Flowroute for the SIP provider).

 What seems to be a pattern here is cell phones.  When we receive a call
 from a cell phone, or from certain people on certain phone systems, they
 consistently complain of echo in the call.  Its far less regular when we
 originate the call, which suggested to me that the problem might be with
 Broadvoice.  But I'm now hearing that us calling back the party doesn't
 always solve the problem either.

 We upgraded our Internet feed (we're on a cable Internet through our
 cable company, with 12mb/s down, 1.5mb/s up) and that seems to have
 helped but not solved this problem.  From what I can see, its some form
 of latency issue.  We use IPCop as a firewall for our Internet access,
 but have turned off any IDS on it so that its running fast.  I can play
 online computer games through the network with no issues at all, so I
 don't think its slowing down the traffic and if it was I'd expect this
 problem to be occurring consistently on all calls.

 Are there any tweaks that I can do with Asterisk to increase the network
 performance to reduce these issues?  Have others who have experienced
 this been able to identify the issues to external VoIP SIP providers
 only, or does our system have something to do with all of this?  At the
 time of the calls coming in, IPCop is telling me that we don't have more
 than 100K/s of bandwidth in use, and according to the network bandwidth
 graphs there, even with 2 people on the phone at the same time, the
 bandwidth never seems to exceed 300K/s, so I think we have plenty of
 headroom for this.  I checked with our cable provider for issues with
 modem latency, and they couldn't detect anything.  Again, I'm not
 experiencing any lag issues with computer games, particularly those that
 are heavy in interactivity, so I don't think that is the reason.

 Any suggestions as to what could be tweaked would be greatly appreciated.
  
Steve


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[asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)

[James Hankins]

I'm looking into doing an HA setup for a Asterisk 1.4 install on  
Centos.  I've seen a number of different pointers to packages for this  
some of which are packages that seem quite dated from an update  
perspective (Ultra Monkey links I've seen haven't been updated in a  
while).  What is the current best practice on this for this platform?   
My first foray into any of the Linux HA setups but not afraid of the  
command line.

Jim



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Re: [asterisk-users] What are the reasons for VoIP echo?

[Martin]

> You might like to know that a number of people have used the open source
> OSLEC canceller to replace the rather broken one Linksys put in their ATAs.

there's no way to cancel the sip device in Asterisk, is there ? other
than going through zap ...
I wrote once a virtual zaptel driver device that could be used for
this purpose ...

Martin

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Re: [asterisk-users] What are the reasons for VoIP echo?

[Peder]

On 10/02/2009 08:36 AM, Martin wrote:
> if a user calling you hears echo of himself then it's the fault of
> your sip device/sip phone.
> The manufacturer must be using a cheap or an open source echo canceller
...
>
> try getting a different sip device made by some 'normal' company like
> polycom or linksys/cisco
>
> You might like to know that a number of people have used the open source 
> OSLEC canceller to replace the rather broken one Linksys put in their
ATAs.

How would you do that for just those ATAs and not every phone in your
network?



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[asterisk-users] Extra Sounds Missing on 1.6.1.6 install

[Mark Hulber]

It looks like there's a problem with the location or naming of the Extra 
SLN16 sounds:

--14:11:43-- 

http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
Resolving downloads.digium.com... 76.164.171.232
Connecting to downloads.digium.com|76.164.171.232|:80... connected.
HTTP request sent, awaiting response... 301 Moved Permanently
Location:

http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
[following]
--14:11:44-- 

http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
Resolving downloads.asterisk.org... 76.164.171.233
Connecting to downloads.asterisk.org|76.164.171.233|:80... connected.
HTTP request sent, awaiting response... 404 Not Found
14:11:44 ERROR 404: Not Found.
make[1]: ***
[/var/lib/asterisk/sounds/.asterisk-extra-sounds-fr-SLN16-1.4.9] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.6.1.6/sounds'
make: *** [datafiles] Error 2
[r...@asterisk asterisk]# make menuselect
make[1]: Entering directory `/usr/src/asterisk-1.6.1.6'


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Re: [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)

[Jonathan Thurman]

I have been working on a HA procedure for Asterisk on CentOS 5.3, but
haven't had time to publish it.  It is a little complex, but here are
the components used:

- CentOS 5.3
- Asterisk 1.6 (version doesn't matter)
- MySQL
- Cluster services
- GFS2
- DRBD

A basic run-down is:

* Two servers configured with DRBD in Master-Master mode.  All data is
replicated between the two so in case of a failure there should be
very limited data loss (voicemail) if any at all.

* MySQL and Asterisk run on the same node.  If you have an external
MySQL server / don't use MySQL, then this is not an issue.  The MySQL
data directory is also mounted on a GFS2/DRBD partition.  The most
important thing here is to use INNODB, NOT MYISAM!  MyISAM doesn't
take kindly fail-over...

* Using Cluster services enables you to create GFS2 file systems (on
top of DRBD) so that both nodes can see the data at the same time.
This is important to reduce the time required for fail-over.  Cluster
services also handles starting/stopping the services, and migrating
the Virtual IP address between nodes.

* DHCP (if needed) runs on both nodes, as DHCP has native support for
fail-over configuration.

It's pretty easy to get installed and running.  I also create RPMS for
Asterisk, so that the version on each service is the exact same.  I
can upgrade one node, use the cluster manager to fail-over to the
other node (during a maintenance window of course!).

The biggest issue now is that the CentOS Repo is somewhat broken for
Cluster... but there is a work around on the bug tracker for CentOS.
Hopefully that will be resolved soon.

Let me know off list if you need any help!

-Jonathan



On Fri, Oct 2, 2009 at 10:58 AM, James Hankins
 wrote:
> I'm looking into doing an HA setup for a Asterisk 1.4 install on
> Centos.  I've seen a number of different pointers to packages for this
> some of which are packages that seem quite dated from an update
> perspective (Ultra Monkey links I've seen haven't been updated in a
> while).  What is the current best practice on this for this platform?
> My first foray into any of the Linux HA setups but not afraid of the
> command line.
>
> Jim
>
>
>
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[asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[Michelle Dupuis]

Has anyone written an app that monitors SIP/IAX registration attempts?  A
couple of clients are being flooded with SIP registrations (but the source
IP changes every few hours so IPtables won't do)..
 
I would think that any attempt to reg 5 times with a bad password should
cause a 5 minute timeout until reg is considered again.  Has anyone written
such an app?  The name app_hackblock is my contribution to the project :)
 
MD
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Re: [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)

[Fred Posner]

> * Two servers configured with DRBD in Master-Master mode.  All data is
> replicated between the two so in case of a failure there should be
> very limited data loss (voicemail) if any at all.

If you put the asterisk spool, lib, and config files on the DRBD then
you shouldn't lose voicemail or any configuration.

Fred Posner

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Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[Doug Lytle]

Michelle Dupuis wrote:
> Has anyone written an app that monitors SIP/IAX registration 
> attempts?  A couple of clients are being flooded with SIP 
> registrations (but the source IP changes every few hours so IPtables 
> won't do)..
>  
> I would think that any attempt to reg 5 times with a bad password 
> should cause a 5 minute timeout until reg is considered again.  Has 
> anyone written such an app?  The name app_hackblock is my contribution 
> to the project :)

You may want to take a look at this:

http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk

Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Extra Sounds Missing on 1.6.1.6 install

[Mark Hulber]

Looks like the Makefile is broken and putting SLN16 instead of sln16.

Mark Hulber wrote:
> It looks like there's a problem with the location or naming of the Extra 
> SLN16 sounds:
>
> --14:11:43-- 
> 
> http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
> Resolving downloads.digium.com... 76.164.171.232
> Connecting to downloads.digium.com|76.164.171.232|:80... connected.
> HTTP request sent, awaiting response... 301 Moved Permanently
> Location:
> 
> http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
> [following]
> --14:11:44-- 
> 
> http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
> Resolving downloads.asterisk.org... 76.164.171.233
> Connecting to downloads.asterisk.org|76.164.171.233|:80... connected.
> HTTP request sent, awaiting response... 404 Not Found
> 14:11:44 ERROR 404: Not Found.
> make[1]: ***
> [/var/lib/asterisk/sounds/.asterisk-extra-sounds-fr-SLN16-1.4.9] Error 1
> make[1]: Leaving directory `/usr/src/asterisk-1.6.1.6/sounds'
> make: *** [datafiles] Error 2
> [r...@asterisk asterisk]# make menuselect
> make[1]: Entering directory `/usr/src/asterisk-1.6.1.6'
>
>
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Re: [asterisk-users] Extra Sounds Missing on 1.6.1.6 install

[Kyle Kienapfel]

apache is CaSeSeNsItVe where did you get the link with "SLN" capitalized?
http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-sln16-1.4.9.tar.gz

On Fri, Oct 2, 2009 at 11:22 AM, Mark Hulber wrote:

> It looks like there's a problem with the location or naming of the Extra
> SLN16 sounds:
>
>--14:11:43--
>
> http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
>Resolving downloads.digium.com... 76.164.171.232
>Connecting to downloads.digium.com|76.164.171.232|:80... connected.
>HTTP request sent, awaiting response... 301 Moved Permanently
>Location:
>
> http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
>[following]
>--14:11:44--
>
> http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
>Resolving downloads.asterisk.org... 76.164.171.233
>Connecting to downloads.asterisk.org|76.164.171.233|:80... connected.
>HTTP request sent, awaiting response... 404 Not Found
>14:11:44 ERROR 404: Not Found.
>make[1]: ***
>[/var/lib/asterisk/sounds/.asterisk-extra-sounds-fr-SLN16-1.4.9] Error 1
>make[1]: Leaving directory `/usr/src/asterisk-1.6.1.6/sounds'
>make: *** [datafiles] Error 2
>[r...@asterisk asterisk]# make menuselect
>make[1]: Entering directory `/usr/src/asterisk-1.6.1.6'
>
>
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Re: [asterisk-users] Extra Sounds Missing on 1.6.1.6 install

[Jason Parker]

Mark Hulber wrote:
> It looks like there's a problem with the location or naming of the Extra 
> SLN16 sounds:
> 

This has already been fixed in the 1.6.1 branch.  It should make its way into
the next releases.

See 1.6.1 revision 212386.

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Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[C F]

Couple of old posts:
http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html


On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis  wrote:
> Has anyone written an app that monitors SIP/IAX registration attempts?  A
> couple of clients are being flooded with SIP registrations (but the source
> IP changes every few hours so IPtables won't do)..
>
> I would think that any attempt to reg 5 times with a bad password should
> cause a 5 minute timeout until reg is considered again.  Has anyone written
> such an app?  The name app_hackblock is my contribution to the project :)
>
> MD
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Re: [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)

[Jonathan Thurman]

On Fri, Oct 2, 2009 at 11:41 AM, Fred Posner  wrote:
>> * Two servers configured with DRBD in Master-Master mode.  All data is
>> replicated between the two so in case of a failure there should be
>> very limited data loss (voicemail) if any at all.
>
> If you put the asterisk spool, lib, and config files on the DRBD then
> you shouldn't lose voicemail or any configuration.

If someone is in the middle of recording a message, and the server
fails, you will probably lose that message.  That's all I was getting
at.

-Jonathan

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Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[Michelle Dupuis]

Good post.  One of the recommendations is to limit the number of calls per
sip entity.  Is there an easy way to do that in sip.conf? 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, October 02, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Couple of old posts:
http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html


On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis  wrote:
> Has anyone written an app that monitors SIP/IAX registration attempts?  
> A couple of clients are being flooded with SIP registrations (but the 
> source IP changes every few hours so IPtables won't do)..
>
> I would think that any attempt to reg 5 times with a bad password 
> should cause a 5 minute timeout until reg is considered again.  Has 
> anyone written such an app?  The name app_hackblock is my contribution 
> to the project :)
>
> MD
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: 
> http://www.astricon.net
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] TDM410P - False Answer Supervision

[Nitesh Divecha]

Anyone else having this issue with TDM410P card or anyone with a solution?

Please advise... Thanks

Cheers,
Nitesh



Martin wrote:
> Are you in US ?
>
> do you have the proper keywords in zapata.conf/chan_dahdi.conf like
> callprogress=yes etc ?
>
> Martin
>
> On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha  
> wrote:
>   
>> Danny,
>>
>> Thanks for your reply...
>>
>> Yes these are POTS line and I am not calling myself... Any other
>> suggestions?
>>
>> Cheers,
>> Nitesh
>>
>>
>> Danny Nicholas wrote:
>> 
>>> Assuming you're using POTS, you probably won't have much luck with this.  If
>>> you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk
>>> won't process the line until you pick up and punch a dtmf key.  If you are
>>> using E1 or PRI, there is more hope for you.
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
>>> Sent: Thursday, October 01, 2009 4:42 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [asterisk-users] TDM410P - False Answer Supervision
>>>
>>> Hello All,
>>>
>>> Can anyone help me with False Answer Supervision problem with TDM410P
>>> card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
>>> everything works fine except the Answer supervision...
>>>
>>> When the call hits Asterisk it sends the call to one of the TDM410 card
>>> and the call is answered immediately while the call is still in
>>> progress... Here is the debug output: -
>>>
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3602045'
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3602045
>>> [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3602045w
>>> [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9223421808-091b3f50
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> The connect message is sent back immediately when " DAHDI/8-1 answered
>>> SIP/9223421808-091b3f50" while the call is still in progress... If the
>>> call is hang up without answer the sender gets Normal Code 16 while it
>>> suppose to be "Abandoned Call".
>>>
>>>
>>>
>>> The Polarity Reversal only works when call is ANSWERED... Here is the
>>> debug log: -
>>>
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3312808'
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3312808
>>> [Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3312808w
>>> [Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9765782184-091b9678
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event:
>>> Ignore switch to REVERSED Polarity on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event:
>>> Ignoring Polarity switch to IDLE on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event:
>>> Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0,
>>> aonp= 1, honp= 0, pdelay= 600, tv= 301564043
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> Please help...
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>> Register Now: http://www.astricon.net
>>>
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>>>
>>>   
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>
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> as

Re: [asterisk-users] What are the reasons for VoIP echo?

[Steve Underwood]

On 10/03/2009 02:18 AM, Martin wrote:
>> You might like to know that a number of people have used the open source
>> OSLEC canceller to replace the rather broken one Linksys put in their ATAs.
>>  
> there's no way to cancel the sip device in Asterisk, is there ? other
> than going through zap ...
> I wrote once a virtual zaptel driver device that could be used for
> this purpose ...
>
To be more specific, peopel have replaced SPA3102s, with a hopeless echo 
cancellers on the PSTN port, with a Blackfin box using OSLEC and running 
Asterisk. OSLEC works extremely well on almost any consumer line.

Steve


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Re: [asterisk-users] TDM410P - False Answer Supervision

[Danny Nicholas]

Since POTS supervision is questionable at best, the best option IMO would be
to use something like ForkCDR to let you know that the call has really been
answered.  Just do the ForkCDR on the polarity reversal.  This will give you
two CDR events for a good call and one for a bad call.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
Sent: Friday, October 02, 2009 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM410P - False Answer Supervision

Anyone else having this issue with TDM410P card or anyone with a solution?

Please advise... Thanks

Cheers,
Nitesh



Martin wrote:
> Are you in US ?
>
> do you have the proper keywords in zapata.conf/chan_dahdi.conf like
> callprogress=yes etc ?
>
> Martin
>
> On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha 
wrote:
>   
>> Danny,
>>
>> Thanks for your reply...
>>
>> Yes these are POTS line and I am not calling myself... Any other
>> suggestions?
>>
>> Cheers,
>> Nitesh
>>
>>
>> Danny Nicholas wrote:
>> 
>>> Assuming you're using POTS, you probably won't have much luck with this.
If
>>> you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and
asterisk
>>> won't process the line until you pick up and punch a dtmf key.  If you
are
>>> using E1 or PRI, there is more hope for you.
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh
Divecha
>>> Sent: Thursday, October 01, 2009 4:42 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [asterisk-users] TDM410P - False Answer Supervision
>>>
>>> Hello All,
>>>
>>> Can anyone help me with False Answer Supervision problem with TDM410P
>>> card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
>>> everything works fine except the Answer supervision...
>>>
>>> When the call hits Asterisk it sends the call to one of the TDM410 card
>>> and the call is answered immediately while the call is still in
>>> progress... Here is the debug output: -
>>>
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3602045'
>>> [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3602045
>>> [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3602045w
>>> [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9223421808-091b3f50
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> The connect message is sent back immediately when " DAHDI/8-1 answered
>>> SIP/9223421808-091b3f50" while the call is still in progress... If the
>>> call is hang up without answer the sender gets Normal Code 16 while it
>>> suppose to be "Abandoned Call".
>>>
>>>
>>>
>>> The Polarity Reversal only works when call is ANSWERED... Here is the
>>> debug log: -
>>>
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing
>>> '3312808'
>>> [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring
>>> dialing...
>>> -- Called G2/3312808
>>> [Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent
>>> deferred digit string: T3312808w
>>> [Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done
>>> dialing, but waiting for progress detection before doing more...
>>> -- DAHDI/8-1 answered SIP/9765782184-091b9678
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event:
>>> Ignore switch to REVERSED Polarity on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event:
>>> Ignoring Polarity switch to IDLE on channel 8, state 6
>>> [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event:
>>> Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0,
>>> aonp= 1, honp= 0, pdelay= 600, tv= 301564043
>>> -- Hungup 'DAHDI/8-1'
>>> =
>>>
>>> Please help...
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>> Register Now: http://www.astricon.net
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>>>
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>>> Register Now: http://www.astricon.net
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>   

Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[Danny Nicholas]

Sipregisterattempts would seem to be the simplest way to do this.  It is 0
by default, changing it to 5 would stop the hacker after 5 tries.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, October 02, 2009 2:24 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Good post.  One of the recommendations is to limit the number of calls per
sip entity.  Is there an easy way to do that in sip.conf? 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, October 02, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Couple of old posts:
http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html


On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis  wrote:
> Has anyone written an app that monitors SIP/IAX registration attempts?  
> A couple of clients are being flooded with SIP registrations (but the 
> source IP changes every few hours so IPtables won't do)..
>
> I would think that any attempt to reg 5 times with a bad password 
> should cause a 5 minute timeout until reg is considered again.  Has 
> anyone written such an app?  The name app_hackblock is my contribution 
> to the project :)
>
> MD
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: 
> http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] TDM410P - False Answer Supervision

[Nitesh Divecha]

Danny,

Thanks for your reply...

ForkCDR will take care the problem locally but what about the originator 
of the call? Calls are originated from the Nextone Softswitch (Customer) 
to Asterisk and while the call is still in progress Asterisk has already 
sent back "Answered" and the call is accounted Normal code 16 on Nextone 
Billing (Customer). So all the time ASR is at 100% from the 
originator...  which is an issue we are facing...

Cheers,
Nitesh


Danny Nicholas wrote:
> Since POTS supervision is questionable at best, the best option IMO would be
> to use something like ForkCDR to let you know that the call has really been
> answered.  Just do the ForkCDR on the polarity reversal.  This will give you
> two CDR events for a good call and one for a bad call.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
> Sent: Friday, October 02, 2009 2:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TDM410P - False Answer Supervision
>
> Anyone else having this issue with TDM410P card or anyone with a solution?
>
> Please advise... Thanks
>
> Cheers,
> Nitesh
>
>
>
> Martin wrote:
>   
>> Are you in US ?
>>
>> do you have the proper keywords in zapata.conf/chan_dahdi.conf like
>> callprogress=yes etc ?
>>
>> Martin
>>
>> On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha 
>> 
> wrote:
>   
>>   
>> 
>>> Danny,
>>>
>>> Thanks for your reply...
>>>
>>> Yes these are POTS line and I am not calling myself... Any other
>>> suggestions?
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>> Danny Nicholas wrote:
>>> 
>>>   
 Assuming you're using POTS, you probably won't have much luck with this.
 
> If
>   
 you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and
 
> asterisk
>   
 won't process the line until you pick up and punch a dtmf key.  If you
 
> are
>   
 using E1 or PRI, there is more hope for you.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh
 
> Divecha
>   
 Sent: Thursday, October 01, 2009 4:42 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] TDM410P - False Answer Supervision

 Hello All,

 Can anyone help me with False Answer Supervision problem with TDM410P
 card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
 everything works fine except the Answer supervision...

 When the call hits Asterisk it sends the call to one of the TDM410 card
 and the call is answered immediately while the call is still in
 progress... Here is the debug output: -

 [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing
 '3602045'
 [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring
 dialing...
 -- Called G2/3602045
 [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event: Sent
 deferred digit string: T3602045w
 [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event: Done
 dialing, but waiting for progress detection before doing more...
 -- DAHDI/8-1 answered SIP/9223421808-091b3f50
 -- Hungup 'DAHDI/8-1'
 =

 The connect message is sent back immediately when " DAHDI/8-1 answered
 SIP/9223421808-091b3f50" while the call is still in progress... If the
 call is hang up without answer the sender gets Normal Code 16 while it
 suppose to be "Abandoned Call".



 The Polarity Reversal only works when call is ANSWERED... Here is the
 debug log: -

 [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing
 '3312808'
 [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring
 dialing...
 -- Called G2/3312808
 [Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event: Sent
 deferred digit string: T3312808w
 [Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event: Done
 dialing, but waiting for progress detection before doing more...
 -- DAHDI/8-1 answered SIP/9765782184-091b9678
 [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event:
 Ignore switch to REVERSED Polarity on channel 8, state 6
 [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4931 dahdi_handle_event:
 Ignoring Polarity switch to IDLE on channel 8, state 6
 [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4934 dahdi_handle_event:
 Polarity Reversal event occured - DEBUG 2: channel 8, state 6, pol= 0,
 aonp= 1, honp= 0, pdelay= 600, tv= 301564043
 -- Hungup 'DAHDI/8-1'
 =

 Please help...

 Cheers,
 Nitesh


 ___

Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[Michelle Dupuis]

I can't find any reference for this - is there a doc / link? 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, October 02, 2009 3:59 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Sipregisterattempts would seem to be the simplest way to do this.  It is 0
by default, changing it to 5 would stop the hacker after 5 tries.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, October 02, 2009 2:24 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Good post.  One of the recommendations is to limit the number of calls per
sip entity.  Is there an easy way to do that in sip.conf? 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, October 02, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Couple of old posts:
http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html


On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis  wrote:
> Has anyone written an app that monitors SIP/IAX registration attempts?  
> A couple of clients are being flooded with SIP registrations (but the 
> source IP changes every few hours so IPtables won't do)..
>
> I would think that any attempt to reg 5 times with a bad password 
> should cause a 5 minute timeout until reg is considered again.  Has 
> anyone written such an app?  The name app_hackblock is my contribution 
> to the project :)
>
> MD
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>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: 
> http://www.astricon.net
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] TDM410P - False Answer Supervision

[Danny Nicholas]

You can recalculate the "Real" ASR by polling the CDR and only counting the
forked calls as Answered.  If the CDR is in a table instead of the CSV that
I use, you could do a PERL/PHP/C script to "fix" it.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh Divecha
Sent: Friday, October 02, 2009 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM410P - False Answer Supervision

Danny,

Thanks for your reply...

ForkCDR will take care the problem locally but what about the originator 
of the call? Calls are originated from the Nextone Softswitch (Customer) 
to Asterisk and while the call is still in progress Asterisk has already 
sent back "Answered" and the call is accounted Normal code 16 on Nextone 
Billing (Customer). So all the time ASR is at 100% from the 
originator...  which is an issue we are facing...

Cheers,
Nitesh


Danny Nicholas wrote:
> Since POTS supervision is questionable at best, the best option IMO would
be
> to use something like ForkCDR to let you know that the call has really
been
> answered.  Just do the ForkCDR on the polarity reversal.  This will give
you
> two CDR events for a good call and one for a bad call.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh
Divecha
> Sent: Friday, October 02, 2009 2:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TDM410P - False Answer Supervision
>
> Anyone else having this issue with TDM410P card or anyone with a solution?
>
> Please advise... Thanks
>
> Cheers,
> Nitesh
>
>
>
> Martin wrote:
>   
>> Are you in US ?
>>
>> do you have the proper keywords in zapata.conf/chan_dahdi.conf like
>> callprogress=yes etc ?
>>
>> Martin
>>
>> On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha 
>> 
> wrote:
>   
>>   
>> 
>>> Danny,
>>>
>>> Thanks for your reply...
>>>
>>> Yes these are POTS line and I am not calling myself... Any other
>>> suggestions?
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>> Danny Nicholas wrote:
>>> 
>>>   
 Assuming you're using POTS, you probably won't have much luck with
this.
 
> If
>   
 you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and
 
> asterisk
>   
 won't process the line until you pick up and punch a dtmf key.  If you
 
> are
>   
 using E1 or PRI, there is more hope for you.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nitesh
 
> Divecha
>   
 Sent: Thursday, October 01, 2009 4:42 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] TDM410P - False Answer Supervision

 Hello All,

 Can anyone help me with False Answer Supervision problem with TDM410P
 card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
 everything works fine except the Answer supervision...

 When the call hits Asterisk it sends the call to one of the TDM410 card
 and the call is answered immediately while the call is still in
 progress... Here is the debug output: -

 [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2291 dahdi_call: Dialing
 '3602045'
 [Oct  2 09:39:17] DEBUG[867]: chan_dahdi.c:2369 dahdi_call: Deferring
 dialing...
 -- Called G2/3602045
 [Oct  2 09:39:18] DEBUG[867]: chan_dahdi.c:4874 dahdi_handle_event:
Sent
 deferred digit string: T3602045w
 [Oct  2 09:39:20] DEBUG[867]: chan_dahdi.c:4209 dahdi_handle_event:
Done
 dialing, but waiting for progress detection before doing more...
 -- DAHDI/8-1 answered SIP/9223421808-091b3f50
 -- Hungup 'DAHDI/8-1'
 =

 The connect message is sent back immediately when " DAHDI/8-1 answered
 SIP/9223421808-091b3f50" while the call is still in progress... If the
 call is hang up without answer the sender gets Normal Code 16 while it
 suppose to be "Abandoned Call".



 The Polarity Reversal only works when call is ANSWERED... Here is the
 debug log: -

 [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2291 dahdi_call: Dialing
 '3312808'
 [Oct  2 09:20:05] DEBUG[693]: chan_dahdi.c:2369 dahdi_call: Deferring
 dialing...
 -- Called G2/3312808
 [Oct  2 09:20:06] DEBUG[693]: chan_dahdi.c:4874 dahdi_handle_event:
Sent
 deferred digit string: T3312808w
 [Oct  2 09:20:08] DEBUG[693]: chan_dahdi.c:4209 dahdi_handle_event:
Done
 dialing, but waiting for progress detection before doing more...
 -- DAHDI/8-1 answered SIP/9765782184-091b9678
 [Oct  2 09:20:14] DEBUG[693]: chan_dahdi.c:4911 dahdi_handle_event:
 Ignore switch to REVERSED Polar

Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[Danny Nicholas]

This is where I found this
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf#SIPConfiguration
general

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, October 02, 2009 3:26 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

I can't find any reference for this - is there a doc / link? 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, October 02, 2009 3:59 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Sipregisterattempts would seem to be the simplest way to do this.  It is 0
by default, changing it to 5 would stop the hacker after 5 tries.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, October 02, 2009 2:24 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Good post.  One of the recommendations is to limit the number of calls per
sip entity.  Is there an easy way to do that in sip.conf? 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, October 02, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Couple of old posts:
http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html


On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis  wrote:
> Has anyone written an app that monitors SIP/IAX registration attempts?  
> A couple of clients are being flooded with SIP registrations (but the 
> source IP changes every few hours so IPtables won't do)..
>
> I would think that any attempt to reg 5 times with a bad password 
> should cause a 5 minute timeout until reg is considered again.  Has 
> anyone written such an app?  The name app_hackblock is my contribution 
> to the project :)
>
> MD
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: 
> http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] Restarting of B-channel on span 1

[Chris Miller]

Darrin Henshaw wrote:
> add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
> depending on what you are running. zaptel or dahdi.

Can someone confirm when the default was changed from "never" to
3600 seconds? According to the voip-info wiki, "never" has always
been the default. I would tend to agree, because I've never seen
this behavior on customer systems until recently.

http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf

The current configs/chan_dahdi.conf.sample says the default is 3600.
I don't see anything in the 1.4 changelog about this change.

I ran across this troubleshooting a PRI issue, and was concerned
that this frequent resetting was related to the customer issues.
What happens when a call comes in when a reset is in progress? If
this condition can't be handled gracefully (i.e. without failing a
call), then I would argue the default is not conservative enough.
Just want to know the right way to handle this.

Chris


> On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhally wrote:
>> Hi All,
>>
>> Hope you all are fine and good, Today i have found that Mine all PRI
>> Channels are restating after every interval of one hour, and i have search
>> and psot on
>> fourms and everyone said that this is a normal behaviour.
>> If this is a normal behaviour is there is any way to stop it { i still don't
>> know what is the reson to restart ever hour } . Because this is listed
>> everywhere that this is a normal behaviour, but not one mention {may be i am
>> not able to find it is listed some where} why this is nesessary? and if this
>> is not nessary how to stop it...
>> I think we all already know the message , but posting it for future
>> reference..
>>
>> Thanks a lot .
>> Aman Dhally
>>
>> --
>> ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
>> [Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
>> [Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection
>> disconnected
>> [Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
>> restarted on span 1
>> [Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
>> restarted on span 1
>> [Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
>> restarted on span 1
>> [Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
>> restarted on span 1
>> [Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
>> restarted on span 1
>> [Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
>> restarted on span 1
>> [Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
>> restarted on span 1
>> [Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
>> restarted on span 1
>> [Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
>> restarted on span 1
>> [Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
>> restarted on span 1
>> [Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
>> restarted on span 1
>> [Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
>> restarted on span 1
>> [Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
>> restarted on span 1
>> [Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
>> restarted on span 1
>> [Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
>> restarted on span 1
>> [Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
>> restarted on span 1
>> [Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
>> restarted on span 1
>> [Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
>> restarted on span 1
>> [Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
>> restarted on span 1
>> [Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
>> restarted on span 1
>> [Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
>> restarted on span 1
>> [Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
>> restarted on span 1
>> [Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
>> restarted on span 1
>> [Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
>> restarted on span 1
>> [Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
>> restarted on span 1
>> [Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
>> restarted on span 1
>> [Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
>> restarted on span 1
>> [Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
>> restarted on span 1
>> [Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
>> restarted on span 1
>> [Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
>> re

Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[John A. Sullivan III]

Is that what that does? I assumed that was like a protocol retry.  In
other words, if the registrar does not reply to the registry when it
submits its credentials, it will resubmit them registerattempts number
of times.  I did not think that prevented a registree from submitting
10,000 new sets of credentials.  But that was only my guess - John

On Fri, 2009-10-02 at 14:58 -0500, Danny Nicholas wrote:
> Sipregisterattempts would seem to be the simplest way to do this.  It is 0
> by default, changing it to 5 would stop the hacker after 5 tries.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
> Dupuis
> Sent: Friday, October 02, 2009 2:24 PM
> To: 'Asterisk Users List'
> Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling
> 
> Good post.  One of the recommendations is to limit the number of calls per
> sip entity.  Is there an easy way to do that in sip.conf? 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
> Sent: Friday, October 02, 2009 3:01 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling
> 
> Couple of old posts:
> http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
> http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
> http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html
> 
> 
> On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis  wrote:
> > Has anyone written an app that monitors SIP/IAX registration attempts?  
> > A couple of clients are being flooded with SIP registrations (but the 
> > source IP changes every few hours so IPtables won't do)..
> >
> > I would think that any attempt to reg 5 times with a bad password 
> > should cause a 5 minute timeout until reg is considered again.  Has 
> > anyone written such an app?  The name app_hackblock is my contribution 
> > to the project :)
> >
> > MD
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: 
> > http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now:
> http://www.astricon.net
> 
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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> 
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-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

[Danny Nicholas]

Unfortunately I don't really know since I use POTS for all of my external
traffic.  Maybe Tzafir or another guru can shed more light...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Friday, October 02, 2009 3:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling

Is that what that does? I assumed that was like a protocol retry.  In
other words, if the registrar does not reply to the registry when it
submits its credentials, it will resubmit them registerattempts number
of times.  I did not think that prevented a registree from submitting
10,000 new sets of credentials.  But that was only my guess - John

On Fri, 2009-10-02 at 14:58 -0500, Danny Nicholas wrote:
> Sipregisterattempts would seem to be the simplest way to do this.  It is 0
> by default, changing it to 5 would stop the hacker after 5 tries.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
> Dupuis
> Sent: Friday, October 02, 2009 2:24 PM
> To: 'Asterisk Users List'
> Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg
trolling
> 
> Good post.  One of the recommendations is to limit the number of calls per
> sip entity.  Is there an easy way to do that in sip.conf? 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
> Sent: Friday, October 02, 2009 3:01 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg
trolling
> 
> Couple of old posts:
> http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
> http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
> http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html
> 
> 
> On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis  wrote:
> > Has anyone written an app that monitors SIP/IAX registration attempts?  
> > A couple of clients are being flooded with SIP registrations (but the 
> > source IP changes every few hours so IPtables won't do)..
> >
> > I would think that any attempt to reg 5 times with a bad password 
> > should cause a 5 minute timeout until reg is considered again.  Has 
> > anyone written such an app?  The name app_hackblock is my contribution 
> > to the project :)
> >
> > MD
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: 
> > http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now:
> http://www.astricon.net
> 
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Re: [asterisk-users] help on ${RTPAUDIOQOS}

[Klaus Darilion]

Do you have canreinvite=no in sip.conf? Maybe the variable is only set 
if Asterisk is actually relaying RTP too.

regards
klaus

Asterisk User wrote:
> Hi All,
> 
> While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use 
> it in my dialplan.
> I had 2 sip extensions 555 and 666 and I called from 555 to 666, but 
> unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI.
> 
> Would you please let me know what is wrong with my dialplan and/or what 
> else should be done to get the value of ${RTPAUDIOQOS}?
> 
> Following is my dialplan context where my call landed
> 
> [incoming_vpbx]
> exten => _x.,1,NoOp(A call has come)
> exten => _x.,n,Noop(${RTPAUDIOQOS})
> 
> exten => _x.,n,Dial(SIP/666,30,m)
> exten => _x.,n,Hangup()
> exten => h,1,Noop(***${RTPAUDIOQOS})
> 
> 
> And here is what appeared on CLI...
> -- Executing [...@incoming_vpbx:1] NoOp("SIP/555-b7a80948", "A call 
> has come") in new stack
> -- Executing [...@incoming_vpbx:2] NoOp("SIP/555-b7a80948", 
> "") in new stack
> -- Executing [...@incoming_vpbx:3] Dial("SIP/555-b7a80948", 
> "SIP/666,30,m") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called 666
> -- Started music on hold, class 'default', on SIP/555-b7a80948
> -- SIP/666-089cb090 is ringing
> -- SIP/666-089cb090 answered SIP/555-b7a80948
> -- Stopped music on hold on SIP/555-b7a80948
> -- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090
> -- Executing [...@incoming_vpbx:1] NoOp("SIP/555-b7a80948", 
> "***") in new stack
> 
> 
> Thanking you...
> 
> ---Asterisk User
> 
> 
> 
> 
> 
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[asterisk-users] Asterisk,click2talk, webphone

[Doddle WebPhone]

Hi,This can be useful for Asterisk / TI integrators:

How to create a Free click2call application using Asterisk:

We can build click2talk / webphone application empowering webpages with VoIP
Telephony  using online DoddlePhone and Asterisk

Invoke doddle webphone (http://www.doddlephone.com) as follows:
sipserver=Asterisk_SERVER&username=USER&password=PASSWORD&callto=PHONE_NUMBER_TO_CALL


Just create  sip account on Asterisk, define its route and trigger
click2call as above
Notice that we can set a fixed route / context to the click2talk sip peer.


check out http://www.doddlephone.com for details

Regards
Sergio
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[asterisk-users] Problem sending a DTMF remotely. Please need help...

[Pablo Bernasconi]

Hello,

I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.

isb177*CLI> features show
Builtin Feature   Default Current
---   --- ---
Pickup*8  *8
Blind Transfer#   #8
Attended Transfer *2
One Touch Monitor *1
Disconnect Call   *   **
Park Call
One Touch MixMonitor

Dynamic Feature   Default Current
---   --- ---
(none)

Call parking

Parking extension   :  70
Parking context :  parkedcalls
Parked call extensions:  71-78


My script is:

#!/usr/bin/php -q



The script output is:

Array
(
[0] => Asterisk Call Manager/1.1
[1] => Response: Success
[2] => Message: Authentication accepted
[3] =>
[4] => Response: Success
[5] => Message: DTMF successfully queued
[6] =>
[7] => Response: Success
[8] => Message: DTMF successfully queued
[9] =>
[10] => Response: Goodbye
[11] => Message: Thanks for all the fish.
[12] =>
[13] =>
)


When I run the script I can hear the two digit (only the audio) but nothing
happens, the Transfer menu doesnt start. The Cli shows:


[Oct  2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'login'
  == Manager 'admin' logged on from 127.0.0.1
[Oct  2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'PlayDTMF'
[Oct  2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread
-1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread
-1217414256 in procedure ast_waitfor_nandfds
[Oct  2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided
for write to channel 'SIP/1000-0a292360'
[Oct  2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'PlayDTMF'
[Oct  2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread
-1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread
-1217414256 in procedure ast_waitfor_nandfds
[Oct  2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided
for write to channel 'SIP/1000-0a292360'
[Oct  2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'Logoff'
  == Manager 'admin' logged off from 127.0.0.1



BUT, if I press #8 in the softphone, I can hear the two digit and
inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli
output in this case is:


[Oct  2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of
60 bytes
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#),
at 192.168.0.148
[Oct  2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#'
received on SIP/1000-0a292360
[Oct  2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin
passthrough '#' on SIP/1000-0a292360
[Oct  2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF
begin on channel (SIP/1000-0a292360)
[Oct  2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge
stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#),
at 192.168.0.148
[Oct  2 11:09:20] DTMF[29533]: channel.c:2768 __ast_read: DTMF end '#'
received on SIP/1000-0a292360, duration 80 ms
[Oct  2 11:09:20] DTMF[29533]: channel.c:2808 __ast_read: DTMF end accepted
with begin '#' on SIP/1000-0a292360
[Oct  2 11:09:20] DTMF[29533]: channel.c:2824 __ast_read: DTMF end
passthrough '#' on SIP/1000-0a292360
[Oct  2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF
end on channel (SIP/1000-0a292360)
[Oct  2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge
stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408
[Oct  2 11:09:20] DEBUG[29533]: features.c:1836 ast_feature_interpret:
Feature interpret: chan=SIP/1000-0a292360, peer=SIP/1001-0a026408, code=#,
sense=1, features=2, dynamic=#
[Oct  2 11:09:20] DEBUG[29533]: features.c:2496 ast_bridge_call: Set time
limit to 2000
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct  2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
0008 (len = 4)
[Oct  2 11:09:21] DEBUG[29533]: rtp.c:806 send_dtmf:

Re: [asterisk-users] help on ${RTPAUDIOQOS}

[Asterisk User]

Klaus,

Yes I do have set canreinvite=no in sip.conf.
One more thing I noticed is following two cases when I replaced  exten =>
_x.,n,Dial(SIP/666,30,m) with .exten => _x.,n,Dial(SIP/666,30,me)

(1) When called extension(666) receives and hangs up the call.

 -- Executing [...@incoming_vpbx:1] NoOp("SIP/555-b7918e68", "A call has
come") in new stack
-- Executing [...@incoming_vpbx:2] NoOp("SIP/555-b7918e68",
"") in new stack
-- Executing [...@incoming_vpbx:3] Dial("SIP/555-b7918e68",
"SIP/666,30,me") in new stack
  == Using SIP RTP CoS mark 5
-- Called 666
-- Started music on hold, class 'default', on SIP/555-b7918e68
-- SIP/666-09830108 is ringing
-- SIP/666-09830108 answered SIP/555-b7918e68
-- Stopped music on hold on SIP/555-b7918e68
-- Packet2Packet bridging SIP/555-b7918e68 and SIP/666-09830108
-- Executing [...@incoming_vpbx:1] NoOp("SIP/555-b7918e68",
"**") in new stack
-- Executing [...@incoming_vpbx:1] NoOp("SIP/666-09830108",
"**ssrc=1245221053;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00")
in new stack
  == Spawn extension (incoming_vpbx, 12, 3) exited non-zero on
'SIP/555-b7918e68'


(2)When called extension(666) receives and caller extension(555) hangs up
the call.

-- Executing [...@incoming_vpbx:1] NoOp("SIP/555-b7918e68", "A call has
come") in new stack
-- Executing [...@incoming_vpbx:2] NoOp("SIP/555-b7918e68",
"") in new stack
-- Executing [...@incoming_vpbx:3] Dial("SIP/555-b7918e68",
"SIP/666,30,me") in new stack
  == Using SIP RTP CoS mark 5
-- Called 666:00*CLI>
-- Started music on hold, class 'default', on SIP/555-b7918e68
-- SIP/666-09830108 is ringing
-- SIP/666-09830108 answered SIP/555-b7918e68
-- Stopped music on hold on SIP/555-b7918e68
-- Packet2Packet bridging SIP/555-b7918e68 and SIP/666-09830108
-- Started music on hold, class 'default', on SIP/555-b7918e68
-- Stopped music on hold on SIP/555-b7918e68
-- Executing [...@incoming_vpbx:1] NoOp("SIP/555-b7918e68",
"**ssrc=1405826681;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=101;rlp=0;rtt=0.00")
in new stack
-- Executing [...@incoming_vpbx:1] NoOp("SIP/666-09830108",
"**") in new stack
  == Spawn extension (incoming_vpbx, 12, 3) exited non-zero on
'SIP/555-b7918e68'



So it looks like it has something to do with the way a call is hungup.
Has anybody else any idea?

Thanks,

---Asterisk User
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Re: [asterisk-users] Music On Hold

[Cyprus VoIP]

 > What does your musiconhold.conf look like?
 >


[general]

[default]
mode=files
directory=/var/lib/asterisk/moh


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