Re: [asterisk-users] a2billing

[bruce bruce]

You might have to tamper the main a2billing.php or more files for that
feature to work. Or it might cost around $800 in development time.

On Tue, Oct 19, 2010 at 4:34 AM, Baha @ SH  wrote:

> Exactly,
>
> I don’t want that, it’s annoying! I just want it to run if the customer
> balance reach for example < 1 dollar!
>
> Anyway?
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Monday, October 18, 2010 8:17 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] a2billing
>
>
>
> I don't think voucher can be triggered to announce at certain threshold
> ONLY but it will be run everytime at the begining after PIN is asked for. By
> default it's set to: "Press 8 to fill up with a voucher".
>
>
>
> System Settings is the last in the menu.
>
>
>
> -Bruce
>
> On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH  wrote:
>
> I am sorry , but where is System Settings??? And what is the parameter
> name?
>
> And also, id like to mention that the voucher is working, only when balance
> is below minimum balance it does not go to voucher ivr.
>
>
>
> Thanks, awaiting,,,
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Monday, October 18, 2010 12:46 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] a2billing
>
>
>
> Turn on the voucher feature in System Settings and it will tell the user
> right after the PIN authentication or CLID authentication that their balance
> is below threshold and they should pay.
>
>
>
> -Bruce
>
> On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH  wrote:
>
> Not sure if a2billing can be shared here, but ill give a shot
>
> If the credit < min_credit the IVR play: sorry you have 0 credit and
> hangup,
> I want it to FW me to the IVR to add voucher, please let me know: here is
> log:
>
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:76]:[CallerID:]:[CN:]:[MODE
> : standard]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
> SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
> activated, inuse, simultaccess, typepaid, creditlimit, language,
> removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
> expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
> UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
> cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
> cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
> tariff=cc_tariffgroup.id WHERE username='9971524976']
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
> active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
> language=en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
> prepaid-zero-balance (cardnumber:9971524976)]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
> callingcard_ivr_authenticate]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
> (cia_res:-2)]]
> [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]
>
>
>
>
>
> --
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Re: [asterisk-users] a2billing

[bruce bruce]

I don't think voucher can be triggered to announce at certain threshold ONLY
but it will be run everytime at the begining after PIN is asked for. By
default it's set to: "Press 8 to fill up with a voucher".

System Settings is the last in the menu.

-Bruce

On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH  wrote:

> I am sorry , but where is System Settings??? And what is the parameter
> name?
>
> And also, id like to mention that the voucher is working, only when balance
> is below minimum balance it does not go to voucher ivr.
>
>
>
> Thanks, awaiting,,,
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Monday, October 18, 2010 12:46 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] a2billing
>
>
>
> Turn on the voucher feature in System Settings and it will tell the user
> right after the PIN authentication or CLID authentication that their balance
> is below threshold and they should pay.
>
>
>
> -Bruce
>
> On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH  wrote:
>
> Not sure if a2billing can be shared here, but ill give a shot
>
> If the credit < min_credit the IVR play: sorry you have 0 credit and
> hangup,
> I want it to FW me to the IVR to add voucher, please let me know: here is
> log:
>
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:76]:[CallerID:]:[CN:]:[MODE
> : standard]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
> SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
> activated, inuse, simultaccess, typepaid, creditlimit, language,
> removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
> expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
> UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
> cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
> cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
> tariff=cc_tariffgroup.id WHERE username='9971524976']
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
> active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
> language=en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
> prepaid-zero-balance (cardnumber:9971524976)]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
> callingcard_ivr_authenticate]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
> (cia_res:-2)]]
> [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]
>
>
>
>
>
> --
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Re: [asterisk-users] a2billing

[bruce bruce]

Turn on the voucher feature in System Settings and it will tell the user
right after the PIN authentication or CLID authentication that their balance
is below threshold and they should pay.

-Bruce

On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH  wrote:

> Not sure if a2billing can be shared here, but ill give a shot
>
> If the credit < min_credit the IVR play: sorry you have 0 credit and
> hangup,
> I want it to FW me to the IVR to add voucher, please let me know: here is
> log:
>
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:76]:[CallerID:]:[CN:]:[MODE
> : standard]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ;
> SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00]
> [18/10/2010 07:01:12]:[file:a2billing.php -
> line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff,
> activated, inuse, simultaccess, typepaid, creditlimit, language,
> removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate),
> expiredays, nbused, UNIX_TIMESTAMP(firstusedate),
> UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname,
> cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign,
> cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON
> tariff=cc_tariffgroup.id WHERE username='9971524976']
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 ::
> active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 ::
> language=en]]
> [18/10/2010 07:01:12]:[file:Class.A2Billing.php -
> line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD :
> prepaid-zero-balance (cardnumber:9971524976)]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY :
> callingcard_ivr_authenticate]]
> [18/10/2010 07:01:14]:[file:a2billing.php -
> line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED
> (cia_res:-2)]]
> [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]]
>
>
>
>
>
> --
> _
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[asterisk-users] How to check if Agent is logged into a specific Queue using dial-plan?

[bruce bruce]

Hi,

I have this on an Aastra phone:

Button 1:Login English Queue
Button 2:Login French Queue
Button 3:Logout both English and French


I am out of buttons and using only three buttons I want my third button to
be smarter. Currently the third button does a QueueRemoveMember to both
English and French Queue at the same time. I want this button to be smarter
to and to check and see if the Agent is logged into only English to only do
a Remove on English or if the Agent is only logged into French to only log
out French. Same goes for both, if both are logged in to log out both.

Currently I have this for Third Button:

exten => 99,1,Answer
exten =>
99,n,RemoveQueueMember(900|Local/${CALLERID(num)}...@from-internal/n)
exten =>
99,n,RemoveQueueMember(899|Local/${CALLERID(num)}...@from-internal/n)
exten => 99,n,Hangup

900 is English and 800 is Spanish Queue numbers.

P.S. Is there a way to do exten = s rather than exten = 99 as someone
from outside might find out about the 99 extension and try to log into
it? This is for an Aastra phone with XML support.

Thanks
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Re: [asterisk-users] fraud advice

[bruce bruce]

Jeff,

I suggest talking to your PSTN/VoIP provider. We had a large amount going
through TATA communications and have not accepted their word for payment
because they had a duty to not allow traffic if our credit went down to $1k
while the calls charged were actually more than that.

Unfortunately, probably there is no one you can complain to. But it also
sickens me at how badly Asterisk is made to not cope with situations like
this and worse than that is FreePBX.

I suggest checking your contract terms with your provider as they might have
some sort of restrictions. At the very least PSTN providers try to bring the
price per minute lowered to their buy rate which is usually less than half
of the original bill.

Regards,
Bruce

On Thu, Oct 14, 2010 at 9:10 PM, Jeff LaCoursiere  wrote:

>
> Hi,
>
> Embarrassed as I am to write this, I am hoping for some advice.  One of
> our very first PBX installs, now six years old, was "taken advantage of"
> over the past few weeks.  A victim of sipvicious, I assume, that managed
> to guess one of the SIP passwords.  4000 calls to various middle eastern
> destinations have been placed, which ended up being sent over our
> customer's PSTN trunk, and of course there was no warning until the bill
> came today.  Unfortunately the bill only covered the first few days of
> this fiasco, and was only $700.  I am afraid the one that is on the way
> will be tens of thousands.  ONE CALL on the bill that just arrived was
> $200 (80 minutes to Sierra Leone).
>
> I'm sure this started out as a single scan.  It must have been posted,
> because I have at least ten IP addresses now that were placing calls via
> the same peer.  They are from all over the world.
>
> So what is the accepted procedure?  I'm in the US Virgin Islands, so do I
> go to the FBI?  Police?  Is their some telecom fraud body to report such
> things to?  Does any one ever get any relief from such events?
>
> I'm basically sick to my stomach right now.
>
> j
>
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Re: [asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets

[bruce bruce]

And that is exactly what is done on the device: Nat=yes but Asterisk still
sees the SIP packet coming in to register with a local IP an so it responds
to a local IP which doesn't even exist on the Asterisk network. This is what
frustrates me that it's not so straight forward to Asterisk to obtain the
proper public IP of the device from the IP packet headers rather than the
SIP packets.

Thanks

On Sat, Oct 9, 2010 at 8:27 AM, Kevin P. Fleming wrote:

> On 10/08/2010 10:16 PM, bruce bruce wrote:
>
> > I said previously, Asterisk receives packets like extens...@192.168.0.10
> > <mailto:extens...@192.168.0.10> is trying to register to it. So,
> > Asterisk sends out to local LAN an ACK which obviously is not right.
> > SPA-2102 should send SIP request like extens...@123.123.123.123
> > <mailto:extens...@123.123.123.123> (public IP).
>
> If you set 'nat=yes' in the sip.conf peer entry for that device in
> Asterisk, Asterisk will reply to the IP address and port number the
> REGISTER request was received from, not the address in the Contact
> header provided in the request itself. It will also record this address
> and port number as the location of that peer for future INVITE messages
> to be sent to it.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
> _
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Re: [asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets

[bruce bruce]

Thanks for the feedback.

I said previously, Asterisk receives packets like extens...@192.168.0.10 is
trying to register to it. So, Asterisk sends out to local LAN an ACK which
obviously is not right. SPA-2102 should send SIP request like
extens...@123.123.123.123 (public IP).

Thanks

On Fri, Oct 8, 2010 at 3:32 PM, Kevin P. Fleming wrote:

> On 10/06/2010 02:50 PM, bruce bruce wrote:
> > Hi Guys,
> >
> > This is such an annoying issue whenever it comes up. The sender and
> > receive always receive the source public IP no matter what in the IP
> > packets but then SIP packets go out with something like 192.168.0.20.
> >
> > In this instance, a Bell Canada DSL modem is installed and I saw the
> > SPA-2102 register properly but only to fail later on due to sending it's
> > LAN IP to the Asterisk server.
> >
> > So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also
> > put the device on DMZ in the Bell Canada DSL modem and still the same
> issue.
> >
> > I am wondering when would manufacturers finally fully comply the SIP
> RFC?!
>
> Exactly how is this behavior non-compliant with "the" (sic) SIP RFC?
> There is nothing in any SIP RFC that mandates that a SIP UA must be
> aware of multiple IP addresses over which it can be reached, and select
> the proper one to include in SIP requests and responses.
>
> In fact, many SIP UAs, Asterisk included, work just fine behind NAT
> devices without ever knowing what their external IP addresses are.
>
> If you had actually described how the device failed, we might be able to
> tell you what you could do to resolve the problem. In general, Asterisk
> works just fine with endpoints that are behind NAT devices and never
> send their external IP addresses in their SIP messages... there are
> probably millions of devices working that way every day.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets

[bruce bruce]

Kyle,

Got an empty response from you. Were you intending to give your feedback?

Regards,
Bruce

On Wed, Oct 6, 2010 at 8:10 PM, Kyle Kienapfel wrote:

>
>
> On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce  wrote:
>
>> Hi Guys,
>>
>> This is such an annoying issue whenever it comes up. The sender and
>> receive always receive the source public IP no matter what in the IP packets
>> but then SIP packets go out with something like 192.168.0.20.
>>
>> In this instance, a Bell Canada DSL modem is installed and I saw the
>> SPA-2102 register properly but only to fail later on due to sending it's LAN
>> IP to the Asterisk server.
>>
>> So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also
>> put the device on DMZ in the Bell Canada DSL modem and still the same issue.
>>
>> I am wondering when would manufacturers finally fully comply the SIP RFC?!
>>
>> I don't have the luxury to put SIP proxy, do a VPN, install a server or
>> anything on client site.
>>
>> Diagram:
>>
>> Asterisk Server <= Internet = Bell Canada Modem => SPA2102
>>
>> Please send me your suggestions on how to fix this if you have this type
>> of experience with SPA-2102
>>
>> Thanks for the input,
>> Bruce
>>
>>
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>
> Are you using stun?
> http://en.wikipedia.org/wiki/Session_Traversal_Utilities_for_NAT
>
>
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

Glad to hear it helped you Dennison.

VPN is such a confusing beast to lots of people I think and hence the
responses to this thread were all sort of work around and sometimes
off-topic. It's also not well documented or maybe the feature is not widely
used within the Asterisk community. I think it would be very good if some
standard guidelines become available from the Asterisk side on this.

Good day,
Bruce


On Wed, Oct 6, 2010 at 7:33 PM, Dennison Williams <
dennison.willi...@gmail.com> wrote:

> On 09/22/2010 08:36 AM, Carlos Chavez wrote:
> > Do you have a localnet statement in your sip.conf?  That and using
> > nat=no will make sure Asterisk does not replace the IP address in the
> > Invite.
> >
>
> I just wanted to give a +1 for this response.  I am using openvpn to
> connect road warriors and remote offices to a central asterisk server.
> When setting up the configuration for the road warriors I created a new
> subnet for them, but forgot to include their subnet as a localnet
> directive in sip.conf.  The result was that sip clients on the road
> warrior network would be able to register, but then when initiating a
> sip call the 200 response (to the INVITE from the client) from the
> asterisk server would include a contact address for some external ip
> that I did not recognize.  This hint here allowed me to fix this bug,
> now calls from the road warrior subnet are coming in fine.  Thanks!
>
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[asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets

[bruce bruce]

Hi Guys,

This is such an annoying issue whenever it comes up. The sender and receive
always receive the source public IP no matter what in the IP packets but
then SIP packets go out with something like 192.168.0.20.

In this instance, a Bell Canada DSL modem is installed and I saw the
SPA-2102 register properly but only to fail later on due to sending it's LAN
IP to the Asterisk server.

So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put
the device on DMZ in the Bell Canada DSL modem and still the same issue.

I am wondering when would manufacturers finally fully comply the SIP RFC?!

I don't have the luxury to put SIP proxy, do a VPN, install a server or
anything on client site.

Diagram:

Asterisk Server <= Internet = Bell Canada Modem => SPA2102

Please send me your suggestions on how to fix this if you have this type of
experience with SPA-2102

Thanks for the input,
Bruce
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Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

[bruce bruce]

Thanks for the input guys.

So, the IP is resolved only when IPTABLES is loaded or reloaded. Therefore,
the best approach would be to ping the hostname every let's say 3 seconds
and see if the IP is still the same and if it is then move on, otherwise
update the iptables with the new IP address. This sounds it would work but I
am not sure how fast DynDns can resolve the IP for me (delay) and I am
looking to connect 40 PAP2T to this system. So, all in all that is 40
queries to DynDNS each 3 seconds.

As I mentioned earlier, wouldn't it be more solid if I run my own Dynamic
DNS server on the same box as Asterisk (is that even possible?) and what
sort of other security holes would I be exposing doing that?

Thanks again for all the great input.

-Bruce

On Sun, Oct 3, 2010 at 8:01 AM, Steve Edwards wrote:

> On Sat, 2 Oct 2010, Kyle Kienapfel wrote:
>
> > You're not going to be able to put a dns hostname in the iptables, but
> > you could have a script that runs at times and gets the ip address for
> > your dynamic hostname and allows that.
>
> Almost.
>
> You can put a host name in iptables, but it is resolved when loaded.
>
> You could restart iptables when your dynamic host name changes and it will
> be resolved correctly with the new IP address.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

[bruce bruce]

Thanks Roger.

I will be trying this box to see what I can do. Otherwise, I'd probably have
to find a list of all of the Rogers (The ISP providing internet to these
boxes) IPs to at least limit the attacks to Rogers ISP.

hmmm


Or maybe secure is using DNS like this:
 sdlfjds...@$523k4j98sd7fkjh324#@$832.dyndns.org

isn't that a security feature in itself?

Thanks



On Sat, Oct 2, 2010 at 4:32 PM, Roger Burton West wrote:

> On Sat, Oct 02, 2010 at 04:09:33PM -0400, bruce bruce wrote:
> >Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind
> to
> >the PAP2T? do you think the devices comes in with it's external IP rather
> >than the dyndns domain?
>
> Yes. An IP datagram carries only the source and destination IP
> addresses, not the DNS names associated with them. Your firewall _may_
> be able to accept a DNS name to block or allow rather than an IP
> address, but most don't, and doing so makes you vulnerable to DNS
> spoofing attacks.
>
> To go further would be thoroughly off-topic for this list.
>
> Roger
>
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Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

[bruce bruce]

Yeah, you are missing all :-)

Sorry, read the thread again.

On Sat, Oct 2, 2010 at 5:05 PM, sean darcy  wrote:

> On 10/02/2010 04:09 PM, bruce bruce wrote:
> > Can't I in my ip tables just accept the pap2t.dyndns.org
> > <http://pap2t.dyndns.org> if that is bind to the PAP2T? do you think the
> > devices comes in with it's external IP rather than the dyndns domain?
> >
> > Thanks
> >
> > On Sat, Oct 2, 2010 at 3:43 PM, bruce bruce  > <mailto:bruceb...@gmail.com>> wrote:
> >
> > I was confusing the asterisk server side of sip_nat with the PAP2T.
> > So, PAP2T can only register to DynDNS and that's all.
> >
> > What sort of a script would I be looking for? something to query
> > DynDNS for the new IP of the device to add to firewall? This might
> > however bring down time if inquiry is not successful.
> >
> > Or can I setup my own Dyndns server on the Asterisk server and have
> > those PAP2T units registered to it and then work it from there when
> > their IPs change?
> >
> >     Thanks
> >
> > On Sat, Oct 2, 2010 at 3:32 PM, jon pounder  > <mailto:j...@inline.net>> wrote:
> >
> > On 10/02/2010 03:31 PM, bruce bruce wrote:
> >> Hi,
> >>
> >> Can you please explain the DynDNS part. How would I put that
> >> in my Asterisk server as an identified party? Usually it comes
> >> to me with IP address (dynamic). Or do add something like this
> >> in sip_nat.conf:
> >>
> >> externip=mybox.dyndns.org <http://mybox.dyndns.org>
> >> localnet=192.168.0.0/255.255.255.0
> >> <http://192.168.0.0/255.255.255.0>
> >
> > every time the address changes you have to have some script to
> > make the change in your firewall.
> >
> >>
> >> ???
> >>
> >> Thansk again,
> >>
> >> On Sat, Oct 2, 2010 at 2:59 PM, jon pounder  >> <mailto:j...@inline.net>> wrote:
> >>
> >> On 10/02/2010 02:56 PM, bruce bruce wrote:
> >> > Hi Everyone
> >> >
> >> > I think PAP2T supports DynDNS and other Dynamic DNS
> >> providers. I have
> >> > a box that needs to be secured at all times. Currently
> >> it's not
> >> > connected to the internet. If it were connected, I would
> >> have iptables
> >> > block any and all traffic from outside but I want a
> >> single device -
> >> > Linksys PAP2T - to be able to connect back to the
> >> server. That is a
> >> > stand alone device and doesn't support VPN and I don't
> >> have the luxury
> >> > of putting a VPN client on the PAP2T side to connect
> >> back to the
> >> > server. Is there any way I can DynDNS on the PAP2T to
> >> somehow notify
> >> > the Asterisk Server that it's a safe device coming in?
> >> >
> >> > I do use fail2ban but that is not what I am looking for
> >> at this
> >> > moment. And since the IP is dynamic on the PAP2T, I
> >> can't just use the
> >> > iptables to let it in as it might change all a sudden.
> >> >
> >> > Thanks
> >> do the dyndns on whatever router is in front of the pap2t
> >> or
> >> get some other box that supports it.
> >>
> >>
> >> other than that you are looking for some sort of magic
> bullet
> >>
> >> --
> >>
> _
> >> -- Bandwidth and Colocation Provided by
> >> http://www.api-digital.com --
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> >> every Thurs:
> >> http://www.asterisk.org/hello
> >>
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> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
>

Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

[bruce bruce]

Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind to
the PAP2T? do you think the devices comes in with it's external IP rather
than the dyndns domain?

Thanks

On Sat, Oct 2, 2010 at 3:43 PM, bruce bruce  wrote:

> I was confusing the asterisk server side of sip_nat with the PAP2T. So,
> PAP2T can only register to DynDNS and that's all.
>
> What sort of a script would I be looking for? something to query DynDNS for
> the new IP of the device to add to firewall? This might however bring down
> time if inquiry is not successful.
>
> Or can I setup my own Dyndns server on the Asterisk server and have those
> PAP2T units registered to it and then work it from there when their IPs
> change?
>
> Thanks
>
> On Sat, Oct 2, 2010 at 3:32 PM, jon pounder  wrote:
>
>>  On 10/02/2010 03:31 PM, bruce bruce wrote:
>>
>> Hi,
>>
>>  Can you please explain the DynDNS part. How would I put that in my
>> Asterisk server as an identified party? Usually it comes to me with IP
>> address (dynamic). Or do add something like this in sip_nat.conf:
>>
>>  externip=mybox.dyndns.org
>> localnet=192.168.0.0/255.255.255.0
>>
>>
>> every time the address changes you have to have some script to make the
>> change in your firewall.
>>
>>
>>  ???
>>
>>  Thansk again,
>>
>> On Sat, Oct 2, 2010 at 2:59 PM, jon pounder  wrote:
>>
>>>  On 10/02/2010 02:56 PM, bruce bruce wrote:
>>> > Hi Everyone
>>> >
>>> > I think PAP2T supports DynDNS and other Dynamic DNS providers. I have
>>> > a box that needs to be secured at all times. Currently it's not
>>> > connected to the internet. If it were connected, I would have iptables
>>> > block any and all traffic from outside but I want a single device -
>>> > Linksys PAP2T - to be able to connect back to the server. That is a
>>> > stand alone device and doesn't support VPN and I don't have the luxury
>>> > of putting a VPN client on the PAP2T side to connect back to the
>>> > server. Is there any way I can DynDNS on the PAP2T to somehow notify
>>> > the Asterisk Server that it's a safe device coming in?
>>> >
>>> > I do use fail2ban but that is not what I am looking for at this
>>> > moment. And since the IP is dynamic on the PAP2T, I can't just use the
>>> > iptables to let it in as it might change all a sudden.
>>> >
>>> > Thanks
>>>  do the dyndns on whatever router is in front of the pap2t
>>> or
>>> get some other box that supports it.
>>>
>>>
>>> other than that you are looking for some sort of magic bullet
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
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>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> _
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>
>
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Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

[bruce bruce]

I was confusing the asterisk server side of sip_nat with the PAP2T. So,
PAP2T can only register to DynDNS and that's all.

What sort of a script would I be looking for? something to query DynDNS for
the new IP of the device to add to firewall? This might however bring down
time if inquiry is not successful.

Or can I setup my own Dyndns server on the Asterisk server and have those
PAP2T units registered to it and then work it from there when their IPs
change?

Thanks

On Sat, Oct 2, 2010 at 3:32 PM, jon pounder  wrote:

>  On 10/02/2010 03:31 PM, bruce bruce wrote:
>
> Hi,
>
>  Can you please explain the DynDNS part. How would I put that in my
> Asterisk server as an identified party? Usually it comes to me with IP
> address (dynamic). Or do add something like this in sip_nat.conf:
>
>  externip=mybox.dyndns.org
> localnet=192.168.0.0/255.255.255.0
>
>
> every time the address changes you have to have some script to make the
> change in your firewall.
>
>
>  ???
>
>  Thansk again,
>
> On Sat, Oct 2, 2010 at 2:59 PM, jon pounder  wrote:
>
>>  On 10/02/2010 02:56 PM, bruce bruce wrote:
>> > Hi Everyone
>> >
>> > I think PAP2T supports DynDNS and other Dynamic DNS providers. I have
>> > a box that needs to be secured at all times. Currently it's not
>> > connected to the internet. If it were connected, I would have iptables
>> > block any and all traffic from outside but I want a single device -
>> > Linksys PAP2T - to be able to connect back to the server. That is a
>> > stand alone device and doesn't support VPN and I don't have the luxury
>> > of putting a VPN client on the PAP2T side to connect back to the
>> > server. Is there any way I can DynDNS on the PAP2T to somehow notify
>> > the Asterisk Server that it's a safe device coming in?
>> >
>> > I do use fail2ban but that is not what I am looking for at this
>> > moment. And since the IP is dynamic on the PAP2T, I can't just use the
>> > iptables to let it in as it might change all a sudden.
>> >
>> > Thanks
>>  do the dyndns on whatever router is in front of the pap2t
>> or
>> get some other box that supports it.
>>
>>
>> other than that you are looking for some sort of magic bullet
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
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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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

[bruce bruce]

Hi,

Can you please explain the DynDNS part. How would I put that in my Asterisk
server as an identified party? Usually it comes to me with IP address
(dynamic). Or do add something like this in sip_nat.conf:

externip=mybox.dyndns.org
localnet=192.168.0.0/255.255.255.0

???

Thansk again,

On Sat, Oct 2, 2010 at 2:59 PM, jon pounder  wrote:

> On 10/02/2010 02:56 PM, bruce bruce wrote:
> > Hi Everyone
> >
> > I think PAP2T supports DynDNS and other Dynamic DNS providers. I have
> > a box that needs to be secured at all times. Currently it's not
> > connected to the internet. If it were connected, I would have iptables
> > block any and all traffic from outside but I want a single device -
> > Linksys PAP2T - to be able to connect back to the server. That is a
> > stand alone device and doesn't support VPN and I don't have the luxury
> > of putting a VPN client on the PAP2T side to connect back to the
> > server. Is there any way I can DynDNS on the PAP2T to somehow notify
> > the Asterisk Server that it's a safe device coming in?
> >
> > I do use fail2ban but that is not what I am looking for at this
> > moment. And since the IP is dynamic on the PAP2T, I can't just use the
> > iptables to let it in as it might change all a sudden.
> >
> > Thanks
> do the dyndns on whatever router is in front of the pap2t
> or
> get some other box that supports it.
>
>
> other than that you are looking for some sort of magic bullet
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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[asterisk-users] Attempts to hack Asterisk - What do these lines means

[bruce bruce]

Hi Everyone,

Like always, here are IPs from China that try to hack an Asterisk server.
Can someone please explain what is happening or what the hacker is trying to
reach:

02/10/2010 11:10 SIP/113.105.152.51-00fb sip "sip"  s ANSWERED 13
02/10/2010 11:10 SIP/113.105.152.51-00fe sip "sip"  s ANSWERED 13
02/10/2010 11:10 SIP/113.105.152.51-00fc sip "sip"  s ANSWERED 13
02/10/2010 11:10 SIP/113.105.152.51-00fd sip "sip"  s ANSWERED 13
02/10/2010 11:10 SIP/113.105.152.51-00ff sip "sip"  s ANSWERED 13
02/10/2010 11:10 SIP/113.105.152.51-0100 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0101 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0102 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0103 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0104 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0105 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0106 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0107 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0108 sip "sip"  s ANSWERED 13
02/10/2010 11:17 SIP/222.73.204.198-0109 sip "sip"  s ANSWERED 13


Thanks
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[asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

[bruce bruce]

Hi Everyone

I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box
that needs to be secured at all times. Currently it's not connected to the
internet. If it were connected, I would have iptables block any and all
traffic from outside but I want a single device - Linksys PAP2T - to be able
to connect back to the server. That is a stand alone device and doesn't
support VPN and I don't have the luxury of putting a VPN client on the PAP2T
side to connect back to the server. Is there any way I can DynDNS on the
PAP2T to somehow notify the Asterisk Server that it's a safe device coming
in?

I do use fail2ban but that is not what I am looking for at this moment. And
since the IP is dynamic on the PAP2T, I can't just use the iptables to let
it in as it might change all a sudden.

Thanks
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Re: [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...

[bruce bruce]

Thanks guys. Amazing feedback.

Sounds like 2.5" is a better choice for being less in size (easier access
for voicemail for example), as fast as 3.5" HDD in RPM, and allows 6 HDDs
per Node which allows more RAID choice.

However, it does come to be more expensive.

Thanks again,

On Mon, Sep 27, 2010 at 10:57 AM, Benny Amorsen

> wrote:

> bruce bruce  writes:
>
> > Other than the price difference (2.5" is more expensive and can't find
> > many of the 1TB or so) is there any preference, advantage, or
> > disadvatage of chosing 2.5" HDD or 3.5" when it comes to the server
> > operations or Asterisk operation?
>
> There is no difference. Pick the server which offers the disk bandwidth
> and I/O's per second which you need.
>
> Do you really need 1TB disks? If you do, be careful what you place on
> those disks. Reading e.g. a voice mail or a speak off a large slow
> platter which is busy writing CDR's does not sound good at all.
>
>
> /Benny
>
>
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[asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...

[bruce bruce]

Hi Everyone,

I am stack between two identical systems (2U Twin2, 4 nodes, SuperMicro)
servers that have the same exact specs except for HDDs. These nodes will all
either have Asterisk installed with CentOS or will have Asterisk install in
virtual environment.

Option 1: *12* x 3.5" HDD (3 HDDs per node)
Option 2: *24* x 2.5" HDD (6 HDDs per node)
**both options come to the same price.

Other than the price difference (2.5" is more expensive and can't find many
of the 1TB or so) is there any preference, advantage, or disadvatage of
chosing 2.5" HDD or 3.5" when it comes to the server operations or Asterisk
operation?

Each node of this server will be running CentOS 5.5 either in 64 or 32 bit +
Asterisk or they will be used for virtual environment where multiple
instance of Asterisk will be installed within CentOS XEN.

Your input is much appreciated.

Thanks
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)

[bruce bruce]

Thanks for the detailed info. Problem was solved by including Server B
subnet as the localnet of the Server A (OpenVPN server) and setting each
extension NAT=NO.

Your points are good guides for future problem diagnoses.

Thanks again,
Bruce

On Thu, Sep 23, 2010 at 1:56 PM, Dave Platt  wrote:

>
> > I don't think it's an endpoint issue. I think the SIP packet headers get
> > over-written by the tunnel (openvpn) protocol.
>
> I'd be rather astonished if OpenVPN itself were responsible for this.
> As far as I know, OpenVPN doesn't do higher-level-protocol rewriting
> of any sort.  It just provides the "bit pipe" through the tunnel.
>
> I'd suggest several other possible culprits:
>
> (1) Server B might be doing higher-level protocol rewriting (i.e.
>SIP border gateway stuff) prior to routing the SIP packets
>through the OpenVPN tunnel.
>
>This might happen if Server B were configured to use the
>Linux "iptables" features, with a SIP protocol module and
>Network Address Translation features.
>
>The fix would be to disable NAT and boundary processing in
>Server B's routing functions.
>
> (2) The SIP endpoints (phones) might be configured to "discover
>their external address", via STUN or a similar mechanism.
>
>The fix would be to change the endpoint device configuration.
>
> I think you'll need to use Wireshark or a similar sniffer, to see
> what the SIP traffic looks like at several points along the path,
> so you can locate the earliest point at which the wrong address is
> in the SIP packet payload.
>
> Several examination points come to mind:
>
> -  Right after the packet leaves the endpoint device.  I'd suggest
>   using a laptop running Wireshark as a passive packet sniffer...
>   connect the endpoint device and the laptop to an Ethernet hub
>   (not a switch!) and sniff the packets before any router gets
>   its hands on them.
>
> -  As the packets enter Server B - use Wireshark on Server B and
>   have it tap into the incoming Ethernet interface.
>
> -  As the packets are pushed out of Server B's routing layer into
>   the OpenVPN tunnel.  Use Wireshark to monitor the "tap" or
>   "tun" virtual interface, to which the kernel transmits the packets
>   that OpenVPN is to convey.
>
> -  As the packets come out of the tap/tun device on Server A.
>
> In scenario (1) I described above, you'd see the packets be correct
> at the first and second Wireshark sniffing points, and incorrect at the
> third and fourth (i.e. the modifications are being performed in
> Server B's routing/NAT'ing layer).
>
> In Scenario (2), they'd be incorrect at every point, including just
> after they come out of the IP-phone.
>
> In the scenario you described, they'd be correct at the first, second,
> and third points, and wrong at the fourth.
>
>
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[asterisk-users] Installing Asterisk + FreePBX from Repsitory spits out some warnings and errors for ever

[bruce bruce]

Hello,

This is what what I see after a Yum install asterisk16 asterisk16-config
freepbx:

Use of uninitialized value in string ne at
/var/www/html/panel/op_server.plline 4997.
Use of uninitialized value in substitution (s///) at /var/www/html/panel/
op_server.pl line 5439.
Use of uninitialized value in substitution (s///) at /var/www/html/panel/
op_server.pl line 5440.
Use of uninitialized value in substitution (s///) at /var/www/html/panel/
op_server.pl line 5441.
Use of uninitialized value in substitution (s///) at /var/www/html/panel/
op_server.pl line 5442.
Use of uninitialized value in concatenation (.) or string at
/var/www/html/panel/op_server.pl line 5444.


Usually these error stay on the /var/log/messages for ever. I mean they
repeat. Is there a problem with these?

Thanks
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

Calls are not going outside of the network. I had to setup up the subnet of
the other side (openvpn client) as the localnet of the Asterisk server for
Asterisk to not handle it with NAT or hand shake it with external IP.

Thanks,
-Bruce

On Wed, Sep 22, 2010 at 1:58 PM, Paul Belanger  wrote:

> On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce  wrote:
> > Thanks, but Carlos Chavez was right on point. This fixed the problem:
> > externip=123.123.123.123
> > localnet=192.168.100.0/255.255.255.0
> > nat=no in each extension.
> >
> So now I am confused, If you have a VPN setup between sites, why are
> calls going outside the VPN?  Or do you have remote agents that are
> not using a VPN?
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

Thanks, but Carlos Chavez was right on point. This fixed the problem:

externip=123.123.123.123
localnet=192.168.100.0/255.255.255.0

nat=no in each extension.

Maybe combination of both or only the localnet just fixed it.

Thanks,
Bruce

On Wed, Sep 22, 2010 at 1:35 PM, Steve Edwards wrote:

> Un-top-posting...
>
> >   On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce 
> wrote:
> >   > Any feed back is appreciated.
>
> > On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger <
> paul.belan...@polybeacon.com> wrote:
>
> > Then configure you endpoints to use the 192.168.100.0/24 network. This
> > is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is sending
> > the INVITE message.
>
> On Wed, 22 Sep 2010, bruce bruce wrote:
>
> > I don't think it's an endpoint issue. I think the SIP packet headers get
> > over-written by the tunnel (openvpn) protocol.
>
> Would wireshark shed some light?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

Thanks for that Carlos. I am playing with that right now. What do you
suggest localnet should say?

Server A = OpenVPN Server:
localnet=127.0.01
localnet=192.168.100.0/255.255.255.0

Where 192.168.100.0 is the DHCPd subnet of Server B (the openvpn client)

Server A doesn't have any localnet other than the loop back and then a Vnet
to internet (public ip address).

Thanks,
Bruce

On Wed, Sep 22, 2010 at 11:36 AM, Carlos Chavez wrote:

> Do you have a localnet statement in your sip.conf?  That and using
> nat=no will make sure Asterisk does not replace the IP address in the
> Invite.
>
> On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote:
> > Hi Everyone,
> >
> >
> > I have setup an OpenVPN tunnel between Server A (running Asterisk) and
> > Server B suppling it's SIP Phones with DHCP pool of IPs.
> >
> >
> > So, the tunnel is established nicely and everyone can ping others.
> > "sip show peers" shows the local subnet of the SIP Phones registered
> > (192.168.100.0/24).
> >
> >
> > But there is the old bad one-way audio. Calls also drop after few
> > seconds. In the SIP debug I can see that asterisk uses it's external
> > public IP address to communicate to endpoints that are known to it as
> > the 192.168.100.0/24 endpoints and the endpoints identify themselves
> > with the OpenVPN tunnel IP address scheme in one part of the sip
> > handshake. How can this be fixed? After all, with the OpenVPN this
> > should all look like an internal network to Asterisk.
> >
> >
> > I have added my comments followed by # to lines below that are
> > problematic.
> >
> >
> > <--- SIP read from UDP:192.168.100.5:5060 --->#This line is good
> > as it uses the local DHCP supplied network address scheme
> > INVITE sip:2...@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we
> > inviting Ext. 203 with it's OpenVPN IP while it's on the same network
> > of 192.168.50.0/24 as 202?
> > Via: SIP/2.0/UDP
> > 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6
> Max-Forwards: 70
> > From: "SIP Phone - Ext. 202" ;tag=6d6f8c4226
> >#BAD line again. Should be 
> > SIP:2...@192.168.100.6
> > To: "203"  #Bad again
> > Call-ID: 43af67a634e06e75
> > CSeq: 32058 INVITE
> > Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
> > PRACK, SUBSCRIBE, INFO
> > Allow-Events: talk, hold, conference, LocalModeStatus
> > Contact: "SIP Phone - Ext. 202"
> > ;
> > +sip.instance=""
> > Supported: gruu, path, timer, 100rel, replaces
> > User-Agent: Aastra 55i/2.5.2.1500
> > Content-Type: application/sdp
> > Content-Length: 594
> >
> >
> > Basically the phones should only send with FROM their local
> > 192.168.100.0/24 address and Asterisk should only send ANSWER and ACK
> > back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24
> > (which is the openvpn client ip).
> >
> >
> > Once above is fixed, I think all the audio and call cut will go away.
> > I hate to use a sip proxy in this situation since I already have an
> > openvpn connection.
> >
> >
> > Any feed back is appreciated.
> >
> >
> > Thanks,
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> --
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> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Tel: +52-55-91169161 Ext 2001
>
>
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

I don't think it's an endpoint issue. I think the SIP packet headers get
over-written by the tunnel (openvpn) protocol.

Thanks,
Bruce

On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger  wrote:

> On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce  wrote:
> > Any feed back is appreciated.
> >
> Then configure you endpoints to use the 192.168.100.0/24 network.
> This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is
> sending the INVITE message.
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

Thanks for the feedback. I thought about that but it's not an option for me
right now.

Any other ways folks?

Thanks

On Wed, Sep 22, 2010 at 4:06 AM, Roger Burton West wrote:

> On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
> >I have setup an OpenVPN tunnel between Server A (running Asterisk) and
> >Server B suppling it's SIP Phones with DHCP pool of IPs.
>
> Have you considered running Asterisk on Server B as well, and using IAX
> to trunk between them? This is working well for me.
>
> Roger
>
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[asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[bruce bruce]

Hi Everyone,

I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.

So, the tunnel is established nicely and everyone can ping others. "sip show
peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24
).

But there is the old bad one-way audio. Calls also drop after few seconds.
In the SIP debug I can see that asterisk uses it's external public IP
address to communicate to endpoints that are known to it as the
192.168.100.0/24 endpoints and the endpoints identify themselves with the
OpenVPN tunnel IP address scheme in one part of the sip handshake. How can
this be fixed? After all, with the OpenVPN this should all look like an
internal network to Asterisk.

I have added my comments followed by # to lines below that are problematic.

<--- SIP read from UDP:192.168.100.5:5060 --->#This line is good as it
uses the local DHCP supplied network address scheme
INVITE sip:2...@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we
inviting Ext. 203 with it's OpenVPN IP while it's on the same network of
192.168.50.0/24 as 202?
Via: SIP/2.0/UDP
192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6
Max-Forwards:
70
From: "SIP Phone - Ext. 202" ;tag=6d6f8c4226
 #BAD line again. Should be SIP:2...@192.168.100.6 
To: "203"  #Bad again
Call-ID: 43af67a634e06e75
CSeq: 32058 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "SIP Phone - Ext. 202" ;+sip.instance=""
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 55i/2.5.2.1500
Content-Type: application/sdp
Content-Length: 594

Basically the phones should only send with FROM their local
192.168.100.0/24address and Asterisk should only send ANSWER and ACK
back to
192.168.100.0/24 rather than sending it to 172.16.0.0/24 (which is the
openvpn client ip).

Once above is fixed, I think all the audio and call cut will go away. I hate
to use a sip proxy in this situation since I already have an openvpn
connection.

Any feed back is appreciated.

Thanks,
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Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

[bruce bruce]

Thanks guys. I wasn't able to collect enough SIP debug as the problem was
resolved as I was testing different configuration for the trunk. Probably a
change on the provider side.

John Novack: Unfortunately, it seems that this list has a non-stop list of
people who like to stir up things or try to censor people who bring legit
questions without the consideration that they are not moderators of the list
at any level. They forget to remember that AsteriskNow uses FreePBX as well
and that Asterisk IS the underlying technology for all the flavours. Thanks
for the feedback.

Most I was able to collect was that:

- if the trunk configuration even included "context=from-pstn", the CLI
would show "Executing .@from-sip-external".
- if SIP Anonymous was set to YES then the [from-sip-external] context would
match the peer to the right trunk defined as that is what is expected of
that context from the code. If SIP Anonymous was set to off
@from-sip-external is set to go to ss-noservice.
- Later on when the calls resumed and the problem was fixed, calls were
coming in with "Executing @from-pstn" which should have always been the case
regardless of the SIP Anonymous or not.

I was puzzled because the FISRT line of the CLI was the "Executing
@from-pstn" or "Executing .@from-sip-external" and that made a world
of difference. The latter one not working.

I just couldn't pinpoint where FreePBX failed to read the
"context=from-pstn". If it was something to do with the MySQL database or of
parsing the _custom.conf files as the problem was fixed all a sudden. I
guess now I have to wait and see if it comes back.

Thanks,



On Tue, Sep 14, 2010 at 10:47 AM, Zeeshan Zakaria wrote:

> This might help to answer poster's question. It tells how the allow
> anonymous sip connections work in FreePBX, and shows the code.
>
> http://www.geekzone.co.nz/sbiddle/7183
>
> --
> Zeeshan
>
>
> On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger <
> paul.belan...@polybeacon.com> wrote:
>
>> On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria 
>> wrote:
>> > Poster is having problem when he disallows anonymous sip peers. Do you
>> know
>> > at all how FreePBX deals with anonymous sip peers? Obviously you haven't
>> yet
>> > seen the dialplan for FreePBX.
>> >
>> It's very simple to find the actually issue, if the OP does the following:
>>
>>
>> http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
>>
>> The attached the debug log to thread.
>>
>> --
>> Paul Belanger | dCAP
>> Polybeacon | Consultant
>> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
>> blog.polybeacon.com
>>
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[asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

[bruce bruce]

Hi Everyone,

I have a provider whose DID used to come into the box just fine but recently
stopped working. Nothing has been changed on our end.

Here is what I get when doing "sip set debug peer PROVIDER":

Sending to 123.123.123.123 : 5060 (no NAT)

 That is ALL I am getting with sip debug turned on.

With Allow Anonymous SIP set to YES, then the call comes in properly and you
see the ACK, REQUEST and ACCEPT of sip debug just fine.

This is Elastix with Asterisk 1.4.33.1

Any thoughts?

Thanks
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[asterisk-users] Cisco or Linksys WRP400 reliability?

[bruce bruce]

Hi Everyone,

I see one long post on Cisco community forum where everyone including ISPs
are complaining about silence on FXS port, reboots, frozen state, etcof
WRP400. This is the a wireless router + 2 FXS combo box. I am looking to use
this for home user to connect to hosted Asterisk PBX.

I am looking for some feedback from the community as to how stable the unit
is - or if it is stable at all?

Thanks for your input.
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[asterisk-users] Anyone can share their experience about Thomson TG784 wireless router/ATA?

[bruce bruce]

Hi Everyone,

Wondering if any of you folks ever had trouble using *Thomson
TG784
 *DSL/Wireless Router/FXS ATA combo? I am looking to use this to connect
users from home to a hosted Asterisk PBX.

Any and all inputs are appreciated.

Thanks
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[asterisk-users] VoIP friendly Internet providers in Dallas and Philadelphia

[bruce bruce]

Hi Everyone,

My experience is only with the Canadian providers. What options/providers
are there in Dallas and Philadelphia other than Verizon when it comes to
internet? Something in the order of at least 10mbps down and up - I
understand that and higher bandwidths are easily available in USA due to
vast fiber networks? The connection will be replacing a T1 and will be
support VPN connections to connect office for VoIP.

Thanks
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Re: [asterisk-users] openvz

[bruce bruce]

1- I am interested in this as well. Looking into Proxmox as it provides a
nice interface (do you guys know of any other good one?)

2- Would the conference calls be fine as well? I understanding Asterisk
1.6.x uses a kernel timing source now a days so that ztdummy is not needed
anymore?

3- Would installing from yum repository be just fine?

Thanks

On Fri, Sep 3, 2010 at 10:31 AM, mattias  wrote:

> Outlook?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
> Nicholas
> Sent: Friday, September 03, 2010 3:16 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] openvz
>
>
> >From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
> >Subject: [asterisk-users] openvz
>
> >Can i run asterisk on a openvz vps or do i need a kernel?
> >I dont use dadi
>
> Blind Answer - you should be able to; Asterisk doesn't rebuild the
> kernel. You might have to get some "kernel source" using ZYPPER (in caps
> so Outlook express doesn't change it to zipper).
>
>
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Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost?

[bruce bruce]

Thanks Don for clarification.

There are lots of people on this list that hastily decide to answer without
even reading a post properly. I am sure they won't even read the follow-ups.
They just talk for the sake of talking. Sickens me!

Please note the subject line in my original post:  "To compete with Avaya -
What are their current cost?"
My question is specifically related to Avaya and other propriety call
centers because I want to compete with them with Asterisk.

If you know recent prices please post back. If not, don't bother. Also,
please do not private message me. I will move this post to Biz List as I
just noted I posted to wrong listing.

Thanks to those who tried meaningful posts.



On Thu, Sep 2, 2010 at 3:06 PM, Don Kelly  wrote:

> It could be that I'm entirely confused, but I think he asked what people
> are
> paying for Avaya solutions--so he'd know what competitive pricing would be
> for the open source solution he's prepared to offer.
>
> When someone replied with open-source suggestions, he pointed out that that
> was not the information he was looking for. He did not say that he's not
> interested in providing open source solutions for his clients.
>
> --Don
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
> Nicholas
> Sent: Thursday, September 02, 2010 1:44 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya-
> Whatare their current cost?
>
> He doesn't deserve the responses, but it seems that boundaries are being
> pushed in both sides of the response.  If he thinks he's on the biz list,
> that's one thing, but in the purely open discussion, don't be dissing open
> source either.
>
>
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Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?

[bruce bruce]

I am not interested in open source solutions. I want to know how much the
propriety systems cost in terms of licensing. Specially Avaya now a days per
extension. Exclusive or Inclusive of the hardware for 10 agents as noted.

Thanks

On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider
wrote:

> Hi Bruce,
>
> It all depends what exactly you are in need of. A basic call center
> solution will only cost $500 exclusive of hardware, depending on your need
> you will have to decide what type of servers you need or weather you would
> have handsets or softphones, type of headgears you want, kind of workstation
> you will need.
>
> I work for a company that provides open source call center solution. You
> can visit the website www.crystalconsulting.pk or if you want more detail
> you can email the detail of the requirements on sa...@crystalconsulting.pkor 
> you can email me and I can revert back to you with detail.
>
> Regards,
>
> Shomail
>
> On Fri, Aug 27, 2010 at 2:03 PM,  wrote:
>
>> Hi Everyone,
>>
>> Just a quick estimate of what Call Center Software/Hardware providers
>> charge now a days for a 10 seat and 20 seat with upfront costs and monthly
>> licensing cost?
>>
>> Thanks,
>>
>>
>
>
> --
> Muhammad Shomail Haider
> www.shomail.blogspot.com
> www.facebook.com/shomail
> www.twitter.com/shomail
> www.linkedin.com/in/shomail
>
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Re: [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?

[bruce bruce]

Maybe dvossel can re-open issue # 16753 and fix the warning to show on iax2
debug as well along with core set debug like all other warnings. That way
it's straight forward. That ticket shouldn't have been closed without a fix.

On Thu, Sep 2, 2010 at 4:11 AM, bruce bruce  wrote:

> I'd rather find the problem than upgrade blindly. Upgrading not always
> solves the problem and has the potential to break other things. Thanks for
> the offer though.
>
> Bug # 16753 applies.
>
> call token not required was set in the trunk and problem solved.
>
> There is not warning for this in iax2 debug but there is in core set debug.
> That's a petty.
>
>  -Bruce
>
>
> On Thu, Sep 2, 2010 at 3:19 AM, Tzafrir Cohen wrote:
>
>> On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote:
>> > Hi Everyone,
>> >
>> > I have two servers as the following that are trunked with each other via
>> > IAX2 trunk:
>> >
>> > Server A:
>> > Asterisk 1.4.21.2 (Elastix Flavor)
>>
>> Any chance you could upgrade that? Elastix has newer versions of
>> Asterisk, for starters.
>>
>> >
>> >
>> > Server B (IP # 72.72.72.72):
>> > Asterisk 1.6.2.0 (Vanilla)
>>
>> Is it configured to talk to old IAX2 peers?
>>
>> http://downloads.asterisk.org/pub/security/IAX2-security.html
>> http://downloads.asterisk.org/pub/security/AST-2009-006.html
>>
>> --
>>   Tzafrir Cohen
>> 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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>
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Re: [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?

[bruce bruce]

I'd rather find the problem than upgrade blindly. Upgrading not always
solves the problem and has the potential to break other things. Thanks for
the offer though.

Bug # 16753 applies.

call token not required was set in the trunk and problem solved.

There is not warning for this in iax2 debug but there is in core set debug.
That's a petty.

-Bruce

On Thu, Sep 2, 2010 at 3:19 AM, Tzafrir Cohen wrote:

> On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote:
> > Hi Everyone,
> >
> > I have two servers as the following that are trunked with each other via
> > IAX2 trunk:
> >
> > Server A:
> > Asterisk 1.4.21.2 (Elastix Flavor)
>
> Any chance you could upgrade that? Elastix has newer versions of
> Asterisk, for starters.
>
> >
> >
> > Server B (IP # 72.72.72.72):
> > Asterisk 1.6.2.0 (Vanilla)
>
> Is it configured to talk to old IAX2 peers?
>
> http://downloads.asterisk.org/pub/security/IAX2-security.html
> http://downloads.asterisk.org/pub/security/AST-2009-006.html
>
> --
>   Tzafrir Cohen
> 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] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?

[bruce bruce]

Hi Everyone,

I have two servers as the following that are trunked with each other via
IAX2 trunk:

Server A:
Asterisk 1.4.21.2 (Elastix Flavor)


Server B (IP # 72.72.72.72):
Asterisk 1.6.2.0 (Vanilla)

Server B can place calls to Server A but when trying to place calls from
Server A to Server B this is what I am getting:


pbx*CLI> originate iax2/mel/14161234567 extension s...@null

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW

   Timestamp: 3ms  SCall: 16389  DCall: 0 [72.72.72.72:4569]
   VERSION : 2
   CALLED NUMBER   : 14161234567
   CODEC_PREFS : (gsm)
   CALLING NUMBER  :
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME:
   LANGUAGE: en
   USERNAME: mel
   FORMAT  : 2
   CAPABILITY  : 57346
   ADSICPE : 2
   DATE TIME   : 2010-09-02  02:14:50

Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
   Timestamp: 3ms  SCall: 1  DCall: 16389 [72.72.72.72:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK

   Timestamp: 3ms  SCall: 16389  DCall: 1 [72.72.72.72:4569]
-- Hungup 'IAX2/mel-16389'



As you can see above, Subclass: REJECT comes back wtih no cause code.
Usually there is a cause code to debug but in this case there is no Cause
code. Trunks on both sides in the context=from-internal so it's not an
Inbound Route issue.

Any pointers are much appreciated.

-Bruce
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[asterisk-users] OrderlyStats or QueueMetrics

[bruce bruce]

Hi Everyone,

There are a few things I like in OrderlyStats, specially some graph
presentations and the fact that if agent puts someone on HOLD or PAUSE it
shows fine.

1 -But I see a lot of similarities in pricing, descriptions, wording on both
sites. Were these same projects forked out? or is it still both owned by the
same company?

2- What are the main differences between the two? - so that we can make a
better choice.

3- What part of these two products are open source? and what part is not
opensource? - just the license part?

4- Any better alternatives?

Thanks
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Re: [asterisk-users] Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?

[bruce bruce]

Thanks. That is one thing I really HATE about AASTRA - them confusing the
user with providing different setting level on the WEB UI and the PHONE UI -
very stupid.

But thank you and it works just fine.

-Bruce

On Thu, Aug 26, 2010 at 4:09 AM, Gareth Blades
wrote:

> bruce bruce wrote:
> > Hi Everyone,
> >
> > I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i
> > receiver port and I get a tone. But when I connect it to the headset
> > port there is no tone. I am running firmware 2.4 and I can't seem to
> > find that DHSG, EHS or whatever the setting maybe called to enable to
> > get this headset work with the phone. Can anyone quickly tell me where
> > the audio options are on this phone?
> >
> > Thanks,
> > Bruce
> >
>
> Press the tools button.
> Press 2 (Preferences)
> Press 5 (Set Audio)
> You now have 3 options to set the handsfree/headset mode, mic volume and
> DHSG
>
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[asterisk-users] Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?

[bruce bruce]

Hi Everyone,

I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i
receiver port and I get a tone. But when I connect it to the headset port
there is no tone. I am running firmware 2.4 and I can't seem to find that
DHSG, EHS or whatever the setting maybe called to enable to get this headset
work with the phone. Can anyone quickly tell me where the audio options are
on this phone?

Thanks,
Bruce
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Re: [asterisk-users] Opensource Speech recognition for Asterisk

[bruce bruce]

Bob,

Both ZanziIVR and Speechforge have similar look web pages. I guess you have
used one of those to get the speech going as this link:
http://scribblej.com/svn/ probably is not the full thing.

These seem like practical project. Thanks for pointing out. This is what I
was looking for.

Now starts the try to get these installed and tested.

Thanks,
Bruce

On Tue, Aug 24, 2010 at 7:30 AM, Bob Kleiner  wrote:

> > Thanks guys. A lot of info here :-)
> >
> > I am wondering if anyone followed this and it was working for them:
> >
> > http://scribblej.com/svn/
> >
> > ???
>
> Hello Bruce
>
> We successfully deployed it and now saving thousands on commercial ASR
> ports. It seems users are rather happy with it. The recognition seems
> pretty accurate. Of course it has it's own limitations but so any
> other technology. It will not hurt if some of your users will benefit
> from ASR.
>
> > I am not looking for anything fancy. The basic "yes", "no", dialing a
> > number, asking for agent, etc...out of which probably the hardest is a 10
> > digit number to be asked to be dialed.
>
> Yes, that should work. It also supports JSGF grammars, so you should
> be able to recognize digit strings easily.
>
> And if you want something serious, there are at least two open source
> products
> providing ASR over standard MRCP protocol. They also use CMUSphinx, so
> provide the same accuracy
>
> Zanzibar http://www.spokentech.org/writing-speechlets.html
> Cairo http://www.speechforge.org/
>
> Though Cairo is a bit dated.
>
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Re: [asterisk-users] Opensource Speech recognition for Asterisk

[bruce bruce]

Thanks guys. A lot of info here :-)

I am wondering if anyone followed this and it was working for them:

http://scribblej.com/svn/

???

I am not looking for anything fancy. The basic "yes", "no", dialing a
number, asking for agent, etc...out of which probably the hardest is a 10
digit number to be asked to be dialed.

Thanks


On Sun, Aug 22, 2010 at 2:30 AM, Nickolay V. Shmyrev
wrote:

> > Hi Everyone,
> >
> > Has anyone got any opensource speech recognition software to work with
> > Asterisk? Please only list WORKING ones. Not the "theoretically" should
> work
> > ones!
>
> Hi
>
> I definitely suggest you to try CMU Sphinx connector for Asterisk. You
> can find all required information here
>
> http://scribblej.com/svn/
>
> If you need any help with setup, just ask.
>
> --
> Nexiwave - Speech Indexing Solution For Call Centers
> http://nexiwave.com
>
>
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[asterisk-users] Opensource Speech recognition for Asterisk

[bruce bruce]

Hi Everyone,

Has anyone got any opensource speech recognition software to work with
Asterisk? Please only list WORKING ones. Not the "theoretically" should work
ones!

Thanks
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Re: [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?

[bruce bruce]

That is set and here is what I get:

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
   Timestamp: 3ms  SCall: 01217  DCall: 0 [44.55.66.77:4569]
   USERNAME: 9988
   REFRESH : 60

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK

   Timestamp: 3ms  SCall: 01217  DCall: 1 [44.55.66.77:4569]

Any other suggestions. Anyone with a working pfsense configuration that can
share with me?

Thanks,
Bruce


On Wed, Aug 18, 2010 at 3:42 AM, Nasir Iqbal wrote:

> Hi,
>
> Use requirecalltoken=no in your peer configuration
>
> Regards
>
> On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce  wrote:
>
>> Hello Everyone,
>>
>> I am trying to diagnose issue with my IAX2 extension not working.
>>
>> When I have iax2 set debug on all I see is this:
>>
>> *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
>> REGREQ *
>> *   Timestamp: 3ms  SCall: 00130  DCall: 0 [64.229.229.111:64823]
>> *
>> *   USERNAME: 100*
>> *   REFRESH : 60*
>>  *
>> *
>> *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
>> ACK*
>> *   Timestamp: 3ms  SCall: 00130  DCall: 1 [64.229.229.111:64823]
>> *
>>
>>
>> So, all the packets are coming in, but there is no Tx response. Is that
>> normal and is that how IAX2 works according to RFC to not respond back?
>>
>> I have checked my firewall and all is set fine. I have any WAN address to
>> come in through port 4569 to map to the server and it worked last week but
>> now it doesn't.
>>
>> Any suggestions?
>>
>> Thanks
>>
>> --
>>
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
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[asterisk-users] Pfsense and IAX2 - What is the proper firewall NAT setup?

[bruce bruce]

Hi Everyone,

Just trying to connect the Zoiper Communicator to connect to Asterisk which
is behind Pfsense. Here is what I get at debug and it doesn't register.
Error code 16. Can someone please let me know their firewall, NAT, outbound
1-to-1 pfsense settings as it seems to me I am doing something wrong on the
firewall?


*Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ *
*   Timestamp: 3ms  SCall: 00426  DCall: 0 [44.55.66.77:4569]*
*   USERNAME: 100*
*   REFRESH : 60*
*
*
*Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ *
*   Timestamp: 3ms  SCall: 00427  DCall: 0 [44.55.66.77:4569]*
*   USERNAME: 100*
*   REFRESH : 60*
*
*
*Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
   *
*   Timestamp: 3ms  SCall: 00427  DCall: 1 [44.55.66.77:4569]*

44.55.66.77 is my client IP. I see no Tx packets. What is happening?

Thanks for sharing,
Bruce
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[asterisk-users] Asterisk with Motorola Canopy

[bruce bruce]

Hi Everyone,

Can anyone share their experience with Motorola Canopy solution deployment
and Asterisk? Is this a good fit?

Thanks,
Bruce
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Re: [asterisk-users] How to determine which party hangup th e call? cause of Hang-up needed.‏

[bruce bruce]

Sorry, I am not following:

*"**read the value of var ${HANGUPCAUSE} next line to dial command."*
*
*
*Where is that value? Next to dial you mean right when the call was placed?
or check next few lines to find HANGUP cause?*
*
*
*Note: This is using ZAP (analogue) and not PRI.*
*
*
*Thanks,*
*Bruce
*
On Wed, Aug 11, 2010 at 12:33 AM, Faisal Hanif  wrote:

>  read the value of var ${HANGUPCAUSE} next line to dial command.
>
> Regards,
>
> Faisal Hanif
> *VoIP Manager
> ***Vopium A/S
>  On 8/10/2010 9:51 PM, bruce bruce wrote:
>
> Hi Everyone
>
> Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell
> Canada.
>
> User claims that call hangup without any interferance to the phone set.
>
> Is there ANYWAY to find out which party hang-up the call or if the call was
> cut-off due to other reasons?
>
> I checked the *"asteriskcdrb"* table and it's pretty much useless in this
> case as it only logs the duration and other properties but not cause of the
> Hangup.
>
>
>   /var/log/asterisk/full
>
> [Jul 10 10:37:02] VERBOSE[29366] logger.c:   == Manager 'admin' logged off
> from 127.0.0.1
> [Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
> [...@macro-dialout-trunk:1] Macro("SIP/1007-069a", "hangupcall|") in new
> stack
> [Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
> [...@macro-hangupcall:1] GotoIf("SIP/1007-069a", "1?skiprg") in new
> stack
> [Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Goto
> (macro-hangupcall,s,4)
>
>
>  Thanks,
>
> Bruce
>
>
> --
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[asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?

[bruce bruce]

Hello Everyone,

I am trying to diagnose issue with my IAX2 extension not working.

When I have iax2 set debug on all I see is this:

*Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ *
*   Timestamp: 3ms  SCall: 00130  DCall: 0 [64.229.229.111:64823]*
*   USERNAME: 100*
*   REFRESH : 60*
*
*
*Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
   *
*   Timestamp: 3ms  SCall: 00130  DCall: 1 [64.229.229.111:64823]*


So, all the packets are coming in, but there is no Tx response. Is that
normal and is that how IAX2 works according to RFC to not respond back?

I have checked my firewall and all is set fine. I have any WAN address to
come in through port 4569 to map to the server and it worked last week but
now it doesn't.

Any suggestions?

Thanks
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[asterisk-users] How to determine which party hangup th e call? cause of Hang-up needed.‏

[bruce bruce]

Hi Everyone

Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell
Canada.

User claims that call hangup without any interferance to the phone set.

Is there ANYWAY to find out which party hang-up the call or if the call was
cut-off due to other reasons?

I checked the *"asteriskcdrb"* table and it's pretty much useless in this
case as it only logs the duration and other properties but not cause of the
Hangup.


 /var/log/asterisk/full

[Jul 10 10:37:02] VERBOSE[29366] logger.c:   == Manager 'admin' logged off
from 127.0.0.1
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
[...@macro-dialout-trunk:1] Macro("SIP/1007-069a", "hangupcall|") in new
stack
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
[...@macro-hangupcall:1] GotoIf("SIP/1007-069a", "1?skiprg") in new stack
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Goto
(macro-hangupcall,s,4)


Thanks,

Bruce
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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

[bruce bruce]

Hi Mike,

I am putting the phones on AC Adapter now as I am suspecting the Linksys POE
switch. Once that test is done and if problem still presists, I will be
enabling DHCPMasq and also set the SIP registration time to 1 second on the
phone UI.

-Bruce

On Tue, Aug 3, 2010 at 1:13 PM, Mike  wrote:

>  Hi Bruce,
>
>
>
> Did you ever get a working solution and confirm the underlying issue ? I am
> having the same issue on a set of phones, my next step is replacing the
> router, but I was wondering if you found something else.
>
>
>
> Regards,
>
>
>
> Mike
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Thursday, July 29, 2010 22:36
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Aastra phones occasionally show "No Service" -
> Is there any network setting I can tamper to facilitate a quick DHCP renewal
> on the Aastra phones?
>
>
>
> Hi Everyone,
>
>
>
> I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The
> phones occasionally go into "No Service" mode. The POE switch doesn't seem
> to be the problem as it's tested fine. I think the router sometimes gives up
> and comes back quickly. Or something of that nature. However, the
> connections are maintained if a call is going on because there are peer to
> peer connections between the phones in a network. Anyhow, if the phones are
> restarted they work fine.
>
>
>
> So, I was looking around the Aastra Admin UI to find any timer to lower it
> to 1 second to check and make sure the device always has an ip but I can't
> seem to find anything other than LLDP which is set at 30 and I don't think
> that will be of any help.
>
>
>
> I did a test where I would disconnect the router from the switch and after
> a while phones go into "No Service" but if I plug it back into the switch
> the phones do not come back right away. Maybe something should be dialed on
> the phone or wait long time or restart it to work again.
>
>
>
> Any work around?
>
>
>
> Thanks a lot
>
> --
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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

Yep, I seen that. That is probably the closet thing but looking at he
interface it makes me not try to install it. Maybe too complicated. I
wouldn't want to send customer the whole CDRs but rather a nice Bill like
the telco sends out.

I am currently toying with NCH Invoicing. Those guys make a software for
anything and everything.

Thanks,
Bruce

On Tue, Aug 3, 2010 at 9:17 AM, Zeeshan Zakaria  wrote:

> I know someone who uses a billing solution called 'freeside', and is happy
> with it. Personally I developed my own solution because none could satisfy
> my needs.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-08-03 2:34 AM,  wrote:
>
> Hi,
>
>
> On 08-02-2010 20:55, Gordon Henderson wrote:
>
> > I generated invoices with PHP code - it uses a LaTe...
>
> well, then i must be a geek too, because i also decided to throw some
> php code together to generate PDFs from sql.  It was just quicker this
> way rather than looking and trying a buch of other software.  I'm not
> sure how other (real) softwares work but since then i'm not spending
> even a minute with invoices, it's all in crontab.
>
> regards
> adam
>
>
> --
> _
> -- Bandwidth and Colocati...
>
>
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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

Oh, you seem to be right on. It's actually an install of Elastix. I will be
testing this for sure. Hope it doesn't do any damages though.

I guess the installation material is inside the tar ball?

Thanks

On Tue, Aug 3, 2010 at 2:01 AM, Nasir Iqbal wrote:

> Hi Bruce,
>
> We have build an Invoicing module (ICTInovice) for Elastix. It is Free,
> Open Source, Generate PDF Invoices, and can mail invoices to clients!
>
> You can download it from http://sourceforge.net/projects/ictinvoice/
>
> <http://sourceforge.net/projects/ictinvoice/>Note: Currently ICTInvoice
> only work with Elastix 1.6
>
> Regards
>
> On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce  wrote:
>
>> Hi Everyone,
>>
>> Sorry, if it's not directly related to Asterisk. Some of people on this
>> list might have PBX deployed for their clients. What software do you use to
>> invoice them so the invoice looks like a proper telecom invoice maybe?
>>
>> Prefer:
>> -opensource with Windows binary available.
>> -able to create .pdf invoices rather than printable ones.
>>
>> Thanks
>>
>> --
>> _
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

I agree but the mentioned software is not opensource.
My conditions clearly included opensource.

On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown  wrote:

>   *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Tuesday, 3 August 2010 1:58 PM
> *To:* j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
>
> *Subject:* Re: [asterisk-users] What do you use for Invoicing?
>
>
>
> Maybe good but the first look brought me to a Pay version. Doesn't satisfy
> the opensource condition.
>
>
>
> thanks,
>
>
>
> Open Source software does not necessarily mean free software.
>
>
>
> Nick.
>
>
>
>
>
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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

Sorry, I am not familiar with them.

Wondering if any full package system out there does the job.

Thanks

On Mon, Aug 2, 2010 at 2:55 PM, Gordon Henderson

> wrote:

> On Mon, 2 Aug 2010, bruce bruce wrote:
>
> > Hi Everyone,
> >
> > Sorry, if it's not directly related to Asterisk. Some of people on this
> list
> > might have PBX deployed for their clients. What software do you use to
> > invoice them so the invoice looks like a proper telecom invoice maybe?
> >
> > Prefer:
> > -opensource with Windows binary available.
> > -able to create .pdf invoices rather than printable ones.
>
> I generated invoices with PHP code - it uses a LaTeX template which it
> fills in the gaps, then feeds it through LaTeX and dvi2pdf to generate
> PDFs.
>
> Bit of a geek solution though.
>
> Gordon
>
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Re: [asterisk-users] What do you use for Invoicing?

[bruce bruce]

Maybe good but the first look brought me to a Pay version. Doesn't satisfy
the opensource condition.

thanks,

On Mon, Aug 2, 2010 at 2:39 PM, Jeff LaCoursiere  wrote:

> On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote:
> > Hi Everyone,
> >
> >
> > Sorry, if it's not directly related to Asterisk. Some of people on
> > this list might have PBX deployed for their clients. What software do
> > you use to invoice them so the invoice looks like a proper telecom
> > invoice maybe?
> >
> >
> > Prefer:
> > -opensource with Windows binary available.
> > -able to create .pdf invoices rather than printable ones.
> >
>
> Its partially open source (you get the source to everything but the
> financial routines), and it runs on Unix rather than Windows, though you
> do have a web interface.  Checkout BillMax: www.billmax.com
>
> They have some extensions that create PDF invoices in "telecom" style.
> Its pretty powerful otherwise for doing any kind of recurring billing.
>
> I wrote the initial version, but I am not associated with the company
> anymore.
>
> j
>
> >
> > Thanks
> > --
> > _
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>
>
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[asterisk-users] What do you use for Invoicing?

[bruce bruce]

Hi Everyone,

Sorry, if it's not directly related to Asterisk. Some of people on this list
might have PBX deployed for their clients. What software do you use to
invoice them so the invoice looks like a proper telecom invoice maybe?

Prefer:
-opensource with Windows binary available.
-able to create .pdf invoices rather than printable ones.

Thanks
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Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

[bruce bruce]

2 users. So, it's probably never used as a free version as probably there
are no 2 seat call centers that can survive this economy. But, it should
great for testing.

On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen
wrote:

> On 7/30/2010 5:49 AM, Lenz Emilitri wrote:
> > QueueMetrics is actually free (as in beer) for very small call centers
> and
> > individual hackers.
>
> Oh really! I didn't know that! Very nice.
>
> What is considered a "small" call centre? Are we talking up to around 5
> agents or something? Is there a limit on the number of queues?
>
> (I'm sure there is a page on the website that answers most of these
> questions, heh :))
>
> Leif Madsen.
>
>
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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

[bruce bruce]

Now that I check again, I see that DNSMasq for DHCP and DNSMasq for DNS is
NOT enabled. Which one should I enable and also can you please detail what
DNSMasq really does?

With my situation, only some of the phones go off randomly. But it is a few
specific extensions I think. Not the server itself. At best, I can think of
a cable or two jacked improperly into the patch panel and that's all which
MAYBE the cause for failing of DNS.

Thanks,
Bruce

On Fri, Jul 30, 2010 at 12:03 PM, bruce bruce  wrote:

> DNSMasq has always been enabled. It's only one check box and if I didn't
> have it enabled phones won't work. So, that is set. Any other suggestions?
> including things regarding DNSMasq?
>
> Thanks
>
>
> On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton 
> wrote:
>
>> On 30/07/10 16:15, bruce bruce wrote:
>> > Adria,
>> >
>> > How can I build a dns cache into my lan? I am using a Linksys 48 port
>> > POE switch and running a micro DD-WRT firmware on a linksys router.
>> >
>>
>> DD-WRT supports DNSMasq which would do exactly what you need.
>>
>> DC
>>
>>
>>
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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

[bruce bruce]

DNSMasq has always been enabled. It's only one check box and if I didn't
have it enabled phones won't work. So, that is set. Any other suggestions?
including things regarding DNSMasq?

Thanks

On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton wrote:

> On 30/07/10 16:15, bruce bruce wrote:
> > Adria,
> >
> > How can I build a dns cache into my lan? I am using a Linksys 48 port
> > POE switch and running a micro DD-WRT firmware on a linksys router.
> >
>
> DD-WRT supports DNSMasq which would do exactly what you need.
>
> DC
>
>
>
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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

[bruce bruce]

Adria,

How can I build a dns cache into my lan? I am using a Linksys 48 port POE
switch and running a micro DD-WRT firmware on a linksys router.

Gareth,

I think the registration time is part of the reason. I have lowered it less
than 10 seconds.

Thanks

On Fri, Jul 30, 2010 at 8:21 AM, Adrià Vidal  wrote:

> try to have a dns cache into your LAN, Aastra phone are prone to fail when
> have any little DNS error.
>
>
> --
> --
> Adrià Vidal
>
>
>
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Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

[bruce bruce]

Is it easy to install along with FreePBX as well?

Thanks

On Fri, Jul 30, 2010 at 5:49 AM, Lenz Emilitri  wrote:

> QueueMetrics is actually free (as in beer) for very small call centers and
> individual hackers.
> l.
>
> 2010/7/28 Zeeshan Zakaria 
>
> There is none for free.
>>
>> Zeeshan A Zakaria
>>
>> --
>> www.ilovetovoip.com
>>
>> On 2010-07-27 6:12 PM, "bruce bruce"  wrote:
>>
>> :-) I knew someone would bring up FreePBX. I have FreePBX installed and
>> it's not good for Queues at all. It's using the reporting tool from Areski
>> and Areski has recently released an upgrade to it which again is not what I
>> want.
>>
>> There are few other programs that do this but really none that are neat in
>> interface or useful in features.
>>
>> I guess no one else has any thoughts on this? Maybe there is none
>> available?
>>
>> Thanks,
>> Bruce
>>
>>
>>
>> On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg 
>> wrote:
>> >
>> > On Mon, Jul 26...
>>
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>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
>
>
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Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?

[bruce bruce]

Thank Martin,

That makes absolute sense. I have turned busy detect off for now and haven't
heard complains or lines remaining open for a Day. I am in Canada. I just
checked chan_dahdi.conf and I don't see callprogress there at all. So, I
guess the lines are fine for hanging up by themselves. Hope this doesn't
give me probs in future.

Thanks,
Bruce

On Fri, Jul 30, 2010 at 6:18 AM, Martin  wrote:

> Either turn off busydetect or increase the busycount to 5-7 or even
> more ... (10 should be conservative)
> busydetect looks for cadence or patterns of the same length ... beep
> on [X ms] beep off [Y ms]
> so you can afford to increase busycount and have a few second longer
> calls / the line is kept longer offhook
> but you don't get false busy detections
>
> Also in US/Canada callprogress will do a better job then busydetect
> since it looks for specific frequencies of the busy signal
> and not just noise/beep then silence ... If you're somewhere else then
> you can hire a coder to tweak callprogress algorithm
> to your country's busy signal frequencies ... Just record the busy
> signal with ztmonitor and send to someone for code patch...
>
> regards
> Martin
>
> On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce  wrote:
> > Hmmwhat about call waiting?
> > You mean, when a call comes in on that specific line, it generate two
> beep
> > tones and hence the system hangs up thinking it's end of the call?
> > Interesting!!!
> > If it is call-waiting do I have to set all of the following off for it to
> > not give me problem again:
> > callwaiting=yes
> > usecallingpres=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > canpark=yes
> > cancallforward=yes
> > busydetect=yes
> > busycount=3
> > Please elaborate a bit if I am off-topic.
> > Regards,
> > Bruce
> > On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas 
> wrote:
> >>
> >> From: asterisk-users-boun...@lists.digium.com
> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce
> bruce
> >> Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden?
> >> Couldbusy detect be the problem?
> >>
> >>
> >>
> >> I am getting a complain that call on analogue lines (Sangoam A400D)
> drops
> >> all of a sudden. Here is what I see in logs:
> >>
> >>
> >>
> >> Could be callwaiting?
> >>
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Re: [asterisk-users] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1

[bruce bruce]

This was a static IP. Further checks into the server prevails that there are
no logs of what happened on the 24th and 25th even in /var/log/messages.
This makes me believe that a hardware lockup has happened and according to
people on CentOS forum this is VERY HARD to diagnose as there will be no
logs. Even MS Blue Screed Of Death does a better job of logging at instances
like this :-(

On Thu, Jul 29, 2010 at 10:13 PM, Lyle Giese  wrote:

>  Lyle Giese wrote:
>
> bruce bruce wrote:
>
> I am not sure why it would be sleeping. I have never dealt with putting a
> linux server to sleep. It is connected to a UPS, but I don't think it has
> been put to sleep by the UPS as the USB cable from UPS is not connected to
> it.
>
>  Can you please elaborate on what you mean by AMI:Ping? Is there a service
> that you recommand that does this or are there any opensource monitoring
> tools out there that I can use?
>
>  But my main question remains why there are no activities on 24th and
> 25th?
>
>
>  This is what I see in the /var/log/messages.1:
>
>  Jul 23 17:11:55 elastix last message repeated 20 times
> Jul 23 17:22:51 elastix last message repeated 38 times
> Jul 23 17:30:39 elastix last message repeated 26 times
> Jul 23 17:30:39 elastix last message repeated 45 times
> Jul 23 19:09:42 elastix ntpd[3113]: synchronized to 216.216.216.216,
> stratum 2
> Jul 23 20:17:44 elastix ntpd[3113]: synchronized to 216.216.216.216,
> stratum 2
> Jul 23 21:29:16 elastix dhclient: DHCPREQUEST on eth0 to 192.168.1.254 port
> 67
> Jul 23 21:29:16 elastix dhclient: DHCPACK from 192.168.1.254
> Jul 23 21:29:16 elastix dhclient: bound to 192.168.1.100 -- renewal in
> 37640 seconds.
> Jul 26 09:22:37 elastix syslogd 1.4.1: restart.
> Jul 26 09:22:37 elastix kernel: klogd 1.4.1, log source = /proc/kmsg
> started.
> Jul 26 09:22:37 elastix kernel: Linux version 2.6.18-164.el5 (
> mockbu...@builder16.centos.org) (gcc version 4.1.2 20080704 (Red Hat
> 4.1.2-46)) #1 SMP Thu Se$
> Jul 26 09:22:37 elastix kernel: BIOS-provided physical RAM map:
>
>  Morning of the 26th at 9:22 the server was restarted because it was
> un-reachable from outside and hence the restart log but where is the 24th,
> and 25th?
>
>  Thanks,
> Bruce
>
> On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger <
> paul.belan...@polybeacon.com> wrote:
>
>> On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce  wrote:
>>  > See the jump from Jul 23rd to Jul 26th. Is this an indication of
>> Asterisk
>> > being down?
>> >
>>  No, it just means there was no logger activity for those days.  You
>> need to add a monitoring solution to your Asterisk box (IE: AMI:
>> Ping).
>>
>> --
>> Paul Belanger | dCAP
>> Polybeacon | Consultant
>> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
>> blog.polybeacon.com
>>
>> --
>>
>   It's 'well known' that Asterisk gets confused and runs around in a very
> tight loop when DNS resolution is failing.  Asterisk does a lot of DNS
> queries and when the Internet goes down, that puts Asterisk into a loop.
>
> Depending on your machine, I am guessing that Asterisk locked up or dropped
> out on the 23rd and the restart on the 26th brought it back to life.
>
> Nagios is a good choice for monitoring servers and services.  I use it here
> to monitor all the servers and SIP on my Asterisk box.
>
> Lyle Giese
> LCR Computer Services, Inc.
>
>  While the above comment about DNS holds, I also realized that most likely
> your Asterisk machine lost it's only ip address when the DSL went down.
> That may also have caused Asterisk to exit.  I think most(if not all) admins
> here would never have a dynamic ip address on an Asterisk server.
>
>
> Lyle Giese
> LCR Computer Services, Inc.
>
>
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[asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?

[bruce bruce]

Hi Everyone,

I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones
occasionally go into "No Service" mode. The POE switch doesn't seem to be
the problem as it's tested fine. I think the router sometimes gives up and
comes back quickly. Or something of that nature. However, the connections
are maintained if a call is going on because there are peer to peer
connections between the phones in a network. Anyhow, if the phones are
restarted they work fine.

So, I was looking around the Aastra Admin UI to find any timer to lower it
to 1 second to check and make sure the device always has an ip but I can't
seem to find anything other than LLDP which is set at 30 and I don't think
that will be of any help.

I did a test where I would disconnect the router from the switch and after a
while phones go into "No Service" but if I plug it back into the switch the
phones do not come back right away. Maybe something should be dialed on the
phone or wait long time or restart it to work again.

Any work around?

Thanks a lot
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Re: [asterisk-users] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1

[bruce bruce]

I am not sure why it would be sleeping. I have never dealt with putting a
linux server to sleep. It is connected to a UPS, but I don't think it has
been put to sleep by the UPS as the USB cable from UPS is not connected to
it.

Can you please elaborate on what you mean by AMI:Ping? Is there a service
that you recommand that does this or are there any opensource monitoring
tools out there that I can use?

But my main question remains why there are no activities on 24th and 25th?


This is what I see in the /var/log/messages.1:

Jul 23 17:11:55 elastix last message repeated 20 times
Jul 23 17:22:51 elastix last message repeated 38 times
Jul 23 17:30:39 elastix last message repeated 26 times
Jul 23 17:30:39 elastix last message repeated 45 times
Jul 23 19:09:42 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum
2
Jul 23 20:17:44 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum
2
Jul 23 21:29:16 elastix dhclient: DHCPREQUEST on eth0 to 192.168.1.254 port
67
Jul 23 21:29:16 elastix dhclient: DHCPACK from 192.168.1.254
Jul 23 21:29:16 elastix dhclient: bound to 192.168.1.100 -- renewal in 37640
seconds.
Jul 26 09:22:37 elastix syslogd 1.4.1: restart.
Jul 26 09:22:37 elastix kernel: klogd 1.4.1, log source = /proc/kmsg
started.
Jul 26 09:22:37 elastix kernel: Linux version 2.6.18-164.el5 (
mockbu...@builder16.centos.org) (gcc version 4.1.2 20080704 (Red Hat
4.1.2-46)) #1 SMP Thu Se$
 Jul 26 09:22:37 elastix kernel: BIOS-provided physical RAM map:

Morning of the 26th at 9:22 the server was restarted because it was
un-reachable from outside and hence the restart log but where is the 24th,
and 25th?

Thanks,
Bruce

On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger  wrote:

> On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce  wrote:
> > See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk
> > being down?
> >
> No, it just means there was no logger activity for those days.  You
> need to add a monitoring solution to your Asterisk box (IE: AMI:
> Ping).
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
> --
> _
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[asterisk-users] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1

[bruce bruce]

Hi Everyone,

This is probably more related to Linux than to Asterisk. Analogue channels
on a system were un-responsive on Monday morning. Apparently something
happened over the weekend and the router went off or it lost it's DSL
connection.

[Jul 23 22:50:01] VERBOSE[12437] logger.c: -- Remote UNIX connection
[Jul 23 22:50:01] VERBOSE[27087] logger.c: -- Remote UNIX connection
disconnected
[Jul 23 22:55:01] VERBOSE[12437] logger.c: -- Remote UNIX connection
[Jul 23 22:55:01] VERBOSE[27093] logger.c: -- Remote UNIX connection
disconnected
[Jul 23 23:00:01] VERBOSE[12437] logger.c: -- Remote UNIX connection
[Jul 23 23:00:02] VERBOSE[27099] logger.c: -- Remote UNIX connection
disconnected
[Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Event Logger Started
/var/log/asterisk/event_log
[Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Dynamic Loader Starting:
[Jul 26 09:22:59] VERBOSE[3529] logger.c:   == Parsing
'/etc/asterisk/modules.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c:
Found
[Jul 26 09:22:59] VERBOSE[3529] logger.c:   == Parsing
'/etc/asterisk/dnsmgr.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found
[Jul 26 09:22:59] VERBOSE[3529] logger.c:   == Parsing
'/etc/asterisk/http.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found

See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk
being down? I don't see any of that but yet no calls are on the report for
July 24th and 25th indicating to me that Analogue channels, or Asterisk, or
the server was down during this time as this office always receives calls on
the weekend to the IVR.

Where are the logs for eth0 so that I can check to see why this happened and
if indeed it was a drop in internet connection. If so, and this is the known
bug for Asterisk stop working due to internet drop, why is it not listed in
the log file posted above?

Thanks,
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Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?

[bruce bruce]

Furthermore, these are lines in Hunt, so, I am not sure if Call-Waiting is
turned ON on these lines at all. But it's definitely an interesting idea.

On Wed, Jul 28, 2010 at 5:54 PM, bruce bruce  wrote:

> Hmmwhat about call waiting?
> You mean, when a call comes in on that specific line, it generate two beep
> tones and hence the system hangs up thinking it's end of the call?
>
> Interesting!!!
>
> If it is call-waiting do I have to set all of the following off for it to
> not give me problem again:
> *callwaiting=yes*
> *usecallingpres=yes*
> *callwaitingcallerid=yes*
> *threewaycalling=yes*
> *transfer=yes*
> *canpark=yes*
> *cancallforward=yes*
> *busydetect=yes*
> *busycount=3*
>
> Please elaborate a bit if I am off-topic.
>
> Regards,
> Bruce
>
> On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas  wrote:
>
>>   *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
>> *Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden?
>> Couldbusy detect be the problem?
>>
>>
>>
>> I am getting a complain that call on analogue lines (Sangoam A400D) drops
>> all of a sudden. Here is what I see in logs:
>>
>>
>>
>> Could be callwaiting?
>>
>> --
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>
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Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?

[bruce bruce]

Hmmwhat about call waiting?
You mean, when a call comes in on that specific line, it generate two beep
tones and hence the system hangs up thinking it's end of the call?

Interesting!!!

If it is call-waiting do I have to set all of the following off for it to
not give me problem again:
*callwaiting=yes*
*usecallingpres=yes*
*callwaitingcallerid=yes*
*threewaycalling=yes*
*transfer=yes*
*canpark=yes*
*cancallforward=yes*
*busydetect=yes*
*busycount=3*

Please elaborate a bit if I am off-topic.

Regards,
Bruce

On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas  wrote:

>   *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden?
> Couldbusy detect be the problem?
>
>
>
> I am getting a complain that call on analogue lines (Sangoam A400D) drops
> all of a sudden. Here is what I see in logs:
>
>
>
> Could be callwaiting?
>
> --
> _
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[asterisk-users] Why do Zaptel calls drop all of a sudden? Could busy detect be the problem?

[bruce bruce]

Hi Guys,

I am getting a complain that call on analogue lines (Sangoam A400D) drops
all of a sudden. Here is what I see in logs:

[Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy,
avgtone: 75, avgsilence 135
[Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing
[...@macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new
stack
[Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing
[...@macro-hangupcall:1] GotoIf("SIP/2111-b6a400b0", "1?skiprg") in new stack
[Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Goto
(macro-hangupcall,s,4)


This is running 1.4.26.1 (Elastix)

Should I turn of busy detect in chan_dahdi.conf? or is this a known bug and
has a workaround?

Thanks,
Bruce
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Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

[bruce bruce]

:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's
not good for Queues at all. It's using the reporting tool from Areski and
Areski has recently released an upgrade to it which again is not what I
want.

There are few other programs that do this but really none that are neat in
interface or useful in features.

I guess no one else has any thoughts on this? Maybe there is none available?

Thanks,
Bruce

On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg wrote:

> On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce  wrote:
> > I seem to not be able to find any good open source Asterisk Queue
> Analyzer
> > and Asterisk Log Analyzer on the web.
>
> google 'freepbx'
>
> It does some of what you want. For the rest of what you want, strongly
> consider paying a professional consultant.
>
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[asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

[bruce bruce]

Hi Guys,

I seem to not be able to find any good open source Asterisk Queue Analyzer
and Asterisk Log Analyzer on the web.

The Asterisk Queue Analyzer is to serve as the graphic tool for call center
or pbx admins. It will pull the info in queue.log and in MySQL asterisk CDR
to create a graphic bar or to report on each extension that received the
queue calls, etc...

The Asterisk Log Analyzer is to analyze the log and to show any serious
errors or as bonus maybe send out e-mails to admin and to e-mail any
downtime during the day.

Please note that I am not talking about making my own scripts to analyze and
output this data as I know it exists in the system but rather am looking for
a project that has done it already.


Thanks,
Bruce
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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[bruce bruce]

I am having this issue with PRI. But I do not use conference rooms. Our
system is a simple queue and extensions.

-Bruce

On Fri, Jul 23, 2010 at 6:13 PM, Maurizio Faccio adinet <
mauf...@adinet.com.uy> wrote:

> You're right but it do not detect that I hungs on my side of the line.
> I think that in some way we are going into a conference in some unwanted
> way with the two dadhi channels and when i hang up both lines stay bridged.
> I think that the trouble appears when i dial a number in an analog
> phone, hook quickly (seems like a flash), and dial again.
>
> I am wondering that if I change my lines to a pri I solve this trouble
> but now I do not see clear at all. (my analog telco cannot bring me
> polarity reversal on hang up for signaling
>
> Thank you in advance
>
> Maurizio
>
>
>
> El 23/07/2010 06:09 p.m., Tzafrir Cohen escribió:
> > On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote:
> >
> >> I guess same trouble with Elastix 1.5.2-2.3
> >> dahdi 2.1.0.4  19
> >> Asterisk 1.4.25.1
> >> Digium TDM 2400
> >>
> > That's an analog card.
> >
> > With an analog trunk, you're not guaranteed to know if the remote CO has
> > hung up the line.
> >
> >
>
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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[bruce bruce]

Well, what about PRI? Why should this stay on? Isn't the native bridge just
a bridge channel that should go down automatically if the actually Dahdi/ZAP
channel is down and there are no SIP channels on either?

Thanks,
Bruce

On Fri, Jul 23, 2010 at 5:09 PM, Tzafrir Cohen wrote:

> On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote:
> >I guess same trouble with Elastix 1.5.2-2.3
> >dahdi 2.1.0.4  19
> > Asterisk 1.4.25.1
> > Digium TDM 2400
>
> That's an analog card.
>
> With an analog trunk, you're not guaranteed to know if the remote CO has
> hung up the line.
>
> --
>   Tzafrir Cohen
> 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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Re: [asterisk-users] POE Splitters

[bruce bruce]

You can also use Ethernet Over Power Lines solution or wireless :-)

On Fri, Jul 23, 2010 at 8:55 AM, David Backeberg wrote:

> On Fri, Jul 23, 2010 at 8:46 AM, Matt  wrote:
> > It's not necessarily this simple.  There is an approximately 50-75foot
> cable
> > run through ceilings and walls (CAT5) to the location where the phones
> will
> > be.  At the phone location there is no power.
>
> You always have options. You just have to decide what is more difficult:
>
> * moving the phone/devices somewhere else. Easiest solution.
> * having an electrician pull AC power to the location, then use DC
> power bricks or PoE switch
> * having a data cable person pull more ethernet to the location
>
> If you already have one ethernet cable that managed to make that 50-75
> foot run, then clearly it can be done, and a professional could even
> use that cable to yank three more along the same run, and then you're
> all set.
>
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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[bruce bruce]

This is running Elastix (FreePBX), so I am pretty sure there is Hangup()
requested. At least this doesn't happen ALL THE TIME. So, something is
getting stuck.

Thanks,
Bruce

On Fri, Jul 23, 2010 at 9:10 AM, Paul Belanger  wrote:

> On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce  wrote:
> > Any help is appreciated.
> >
> Are you explicitly calling Hangup() within your dial-plans?
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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[asterisk-users] Why does a bridged channel stay open for 4 hours?

[bruce bruce]

Hi Everyone,

Using a PRI with Sangoma A101D and Asterisk 1.4.2.x.

I notice that occasionally after a call is disconnected and both the phone
devices and the the channel is down but the bridge stays open for hours.

Channel  Location State   Application(Data)
Local/9054445...@fro (None)   Up  Bridged Call(Zap/4-1)


905444 is the inbound DID on the PRI. "core show channels" show Zap 4 in
use but if try to save the stream using "ztmonitor -f" it doesn't give me
anything because the channel is actually down. It's just the bridge that
stayed open for more than 4 hours.

Any help is appreciated.

Thanks,
Bruce
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Re: [asterisk-users] POE Splitters

[bruce bruce]

The Aastra 53i draws only 2 Watts from a Linksys 24 port POE switch. 25
phones is around 55 Watts.

-Bruce

On Thu, Jul 22, 2010 at 5:16 PM, Andrew Latham  wrote:

> The Snom 360 phone in front of me draws 4w...
>
>
> ~
> Andrew "lathama" Latham
> lath...@gmail.com
>
> * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
> * Learn more about Linux http://en.wikipedia.org/wiki/Linux
> * Learn more about Tux http://en.wikipedia.org/wiki/Tux
>
>
>
> On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons 
> wrote:
> > There is no such device -- it's outside of the POE spec.
> >
> > Class 3 devices are allowed to consume at max 15.4W. Most phones are
> class 3
> > devices. The math just doesn't work out. Even if you used the draft
> standard
> > for class 4 (~30W), you could still power max 2 devices at 15W/ea.
> >
> > -Dave
> >
> > On Thu, Jul 22, 2010 at 2:46 PM, Matt  wrote:
> >>
> >> I've got an interesting situation where I have one cable run from the
> feed
> >> area to the service area.   I have three devices that I need to power at
> the
> >> service area.  Is anyone aware of a device that will take the POE from
> the
> >> cable run and then allow me to split it to two or three devices at the
> >> service end?
> >>
> >> When I search for splitter all I get are the injectors, but I figure
> >> someone has to make something I realize I'll need a power adapter
> with
> >> enough amps to power the full load at the end.
> >>
> >> --
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Re: [asterisk-users] Does Flash Operator Panel allow for dragging a call into a parking lot?

[bruce bruce]

It's doable with a work around. Create a misc extension with followme set to
##70# which point to your parking lots and failed destination to Misc
parking extension.

Regards,
Bruce

On Sun, Jul 18, 2010 at 3:38 PM, Doug Lytle  wrote:

> bruce bruce wrote:
> > Hi Everyone,
> >
> > If I receive a call on a ZAP line and pickup the call and drag and
> > drop it (by mouse) into a Parking Lot through FOP, it just hangs up.
> > Is this feature supported by FOP?
> >
>
>
> I don't believe so, how would Asterisk know what phone to ring on timeout?
>
> 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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[asterisk-users] Does Flash Operator Panel allow for dragging a call into a parking lot?

[bruce bruce]

Hi Everyone,

If I receive a call on a ZAP line and pickup the call and drag and drop it
(by mouse) into a Parking Lot through FOP, it just hangs up. Is this feature
supported by FOP?

Thanks,
Bruce
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Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

[bruce bruce]

Yes, thanks. I think lots of manufacturers just boost the number of speakers
really needed but again this really depends on the environment noise level.

Regards,
Bruce

On Wed, Jul 14, 2010 at 11:50 PM, C F  wrote:

> I'm happy to hear it worked out so well with so little. :)
>
> On Wed, Jul 14, 2010 at 12:39 AM, bruce bruce  wrote:
> > Thanks for the input guys. For other refrence, a CyberData Voip Amplifier
> > which supplies 10 Watt to each of the two bogen 30 Watt speakers did the
> job
> > for a 35, square feet warehouse with environmental noise level of
> > slightly higher than standard but not those of industrial.
> > Only two speakers and done deal. Though I know that three speaker would
> have
> > been the perfect solution but 4 would cover every single little corner
> and
> > be an overkill.
> > -Bruce
> >
> > On Tue, Jul 13, 2010 at 8:47 PM, C F  wrote:
> >>
> >> I agree with horns you'll usually get better coverage. I have done
> >> this in the past with 5 speakers for a 30k sq ft warehouse very good
> >> coverage. Using bogen horns. This was for a 300ft by 100ft warehouse.
> >> Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft
> >> side I installed a horn every 60ft alternating facing one north and
> >> the other south, which ended up 3 facing one way and 2 the other. You
> >> can get double horn speakers which will face 2 sides. I wouldn't mount
> >> them on the wall specifically not so low as fork lifts and what not
> >> will damage them.
> >>
> >>
> >> On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce 
> wrote:
> >> > Well, these are horn speakers with 30 Watt which will receive 10 Watt
> >> > only
> >> > from Amplifer. I am not connecting them to ceiling so maybe 10 feet
> off
> >> > the
> >> > ground. I guess my coverage would be better???
> >> > Based on your calculations for for 40k sqfeet that would be 33
> speakers.
> >> > I
> >> > think that's way too much of an overkill.
> >> > thanks,
> >> > Bruce
> >> >
> >> > On Mon, Jul 12, 2010 at 1:05 AM, C F  wrote:
> >> >>
> >> >> In my experience using height for radius works, for example if you
> >> >> have a 20 ft high ceiling then the coverage for one speaker would be
> >> >> 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft
> >> >> has never killed anyone, but this really depends on the power of the
> >> >> speaker, I usually deal with 70v speakers tapped at 16 or 8 watts
> >> >> depending on how many speakers I put on one amplifier and the output
> >> >> wattage of that amplifier.
> >> >>
> >> >>
> >> >>
> >> >> On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce 
> >> >> wrote:
> >> >> > Hi Guys,
> >> >> > I am looking to buy a 25 Watt output CyberData VoIP amplifier and
> to
> >> >> > use
> >> >> > 2
> >> >> > Bogen sp308a speakers with it for a 40, 000 squar feet area and 21
> >> >> > feet
> >> >> > height. Is that enough? Is there calculator online I can use to
> >> >> > determine
> >> >> > the number of speakers needed? I guess these speakers go in chain
> so
> >> >> > I
> >> >> > am
> >> >> > not sure if the full capacity of the speaker (30 watt) will be
> used.
> >> >> > I appreciate your advice.
> >> >> > Thanks,
> >> >> > Bruce
> >> >> > --
> >> >> >
> _
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Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

[bruce bruce]

Thanks for the input but that won't be good because people are not going to
remember two extensions for one person.

The sip header should be able to carry alert_info to internal extensions
really easily. Anyone else got a thought?

Thanks again,

On Wed, Jul 14, 2010 at 5:44 PM, Ira  wrote:

> At 11:44 AM 7/14/2010, you wrote:
> >Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra
> >phones, how can one receive distinctive ring tones for INTERNAL calls
> ONLY?
>
> It's ugly, but you could give the phone two different SIP IDs and
> give those different ringtones.
>
> Ira
>
>
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[asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

[bruce bruce]

Hi Everyone,

Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
how can one receive distinctive ring tones for INTERNAL calls ONLY?

Even though FreePBX Inbound has an option for Alert_INFO but that doesn't
work when the call comes into an IVR or Queue. The calls has to go directly
to extension for external ringtone to be different. So, I am looking for
internal calls ringtones to be different rather than external call
ringtones.

Anyone has got this working?

Thanks,
Burce
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Re: [asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

[bruce bruce]

Thanks for the input guys. I don't use .xml files for Aastra. Everything is
done on the UI.

#voicemail show users:

*ContextMbox  User  Zone   NewMsg*
*|default007   Alex 2*
*default2100  Peter1*
*
*
This system is using FreePBX, so I checked the Device and Users in asterisk
tables and they have "default" for voicemail setup which I think is right. I
can also see the msg.txt in a folder that has new voicemail waiting.
However, I am not sure about privileges though. Here is it:

[r...@elastix INBOX]# ls -la
total 284
drwxrwxr-x 2 asterisk asterisk   4096 Jul 14 09:57 .
drwxrwxr-x 8 asterisk asterisk   4096 Jul 13 11:57 ..
*-rw-rw-r-- 1 asterisk asterisk282 Jul 12 18:46 msg.txt*
-rwxrwxr-x 1 asterisk asterisk  83244 Jul 12 18:46 msg.wav
-rwxrwxr-x 1 asterisk asterisk   8510 Jul 12 18:46 msg.WAV
*-rw--- 1 asterisk asterisk261 Jul 14 09:57 msg0001.txt*
-rwx-- 1 asterisk asterisk 150124 Jul 14 09:57 msg0001.wav
-rwx-- 1 asterisk asterisk  15270 Jul 14 09:57 msg0001.WAV

Thanks,
Bruce

On Wed, Jul 14, 2010 at 12:25 PM, Steve Johnson  wrote:

> On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce  wrote:
> > Hi Guys,
> > Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i,
> and
> > 6730i, but none of them indicate the voic-email. Where should I look for
> > trouble to find the root issue for MWI?
>
> (1) Check from the CLI> voicemail show users
>
> to ensure that the proper mailboxes have been set up and there is new
> mail in them.  If this is not right, check the voicemail.conf entry
> for this mailbox.
>
> (2) Check the phone device configuration (in sip.conf) to ensure that
> the phone has a mailbox=xxx entry.
>
> for example:
>
> ;entry in sip.conf for extension 115
> [115]
> context=yourcontext
> mailbox=115
> ...
>
> Restart asterisk if you've made changes and re-test.
>
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[asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

[bruce bruce]

Hi Guys,

Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
6730i, but none of them indicate the voic-email. Where should I look for
trouble to find the root issue for MWI?

Thanks,
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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[bruce bruce]

I am stuck with the same problem but I have used asterisk yum repository and
it worked by itself without me worrying for kernel stuff.

However, I need to install speex codec and now I am stuck as it doesn't get
picked up by the yum asterisk install somehow. I have lib speex and speex
already installed and when doing "yum install asterisk16" I don't see speex
in "core show translation" Is there anything specific I have to do?

Do I have to build from source as well?

-Sorry, didn't mean to hijack the thread.

Thanks,
Bruce

On Wed, Jul 14, 2010 at 5:08 AM, Chandrakant Solanki <
solanki.chandrak...@gmail.com> wrote:

> Hi
>
> If you install rpm from any location it goes to its default location.
>
> You just go for above steps. For kernel you can go for http://kernel.org
>
> --
> Regards,
>
> Chandrakant Solanki
>
>
> On Wed, Jul 14, 2010 at 2:06 PM, liuxin  wrote:
>
>> Hi.
>> The best easy way is:
>> copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
>> then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm
>>
>> 2010/7/14 Gareth Blades 
>>
>>  Thermal Wetland wrote:
>>> > I have a virtual server with godaddy but can not compile DAHDI as it
>>> > complains that I do not have the correct kernel source.
>>> >
>>> > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
>>> > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
>>> > latest version
>>> > Nothing to do
>>> >
>>> > uname -a returns:
>>> > Linux 
>>> > ip-XXX-XXX-XXX-XXX.ip.secureserver.net
>>> > >
>>> 2.6.18-028stab064.7 #1
>>> > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
>>> >
>>> > When I try to compile DAHDI it fails with:
>>> > make[2]: Leaving directory
>>> >
>>> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
>>> > You do not appear to have the sources for the 2.6.18-028stab064.7
>>> kernel
>>> > installed.
>>> >
>>> > Is there a way to trick DAHDI to use the installed kernel?
>>> >
>>> > Thanks for the help!
>>> >
>>> > --
>>> > -Thermal
>>> >
>>>
>>> What kernel versions do you have installed?
>>>
>>> If you are currently running an older kernel but installed a newer
>>> kernel and sources but havent rebooted to activate the new one yet then
>>> it may still be trying to locate the source for the older running kernel.
>>>
>>>
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>>
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Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

[bruce bruce]

Thanks for the input guys. For other refrence, a CyberData Voip Amplifier
which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job
for a 35, square feet warehouse with environmental noise level of
slightly higher than standard but not those of industrial.

Only two speakers and done deal. Though I know that three speaker would have
been the perfect solution but 4 would cover every single little corner and
be an overkill.

-Bruce

On Tue, Jul 13, 2010 at 8:47 PM, C F  wrote:

> I agree with horns you'll usually get better coverage. I have done
> this in the past with 5 speakers for a 30k sq ft warehouse very good
> coverage. Using bogen horns. This was for a 300ft by 100ft warehouse.
> Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft
> side I installed a horn every 60ft alternating facing one north and
> the other south, which ended up 3 facing one way and 2 the other. You
> can get double horn speakers which will face 2 sides. I wouldn't mount
> them on the wall specifically not so low as fork lifts and what not
> will damage them.
>
>
> On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce  wrote:
> > Well, these are horn speakers with 30 Watt which will receive 10 Watt
> only
> > from Amplifer. I am not connecting them to ceiling so maybe 10 feet off
> the
> > ground. I guess my coverage would be better???
> > Based on your calculations for for 40k sqfeet that would be 33 speakers.
> I
> > think that's way too much of an overkill.
> > thanks,
> > Bruce
> >
> > On Mon, Jul 12, 2010 at 1:05 AM, C F  wrote:
> >>
> >> In my experience using height for radius works, for example if you
> >> have a 20 ft high ceiling then the coverage for one speaker would be
> >> 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft
> >> has never killed anyone, but this really depends on the power of the
> >> speaker, I usually deal with 70v speakers tapped at 16 or 8 watts
> >> depending on how many speakers I put on one amplifier and the output
> >> wattage of that amplifier.
> >>
> >>
> >>
> >> On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce 
> wrote:
> >> > Hi Guys,
> >> > I am looking to buy a 25 Watt output CyberData VoIP amplifier and to
> use
> >> > 2
> >> > Bogen sp308a speakers with it for a 40, 000 squar feet area and 21
> feet
> >> > height. Is that enough? Is there calculator online I can use to
> >> > determine
> >> > the number of speakers needed? I guess these speakers go in chain so I
> >> > am
> >> > not sure if the full capacity of the speaker (30 watt) will be used.
> >> > I appreciate your advice.
> >> > Thanks,
> >> > Bruce
> >> > --
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> >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> >> >   http://www.asterisk.org/hello
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> >> > To UNSUBSCRIBE or update options visit:
> >> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >
> >>
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[asterisk-users] How to install speex codec for Asterisk that is downloaded from Digium Yum Repository?

[bruce bruce]

Hi Everyone,

I have done "yum install speex libspeex-devel speex-devel" and it was
succesful on CentOS. I then tried "yum install asterisk16 asterisk16-addons
asterisk16-configs" but "core show translation" doesn't show speex loaded.
Is there a way to or an option that I can append to the asterisk install to
make sure it compiles with speex in mind?

Thanks,
Bruce
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Re: [asterisk-users] My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?

[bruce bruce]

Thanks for that Tim. *Wondering how I can trigger that reload?* I have tried
dialplan reload and reload but that doesn't work. Obviously amportal reload
wouldn't be doable in this case even if it works because the system will go
down.

Thanks,
Bruce

On Mon, Jul 12, 2010 at 2:13 PM, Tim Nelson  wrote:

> - "bruce bruce"  wrote:
> > Hi Everyone,
>
> >
> I have done some php coding to come up with my own FollowME module for
> FreePBX. The need for this has some security considerations behind it.
> >
> This is what my code does at core:
>
> >
> $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre,
> grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id,
> ringing, pre_ring) VALUES
> ('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]','$grplist','0','$postdest','','','0','0','Ring','$_POST[pre_ring]')";
>
> This all conforms with the fields that are filled up by FreePBX followme
> module but it seems that this is not all becuase the followme doesn't work
> when I do it this way. It only works if I press submit and confirm the
> orange bar.
> >
> For one thing, I think the Orange Reload bar does something that I can't
> seem to find and that my php code doesn't do. I tried doing a manual
> "reload" and "dialplan reload" but that wouldn't do the job.
>
> >
> Can someone please shed some light if you know where I am stuck and had to
> tackle the issue yourself?
>
> >
> Thanks,
> Bruce
>
> --
>
> FreePBX development may be best discussed on the FreePBX forums [1].
>
> As an aside, the 'Orange Reload Bar' takes all of the information in the
> FreePBX MySQL database, generates the dialplan code, and then replaces your
> extensions.conf, sip.conf, iax.conf, etc with the appropriate information.
>
> [1] http://www.freepbx.org/forums
> <http://www.freepbx.org/forums>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
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Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

[bruce bruce]

Well, these are horn speakers with 30 Watt which will receive 10 Watt only
from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the
ground. I guess my coverage would be better???

Based on your calculations for for 40k sqfeet that would be 33 speakers. I
think that's way too much of an overkill.

thanks,
Bruce

On Mon, Jul 12, 2010 at 1:05 AM, C F  wrote:

> In my experience using height for radius works, for example if you
> have a 20 ft high ceiling then the coverage for one speaker would be
> 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft
> has never killed anyone, but this really depends on the power of the
> speaker, I usually deal with 70v speakers tapped at 16 or 8 watts
> depending on how many speakers I put on one amplifier and the output
> wattage of that amplifier.
>
>
>
> On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce  wrote:
> > Hi Guys,
> > I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use
> 2
> > Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet
> > height. Is that enough? Is there calculator online I can use to determine
> > the number of speakers needed? I guess these speakers go in chain so I am
> > not sure if the full capacity of the speaker (30 watt) will be used.
> > I appreciate your advice.
> > Thanks,
> > Bruce
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >   http://www.asterisk.org/hello
> >
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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[asterisk-users] My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?

[bruce bruce]

Hi Everyone,

I have done some php coding to come up with my own FollowME module for
FreePBX. The need for this has some security considerations behind it.
This is what my code does at core:

$sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
pre_ring) VALUES
('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]','$grplist','0','$postdest','','','0','0','Ring','$_POST[pre_ring]')";

This all conforms with the fields that are filled up by FreePBX followme
module but it seems that this is not all becuase the followme doesn't work
when I do it this way. It only works if I press submit and confirm the
orange bar.
For one thing, I think the Orange Reload bar does something that I can't
seem to find and that my php code doesn't do. I tried doing a manual
"reload" and "dialplan reload" but that wouldn't do the job.

Can someone please shed some light if you know where I am stuck and had to
tackle the issue yourself?

Thanks,
Bruce
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Re: [asterisk-users] How can get user inputs from called party after dial?

[bruce bruce]

I was under the impression that he is new to Asterisk. No need to fuss.
Hence the ":-)"

On Sat, Jul 10, 2010 at 3:35 PM, Steve Edwards wrote:

> On Sat, 10 Jul 2010, bruce bruce wrote:
>
> > You need to do some reading :-)
>
> Now that is funny -- maybe you could take your own advice and look at
>
>http://www.php.net/docs.php
>
> instead of posting "please help me debug code I'm too lazy to even see if
> PHP says it is syntactically correct and the only relevance it has to
> Asterisk is I'm trying to concatenate some strings and make sure it could
> be a phone number" requests.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] How can get user inputs from called party after dial?

[bruce bruce]

For dial you do this:

[first-Dialplan]
exten => s,1,Answer
exten => s,n,Dial(SIP/provider/111222)
exten => s,n,Playback(Welcome)
exten => s,n,Read(numb,,10)
exten => s,n,NoOp(${numb})

-Bruce

On Sat, Jul 10, 2010 at 2:51 PM, bruce bruce  wrote:

> You need to do some reading :-)
>
> I will give you a quick teach here. At the end of file
> /etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
> /etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
> this:
>
> [first-Dialplan]
> exten => s,1,Answer
> exten => s,n,Playback(Welcome)
> exten => s,n,Read(numb,,10)
> exten => s,n,NoOp(${numb})
>
> And send your inbound route to context first-Dialplan so that it's
> triggered when a call comes in. Then on terminal do a "asterisk -r"
> and you will see the NoOp show the DTMF number entered. From there on you
> can do anything you want with the variable ${numb}
>
> If any part of above is unclear to you, you must consult your friend,
> google, for examples of Asterisk dialplan.
>
> -Bruce
>
>
> On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman wrote:
>
>>  Thanks, but I'm missing something here, the dial command is where?
>>
>>
>>
>> I need to do something like:
>>
>> Dial(1234)
>>
>> Read(1 digit)
>>
>> DoSomthing(based on digit from 1234)
>>
>>
>>
>> And as far as I understand the Dial start the call and only come back (ig
>> you use the g option) after call finished.
>>
>>
>>
>> Eyal
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
>> *Sent:* Saturday, July 10, 2010 9:30 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] How can get user inputs from called party
>> after dial?
>>
>>
>>
>> You need read():
>>
>> http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
>>
>>
>>
>> It's as easy as:
>>
>>
>>
>> exten => s,n,Read(variable,,11)
>>
>> exten => s,n,NoOp(${variable})
>>
>>
>>
>> Above will take up to 11 digits input by user and will display it back in
>> NoOP on Asterisk CLI.
>>
>>
>>
>> -Bruce
>>
>> On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman 
>> wrote:
>>
>> Hi,
>>
>> I want to dial a party, play him a message and wait for his input, i.e.
>> DTMF digits and use them to control the rest of the dial plan.
>>
>>
>>
>> How do I do it?
>>
>>
>>
>> If I use Dial it will not return until the end of the call, isn't it?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Eyal
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> No virus found in this incoming message.
>> Checked by AVG - www.avg.com
>> Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
>> 09:36:00
>>
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>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
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Re: [asterisk-users] How can get user inputs from called party after dial?

[bruce bruce]

You need to do some reading :-)

I will give you a quick teach here. At the end of file
/etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
/etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
this:

[first-Dialplan]
exten => s,1,Answer
exten => s,n,Playback(Welcome)
exten => s,n,Read(numb,,10)
exten => s,n,NoOp(${numb})

And send your inbound route to context first-Dialplan so that it's triggered
when a call comes in. Then on terminal do a "asterisk -r" and you
will see the NoOp show the DTMF number entered. From there on you can do
anything you want with the variable ${numb}

If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.

-Bruce

On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman  wrote:

>  Thanks, but I'm missing something here, the dial command is where?
>
>
>
> I need to do something like:
>
> Dial(1234)
>
> Read(1 digit)
>
> DoSomthing(based on digit from 1234)
>
>
>
> And as far as I understand the Dial start the call and only come back (ig
> you use the g option) after call finished.
>
>
>
> Eyal
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Saturday, July 10, 2010 9:30 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] How can get user inputs from called party
> after dial?
>
>
>
> You need read():
>
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
>
>
>
> It's as easy as:
>
>
>
> exten => s,n,Read(variable,,11)
>
> exten => s,n,NoOp(${variable})
>
>
>
> Above will take up to 11 digits input by user and will display it back in
> NoOP on Asterisk CLI.
>
>
>
> -Bruce
>
> On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman 
> wrote:
>
> Hi,
>
> I want to dial a party, play him a message and wait for his input, i.e.
> DTMF digits and use them to control the rest of the dial plan.
>
>
>
> How do I do it?
>
>
>
> If I use Dial it will not return until the end of the call, isn't it?
>
>
>
> Thanks,
>
>
>
> Eyal
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
> 09:36:00
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] PHP can't insert - Can someone please help

[bruce bruce]

Here is the steel strong sanitizer:

$npaa = "$_POST[anpa]";
$nxxa = "$_POST[anxx]";
$blocka = "$_POST[ablock]";

# Sanitize
$blocka_san = strspn("$blocka", "0123456789");

*if ($blocka_san==4 && is_numeric($npaa) && is_numeric($nxxa) &&
is_numeric($blocka) && $npaa>=200 && $nxxa>=200 && $npaa!=900 &&
$npaa!=911) *
*
*
*  {*

  echo "Number passed sanitization";

  }

What do you think? :-)

-Bruce
On Sat, Jul 10, 2010 at 2:17 PM, bruce bruce  wrote:

> Thanks again. Apparently all POST variables come through as strings. The
> function you pointed out is I think already built in php as
>
> is_numeric() <http://www.php.net/manual/en/function.is-numeric.php>.
>
> <http://php.net/manual/en/function.is-int.php>
> http://www.php.net/manual/en/function.is-numeric.php
>
> <http://www.php.net/manual/en/function.is-numeric.php>If that runs TRUE
> and if I keep my >=200 and !=911 or !900 I should be safe from SQL
> injections. And along with dial-out routes rules, I think I can make this
> stronger.
>
> I have my html/php file set so that the input field only takes 3 digit 3
> digit 4 digit (NPA, NXX, Block) so your purposal of: *'201,0); drop
> database YOUR_DATABASE'; *would fail due to big length and also I tested
> with inputing letters and my IF function caught it and exited.
>
> Further more, everything else (other than phone input fields) is drop down
> boxes with specific numbers or letters inserted in them. I should be 100%
> safe with those right?
>
> By using form POST there should be no other loop holes left opened right?
> It's not like php $_GET so people can't try typing to the browser in this
> format:
>
> http://www.w3schools.com/welcome.php?fname=Peter&age=37
>
> Thanks a lot,
> Bruce
>
> On Sat, Jul 10, 2010 at 1:41 PM, Gerald A  wrote:
>
>> Hi Bruce,
>>
>> On Sat, Jul 10, 2010 at 11:12 AM, bruce bruce wrote:
>>
>>> Further to my last post, I added this to santize. I also created a new
>>> mysql user with access to only findmefollow portion of the asterisk table
>>> for limited access and assigned only two simultaneous connections with only
>>> 10 changes queries per hour (as I know that no more queries will be put
>>> through probably)
>>>
>>> if ($npaa>=200 && $nxxa>=200 && $npaa!=900 && $npaa!=911)
>>>
>>> Should that suffice against SQL injections? The if condition changes the
>>> string to number so it removes the chance of people adding
>>> other characters and it also sticks to format NPAN or 2XX2.
>>>
>>
>> There are two things -- the first is, who call this script? If it's
>> something you control 100%, you can mitigate the risk a bit. I don't really
>> like this tact, because if the script gets repurposed, you end up with
>> something that could be very dangerous.
>>
>> The second thing is simple -- most people think small here, but you have
>> to think big and know a bit about how PHP works. PHP strings are pretty
>> amazing things, and one of the pesky things is that you can put all kinds of
>> things in it. Now, if that string variable is created as a result of a form
>> input, then that string can be anything. For a moment, think about if it
>> $npaa = '201,0); drop database YOUR_DATABASE'; Now, that is pretty nasty,
>> and it would muck up further SQL injections, but now you get the idea. You
>> should always check to make sure the data you are getting is what you are
>> expecting, and exclude what you aren't.
>>
>> So, are your tests sufficient? I can't remember off the top of my head if
>> the string -> integer only considers the first number, or it considers the
>> whole string. (PHP usually errs on the side of ease of use, so I think my
>> snippet above would still pass your test). If your expecting only numbers,
>> I'd write a function that ensures that only numbers are parts of the input.
>> (And not just for the 3 above variables).
>> Really, you should never see $_POST("var") (or any PHP CGI variable) that
>> derives directly from user input.
>>
>> It takes a few minutes extra, but it'll save hours of sorting later if you
>> get hit by a SQL injection.
>>
>> Hope this helps,
>> Gerald
>>
>
>
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Re: [asterisk-users] How can get user inputs from called party after dial?

[bruce bruce]

You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read

It's as easy as:

exten => s,n,Read(variable,,11)
exten => s,n,NoOp(${variable})

Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.

-Bruce

On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman  wrote:

> Hi,
> I want to dial a party, play him a message and wait for his input, i.e.
> DTMF digits and use them to control the rest of the dial plan.
>
> How do I do it?
>
> If I use Dial it will not return until the end of the call, isn't it?
>
> Thanks,
>
> Eyal
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
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Re: [asterisk-users] PHP can't insert - Can someone please help

[bruce bruce]

Thanks again. Apparently all POST variables come through as strings. The
function you pointed out is I think already built in php as

is_numeric() <http://www.php.net/manual/en/function.is-numeric.php>.

<http://php.net/manual/en/function.is-int.php>
http://www.php.net/manual/en/function.is-numeric.php

<http://www.php.net/manual/en/function.is-numeric.php>If that runs TRUE and
if I keep my >=200 and !=911 or !900 I should be safe from SQL injections.
And along with dial-out routes rules, I think I can make this stronger.

I have my html/php file set so that the input field only takes 3 digit 3
digit 4 digit (NPA, NXX, Block) so your purposal of: *'201,0); drop database
YOUR_DATABASE'; *would fail due to big length and also I tested with
inputing letters and my IF function caught it and exited.

Further more, everything else (other than phone input fields) is drop down
boxes with specific numbers or letters inserted in them. I should be 100%
safe with those right?

By using form POST there should be no other loop holes left opened right?
It's not like php $_GET so people can't try typing to the browser in this
format:

http://www.w3schools.com/welcome.php?fname=Peter&age=37

Thanks a lot,
Bruce

On Sat, Jul 10, 2010 at 1:41 PM, Gerald A  wrote:

> Hi Bruce,
>
> On Sat, Jul 10, 2010 at 11:12 AM, bruce bruce  wrote:
>
>> Further to my last post, I added this to santize. I also created a new
>> mysql user with access to only findmefollow portion of the asterisk table
>> for limited access and assigned only two simultaneous connections with only
>> 10 changes queries per hour (as I know that no more queries will be put
>> through probably)
>>
>> if ($npaa>=200 && $nxxa>=200 && $npaa!=900 && $npaa!=911)
>>
>> Should that suffice against SQL injections? The if condition changes the
>> string to number so it removes the chance of people adding
>> other characters and it also sticks to format NPAN or 2XX2.
>>
>
> There are two things -- the first is, who call this script? If it's
> something you control 100%, you can mitigate the risk a bit. I don't really
> like this tact, because if the script gets repurposed, you end up with
> something that could be very dangerous.
>
> The second thing is simple -- most people think small here, but you have to
> think big and know a bit about how PHP works. PHP strings are pretty amazing
> things, and one of the pesky things is that you can put all kinds of things
> in it. Now, if that string variable is created as a result of a form input,
> then that string can be anything. For a moment, think about if it $npaa =
> '201,0); drop database YOUR_DATABASE'; Now, that is pretty nasty, and it
> would muck up further SQL injections, but now you get the idea. You should
> always check to make sure the data you are getting is what you are
> expecting, and exclude what you aren't.
>
> So, are your tests sufficient? I can't remember off the top of my head if
> the string -> integer only considers the first number, or it considers the
> whole string. (PHP usually errs on the side of ease of use, so I think my
> snippet above would still pass your test). If your expecting only numbers,
> I'd write a function that ensures that only numbers are parts of the input.
> (And not just for the 3 above variables).
> Really, you should never see $_POST("var") (or any PHP CGI variable) that
> derives directly from user input.
>
> It takes a few minutes extra, but it'll save hours of sorting later if you
> get hit by a SQL injection.
>
> Hope this helps,
> Gerald
>
-- 
_
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Re: [asterisk-users] PHP can't insert - Can someone please help

[bruce bruce]

Further to my last post, I added this to santize. I also created a new mysql
user with access to only findmefollow portion of the asterisk table for
limited access and assigned only two simultaneous connections with only 10
changes queries per hour (as I know that no more queries will be put through
probably)

if ($npaa>=200 && $nxxa>=200 && $npaa!=900 && $npaa!=911)

Should that suffice against SQL injections? The if condition changes the
string to number so it removes the chance of people adding
other characters and it also sticks to format NPAN or 2XX2.

Thanks

On Sat, Jul 10, 2010 at 10:21 AM, bruce bruce  wrote:

> Thank you for the amazing reply. First few lines of your e-mail was EXACTLY
> getting me to where I made a mistake. I guess I didn't take the () and ' '
> at their face value and was looking somewhere else for the problem.
>
> For sanatizing you mean checking the numbers to make sure they are valid
> numbers and not alphabet or other charecters? or, are you pointing the fact
> that I am keeping mysql root password in plain .php file? I have done an
> include of a php file which has mysql root password and that is insert as an
> #incldue in the html file. So, if someone checks source for html can't see
> mysql root password. Even though root is user on mysql is to accept only
> from localhost.
>
> I would really appreciate it if you can weigh in on it a bit.
>
> Thanks,
> Bruce
>
>
> On Sat, Jul 10, 2010 at 7:42 AM, Gerald A  wrote:
>
>> Hi Bruce,
>>
>> First, your problem isn't PHP, it seems to be SQL and I'm guessing MySQL
>> at that.
>>
>> Next, you seem to be accepting user input and not sanatizing it. DANGER
>> WILL ROBINSON!!!
>> This is bad, because it leaves you open to something known as a "SQL
>> injection attack".
>>
>> Now, as to syntax:
>>
>> On Sat, Jul 10, 2010 at 12:07 AM, bruce bruce wrote:
>>
>>>
>>> I am making another module for Voicemail. I have three fields in a POST
>>> form that have to be connected together to make it a single 10 digit number
>>> but there is something wrong in my syntax probably.
>>>
>>>
>>> $npaa = "('$_POST[anpa]')";
>>> $nxxa = "('$_POST[anxx]')";
>>> $blocka = "('$_POST[ablock]')";
>>>
>>> *$grplist = $npaa.$nxxa.$blocka;*
>>>
>>
>> Ok, so suppose arpa=111, anxx=222 and ablock=.
>> grplist would then be ('111')('333')('').
>>
>>  $sql="INSERT INTO findmefollow(grpnum, strategy, grptime, grppre,
>>> grplist, annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id,
>>> ringing, pre_ring)
>>> VALUES 
>>> ('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]',$grplist,'0','$_POST[postdest]','','','0','0','Ring','$_POST[pre_ring]')";
>>>
>>>
>>> It seems that $grplist is the problem. Can someone please point what is
>>> wrong?
>>>
>>> Error:
>>> Error: You have an error in your SQL syntax; check the manual that
>>> corresponds to your MySQL server version for the right syntax to use near
>>> '('333')(''),'0','ext-local,vmb2000,1','','','0','0','Ring','0')' at
>>> line 3
>>>
>>
>> Look closesly, grasshopper. See it? (Does the hint above help?) Hmmm, ok.
>>
>> Let's write the line as SQL:
>> INSERT INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
>> annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
>> pre_ring)
>> VALUES 
>> ('0','ringall','0','0',('111')('333')(''),'0','0','','','0','0','Ring','0')";
>>
>> Clear now? You are trying to insert the raw value -->
>> ('111')('333')('') <-- into your database. This can't make any sense
>> except as string, And this isn't one.
>>
>> I think what you might have meant is to quote the _whole thing_ as a
>> string, and not the individual pieces. Then:
>> $grplist = "'(".$npaa.$nxxa.$blocka.")'";
>> and
>> $blocka = "($_POST[ablock])";  # and for all of them above
>>
>> This would make the value '(111)(333)()', which should work fine.
>>
>> Now, if you really meant to add in the quotes, you'll have to "quote the
>> quotes", which can be hard to do in good times.
>>
>> Hope this helps,
>> Gerald.
>>
>
>
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Re: [asterisk-users] PHP can't insert - Can someone please help

[bruce bruce]

Thank you for the amazing reply. First few lines of your e-mail was EXACTLY
getting me to where I made a mistake. I guess I didn't take the () and ' '
at their face value and was looking somewhere else for the problem.

For sanatizing you mean checking the numbers to make sure they are valid
numbers and not alphabet or other charecters? or, are you pointing the fact
that I am keeping mysql root password in plain .php file? I have done an
include of a php file which has mysql root password and that is insert as an
#incldue in the html file. So, if someone checks source for html can't see
mysql root password. Even though root is user on mysql is to accept only
from localhost.

I would really appreciate it if you can weigh in on it a bit.

Thanks,
Bruce

On Sat, Jul 10, 2010 at 7:42 AM, Gerald A  wrote:

> Hi Bruce,
>
> First, your problem isn't PHP, it seems to be SQL and I'm guessing MySQL at
> that.
>
> Next, you seem to be accepting user input and not sanatizing it. DANGER
> WILL ROBINSON!!!
> This is bad, because it leaves you open to something known as a "SQL
> injection attack".
>
> Now, as to syntax:
>
> On Sat, Jul 10, 2010 at 12:07 AM, bruce bruce  wrote:
>
>>
>> I am making another module for Voicemail. I have three fields in a POST
>> form that have to be connected together to make it a single 10 digit number
>> but there is something wrong in my syntax probably.
>>
>>
>> $npaa = "('$_POST[anpa]')";
>> $nxxa = "('$_POST[anxx]')";
>> $blocka = "('$_POST[ablock]')";
>>
>> *$grplist = $npaa.$nxxa.$blocka;*
>>
>
> Ok, so suppose arpa=111, anxx=222 and ablock=.
> grplist would then be ('111')('333')('').
>
>  $sql="INSERT INTO findmefollow(grpnum, strategy, grptime, grppre,
>> grplist, annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id,
>> ringing, pre_ring)
>> VALUES 
>> ('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]',$grplist,'0','$_POST[postdest]','','','0','0','Ring','$_POST[pre_ring]')";
>>
>>
>> It seems that $grplist is the problem. Can someone please point what is
>> wrong?
>>
>> Error:
>> Error: You have an error in your SQL syntax; check the manual that
>> corresponds to your MySQL server version for the right syntax to use near
>> '('333')(''),'0','ext-local,vmb2000,1','','','0','0','Ring','0')' at
>> line 3
>>
>
> Look closesly, grasshopper. See it? (Does the hint above help?) Hmmm, ok.
>
> Let's write the line as SQL:
> INSERT INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
> annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
> pre_ring)
> VALUES 
> ('0','ringall','0','0',('111')('333')(''),'0','0','','','0','0','Ring','0')";
>
> Clear now? You are trying to insert the raw value -->
> ('111')('333')('') <-- into your database. This can't make any sense
> except as string, And this isn't one.
>
> I think what you might have meant is to quote the _whole thing_ as a
> string, and not the individual pieces. Then:
> $grplist = "'(".$npaa.$nxxa.$blocka.")'";
> and
> $blocka = "($_POST[ablock])";  # and for all of them above
>
> This would make the value '(111)(333)()', which should work fine.
>
> Now, if you really meant to add in the quotes, you'll have to "quote the
> quotes", which can be hard to do in good times.
>
> Hope this helps,
> Gerald.
>
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