Re: [asterisk-users] How to query Microsoft SQL server for caller-id source

[Nick Brown]

Is there a need to do it within the dialplan? If not you will find it easier to 
do it within AGI. Either connecting directly to the DB or in our case our 
developer build a web service which I make SOAP calls to.

Nick.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen
Sent: Tuesday, 13 December 2011 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to query Microsoft SQL server for caller-id source

Any suggestions from people who have done this before?

Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
A.A.S. Information Technology
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545

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[asterisk-users] Followme generate ringing instead of MOH

[Nick Brown]

Afternoon All,

Is anyone aware of a way to generate ringing as opposed to starting music on 
hold for the party originating a call with followme?

I'm assuming its doable as it looks like FreePBX users get the option (Not to 
say that FreePBX haven't got their own followme implementation though).

Cheers
Nick.


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[asterisk-users] Cross Queue Priorities

[Nick Brown]

Morning All,

My Google skills may be failing me as I can see people asking this but no 
useful responses, I need a way to prioritise calls across queues - I can think 
of ways to do this but they are far from elegant and this seems like such a 
simple request I am sure I am missing something obvious.

All my queues are of equal weight (Ie. A caller in Queue A can be just as 
important as a caller in Queue B) but not all my callers are of equal priority 
- Ie. A caller in Queue A with a priority of 100 needs to reach an agent before 
a priorty 50 call in Queue B, keeping in mind that a single agent can be in 
both Queue A and Queue B.

Would appreciate any input on this at all!

Cheers
Nick.
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Re: [asterisk-users] PRI errors no D channel

[Nick Brown]

Depends what its connected to

Nick.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, 12 August 2010 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI errors no D channel

Jerry Geis wrote:
> signalling=pri_net
>

It needs to be pri_cpe

Doug

-- 
Ben Franklin quote:

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


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Re: [asterisk-users] What do you use for Invoicing?

[Nick Brown]

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] 1 second Audio Lag

[Nick Brown]

Do you see the issue when calling between two softphones? Do you see the issue 
if you call from your mobile into an echo test?

Setting TOS flags on packets will make no difference unless the gear in between 
is configured to treat them differently. Not that I envision this is the issue 
at all.

Nick.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of colin mcdermott
Sent: Tuesday, 27 July 2010 5:47 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 1 second Audio Lag


Hi All (reposting after 24 hours). 

I will do a test call from a soft phone to my mobile. I can speak into my
headset and the audio is heard instantly. But if I speak into my mobile
there is a 1-2 second delay in the Audio. I am using SIP.

I am only finding it in the Zoiper Softphones that we are using. We are able to 
make a call without lag on the X-lite softphone no problem. Sadly the customer 
is Quite attached to the Zoiper.

I have set QOS = CS5 for both SIP and RTP packets. Altering these settings has 
no effect to the lag issue.

We have three 24 port Gigabit switches, with the top switch connecting in
the Asterisk Box. Even the stations plugged into the TOP switch have this
delay and to the same extent as the other switches. No routers on the loop

I have tried switching the stations to IAX. No effect. I have tried using
GSM instead of G711 (alaw). No effect. I have about 30 stations. No change 
under heavy or light load.

I have done a Wireshark trace on the stations and no issues detected when I go 
analyse on the RTP packets. All sequencing is correct.

Is Zoiper any good? Anyone else had these problems? 

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Re: [asterisk-users] "Register Attacks" End of ENUM ?

[Nick Brown]

Blocking SIP traffic is still going to break ENUM. 

The problem with your suggestion Norbert is that Asterisk still would have to 
process the requests at an application layer, providing no real advantage to 
users of boxes with no grunt. 

You could potentially write something to do inspection on the packets, there 
are a handful of L7 Linux switch projects around. Of course - still relatively 
resource intensive.

Fail2Ban is probably the best solution.

What someone needs to offer is an ENUM gateway service :-)

Nick.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys
Sent: Tuesday, 27 July 2010 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] "Register Attacks" End of ENUM ?

On Sun, Jul 25, 2010 at 3:11 AM, Norbert Zawodsky  wrote:
> Hello again!
>
> after it being "relatively quiet" her for the last weeks, my Astrerisk
> server was the target of 3 of that nasty REGISTER attacks during the
> last days. While I can see not much danger coming from these attacks (I
> use very long, complicated random generated passwords), they are still
> very annoying, because they always lead to my server crashing. (I think
> it's some out of memory condition because its a very tiny server. Slow
> CPU, not much memory...)
>
> Now, as a quick-fix I had the idea to use iptables'  --scr-range rule
> to close the whole adress-range from 0.0.0.0 to 255.255.255.255 EXCEPT
> that small range of my VOIP provider. This should keep out all attacks.
> (At least, I think so). But I'm not a iptables-guru at all !!
>
> But the side-effect would be that ENUM wouldn't work any more.
>
> I still think that the best, clean solution would be, if some mechanism
> was built into asterisk (maybe sip.conf was the right place ???) where
> you could configure from which source (ip-range, ethernet-port or
> whatever...) asterisk  will accept or ignore REGISTER requests. For
> example, in my small installation, valid REGISTERs can only originate
> from the internal LAN, never from the "outside world". So I could
> restrict the range for valid REGISTERs to 192.168.1.0/24.
>
> AFAIK incoming calls would start the conversation with INVITE and those
> still may come from "the outside" (=any IP adress).
>
> Another thought makes me feel nervous: What if some sick brain gets the
> idea of sending INVITEs instead of those REGISTERs...
>
> Norbert

If all you need is block the SIP traffic from external sources, you
may do the following:
# iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT
# iptables -A INPUT -p udp --dport 5060 -j DROP

# iptables-save > /etc/iptables.up.rules
and somewhere in init scripts (depending on your lsb release):
# iptables-restore < /etc/iptables.up.rules

fail2ban is more suitable if you have external environment (plus it's
more complicated than just these 2 rules).

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[asterisk-users] SIP TOS Not being set

[Nick Brown]

Hi All,

Facing an issue at the moment with setting the TOS on packets - the 
documentation is a bit light, however is straightforward so unsure if this is a 
configuration issue or a bug. 

Following is set in sip.conf;
tos_sip=CS3
tos_audio=EF

And is reflected in the CLI;
IP ToS SIP: CS3
IP ToS RTP audio:   EF

However a packet capture shows the following;

RTP Packet looks good;
11:39:59.554679 IP (tos 0xb8, ttl  64, id 0, offset 0, flags [DF], proto: UDP 
(17), length: 200) LOCAL.26392 > REMOTE.8768: UDP, length 172

Signaling Packet not so good;
11:39:59.633869 IP (tos 0x0, ttl  64, id 35957, offset 0, flags [none], proto: 
UDP (17), length: 479) LOCAL.sip > REMOTE.sip: SIP, length: 451

Seeing the same behavior on 1.4.28 and 1.6.2.9, separate servers. The packet 
captures are from the box itself so will not be affected by anything upstream.

Anyone able to advise if they see the same problem?

Cheers
Nick.
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Re: [asterisk-users] SIP port 5060 closed - how do I open it?

[Nick Brown]

Zaheer,

On 28/11/07 9:28 AM, "Zaheer K. Master" <[EMAIL PROTECTED]> wrote:

> Yes I have a sip.conf, contents as follows:

>From the CLI can you confirm SIP is running by pasting the results of
'module show like sip'

Cheers
Nick.



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Re: [asterisk-users] Calling with hidden callerid

[Nick Brown]

You can set callerid within the [general] section of your sip.conf. This
should work for you.


On 23/11/07 8:02 AM, "Mike" <[EMAIL PROTECTED]> wrote:

> Hi,
>  
> I have a wholesale provider that allows me to put any caller id I want when
> dialing out.  In some cases, I`d like the outgoing callerid to be hidden.  How
> do I do this?
>  
> I`ve set callerid name to "unknown", that works well, but when I put an empty
> number it goes out with the name "asterisk".  Which is NOT what I want.
>  
> Is there a standard way to say "hid my number"?
>  
>  
> Mike
> 
> 
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Regards,
Nick Brown

Ipera Communications Pty Ltd
Level 1, 9 Denison Street, 
Newcastle West NSW 2302
PO Box 2115, Dangar NSW 2309   

Ü P: +61 2 4910 1000
Ü F: +61 2 4910 1099
Ü ABN: 31 090 964 104

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Re: [asterisk-users] Music on Hold -- Error

[Nick Brown]

I posted to the list earlier this week about this very issue. This
reinforces my thought that it is a bug in 1.4.7.

Since upgrading the box to 1.4.13 the issue resolved itself.

I have not opened a issue in the tracker as I hadn¹t had time to try and
replicate the issue.


On 16/11/07 5:32 AM, "Ryan M. Colbert" <[EMAIL PROTECTED]> wrote:

> We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom
> 330¹s for endpoints.  When one of our end points places a call on hold we get
> the following in CLI.  There is no music on hold provided for the caller.  The
> SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and
> allow=ulaw.  Should there be a similar setting on the user.conf entries?
>  
> An interesting note is the IP noted in the CLI message below is neither
> Bandwidth.com nor the end point.
>  
> Thanks for any help!!
>  
> CLI Message:
> [Nov 15 13:21:38] WARNING[11327]: channel.c:2964 set_format: Unable to find a
> codec translation path from ulaw to unknown
> [Nov 15 13:21:38] WARNING[11327]: res_musiconhold.c:702 moh_alloc: Unable to
> set channel 'SIP/4.68.250.148-08d1e7e0' to format 'unknown'
>  
> Ryan M. Colbert
> Director of Information Technology
> Rissman, Barrett, Hurt,
> Donahue & McLain, P.A.
> 201 E. Pine Street, Suite 1500
> Orlando, FL 32801
> (407) 517-3105 ­ Direct Telephone
> (407) 839-0120 - Main Office
> (407) 841-9726 ­ Fax
> http://www.rissman.com/ <http://www.rissman.com/>
>  
> 
> 
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Regards,
Nick Brown

Ipera Communications Pty Ltd
Level 1, 9 Denison Street, 
Newcastle West NSW 2302
PO Box 2115, Dangar NSW 2309   

Ü P: +61 2 4910 1000
Ü F: +61 2 4910 1099
Ü ABN: 31 090 964 104

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Re: [asterisk-users] MOH Codec Issue - Fixed

[Nick Brown]

I recompiled a version of Zaptel to no avail in an attempt to find a quick
fix. This did not help, however have since upgraded the box to Asterisk
1.4.13 and the issue has disappeared. As such I put it down to either being;
1. Zaptel was broken, I should have however recompiled Asterisk after
recompiling Zap (Opposed to being impatient and frustrated), or
2. There is a bug in 1.4.7. I haven't had time to try and reproduce it,
plus it would be a purely academic project as if there was a bug it has
since been fixed.

Thanks for the suggestions Paul.

Nick.


On 13/11/07 4:48 PM, "Paul Hales"  wrote:

> Is it possibly a funny zaptel issue? Paul Hales AsteriskIT > > On Tue,
2007-11-13 at 
> 15:04 +1100, Nick Brown wrote: > >> Afternoon All, > >> > >> Today rolled a
> pre-production box from Trunk back to 1.4.7 (In an > >> attempt to get a
> working SCCP channel). During the process Music On > >> Hold appears to have
> died (Not, just when calling from a SCCP device, > >> but coming in on SIP
> also). > >> > >> CLI is showing > >> > >> -- Executing
> [EMAIL PROTECTED]:2] > >> MusicOnHold("SIP/10.97.1.33-09f0cfc8",
> "sounds") in new stack > >> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964
> set_format: Unable to > >> find a codec translation path from alaw to
> unknown > >> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702
> moh_alloc: > >> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format
> 'unknown' > >> -- Started music on hold, class '?S?', on channel > >>
> 'SIP/10.97.1.33-09f0cfc8' > >> [Nov 13 15:00:14] WARNING[5461]:
> res_musiconhold.c:575 moh0_exec: > >> Unable to start music on hold (class
> 'sounds') on channel > >> SIP/10.97.1.33-09f0cfc8 > >> > >> Have attempted to
> use an alternate Music On Hold context and forced a > >> format= within
> musiconhold.conf. > >> > >> Otherwise all other audio (Playback, voice etc)
> seems fine. > >> > >> Anyone seen this before? Can not see anything in the
> tracker regarding > >> this issue in 1.4.7 specifically. > >> > >> Cheers > 
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Re: [asterisk-users] MOH Codec Issue

[Nick Brown]

It was using the 3 wav's from Freeplay. I have just recompiled and told it
to pull down the ULAW versions, then removed the Wav's however it has made
no difference.

Cheers
Nick

On 13/11/07 3:56 PM, "Paul Hales" wrote:

> 
> What format is your music on hold in?
> 
> PaulH
> 
> 
> On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
>> Afternoon All,
>> 
>> Today rolled a pre-production box from Trunk back to 1.4.7 (In an
>> attempt to get a working SCCP channel). During the process Music On
>> Hold appears to have died (Not, just when calling from a SCCP device,
>> but coming in on SIP also).
>> 
>> CLI is showing
>> 
>> -- Executing [EMAIL PROTECTED]:2]
>> MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack
>> [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to
>> find a codec translation path from alaw to unknown
>> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc:
>> Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown'
>> -- Started music on hold, class '?S?', on channel
>> 'SIP/10.97.1.33-09f0cfc8'
>> [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec:
>> Unable to start music on hold (class 'sounds') on channel
>> SIP/10.97.1.33-09f0cfc8
>> 
>> Have attempted to use an alternate Music On Hold context and forced a
>> format= within musiconhold.conf.
>> 
>> Otherwise all other audio (Playback, voice etc) seems fine.
>> 
>> Anyone seen this before? Can not see anything in the tracker regarding
>> this issue in 1.4.7 specifically.
>> 
>> Cheers
>> Nick. 
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> 

Regards,
Nick Brown

Ipera Communications Pty Ltd
Level 1, 9 Denison Street, 
Newcastle West NSW 2302
PO Box 2115, Dangar NSW 2309   

Ü P: +61 2 4910 1000
Ü F: +61 2 4910 1099
Ü ABN: 31 090 964 104


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[asterisk-users] MOH Codec Issue

[Nick Brown]

Afternoon All,

Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to
get a working SCCP channel). During the process Music On Hold appears to
have died (Not, just when calling from a SCCP device, but coming in on SIP
also).

CLI is showing

-- Executing [EMAIL PROTECTED]:2]
MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack
[Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a
codec translation path from alaw to unknown
[Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to
set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown'
-- Started music on hold, class '?S?', on channel
'SIP/10.97.1.33-09f0cfc8'
[Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to
start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8

Have attempted to use an alternate Music On Hold context and forced a
format= within musiconhold.conf.

Otherwise all other audio (Playback, voice etc) seems fine.

Anyone seen this before? Can not see anything in the tracker regarding this
issue in 1.4.7 specifically.

Cheers
Nick.
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[asterisk-users] Dynamic Queue Members - Auto Logoff

[Nick Brown]

Another quick question (Spending the day trying to get this project sorted
and tucked away) If I am dynamically adding queue members, they will not
abide to settings within agents.conf will they?

Ie. I need the equivalent of Autologoff however want my agents to receive
calls when someone joins the queue, not have to sit on hold all day. I see
AgentCallbackLogin has finally been removed.

Has anyone got a work around for this?

Thanks.
Nick.


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Re: [asterisk-users] 7960 Queue Issue

[Nick Brown]

Thanks Eric, this is the case. A bit of a shame that it removes the
functionality for the member to see calls that have not come from a queue
however there is not much choice in the matter.

FWIW to get this option a firmware upgrade was required (Now running
POS3-08-8-00).

Cheers.


On 5/11/07 11:57 AM, "Eric Merkel" <[EMAIL PROTECTED]> wrote:

> On 11/4/07, Nick Brown <[EMAIL PROTECTED]> wrote:
>> Morning All,
>> 
>> Quick question that has me stumped. Have a queue with several members
>> (Statically defined in queues.conf at this stage for testing) who use Cisco
>> 7960's.
>> 
>> The queue is configured to use rrmemory and generally this works correctly.
>> However if a member is already on a call their phone will still ring (The
>> 7960 can show multiple incoming calls for one line). I really don't want
>> members who are on calls to get more calls. Especially when we start logging
>> out members who don't answer.
>> 
>> Asterisk shows;
>> -- Called 1014
>> -- SIP/1014-08f2e4d0 is ringing
>> -- Local/[EMAIL PROTECTED];1 is ringing
>> -- Nobody picked up in 15000 ms
>> 
>> Short of disabling the feature to show multiple incoming calls on the 7960's
>> (Which I don't know if it can be done anyway), has anyone got any
>> suggestions?
>> 
> 
> Yes, you can turn off this in the phone. Go into call preferences on
> the phone and turn off call waiting. Not optimal but can be done.
> 
> -Eric
> 
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[asterisk-users] 7960 Queue Issue

[Nick Brown]

Morning All,

Quick question that has me stumped. Have a queue with several members
(Statically defined in queues.conf at this stage for testing) who use Cisco
7960's.

The queue is configured to use rrmemory and generally this works correctly.
However if a member is already on a call their phone will still ring (The
7960 can show multiple incoming calls for one line). I really don't want
members who are on calls to get more calls. Especially when we start logging
out members who don't answer.

Asterisk shows;
-- Called 1014
-- SIP/1014-08f2e4d0 is ringing
-- Local/[EMAIL PROTECTED];1 is ringing
-- Nobody picked up in 15000 ms

Short of disabling the feature to show multiple incoming calls on the 7960's
(Which I don't know if it can be done anyway), has anyone got any
suggestions?

Thanks in advance!

Nick.


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Re: [asterisk-users] Periodic Announce issue

[Nick Brown]

Scrap that. I've somehow broken all queue announcements including position
and holdtime.

Will repost when I sort out what I've done.


On 24/10/07 11:13 AM, "Nick Brown" <[EMAIL PROTECTED]> wrote:

> Morning All,
> 
> Just wondering if anyone can confirm that peridoic-announce and
> periodic-announce-frequency are still valid options within queues.conf?
> 
> For testing purposes my queue includes;
> 
> periodic-announce-frequency = 10
> periodic-announce = demo-congrats
> 
> When in the queue however I'm not hearing the message, the context we break
> out to works fine, its just the messages that are not being played.
> 
> Watching the CLI shows no attempt to play the file either.
> 
> Queue is configured to use MOH opposed to ringing. Box is currently running
> SVN-trunk-r86585, I don't have access to a release version at the moment to
> see if it is working there.
> 
> Cheers!
> Nick.
> 
> 
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[asterisk-users] Periodic Announce issue

[Nick Brown]

Morning All,

Just wondering if anyone can confirm that peridoic-announce and
periodic-announce-frequency are still valid options within queues.conf?

For testing purposes my queue includes;

periodic-announce-frequency = 10
periodic-announce = demo-congrats

When in the queue however I'm not hearing the message, the context we break
out to works fine, its just the messages that are not being played.

Watching the CLI shows no attempt to play the file either.

Queue is configured to use MOH opposed to ringing. Box is currently running
SVN-trunk-r86585, I don't have access to a release version at the moment to
see if it is working there.

Cheers!
Nick.


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Re: [asterisk-users] Conference Calls with single-line SIP

[Nick Brown]

Yes, that will work fine Zaheer.

On 16/10/07 1:32 AM, "Zaheer Master" <[EMAIL PROTECTED]> wrote:

> Hi all,
> If I have 2 single-line SIP phones, I can still do a conference call using
> Asterisk, right? For example, two people in my office are on the call, along
> with 1 other person at a remote site.
> 
> Regards,
> Zaheer
> 
> 
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[asterisk-users] Skills Based Routing

[Nick Brown]

Morning All,

Has anyone here successfully implemented skills based routing within queues?

The concept behind skills based routing is fairly straight forward, and I
know I could do it with multiple queues, agent penalties and a bit of AGI to
put the call into the right queue.

However doing this is going to require the addition of several extra queues
and isn't a very clean solution.

The other alternative is to write our own queue system with AGI, effort++
though :-)

TIA.

Cheers,
Nick.

 


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