Re: [asterisk-users] shell dialplan application blocking

[Matt Riddell (lists)]

Use AGI

Kind regards,

Matt

> On Jun 4, 2018, at 02:28, Benjamin Marty  wrote:
> 
> I'm calling a script which needs to wait a certain time and also hold the 
> call for this time. But the script dialplan application seems to work non 
> blocking. Is there a way to hold the call/dialplan till the shell script is 
> finished?
> 
>same => n,Set(PUSHRESULT=${SHELL(sendpush.sh)})
> 
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Re: [asterisk-users] Asterisk chan_sip registration attempts

[Matt Riddell (lists)]

Maybe the provider has added an extra gateway and it is not processing accounts 
correctly. 

If they had one before and now two then 40-60% registration fails would show 
that. 

Kind regards,

Matt

> On Oct 10, 2017, at 06:27, Dmitriy Ermakov  wrote:
> 
> Hello!
> 
> Could you help me with Asterisk 11.21.2 and AsteriskNow platform.
> 
> The problem is:
> 
> My Asterisk PBX has SIP (chan_sip) trunk to provider.
> 
> Asterisk periodically loses trunk registratrion:
> 
> sip show registry:
> 
> Hostdnsmgr Username   Refresh State   
>  Reg.Time 
> X.X.X.X:5060N  105 Unregistered
>
> 
> This happens sometimes once per 4 hours, sometimes once per a week.
> 
> I don't see any patterns.
> 
> sip.conf:
> 
> registerattempts=0
> 
> registertimeout=20
> 
> peer confifuration:
> [-friend]
> disallow=all
> host=192.168.1.1
> defaultuser=
> fromuser=
> callerid=
> secret=
> type=friend
> qualify=yes
> allow=ulaw
> allow=alaw
> nat=no
> rtpkeepalive=10
> dtmfmode=rfc2833
> insecure=port,invite
> context=from-trunk-ISP1
> fromdomain=
> registration string:
> 
> register=:@/
> 
> where:
> 
>  is our ISP-provided phone number
> 
>  is our ISP-provided SIP secret
> 
>  is our ISP SIP server IP address
> 
> I don't have NAT between the ISP and my server (the ISP server IP address is 
> in Asterisk's sip.conf Localnet scope) but as I can see there is ISP's 
> routers between my Asterisk and the ISP SIP server.
> There is not any firewall between the ISP and my Asterisk. The firewall rules 
> on Asterisk host allow any traffic from host to the ISP and allow 5060/UDP 
> from ISP to Asterisk host.
> 
> 
> When I restart Asterisk I can see successful registration to the ISP. And it 
> works. I can make calls in any direction but after some time I have the 
> "Unregistered" status and calls don't work.
> 
> When I make tcpdump I can see about 40-60% failed REGISTER attempts (401 
> forbidden) and 100% failed OPTIONS attempts (is it qualify packets?) with 401 
> forbidden.
> 
> The interesting thing is: the last REGISTER packet from my Asterisk to the 
> ISP has "OK" response and there was not any REGISTER attempts after this 
> packet, only OPTIONS packets.
> The second interesting thing is: this ISP has been working for about 6 or 
> even 12 months before this problem happened.
> 
> 
> What should I check to understand and solve my problem?
> 
> 
> P.S. I'm sorry for my bad English(
> -- 
> Dmitriy Ermakov
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Re: [asterisk-users] softphone instead of desktop phones

[Matt Riddell (lists)]

I use Bria on all of the above. 

Kind regards,

Matt

> On Apr 29, 2017, at 10:35 AM, Thomas  wrote:
> 
> Hello,
> Iam lookong for an Softphone for iPhor oder Android smartphone using togehter 
> with an headset.
> I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP 
> phone.
> 
> Is there an better softphone?
> 
> Or are there softphone solutions for PC desktop MAC or Android with an 
> headset?
> I want to save cost for desktop phones.
> 
> thanks Thomas
> 
> 
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Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell (lists)]

Not really, doing the way below you don't even have to worry about it. They 
both go out at the same instant and as soon as it hits voicemail it disconnects 
the other leg. 

If you wanted you could leave it ringing for twenty minutes and it would still 
have the same effect. 

Kind regards,

Matt

> On Feb 6, 2017, at 12:29 PM, Tech Support  wrote:
> 
> That's the basics, but you have to nail the timing just right. The timing is
> really important to do it the right way.
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Monday, February 06, 2017 12:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Call List Campaign to an IVR
> 
> 
>> On Mon, 6 Feb 2017, Tech Support wrote:
>> 
>>  We were able to develop a feature to send the call to voicemail
> about 90% of the time. That way, an end user could (1) not be bothered by
> having to answer the call, (2)
>>  delete the message without listening to it, or (3) listen to the
> message when it was most convenient for them. That way, they were in control
> and things were done on
>>  their terms.
> 
>> On 6/02/2017, at 11:34 AM, Steve Edwards 
>> wrote:
>> 
>> Love the idea. How?
> 
>> On Mon, 6 Feb 2017, Matt Riddell wrote:
>> 
>> exten => 
>> _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)
> 
> Amazing. Who knew?
> 
> So how/why does this work?
> 
> I see 2 calls going out to my cell. Does the first 'busy out' my number at
> my cell provider so the second goes straight to VM? What part does the
> '0111' play?
> 
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
> 
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Re: [asterisk-users] Surfing the web via Asterisk.

[Matt Riddell (lists)]

This is a really interesting project but I think it's going to be seriously 
hard. You're going to need to parse meaning from a site, and that's not an easy 
thing to do. 

If you're focused on a few of the bigger sites then it might be easier. 

You almost want a middle layer that can parse meaning from a site into xml or 
something. 

Then I'd work on creating objects out of each kind of tag. The problem is that 
navigation may not be the same as you'd see when visiting a site. You're not 
really going to be moving left and right. It would be more like tab works. Next 
item kinda thing. And items wouldn't necessarily be in the same order as you 
see. Pull right/left classes for bootstrap etc would make layout different. 

I would maybe check if there are any libraries that can parse HTML into objects 
first and if not then start building on yourself. The telephony side will be 
easy. You'd use agi or something to navigate the object you create and tts to 
describe current position. The hard part will be parsing the HTML even though 
most HTML is broken :-)

Kind regards,

Matt Riddell

> On Oct 17, 2016, at 9:00 AM, Jonathan H  wrote:
> 
> Has anyone attempted making the web phone accessible? I can only find one 
> company which operated between 1996 and 2000. 
> 
> I was thinking, install Chrome with Chromevox, headless, on a server, and use 
> something like an AGI to send basic keyboard commands to navigate a page, as 
> a screenreader user would, and pipe the audio back to a channel, to be 
> streamed by Asterisk. 
> 
> (Bear with me here - it's a project for blind people involving a telephone 
> and some lateral thinking!)
> 
> And yes, I mean more than just CURL a page, tts it and then read. I'm talking 
> about using the keypad to navigate the headers and landmarks. There are just 
> enough keys to make it viable.
> 
> Of particular interest is the very high quality of the Chromevox screenreader 
> voice from Google.
> 
> Does such a framework exist? I'm aware of a project called Chromium Headless, 
> but some of the links are broken and it doesn't seem to have the 
> audio/extension part
> 
> Failing headless, what about running it on a vps in an x-window environment? 
> Not sure how I'd pass the keyboard presses to it, without using a keyboard...
> 
> Any ideas, or is the whole idea complete madness? Thanks!
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Re: [asterisk-users] Installing Asterisk on MAC native

[Matt Riddell (lists)]

It pretty much just works the same way as Linux. you might need to use brew to 
install a few prerequisites but I've got it running on my MacBook Pro without 
any major problems. 

It's good for testing things but I wouldn't use a MacBook as an office server 
or anything. 

And to be fair most of the time I just use virtualization and spin up Debian in 
parallels. 

Kind regards,

Matt

> On Sep 20, 2016, at 4:07 PM, Glenn Geller (VDOPh)  wrote:
> 
> If you're looking for installing on a MAC, best to start searching for MAC 
> OSX install
> 
> See here: 
> http://www.voip-info.org/wiki/view/Asterisk+Getting+Started+on+MacOSX
> 
> I don't know how old this is, or if it directly applies to your task at hand, 
> but it may be a start.
> 
> Also, if you're just looking for a simple PBX for light usage, there may be 
> other options out there for MAC OS as well.
> 
> Good hunting!
> 
> Glenn Geller
> 
>> On Tue, Sep 20, 2016 at 1:58 PM, Saint Michael  wrote:
>> ​I need to install Asterisk on a MAC, native, no virtualization.
>> Has anybody done this? Are there documents on the Internet?
>> I googled it and all web sites that claimed to help installing Asterisk on
>> a MAC have disappeared. Is it possible at all?
>> Digium should actually have a MAC app in the Apple store with a PBX. It 
>> should be a paid app. I would buy it right away.
>> 
>> ​
>> 
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Matt Riddell (lists)]

There is definitely no way you should put 1000 lines on a single box. To be 
honest I do wonder what you want to do with 1000 lines as your description 
probably changes the recommendations. 

Kind regards,

Matt

> On Feb 17, 2016, at 5:09 PM, Goke Aruna  wrote:
> 
> Thanks Harry.
> I will check and revert. I hope it work perfectly with asterisk.
> Regards
> 
>> On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor  
>> wrote:
>> Hi,
>> 
>> For analog, I really like telco grade channel banks.
>> 
>> I would recommend the adit 600, there is a good market on Ebay, and you can 
>> do 48 channels per adit 600, with 2 T1 interfaces.  Having onsite spares 
>> would not be an issue (cost is low).  You can put two next to each other in 
>> a rack, taking up about 2U of space per 2 channel banks.
>> 
>> You could service this with six eight port T1 cards, or with eleven/twelve 
>> quad T1 cards.  I would distribute across two, three, or even four servers 
>> for redundancy/resiliency and load balancing.
>> 
>> -Harry
>> 
>> 
>>> On 02/17/2016 12:16 AM, Goke Aruna wrote:
>>> 
 On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani  wrote:
 Sangoma 50 port FXS
>>> 
>>> 
>>> Thanks.
>>> Will I now stack 20 boxes in order to achieve the 1000 FXS lines?
>>> Regards
>>> 
>>> 
>> 
>> 
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[asterisk-users] No sound with internal calls depending on which phones

[(lists) Denis BUCHER]

Dear all,

I have a very strange problem :

 * external calls work perfectly,
 * internal calls between some phones too,
 * but internal call between two similar phones don't work !!! (Snom 710)

When we have sound, there are no errors in asterisk. When we do not have 
sound, there is the following error :


 * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
   module loaded, can't setup SRTP session.

This is a working internal call :

  == Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-", 
"SIP/phone1") in new stack

  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Got  RTP packet from192.168.128.99:49646 (type 126, seq 031575, ts 
01, len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read: 
Unknown RTP codec 126 received from '192.168.128.99:49646'

Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
  == Spawn extension (local, 301, 1) exited non-zero on 
'SIP/dbucher-'

This is a non-working call :

  == Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP 
module loaded, can't setup SRTP session.
-- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002", 
"SIP/phone1") in new stack

  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and 
SIP/phone1-0003

Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
  == Spawn extension (local, 301, 1) exited non-zero on 
'SIP/hsolutionspf5-0002'

I tried many options to disable SRTP but without success :

 * canreinvite = no
 * canreinvite = nonat
 * srtpcapable=no
 * encryption=no
 * directmedia=nonat
 * ...or noload => res_srtp.so in modules.conf


Any help would be GREATLY appreciated !

Denis

P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)

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Re: [asterisk-users] No sound with internal calls depending on which phones

[(lists) Denis BUCHER]

Dear Sam, dear jg, dear Mitul, dear all,

Thanks a lot for your advices! I had the same idea, but it still doesn't 
work!


Maybe I changed the wrong option on the GUI configuration ?
I went to menu "Setup" > "Identity 1" > "RTP" > "RTP Encryption:" > 
"off" on both phones.

And in the configuration I see "user_srtp1!: off"

Is this right ?

Denis

Le 12.11.2015 17:05, Sam Basan a écrit :


Snom default configuration is SRTP enabled.

You should disable the SRTP from the phone web GUI configuration

**

**

*Sincerely,*

cid:image001.jpg@01D0D5C4.27A0CBA0

*Sam Basan*

cid:image003.png@01C918DA.6B3E4530

*From:*Mitul Limbani [mailto:mi...@enterux.in]
*Sent:* Thursday, November 12, 2015 5:25 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
*Subject:* Re: [asterisk-users] No sound with internal calls depending 
on which phones


You might have to disable srtp negotiations inside the phone web ui 
options.


Mitul

On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" 
<dbuche...@hsolutions.ch <mailto:dbuche...@hsolutions.ch>> wrote:


Dear all,

I have a very strange problem :

  * external calls work perfectly,
  * internal calls between some phones too,
  * but internal call between two similar phones don't work !!!
(Snom 710)

When we have sound, there are no errors in asterisk. When we do
not have sound, there is the following error :

  * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp:
No SRTP module loaded, can't setup SRTP session.

This is a working internal call :

  == Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-",
"SIP/phone1") in new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and
SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Got  RTP packet from 192.168.128.99:49646
<http://192.168.128.99:49646> (type 126, seq 031575, ts
01, len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190
ast_rtp_read: Unknown RTP codec 126 received from
'192.168.128.99:49646 <http://192.168.128.99:49646>'
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
  == Spawn extension (local, 301, 1) exited non-zero on
'SIP/dbucher-'

This is a non-working call :

  == Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp:
No SRTP module loaded, can't setup SRTP session.
-- Executing [301@local:1]
Dial("SIP/hsolutionspf5-0002", "SIP/phone1") in new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and
SIP/phone1-0003
Sent RTP P2P packet to 192.168.128.228:65494
<http://192.168.128.228:65494> (type 00, len 000160)
Sent RTP P2P pac

Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Matt Riddell (lists)]

There was a product called something like red box or similar that I saw around 
5 years ago. Probably not entirely helpful but maybe Google will help. 

Kind regards,

Matt

 On Aug 3, 2015, at 9:50 AM, Eric Klein eric.kl...@greenfieldtech.net wrote:
 
 Hi all,
 
 Strange request, I have a customer where we are putting an Asterisk PBX in 
 front of a legacy (non-VoIP) PBX. One of the requirements it that the 
 Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the 
 carrier) with the ability to go to pass through should the Asterisk PBX 
 (software or hardware level) fail.
 
 I did not see this feature in the Digium, Sangoma, Allo, or OpenVox cards.
 
 Does anyone know of a card that will do this? I know that Digium has an 
 external box (the r850) that does something similar for 2 PBXs making them 
 high availability, but in this case I only have the 1 Asterisk box acting as 
 a gateway and passing some calls out over SIP and IAX2. 
 
 Any suggestions would be appreciated.
 
 Thanks
 Eric
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Re: [asterisk-users] How to use TRUNK only if IAX fails?

[Matt Riddell (lists)]

The command he gave you was in Asterisk. Why do you not want to call it to try 
it?

Then you can fail over to the other trunk if the IAX link is down. 

Kind regards,

Matt

 On May 30, 2015, at 2:03 AM, Ashwin Surendran 
 ashwin.surend...@now-health.com wrote:
 
 Many Thanks Carlos, I was hoping to check whether the remote server is
 available before I issue the dial in my dial plan.
 
 Is there a better way to do it in asterisk without using unix commands?
 
 
 Many Thanks,
 Ashwin
 
 On 5/30/15, 2:06 AM, Carlos Chavez cur...@telecomabmex.com wrote:
 
 On 5/29/15 1:16 PM, Ashwin Surendran wrote:
 Hi,
 I have multiple Asterisk servers in various parts of the world all
 connected using dedicated VPN¹s.
 
 Each of these servers have iax and dahdi TRUNK configured on them.
 
 Occasionally the VPN¹s fail.
 
 What I want to be able to do is on my dial plan, use IAX if the asterisk
 server can reach the remote server using the internet OR, use TRUNK only
 if it can¹t use IAX.
 
 Any ideas on how this can be implemented on the dial plan?
Check the DIALSTATUS variable to see if the IAX trunk failed and
 then dial via DAHDI.
 
 https://wiki.asterisk.org/wiki/display/AST/Dial+Channel+Variables
 
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 +52 (55)9116-91161
 
 
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[asterisk-users] Asterisk call forward for T1 incoming calls

[Al lists]

Is there a way to divert incoming calls on DAHDI T1 channels so telco gets
the diversion and send the call to new number and releasing the channel?
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[asterisk-users] handset forwarding Diversion header cannot be set on Local channels

[Al lists]

is there anyway to change Sip headers in local channels?
if a user sets forward on their handset, calls coming in to the handset get
diversion header added:
Diversion: 202 sip:202@192.168.1.46;reason=deflection

Then asterisk sends the call to local channel:
- Now forwarding SIP/201-0483 to 'Local/33@test' (thanks to
SIP/202-0484)

and not all Telco providers handle diversion header gracefully, some dont
like to see 202 in header.

i tried to set the sip header in target 33@test but asterisk
see's this as local channel and wont do sip add header:
WARNING[13584]: chan_sip.c:20562 func_header_read: This function can only
be used on SIP channels.

is there anyway around this?
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Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

[Al lists]

yes, thanks you!



On Sat, Mar 22, 2014 at 9:13 AM, Paul Belanger paul.belan...@polybeacon.com
 wrote:

 On Fri, Mar 21, 2014 at 11:58 PM, Al lists asteris...@gmail.com wrote:
  looking more into this, looks like this is not a issue, its related to
 users
  changing voicemail password from handset, asterisk rewrites the file.
 
 Right, use passwordlocation = spool, create a secret.conf for each
 mailbox, now when a user changes their password, secret.conf gets
 updated not voicemail.conf.

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[asterisk-users] AppVoicemail overwrites voicemail.conf

[Al lists]

We noticed issues with voicemail and somehow looks like voicemail.conf has
been overwritten:

;!
;! Automatically generated configuration file
;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
;! Generator: AppVoicemail
;! Creation Date: Thu Mar 20 06:48:16 2014
;!


i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not
using realtime.
anyway to prevent AppVoicemail ro auto generate files?
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Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

[Al lists]

passwordlocatio seems to be related to vmsecret

from voicemail.conf sample :

; passwordlocation=spooldir
; Usually the voicemail password (vmsecret) is stored in
; this configuration file.  By setting this option you
can
; specify where Asterisk should read/write the vmsecret.
; Supported options:
;   voicemail.conf:
; This is the default option.  The secret is read
from
; and written to voicemail.conf (or users.conf).
;   spooldir:
; The secret is stored in a separate file in the
user's
; voicemail spool directory in a file named
secret.conf.
; Please ensure that normal Linux users are not
; permitted to access Asterisk's spool directory as
the
; secret is stored in plain text.  If a secret is
not
; found in this directory, the password in
; voicemail.conf (or users.conf) will be used.
; Note that this option does not affect password
storage for
; realtime users, which are still stored in the realtime
; backend.


but the issue i was explaining was voicemail.conf getting overwritten
apparently by appvoicemail



On Fri, Mar 21, 2014 at 5:36 PM, Paul Belanger paul.belan...@polybeacon.com
 wrote:

 On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote:
 
  We noticed issues with voicemail and somehow looks like voicemail.conf
 has
  been overwritten:
 
  ;!
  ;! Automatically generated configuration file
  ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
  ;! Generator: AppVoicemail
  ;! Creation Date: Thu Mar 20 06:48:16 2014
  ;!
 
 
  i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are
 not
  using realtime.
  anyway to prevent AppVoicemail ro auto generate files?
 
 passwordlocation = spooldir

 Read voicemail.conf about how to use it.

 --
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Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

[Al lists]

looking more into this, looks like this is not a issue, its related to
users changing voicemail password from handset, asterisk rewrites the file.



On Fri, Mar 21, 2014 at 9:31 PM, Al lists asteris...@gmail.com wrote:

 passwordlocatio seems to be related to vmsecret

 from voicemail.conf sample :

 ; passwordlocation=spooldir
 ; Usually the voicemail password (vmsecret) is stored
 in
 ; this configuration file.  By setting this option you
 can
 ; specify where Asterisk should read/write the
 vmsecret.
 ; Supported options:
 ;   voicemail.conf:
 ; This is the default option.  The secret is read
 from
 ; and written to voicemail.conf (or users.conf).
 ;   spooldir:
 ; The secret is stored in a separate file in the
 user's
 ; voicemail spool directory in a file named
 secret.conf.
 ; Please ensure that normal Linux users are not
 ; permitted to access Asterisk's spool directory
 as the
 ; secret is stored in plain text.  If a secret is
 not
 ; found in this directory, the password in
 ; voicemail.conf (or users.conf) will be used.
 ; Note that this option does not affect password
 storage for
 ; realtime users, which are still stored in the
 realtime
 ; backend.


 but the issue i was explaining was voicemail.conf getting overwritten
 apparently by appvoicemail



 On Fri, Mar 21, 2014 at 5:36 PM, Paul Belanger 
 paul.belan...@polybeacon.com wrote:

 On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote:
 
  We noticed issues with voicemail and somehow looks like voicemail.conf
 has
  been overwritten:
 
  ;!
  ;! Automatically generated configuration file
  ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
  ;! Generator: AppVoicemail
  ;! Creation Date: Thu Mar 20 06:48:16 2014
  ;!
 
 
  i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are
 not
  using realtime.
  anyway to prevent AppVoicemail ro auto generate files?
 
 passwordlocation = spooldir

 Read voicemail.conf about how to use it.

 --
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 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter:
 https://twitter.com/pabelanger

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Re: [asterisk-users] mixmonitor extension

[Patrick Lists]

On 24-01-14 00:37, Marek Cervenka wrote:

can someone confirm that mp3 is unsupported? is patch available?


Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later 
versions of asterisk you can enable format_mp3 in make menuselect.



what about patch for Opus?

uncle google doesnt know


Did you really google?

http://lmgtfy.com/?q=asterisk+opus

Regards,
Patrick

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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Patrick Lists]

On 16-01-14 21:37, Gergely Kiss wrote:

Dear List,

I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as plan B).


It's SIP everywhere and anyone who requires you, in 2014, to use H.323 
should get a clue. Avoid them or at least demand SIP.



As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.


No idea. Maybe someone else with H.323 experience will respond. AFAIK 
it's a dead-end.


Regards,
Patrick

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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Patrick Lists]

On 17-01-14 01:57, Dan Austin wrote:

Patrick Lists wrote:

On 16-01-14 21:37, Gergely Kiss wrote:

Dear List,

I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as plan B).



It's SIP everywhere and anyone who requires you, in 2014, to use H.323
should get a clue. Avoid them or at least demand SIP

Bah.  There is nothing wrong with a working H.323 stack.  Just assuming
that they will have a working SIP stack because of the date can lead to
heartache.


By itself there is nothing wrong with a working H.323 stack. I just 
would not use it :-) Using H.323 for one provider while any backup or 
alternative providers probably use SIP results in needing two stacks in 
testing  production. It also requires the admins to gain knowledge of a 
legacy protocol. Maybe there are some incumbents or service providers 
with legacy H.323 equipment continuing to offer H.323 service. I get 
that. But for a business building a VoIP PBX from scratch H.323 does not 
make sense from a cost and operations point of view.



As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.



No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.

The ooh323 channel has been fairly reliable in our use case, which involve
connecting to a commercial IP PBX with crud SIP support.  Only you can tell
if it will work for you however, as sadly many times new core features only
get tested against the SIP channel(s), or worse only implemented there as
well.  Our current Asterisk version is 11.5.1


The OP mentioned that his VoIP provider prefers H.323 so it seems to be 
about trunking. IMHO fairly reliable is not something that is 
acceptable for trunking phone service.


H.323 is what Gopher is to HTTP/webservers. When was the last time you 
used a Gopher service? Would you today still buy Gopher based service 
because the service provider prefers it? :-)


Regards,
Patrick

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Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

[Patrick Lists]

Hi Steve,

On 15-01-14 18:53, Steve Edwards wrote:

On Wed, 15 Jan 2014, Patrick Lists wrote:


Would you mind sharing where you get the per country IP ranges from?


I confess I 'brute forced' it by entering '/8s' into ARIN's web page and
noting if the block had been assigned to a 'foreign' NIC -- not really a
reliable and robust methodology, but it worked for me.


If it works... :-)


A great way to kill time while on hold for customer dis-service.


Definitely. If any of the calls lasted more than entering 20 /8s I hope 
it was to cancel the service.


I found another solution: install the geoip kernel module from 
xtables-addons, install the MaxMind GeoIP country database and add some 
rules to the iptables config to block a country.


Regards,
Patrick

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Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

[Patrick Lists]

Hi Steve,

On 14-01-14 10:39, Steven Howes wrote:

On 14 Jan 2014, at 02:19, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:

Thanks for your feedback Paul. The not having outbound trunks is going to be a 
challenge.


Why? it’s what contexts were invented for.


Yes that is indeed what they are for but in the case they find a 
loophole or exploit a bug then not having outbound trunks is much safer.


Regards,
Patrick

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Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

[Patrick Lists]

Hi Steve,

On 15-01-14 02:44, Steve Edwards wrote:

On Tue, 14 Jan 2014, Patrick Lists wrote:


...I guess I'll cook up some dialplan logic that records IP addresses,
keeps track of the amount of failed password attempts etc. and block
the offending IP addresses...


A few iptables rules can protect you from access from China, North
Korea, Iran, Iraq, xxxistan, Russia, Nigeria, and any other country
you're not expecting calls from.

Eliminate 90% of the problem at the front door and you can focus more
clearly on the remaining 10%.


Yes that's one of the tricks in my bag. Unfortunately it seems that the 
IP ranges from ip-deny.com are no longer available and even their 
website has disappeared.


Would you mind sharing where you get the per country IP ranges from?

Regards,
Patrick

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[asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

[Patrick Lists]

Hi all,

I'm looking into adding the ability to call me at m...@mydomain.org on my 
Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow 
this kind of access as securely as possible?


Thanks,
Patrick

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Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

[Patrick Lists]

On 14-01-14 02:36, Paul Belanger wrote:

On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:

Hi all,

I'm looking into adding the ability to call me at m...@mydomain.org on my
Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow
this kind of access as securely as possible?


Well, if you want anybody to call you, you need to leave it open to
the public.  Meaning, you can't really secure it.  Obviously, don't
have any outbound trunks configured on the box so that the only
location some could dial would be your extension.


Thanks for your feedback Paul. The not having outbound trunks is going 
to be a challenge. So next to fail2ban I guess I'll cook up some 
dialplan logic that records IP addresses, keeps track of the amount of 
failed password attempts etc. and block the offending IP addresses 
together with max simultaneous outband calls and anything else I can 
think of to beef up security and limit potential damage.


Thanks,
Patrick

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[asterisk-users] is this expected behaviour?

[Al lists]

i noticed in asterisk 10.12.3, i get messages like this:

[2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite:
Failed to authenticate device 305sip:3...@my.server.ip;tag=0d516e63

but not mentioning attacker ip (to be used for fail2ban)

is this expected?
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Re: [asterisk-users] Problem building dahdi from source

[Patrick Lists]

On 01/03/2014 03:56 PM, Jonas Kellens wrote:

Hello,

I am getting the following error when compiling dahdi :

[snip]

`/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux'
make: *** [all] Error 2


I have the right kernel sources installed :

[root@sip dahdi-linux-complete-2.7.0.1+2.7.0.1]# uname -a
Linux sip 2.6.32-431.1.2.0.1.el6.x86_64 #1 SMP Fri Dec 13 13:06:13 UTC
2013 x86_64 x86_64 x86_64 GNU/Linux


So what am I missing ?


Try DAHDI 2.8.0.1
http://www.asterisk.org/downloads/dahdi

Regards,
Patrick

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Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

[Patrick Lists]

On 12/15/2013 09:55 PM, CDR wrote:
 I have had the issue for years. The problem is that Asterisk
 developers are removed from the business. We desperately need simple
 way to eliminate transcoding when unnecessary. Transcoding brings a
 server to its knees. It is a very simple new setting in sip.conf
 prioritize_matching_codecs=yes

Maybe have a look at FreeSWITCH. It's extremely flexible so may offer
what you want to do.

Regards,
Patrick


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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Patrick Lists]

On 12/14/2013 01:29 AM, Martin wrote:
 If I need to use SIP, from where to get the suitable firmware for
 these Cisco IP Phones 7942G?
 
 
 Be careful, not all versions of SIP firmware work with asterisk. I do
 have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with
 my 7961. Downloaded somewhere. Version 9.x is broken, SIP only works
 over TCP.

I thought that was fixed in the latest 9.x?

 Where do u download the SIP firmware usually for your Cisco IP Phones?

I have a 7961 and just registered at cisco.com then logged in, did a
search and was offered the firmware files for free.

Regards,
Patrick

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Re: [asterisk-users] Asterisk on Windows

[Patrick Lists]

Probably feeding the trolls but here it goes.

On 12/04/2013 04:19 PM, CDR wrote:
 Digium is 100% lost in the map. If they would come up with a Paid
 version of Asterisk, one that would use the .NET framework in Windows,
 something simple to install, they could go public on the product.

IIRC Microsoft no longer invests in the .Net framework which makes it a
bad idea for a product that would live for up to 10 years. Do you really
want to bet your business/company that .Net will be there in 5 to 10 years?

 Linux has a very steep learning curve. A Windows application that
 would do exactly the same would be a home run. 

I find Linux easier than Windows. Installing a package on Linux or
Windows is not the issue. How is a simple 'yum install asterisk' any
more difficult than double clicking on it in Windows? It's what you do
afterwards with the OS and package. Asterisk has a much steeper learning
curve than either. It's easy to mess up the config and suffer the
consequences if the box is Internet facing. Also, Windows has a terrible
reputation when it comes to security. Why would anyone want to use
Windows for an Internet facing service? There's a reason that Google,
Facebook, Twitter and pretty much the rest of the world are powered by
Linux and it's not only because it's cheaper.

Just because you find Windows easier does not make it a good idea.

 Note: I am a Linux
 expert user, but it took me years to get here. And still, moving from
 regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck.

There is probably a saying about people calling themselves experts and
then complain about a move from EL6 to F20 which is puzzling by itself.

 The .NET
 framework and Windows server 2012 are miles away in terms of
 friendliness and on equal footing on performance.

I have yet to see a large Telco or ITSP deploy their services on
Windows. A while back I have seen some attempts. It was hilarious to
hear that the servers had to be restarted every few hours. Performance
totally sucked, components would crash and the solution was, even by
telco standards, ridiculously expensive. So no, they are not on equal
footing when it comes to performance (and other aspects).

 I don´t mean another
 slow cygwin port, I man a native Asterisk for windows. In fact, I
 would invest on the project if somebody wants to do it.

If you really want to use Windows then have a look at FreeSWITCH as it's
available on Windows too. Then there is also Lync and 3CX. Good luck
keeping your Windows boxes from getting hacked with all the financial
and other damage it would cause.

Regards,
Patrick

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Re: [asterisk-users] dahdi-tools 2.8.0-rc4 - udev rules not installed?

[Patrick Lists]

On 12/03/2013 06:35 PM, Russ Meyerriecks wrote:
 This is why we love release candidate feedback! Thanks! I've managed to
 mis-tag rc4 and missed all of Oron's commits.
 
 Cutting a v2.7.0.2 and a (correct) v2.8.0-rc5 today.

Thanks. I'll give rc5 a spin when it arrives and report back if I find
anything else out of the ordinary.

Regards,
Patrick

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Re: [asterisk-users] Problems compiling dahdi modules

[Patrick Lists]

On 12/02/2013 04:19 PM, Russ Meyerriecks wrote:
 This is fixed on the dahdi-linux master branch and will be included in
 the next release:
 
 More info:
 http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary
 http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=5ec9d756aac1a0eb5c1f48eb110e80946b43f41a
 https://issues.asterisk.org/jira/browse/DAHLIN-330

Thanks Russ. Any chance there will be a v2.8.0-rc3 release with these fixes?

Regards,
Patrick

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Re: [asterisk-users] DAHDI 2.7.0.1 and CentOS 6.5

[Patrick Lists]

On 12/02/2013 10:09 PM, Bakko wrote:
 Hello,
 
 during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error:
[snip]

This was discussed earlier today and Russ pointed to the fixes:

http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=5ec9d756aac1a0eb5c1f48eb110e80946b43f41a
https://issues.asterisk.org/jira/browse/DAHLIN-330

The fix will be in 2.8.0-rc3. Either wait for the rc3 or add the patch
to your build (don't know if it works on 2.7.0.1).

Regards,
Patrick

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[asterisk-users] dahdi-tools 2.8.0-rc4 - udev rules not installed?

[Patrick Lists]

Hi,

I just looked at 2.8.0-rc4 and noticed the udev rules/apps change which
are now supposed to be part dahdi-tools. After make, make install and
make config it seems the dahdi.rules are not installed. I couldn't find
a reference to it in the Makefile either. Did I miss something or has
the move to dahdi-linux not yet been completed?

Regards,
Patrick

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Re: [asterisk-users] Voicemail greeting playback issues?

[Patrick Lists]

On 11/26/2013 12:24 AM, Doug Lytle wrote:
 Bryant Zimmerman wrote:
 Hey all

 I believe I found the bug in Asterisk 11.xxx If someone can help me
 verify it.
 
 Actually,
 
 I wouldn't consider it a bug.  I've know for years that you need to
 answer a channel before you play back audio or strange things can and
 will happen.

That's what I do since the 0.x days. IIRC in recent Asterisk versions
some apps answer before doing anything else. Guess the voicemail app is
not one of them. I always answer first followed by a small Wait and then
execute the app.

Regards,
Patrick

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Patrick Lists]

On 10/28/2013 07:29 PM, Eddie Mikell wrote:

All,

The users in our organization are well, quite frankly, sick of phone
service that is being provided.  The choppy phone calls, and drop outs
are detrimental to our sales force.

I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine


That's a good start. Now what have you done to conclude that the 
Asterisk server is not the cause of your problems?



More than enough bandwidth


That's irrelevant. It's about the quality of that bandwidth. Have you 
figured out if there might be a lot of packetloss or are you perhaps on 
a cablelink which is a *shared* medium? Once your link hits the box in 
the street it shares it with others who might be eating up all the 
bandwidth with their torrent downloads etc.? Use tools like iperf, smoke 
ping and mtr to see if there are obvious problems on the route to your 
VoIP provider.



Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Once the packets leave your premises and your ISP/cable company starts 
messing with them a QoS setting is generally not honored so not very 
helpful unless your LAN is congested.



Put analog lines in the conference room to avoid the dropouts -
leave the sip lines in place for day to day use


If those analog lines are cheap, easy to get then as an intermediate 
solution I would order those analog lines as fast as I could. Or fix the 
VoIP problems, whichever is faster.



Hire a consultant


An experienced VoIP consultant should be able to tell you what is or 
could be causing your problems. With your users sick of phone service 
it suprises me that you haven't already hired one.



Ditch the system and buy a pre-packaged system - RingCentral or some
such.


And what if it's your Internet link or the route to your VoIP provider? 
What if your VoIP provider is messing up?



There are no local asterisk professionals who can help, and we are a
little leery of opening up our system to outside consultants.


If you don't want that then you don't want that but given the state your 
users are in I would be less worried about giving a Consultant access to 
the Asterisk box and more worried about my job :-)



Anyone else face the above, and finally abandoned Asterisk for a
commercial system?


I have seen that once years ago where some clueless sales guy had 
totally oversold an ancient Asterisk/Bristuff/ISDN setup which was very 
buggy and crash prone. There was no way to make that work reliably. 
After the supplier failed for months I was brought in to review the 
setup and possibly fix it. Told the customer to cut its losses. So they 
kicked out their supplier and opted for a different setup.



We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
conference rooms.


I don't know how Grandstream is these days. I thought the GXP2100 was ok 
but I guess you already know if there's a problem with those phones from 
the (lack of) intra-office call complaints from your users.



Suggestions welcome.


Hire a Consultant or someone who has been part of this Community for a 
while and is well known on this list or in #asterisk on irc. Provide 
remote access if required. Change passwords afterwards.


If you really don't want to provide remote access then find a reputable 
VoIP provider with a switch physically as close as possible to your 
location, get a DID for a few bucks, hook it up to your Asterisk box and 
route it to a line on your phone, grab your cell, call that DID and see 
if you still have the problem. It wouldn't be the first time that the 
link between you and your VoIP provider just doesn't cut it. Or maybe 
your VoIP provider just sucks and you need to change to a different one. 
Both flowroute.com and voip.ms work well for me (no affiliation). Or 
maybe your Internet link sucks and you need to change your ISP.


Good luck.

Regards,
Patrick

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Re: [asterisk-users] Asterisk TON number

[Patrick Lists]

On 09/25/2013 09:22 AM, Endri Stefani wrote:

Hi

Greeting to all you out there.

I am new at asterisk, I have been working with PLMN platforms
telecommunication for 5 years with NSN and Huawei.

We have recently built an asterisk PBX with Trixbox and connected it to
our MSS using Digium E1 cards(ISDN). Everything went smoothly as there
are tons of information out there, except for the TON number.

If you have worked in Telecommunication you will know the importance of
TON flexibility.

All the posts online suggested to update under Chan_dahdi.conf:

pridialplan = international

prilocaldialplan = international

or other TON value ,restart the platform and then trixbox1*CLI dialplan
reload

I have already done this with no success. Are there other changes I have
to make in order to change dialplan?


Afaik the Trixbox Community Edition is no longer developed. So unless 
you use a commercial Trixbox you are perhaps better off with just the 
latest stable Asterisk (asterisk 11.5.1 or 11.6.0-rc1, dahdi 2.7.0.1, 
libpri 1.4.14, libss7 1.0.2) or if you need a GUI have a look at 
Elastix, the FreePBX distro, etc.


Regards,
Patrick



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Re: [asterisk-users] Asterisk TON number

[Patrick Lists]

On 09/25/2013 01:57 PM, Endri Stefani wrote:

Hi Patrick,

If I use latest stable asterisk will I be able to change dialplan by changing 
pridialplan in chan_dahdi.conf?


AFAIK yes.

You may also want to check out Asterisk The Definitive Guide (4th 
edition is the latest). Paperback version:

http://www.amazon.co.uk/Asterisk-Definitive-Guide-Russell-Bryant/dp/1449332420

And voip-info.org has a ton of information (not always current though):
Search pridialplan on this page for more info:
http://www.voip-info.org/wiki/view/chan_dahdi.conf

Regards,
Patrick

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Re: [asterisk-users] Somewhat-OT: Stupid NAT tricks to learn from Apple?

[Patrick Lists]

Hi Kristian,

On 09/20/2013 03:17 PM, Kristian Kielhofner wrote:

I've been spending some time looking at some of the significant
changes Apple has made to Facetime in iOS 7.  I'm far from an Apple
fanboy but some of them are pretty interesting:

- multiplexing everything over a single UDP port
- deflate compression with SIP
- various /slight/ protocol violations ;)

More here:

http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html

As SDP bodies swell more and more can we hope to build significant
support for multiplexing and deflate compression in the SIP-focused
open source ecosystem?


Thanks for sharing your analysis. Interesting read. Makes me wonder why 
not more vendors/projects are doing port multiplexing. Let's hope it 
will pick up steam now that Apple has implemented it.


Regards,
Patrick

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Re: [asterisk-users] ISDN outgoing caller id

[Patrick Lists]

On 08/27/2013 08:04 PM, Gergo Csibra wrote:

Hi,

is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port bri card with misdn on other.
The first installation have p-t-mp configuration, the second one is
p-t-p. Both configuration is EuroISDN in Hungary.

So, can anybody help me?


Have you checked with your Telco if they allow you to change the 
callerid? If yes, are you setting the callerid to a number that you are 
allowed to use? You can't just set callerid to any number you like. You 
must own the number which you want to set callerid to. I have no 
problem setting the callerid on outgoing calls via chan_capi to one of 
the numbers that the telco assigned to me.


Regards,
Patrick



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Re: [asterisk-users] Am I being hacked?

[Patrick Lists]

On 08/19/2013 08:10 PM, Eric Wieling wrote:

One of Asterisk's dirty little secrets is that it does not show the source IP 
when a device or hacker tries sending a call without registering.  The 
rejection message in the logs do not show the IP of the attacker.   Yes it 
sucks, yes it has been that way for many many years.


Are you aware of a patch that would show the source IP in the console 
and logs?


Regards,
Patrick



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Re: [asterisk-users] Am I being hacked?

[Patrick Lists]

On 08/19/2013 08:55 PM, Steve Edwards wrote:

On Mon, 19 Aug 2013, Ira wrote:


 [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c:
Failed to authenticate device
390sip:3...@xx.xx.xxx.xxx;tag=2762c06e

xx.xx.xxx.xxx is my public I.P.


What kind of filtering are you doing? Iptables?

Rather than playing 'wack-a-mole' with hackers, my first line of defense
is to 'white-list' just the few legitimate connections between my
clients and their SIP providers.

If your situation requires remote and mobile access, can you at least
'black-list' certain countries with a propensity for hacking? Do you
need access from China, North Korea, Iran, etc?

You can eliminate a very large percentage of hacking attempts with just
a few rules. Then you can focus better on the remaining threats.


Agree. The ip blocks from ipdeny.com come in handy either blocking 
countries that have no business accessing your Asterisk box or 
whitelisting countries/ip ranges that do.


Regards,
Patrick


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Re: [asterisk-users] Am I being hacked?

[Patrick Lists]

On 08/19/2013 09:29 PM, Eric Wieling wrote:

Actually, you can try enabling the security logging destination in 
logger.conf.  I believe that may contain the info, but it is new in Asterisk 11.  1.8 and 
earlier does not have this.


Thanks I'll give that a try.

Regards,
Patrick

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Re: [asterisk-users] Echo Cancellation

[Patrick Lists]

On 07/25/2013 11:51 AM, bilal ghayyad wrote:

Hello;

If our Digium Telephony Card does not support echo cancellation like
(1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome
the echo?


Use the free OSLEC echo canceller software module or Digium's commercial 
HPEC echo canceller software module. Google is your friend.


Regards,
Patrick


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Re: [asterisk-users] LUA

[Patrick Lists]

On 07/18/2013 03:56 PM, jacob.e.mi...@l-3com.com wrote:

I am attempting to setup my server to use Lua for the dialplan
(extentions.lua), but I am unable to get the asterisk configure script
to find the installation of Lua on my box.  I have downloaded the Lua
sources from the www.lua.org site, and I have installed via the “make
linux install” command.  I can execute lua scripts via the command line,
but asterisk configure script is unable to find the installation of Lua.


That's probably because Asterisk is not looking in /usr/local.


I am on a closed network, so no access to the internet so I am not able
to just install Lua using yum.


You should have downloaded the lua RPMs to e.g. your laptop, then copy 
them to your Asterisk box with e.g. a USB stick and then install the Lua 
RPMs on your Asterisk box with:


$ sudo yum install ./lua*

You can find the CentOS 6.4 x86_64 Lua RPMs here:

http://mirror.stanford.edu/yum/pub/centos/6.4/os/x86_64/Packages/

Regards,
Patrick


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Re: [asterisk-users] Adjusting confbridge call quality

[Patrick Lists]

On 07/10/2013 06:46 PM, Chris Gentle wrote:
[snip]

and then others can connect via SIP.  For some reason, when the
speaker says words with S's and F's, they almost sound distorted.  Not
quite static but you can tell the quality has been affected.  May just
be a side-effect of 8,000 Hz.  Just wondered if there way some way to
improve that.


The distorted S and F are prevented by a pop filter in front of the mic. 
Are you using a pop filter? Also if you are using a cheap mic, do 
yourself a favor and invest in a decent mic. It will make a world of 
difference.


Regards,
Patrick

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Re: [asterisk-users] Asterisk 11 security log, fail2ban, drive-by SIP attacks

[Patrick Lists]

On 07/08/2013 01:46 PM, Giles Coochey wrote:

Just a note that I did a little work to extend FreePBX distro with some
extra Fail2Ban which deals with some drive-by SIP registration attempts.

My regex is poor to middling, but the steps detailed here:
http://www.coochey.net/?p=61 manage to stop IPs which try to
authenticate against Asterisk which FreePBX were not able to stop before.

I would welcome any improvements anyone would care to submit and I'll
extend the article a little.

The changes need the Asterisk security log feature, which I think was
only introduced in later versions of Asterisk (e.g. v11).


It seems your rule is not yet present in fail2ban 0.8.10.0. The only one 
close to it is:


SECURITY%(__pid_re)s [^:]+: 
SecurityEvent=InvalidAccountID,EventTV=[0-9-]+,Severity=[a-zA-Z]+,Service=[a-zA-Z]+,EventVersion=[0-9]+,AccountID=[0-9]+,SessionID=0x[0-9a-f]+,LocalAddress=IPV[46]/(UD|TC)P/[0-9a-fA-F:.]+/[0-9]+,RemoteAddress=IPV[46]/(UD|TC)P/HOST/[0-9]+$


See 
https://github.com/fail2ban/fail2ban/blob/0.8.10/config/filter.d/asterisk.conf


Might be an idea to submit it for future inclusion.

Regards,
Patrick


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Re: [asterisk-users] Is it possible to provision lock Aastra phones?

[Patrick Lists]

On 07/06/2013 03:35 PM, Bruce B wrote:

Thanks Patrick.

Do the encrypted config files safe guard against hard resets such as
Web Recovery mode - aka holding down 1  # sign at startup? My
main purpose is to lock the sets due to contract terms so I'd rather not
see user steal the phone and break contract without payment.


It's been a quite a while since I setup Aastra provisioning so a bit 
fuzzy on the details but from what I recall it worked pretty well. The 
Admin Guide can probably give you more info.


Regards,
Patrick


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Re: [asterisk-users] Is it possible to provision lock Aastra phones?

[Patrick Lists]

On 07/06/2013 08:15 AM, Bruce B wrote:

Hi everyone;

Is it possible to provision lock Aastra phones to provider so that no
soft or hard reset can unlock them?


Iirc you can use encrypted configs using an app called anacrypt and lock 
them down. The admin guide (3.2.2) has more details in section 2-14, 
5-44 - 5-46 and A-187 - A-189.


http://www.aastra.com/cps/rde/aareddownload?file_id=6950-16962-_P06_XMLdsproject=aastramtype=pdf

http://www.aastra.com/document-library.htm?curr_nav=2curr_fam=Aastra+6750iprod_id=6950#

Regards,
Patrick

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Re: [asterisk-users] Digium Analog card and Asterisk

[Patrick Lists]

On 07/04/2013 05:32 PM, 杨华杰 wrote:

Hi

I just bought some digium analog cards and I would like to build an IVR
system for my customers.

However I am googling and googling , I didn't find any blog and
instruction for beginners like me.  So I come here for help. Any tips or
blogs will help.



http://www.asteriskdocs.org/
https://wiki.asterisk.org/wiki/dashboard.action
http://www.asterisk.org/community/documentation

Have fun!

Regards,
Patrick

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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

[Patrick Lists]

On 06/11/2013 04:44 PM, Jonas Kellens wrote:
[snip]

Ok thanks.

Any idea how I can resolve this ?

Even if there *can* be more than 1 digit, in case there is only 1 digit
it should go faster.


Would it help if they pressed for example 1 followed by the # key?
If not then, as Eric mentioned, redesign your dialplan. Any IVR with a 
double digit amount of options needs some rethinking. IMHO the average 
attention span of a person is such that at option 6 they forgot options 
1 through 5. And if the option explanations last longer than 5 seconds 
it gets even worse.


Regards,
Patrick


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Re: [asterisk-users] OC3/STM-1 Line Card

[Patrick Lists]

On 06/09/2013 06:35 PM, Nick Khamis wrote:

Anyone?


Sangoma has a multiplexer:
http://www.sangoma.com/products/stm1mux-fiber-multiplexer/

Which you could then use with:
http://www.sangoma.com/products/a116-16-span-t1e1j1-board/

And there is this card:
http://www.signalogic.com/index.pl?page=asterisk_ip_pbx

Regards,
Patrick


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Re: [asterisk-users] Which dahdi/libpri combo for BRI/PtmP ?

[Patrick Lists]

On 06/06/2013 05:55 PM, Olivier wrote:

Hi,

I need to rebuild a system which has 4 BRI ports and is connected to
Point-to-multiPoint lines, in a country where telco often drop lines
for energy savings.


I think the dropped D-channel issue should be handled by a very recent 
DAHDI. If there are still issues file a bug. Don't know about PtMP.



I'm planning to use latest 11.4.0 asterisk version along with dahdi and
libpri (no misdn).

Which version are recommended for Dahdi and Libpri ?


Probably the very latest you can find. If the latest stable does not 
work for you then try an RC of an upcoming version or get master from git.



My main requirements are, beside having calls coming in and out:
- keep a meaningful pri show spans status (pri show spans outputs Down
when the line is down (cable unplugged, no q921 traffic, ...))
- avoid sporadic ERROR messages in logs.


Guess you have to test.

Regards,
Patrick

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Re: [asterisk-users] MeetMe exit status?

[Patrick Lists]

On 06/03/2013 06:47 PM, Matthew Jordan wrote:

On 06/02/2013 08:36 PM, Patrick Lists wrote:

Hi,

Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I
know for example if a conf ended normally or if someone gave a wrong
conf number or pin?

Thanks,
Patrick



There is no channel variable that provides that level of granularity.
The closest available is the MEETMESECS channel variable, which tells
you how many seconds the participant was in the conference.

You can find a full list on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/MeetMe+Channel+Variables


Thanks Matt. I'll see if I can use MEETMESECS.

Regards,
Patrick




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[asterisk-users] Missing dahdi-firmware RPM in asterisk-current repo at http://packages.asterisk.org

[Patrick Lists]

Hi,

The dahdi-firmware package seems to be missing in the asterisk-current 
repo on http://packages.asterisk.org


-- Finished Dependency Resolution
Error: Package: dahdi-linux-2.6.2-1_centos6.x86_64 (asterisk-current)
   Requires: dahdi-firmware

Can this please be fixed.

Thanks,
Patrick

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[asterisk-users] MeetMe exit status?

[Patrick Lists]

Hi,

Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I 
know for example if a conf ended normally or if someone gave a wrong 
conf number or pin?


Thanks,
Patrick

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Re: [asterisk-users] dahdi driver not getting install

[Patrick Lists]

On 05/13/2013 01:14 PM, Salaheddine Elharit wrote:

hi

You can download a tarball of the release here:

http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz


At least give the link to the latest release which is not 2.6.2-rc1 but 
2.6.3-rc1:


http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.3-rc1.tar.gz

Or he could use the yum repo from Digium:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
http://packages.asterisk.org/centos/6/current/x86_64/RPMS/

Regards,
Patrick


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Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

[Patrick Lists]

Hi Carlos,

On 04/28/2013 10:56 PM, Carlos Alvarez wrote:

We have a new customer with a lot of old phones like the 9133i.  They
won't register, and we see some very strange behavior with them.  If
the SIP peer exists, they simply fail silently, with no error in the
CLI or the messages log.  Nothing works, but no errors.

If the peer does not exist, it's clear that it's registering improperly:

[2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from
'abc123 sip:abc123@' failed for '68.2.x.x' - No matching peer found

Typically of course we'd expect to see:  sip:abc123@server

We're running the latest available firmware, but it's from 2009.  Any
ideas on this before we just trash all the older phones?


I reviewed one of those a long time ago. I'm afraid all I can remember 
is that it had its fair share of issues. I did a lot of factory resets 
and had to set the config through tftp *or* the UI but not mix both. Did 
you try doing factory resets (page 9-11 in admin guide) and/or 
downgrading to an older firmware release?


Main info page:
http://www.aastra.ca/document-library.htm?curr_fam=Aastra+9000icurr_nav=2prod_id=6441

The 1.4.0 firmware:
https://www.voipon.co.uk/products/download/aastra_9133i.zip

The 1.4.3 firmware:
http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6449-_P07_XMLdsproject=www-aastratelecom-commtype=zip

Release notes of 1.4.3:
http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6427-_P07_XMLdsproject=www-aastratelecom-commtype=pdf

Manual
http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6425-_P07_XMLdsproject=www-aastratelecom-commtype=pdf

Regards,
Patrick

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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

[Patrick Lists]

On 04/04/2013 09:54 PM, Joseph wrote:

+1.7044972383


If that number is his actual number, maybe create a script that calls 
him 10 times an hour, every hour between 00:00 - 07:00am and plays 
screaming monkeys every time he picks up (or his voicemail kicks in).


Regards,
Patrick




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Re: [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support

[Patrick Lists]

On 04/03/2013 02:48 PM, Marshall Henderson wrote:

Hi Tzafrir-

I know where to find the DAHDI source, but I was more asking where to
actually find which chipsets are supported within the source. Any thoughts?


Have you checked the PCI IDs in the source?

Regards,
Patrick


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Re: [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support

[Patrick Lists]

On 04/03/2013 08:34 PM, Marshall Henderson wrote:

Hi Patrick- Yes, I did find the list of PCI IDs (I think). Do these look
right (from wctdm.c):

static DEFINE_PCI_DEVICE_TABLE(wctdm_pci_tbl) = {
 { 0xe159, 0x0001, 0xa159, PCI_ANY_ID, 0, 0, (unsigned long)
wctdm },
 { 0xe159, 0x0001, 0xe159, PCI_ANY_ID, 0, 0, (unsigned long)
wctdm },
 { 0xe159, 0x0001, 0xb100, PCI_ANY_ID, 0, 0, (unsigned long)
wctdme },
 { 0xe159, 0x0001, 0xb1d9, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmi },
 { 0xe159, 0x0001, 0xb118, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmi },
 { 0xe159, 0x0001, 0xb119, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmi },
 { 0xe159, 0x0001, 0xa9fd, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
 { 0xe159, 0x0001, 0xa8fd, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
 { 0xe159, 0x0001, 0xa800, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
 { 0xe159, 0x0001, 0xa801, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
 { 0xe159, 0x0001, 0xa908, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
 { 0xe159, 0x0001, 0xa901, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
#ifdef TDM_REVH_MATCHALL
 { 0xe159, 0x0001, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
#endif
 { 0 }
};

So, next question, how do I take those device IDs and find the
associated chip?


Look up the PCI ID at for example:

http://pciids.sourceforge.net/
http://www.pcidatabase.com/

The 0x159 is the TigerJet chip. The third column is the vendor. So if 
you lookup 0xb100 you will find that it is OpenVox.



Going a step further, after a bit of research it appears some
manufacturers have gone away from dedicated hardware chips like the
TigerJet 320G to FPGA general purpose chips with firmware to control
operation. Is this correct? Any thoughts?


No idea so no thoughts :-)

Regards,
Patrick


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Re: [asterisk-users] ERROR: Unknown signalling method ss7

[Patrick Lists]

On 03/14/2013 11:04 PM, mohsen feyzzadeh wrote:

Hi all
I installed
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.


In case the 12 in Fedora 12 was not a typo, you do realize that Fedora 
12 has been end-of-line for years and has more security holes than Swiss 
cheese? It makes sense to upgrade to the latest version of Fedora (which 
is 18) or switch to CentOS 6.4 which is more suited for server 
applications. You may also want to look at the latest versions of DAHDI 
(2.6.2/2.6.3rc) and Asterisk (11.2.1) assuming both work with an 
appropriate version of libss7.


Regards,
Patrick


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Re: [asterisk-users] digium card and virualbox

[Patrick Lists]

On 03/11/2013 04:18 AM, bilal ghayyad wrote:

I am not mixing. I need this for LAB testing.
How? This PCI passthrough, how to enable it on virualbox?


It's in the VirtualBox manual.

Regards,
Patrick


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Re: [asterisk-users] Laptop error

[Patrick Lists]

On 03/11/2013 12:53 PM, termo termosel wrote:

Hi,

I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in
desktop computer, asterisk starts without problem but if I insert the
same USB in a laptop computer Asterisk doesn't start. Is it possible
because different microprocessors?


Yes. If you made the USB stick on a x86_64 (64 bit) computer and then 
try it on a x86 (32 bit) laptop, it will not work.


Regards,
Patrick




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Re: [asterisk-users] Sending SMS from asterisk

[Patrick Lists]

On 03/11/2013 07:07 PM, Asghar Mohammad wrote:

HI Bilal,
i am using chan_mobile for call termination, you can use it but you need
to tweak chan_mobile.c it is broken from a long time.
let me know if you want give it a try.


If you could send the patches you made to chan_mobile to this mailing 
list then other Asterisk users can benefit from your work and use 
chan_mobile too.


Regards,
Patrick


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Re: [asterisk-users] Serviced Office operator panel

[Patrick Lists]

On 03/12/2013 12:07 AM, Andrew Yager wrote:

Hi,

I'm trying to find (with some desperation now) a decent web based or
application based UI that integrates with an Asterisk based PBX and is
designed for a Serviced Office environment.

Key features we're looking for:


Don't know if it covers your requirements but here's another commercial 
solution: http://www.getisymphony.com/


Regards,
Patrick


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Re: [asterisk-users] Where can get the latest manual our user guide

[Patrick Lists]

On 02/08/2013 06:35 AM, Ding Peng wrote:

Hi, everybody,

  Where can I get the manual or user guide of latest asterisk version,
1.11.x?
I want to know the syntax and usage of all the supported functions or
something like that in the latest version.


You can find one on the O'Reilly website. Don't recall the link so you 
have to google for it. And the Asterisk wiki has a lot of info about 
version 11.


Regards,
Patrick



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Re: [asterisk-users] DECT Solution

[Patrick Lists]

On 01/24/2013 10:37 AM, Zyumbilev, Peter wrote:

Hello,


I need to setup system of aroud 60 DECT phones with asterisk.

So far I found
http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710

However is there some cheap base station(similar to GSM cell) so I can
handle all DECT phones centralized and plug it inside asterisk ?


Aastra has DECT base stations that can hook up to an Asterisk server. 
Last time I set one up it worked fine. You may want to try out several 
different brands of DECT phones and see which one the users like best. 
You don't want to get 50 support calls a day from your users complaining 
about how much the DECT phones suck.


http://www.aastra.com/product-families.htm?curr_cat=DECT+Infrastructurecurr_type=Familymode_f=1mode_c=1mode_l=4

Polycom also has DECT stuff. I doubt it will come cheap.
http://spectralink.polycom.com/dect_communications/index.html

Regards,
Patrick


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Re: [asterisk-users] clicking sound with alaw codec

[Patrick Lists]

On 01/24/2013 09:44 PM, Richard Kenner wrote:
[snip]

When I use alaw, the path from Asterisk to the Alcatel is completely
clean, but the other way has a set of clicks that kind of sound like
old-fashioned audio noise.

[snip]

It's been ages since I experienced that but things to check that come to 
mind in no particular order are:


- DAHDI settings (sync source)
- Asterisk server not properly grounded
- timing is off (check logs)
- shared interrupts (make sure nic/TDM card have their own)
- jitterbuffer settings (try on/off)
- echo cancellation going bonkers (OSLEC?)
- QoS (proper priority for voice packets?)
- PCI slot (if you have a card, try changing the slot it's in)

Use Wireshark to see the difference between a good call and a bad one. 
If you see a lot of time jumps on the bad call then look at your 
network/QoS.


Regards,
Patrick


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Re: [asterisk-users] clicking sound with alaw codec

[Patrick Lists]

On 01/24/2013 11:57 PM, Richard Kenner wrote:

- jitterbuffer settings (try on/off)


I added
   jbenable=yes

and get lots of:

[Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: 
DAHDI/i1/2128518396-6c7 received frame with invalid timing info: 
has_timing_info=1, len=0, ts=371371424, src=RTP
[Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: 
DAHDI/i1/2128518396-6c7 received frame with invalid timing info: 
has_timing_info=1, len=0, ts=371371434, src=RTP


Check https://issues.asterisk.org/jira/browse/ASTERISK-12042

Regards,
Patrick


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Re: [asterisk-users] Details process to configure Asterisk in CENTOS

[Patrick Lists]

On 01/22/2013 08:54 AM, Sakharam Thorat wrote:


Can  anybody send me Detailed process to configure Asterisk in CENTOS ??

Detailed description highly appreciated.


Start by reading the Asterisk book, check asterisk.org and Google around 
to see if your question has already been answered.


Regards,
Patrick


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Re: [asterisk-users] Need Help

[Patrick Lists]

On 01/17/2013 09:05 PM, Joe Ruffolo wrote:

Hi all! In need of some serious help. We currently run Trixbox and Cent
Os on a 2u server for our small business’s phones system.


Afaik Trixbox is no longer maintained and their forum are hardly active 
anymore so it may be a bit of a challenge to get support. If you really 
need a GUI like Trixbox then I suggest you have a look at Elastix which 
is very much alive and has a large community and professional services 
to help you out. See http://www.elastix.org/ Or have a look at Digium's 
Switchvox (payware). Whatever you do/choose, make sure that your box is 
secure if you open it up to the Internet.


Regards,
Patrick


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Re: [asterisk-users] Top Posting

[Patrick Lists]

On 01/02/2013 07:11 PM, Carlos Alvarez wrote:

The number of questions posted here that are easily answered with a
search or which are far too basic and open (how do I make Asterisk work)
is very high these days, and that does kill a list.  A lot of us are
interested in helping people who help themselves, and solving complex
problems.  I've seen many tech lists die off when people stop trying to
help themselves and ask intelligent questions.


Good point Carlos and I share your feeling. On the Postfix mailing list, 
when someone asks a basic how do I ... question, inevitably the 
response is one or more links to a section in the documentation. And 
that works really well. The interesting problems discussed on that ML 
outnumber the questions from those who can't be bothered to try to help 
themselves by spending a couple of minutes reading the docs. I would 
welcome similar responses on this mailing list to improve the S/N ratio.



As to top-posting, another example of when I think it's generally
acceptable is people using tablets.  I have found no way on either my
iOS or Android tablets to quickly/easily post in the traditional manner.
  If I'm faced with spending a few minutes carefully trimming a useful
reply or just not posting it at all, I'm likely to choose the latter if
I'm on a list that says absolutely never top post.


I only use Thunderbird to post but I now have seen several arguments 
that MUAs like Outlook and iOS/Android clients are simply  not capable 
of bottom posting  trimming. Perhaps the list admins could take that 
into account when appropriate.


Regards,
Patrick


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Re: [asterisk-users] Top Posting

[Patrick Lists]

On 01/02/2013 06:20 PM, Steve Totaro wrote:

I became a list member way before any such rule and never had to click
through and agree to these update ToS.

I am grandfathered in.


Just looked it up. I see my first post back in April 2003, yours in 
September 2003 and Jon in March 2003. Wow you find something fun to play 
with and suddenly a decade has passed :-)


Regards,
Patrick


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Re: [asterisk-users] Top Posting

[Patrick Lists]

On 01/02/2013 09:46 PM, jon pounder wrote:

On 01/02/2013 03:22 PM, Patrick Lists wrote:

On 01/02/2013 06:20 PM, Steve Totaro wrote:

I became a list member way before any such rule and never had to click
through and agree to these update ToS.

I am grandfathered in.


Just looked it up. I see my first post back in April 2003, yours in
September 2003 and Jon in March 2003. Wow you find something fun to
play with and suddenly a decade has passed :-)


Are you sure about that ? I know I was doing stuff with asterisk back in
the LSS days and that was around 2001


I only looked at the list archives. LSS definitely predates anything 
else so it's safe to say you are dinosaured in :-)


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

Here's to another decade of fun!

Regards,
Patrick

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Re: [asterisk-users] Top Posting

[Patrick Lists]

On 12/30/2012 04:26 PM, Ron Wheeler wrote:

I participate in a lot of lists and top posting is now the norm since
people want to see quickly if the message is worth reading.


Isn't it a bit of a stretch to extrapolate your experience with your 
lists to top posting being the norm? I am subscribed to several lists 
and bottom posting, proper trimming and commenting inline is the norm there.


Actually the norm is determined by the list rules. If the list rules say 
one must use bottom posting then one should use bottom posting. If 
someone does not like that then don't subscribe, find another source to 
ask a question (the forum, LUG, hire a consultant) or just bottom post.


Questions come before answers.
Answers come after questions.

-1 against changing rule #5.

Regards,
Patrick


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Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files

[Matt Riddell (lists)]

There's no priority in your call file. 

Sent from my iPhone

On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:

 Hello,
 
 I noticed that when i move a call file to outgoing directory, two asterisk 
 threads are dealing with it.
 
 ]# grep FAX_44731.call /var/log/asterisk/full.2
 
 [Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on 
 /var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
 [Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1] 
 System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3] 
 System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or 
 extension must be specified, along with tech and dest in file 
 /var/spool/asterisk/outgoing/FAX_44731.call
 [Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in 
 /var/spool/asterisk/outgoing/FAX_44731.call, deleting
 
 As you see there are two thread dealing with my call file. Now let's inspect 
 the thread 18852.
 
 ]# grep \[18852\] /var/log/asterisk/full.2 
 [Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on 
 DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
 [Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
 [Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME:  NUM: 90312xxx
 [Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer 
 capability: 0x00 - SPEECH
 [Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1] 
 System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:2] 
 SendFAX(DAHDI/i1/0312xxx-b08, /tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new 
 stack
 [Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel 
 'DAHDI/i1/0312xxx-b08' sending FAX:
 [Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
 /tmp/Qg90Ox5YGF5kYkJu.tif
 [Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3] 
 System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel 
 'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
 [Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup 
 'DAHDI/i1/0312xxx-b08'
 [Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to 
 DAHDI/g0/0312xxx
 
 It seems that the thread 18852 executes it normally but the thread 26842 
 deletes my call file. And when I inspected the asterisk log file, i saw that 
 the thread 26842 is deleting all my call files.
 
 Here is my custom_extensions.conf file:
 
 [asteriskgw_fax]
 exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}  
 /var/spool/asterisk/outgoing/FAX_${ID}.call)
 exten = s,2,SendFAX(${FAXFILE},zdfs)
 exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}  
 /var/spool/asterisk/outgoing/FAX_${ID}.call)
 
 And here is a sample of call file:
 
 Channel: DAHDI/g0/0312xxx
 MaxRetries: 0
 RetryTime: 60
 Context: asteriskgw_fax
 Extension: s
 Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
 Set: ID=44884
 Callerid: 90312xxx
 Archive: Yes
 
 
 
 -- 
 Necati DEMİR
 
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Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Patrick Lists]

On 11/13/2012 12:11 AM, Phil Reynolds wrote:
[snip]

It turns out to be a known issue:

https://issues.asterisk.org/jira/browse/ASTERISK-19532

... and can be fixed by applying the patch at:

https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch

I will file the details with Debian too...


Is it an omission that this fix has not been applied to the 11 tree? 
From the looks of ASTERISK-19532 it seems that the fix has only been 
applied to 1.8 and 10.


Regards,
Patrick

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Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Patrick Lists]

On 11/13/2012 07:05 PM, Michael L. Young wrote:
[snip]

Is it an omission that this fix has not been applied to the 11 tree?
  From the looks of ASTERISK-19532 it seems that the fix has only been
applied to 1.8 and 10.



If you click on the link for ASTERISK-19532, there is a tab in the Activity section 
labeled Subversion.  It shows that the patch was applied to 1.8, 10, 11 and 
trunk.


Thanks Michael. Missed that one. Good to know.

Regards,
Patrick


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Re: [asterisk-users] B200p card - use dahdi or mISDN?

[Patrick Lists]

On 10/16/2012 08:50 AM, Sebastian Arcus wrote:

I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
choosing which software to use? Is one better than the other - or does
one have features which are not present in the other?


I would go for DAHDI so you can use the card like you would use any 
Digium card. OpenVOX also seems to focus on DAHDI integration. Looking 
at the OpenVOX site it seems that you will need to use the patched DAHDI 
from here:


http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/

Regards,
Patrick


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Re: [asterisk-users] asterisk installation under a single directory

[Patrick Lists]

On 10/15/2012 09:07 AM, sudeep melekar wrote:
[snip]

i m completely new to asterisk
so any help would be appreciated


If you are totally new to Asterisk I recommend you first read the 
Asterisk book and go through the wiki. Both have sections how to install 
the various Asterisk components.


Regards,
Patrick


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Re: [asterisk-users] Connecting Skype to Asterisk

[Patrick Lists]

On 10/12/2012 11:17 PM, Philip Bennefall wrote:

 From what I gather, it costs extra for each channel even for direct
Skype to Asterisk calls. Since my plan was to use this for business
purposes, I'd need at least something like 30 channels which would be
way out of my monthly budget unfortunately.


If you *really* need this then have a look at FreeSWITCH which has a 
module for Skype calls (in/out) without the need for Skype Connect and 
its fees. Afaik you can use a regular Skype account and iirc even 
multiple Skype accounts.


Regards,
Patrick

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Re: [asterisk-users] LDAP Driver and VoiceMail

[Patrick Lists]

On 10/04/2012 10:00 PM, Phil Daws wrote:

Hello:

I am investigating the possibility of using LDAP for storing certain Asterisk 
configuration parameters.

I have examined res_ldap.conf and see where mailbox can be defined from 
AstAccountMailbox but I do not see where the password can be stored ?


I've never looked at res_ldap but wouldn't a look at the schema tell you 
that?


Regards,
Patrick



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Re: [asterisk-users] CDR Unanswered calls

[Patrick Lists]

On 10/05/2012 11:51 AM, Shanavaz E A wrote:

Hi,
No replies until now. Some one please help... There must be some people
who are using it...
Thanks


No idea but since Asterisk is making you money why don't you hire an 
experienced Asterisk consultant to get it resolved.


Regards,
Patrick


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Re: [asterisk-users] How to log caller IP address in the CDR?

[Patrick Lists]

On 10/05/2012 02:10 PM, Benoit Panizzon wrote:

Hello

We had this situation:

Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
Server was abused to call a large number of expensive destinations.


I'm sorry to hear that. In the Asterisk source there is a doc that 
focuses on security. you might want to read that. Google should give you 
more information about Asterisk/SIP security. Also you may want to 
install something like fail2ban which prevents brute forcing by banning 
originating IP addresses after a few failed attempts.


Regards,
Patrick


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Re: [asterisk-users] FAX via Asterisk

[Patrick Lists]

On 09/27/2012 08:15 AM, Shanavaz E A wrote:
[snip]

Patrick, can you please give the steps to configure fax with iaxmodem
and hylafax. Is it free to use?


It's been years since I set it up so I don't know exactly how to 
configure it anymore. But I do remember that I found some howto/docs via 
Google so try that. And yes both Hylafax and iaxmodem are free.


Regards,
Patrick



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Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/

[Patrick Lists]

On 09/28/2012 03:01 AM, Patrick Archibald wrote:

Hi,

Is there a way to move 100 .call files in to
/var/spool/asterisk/outgoing/ at once and have Asterisk call at
maximum 10 at a time?


Afaik that is not possible. Wouldn't it make more sense to move call 
files in batches of 10 to outgoing/?


Regards,
Patrick



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Re: [asterisk-users] FAX via Asterisk

[Patrick Lists]

On 09/26/2012 05:53 PM, Mark Robinson wrote:

Hello.
I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip,
Aastra 6757i. Everything works as expected.
We also have a FAX machine. We need to be able to use that FAX machine
to send or receive faxes. We are planning to have a dedicated did for
faxes. Before, FAX machine was connected directly to pots line.

Any digestions how to accomplish it?


In addition to Markus' suggestions you could also look at using iaxmodem 
and Hylafax. I've been using that for years and it works great. Once a 
fax is received it is emailed to whatever you configure.
Sending faxes works too in libreOffice. If you have a bunch of DIDs you 
could give people their own direct line and fax number.


Regards,
Patrick


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Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

[Patrick Lists]

On 09/25/2012 11:18 PM, Logan Bibby wrote:

MyISAM would be best, in my opinion. The features that cause the little
bit of performance overhead in InnoDB wouldn't be necessary for CDR storage.


Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not. 
More information here:


http://en.wikipedia.org/wiki/ACID

https://blogs.oracle.com/MySQL/entry/comparing_innodb_to_myisam_performance

Regards,
Patrick


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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Patrick Lists]

On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote:
[snip]

Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)


Your DAHDI and Asterisk versions are old so for starters I would update 
everything to the latest releases. See asterisk.org.


Regards,
Patrick


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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[Patrick Lists]

On 09/14/2012 10:25 AM, A J Stiles wrote:
[snip]

It could be nothing more than a dry solder joint on one of the RJ45s.  For the
sake of five minutes' work with a soldering iron, that's got to be worth
eliminating.


Wouldn't that void your warranty? I would take it up with Digium support 
and let them sort it out.


Regards,
Patrick



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Re: [asterisk-users] Grandstream VoIP phones

[Patrick Lists]

On 01-09-12 04:14, Vladimir Mikhelson wrote:
[snip]

  * Ability to send host name or other CN not equal to the phone IP in
TLS negotiation


Afaik you usually put alternative CNs in SubjectAltName in the 
certificate. Have you tried that?


Regards,
Patrick


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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Patrick Lists]

On 08/30/2012 09:45 AM, Gopalakrishnan N wrote:

Hi,

I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host,
I am not using any virtualbox, still i struck in loading the modules.


Please do not top post.

Install strace and then start asterisk with the command:
# strace asterisk

That should give you some low level info what's going on. More info 
about strace and available options can be found in:


$ man strace

Regards,
Patrick


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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Patrick Lists]

On 27-08-12 08:25, Gopalakrishnan N wrote:

This is really tuff working with OpenSuse. I am clueless how to sort our
this.


Maybe switch to a different distribution? I have used CentOS and RHEL 
for years without any problems and as far as I know both debian and 
ubuntu should work without problems too.


Regards,
Patrick



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Re: [asterisk-users] can we install 10 PCI card on asterisk

[Patrick Lists]

On 27-08-12 14:08, DHAVAL INDRODIYA wrote:

Hi All,

i would like to know if anyone has done or having idea regarding PRI
terminations in asterisk.

i have a requirement where i need to support 80 PRI in one machine i
have found a machine which have 10 PCI slots available

now i am thinking of arranging 8port sangoma card in this pci slots so i
can arrenge 10 card in that.

is it possible to run system like that ? is it good idea , can asterisk
handle 2400 calls if machine size and RAM is good.



I don't think Asterisk can handle that many DAHDI channels and I have 
never heard of an Asterisk box with more that 16 PRI's.


Taking a step back, do you really want to put all your eggs in one 
basket? what if the box fails? That's 2400 channels going down and 
unavailable until you fix it. That will cost a log of money and get you 
angry clients. It makes more sense to spread the lot across different 
servers. Besides that, is your telco willing to provide you with 80x PRI 
or will they insist on aggregating it to several E3 links or something 
higher (STM-1)?


If you really insist on going down this route have a look at FreeSWITCH 
or look at something like Cisco, Alcatel, Telco Systems etc.


Regards,
Patrick


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Re: [asterisk-users] Understanding CHANNEL function values

[Patrick Lists]

On 25-08-12 14:31, Stefan at WPF wrote:

Hello all,

I need some help understand the values of the CHANNEL function, e.g.

txploss // local packets loss
rxploss // remote packets loss
txjitter  // local jitter
rxjitter  // remote jitter


My main problem in understand is that a CHANNEL has two nodes (sender
and receiver), while a typical setup includes at least 3 nodes:
SIP phone - Asterisk - SIP Provider ( - each is a node)

1) So e.g. txploss, is it
- what is lost between SIP phone and Asterisk
- what is lost between Asterisk and SIP Provider
- or probably both?


I would assume that those statistics apply to a leg and not an 
end-to-end connection. So in your example I would assume that a txploss 
value is determined for the leg between the SIP phone and the Asterisk 
server and another txploss value is determined for the leg between the 
Asterisk server and the upstream SIP provider.


Interesting stuff. If you figure it all out, please update this thread 
(and possibly the wiki).


Regards,
Patrick


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Re: [asterisk-users] xmpp / sip

[Patrick Lists]

Hi Hans,

On 24-08-12 10:13, Hans Witvliet wrote:

Hi all,

After making a nice demo-setup for one of our innivationmanagers, he
came up with a completely different stratagy ;-(


Well if you could create it then obviously it's no longer innovative so 
they had to come up with something else :-)



They want to have an Ejabberd server, with xmpp-clients.
When you see a contact coming online, just point and click for making a
phone call.


The concept sounds like what Cisco was using internally. The Asterisk 
Wiki only mentions Tigase so you may want to verify that ejabberd 
supports the required XMPP PubSub stuff.


https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub

Regards,
Patrick


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Re: [asterisk-users] quick questions on version 10

[Matt Riddell (lists)]

I thought this was discussed and it was going to be left in?

Sent from my iPhone

On 23/08/2012, at 2:30 PM, Jerry Geis ge...@pagestation.com wrote:

 The AMI action CoreShowChannels deprecated the CLI concise command
 because the output of the AMI action is extensible without breaking
 existing systems.  The CLI command is not extensible without breaking
 existing systems.
 Richard,
 
 Thanks - I tried the CoreShowChannels AMI and it says:
 
 Response: Follows
 Privilege: Command
 No such command 'CoreShowChannels' (type 'core show help CoreShowChannels' 
 for other possible commands)
 --END COMMAND--
 
 In my manager.conf I have 
 read = system,call,command,agent,user,reporting
 write = system,call,command,agent,user,originate,reporting
 
 Did I miss something?
 
 Jerry
 
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Re: [asterisk-users] Asterisk 1.8 and 11

[Patrick Lists]

On 22-08-12 20:04, Giuseppe Longo wrote:

Just a little questions, what's the difference between asterisk 1.8
and asterisk 11?


Iirc you can check the ChangeLog in the Asterisk 11 sources.

Regards,
Patrick


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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Patrick Lists]

On 14-08-12 08:29, Gopalakrishnan N wrote:

If I change autoload=no then asterisk is starting, but post to that
loading modules even chan_sip.so asterisk hangs. Its strange, only in
OpenSuse I am facing this. In CentOS, Ubuntu its working fine, same
Asterisk version with same hardware.


Please do not top post and properly trim your replies.

Have you made sure that on the OpenSuse box your DNS is configured 
properly? You should be able to lookup your IP address/FQDN both ways. 
So for example 192.168.1.1 (replace with your IP adres) should resolve 
in your.box.com (replace with your FQDN) and vice versa your.box.com 
should resolve into 192.168.1.1. See man dig or man nslookup for 
commands that can do DNS lookups.


Regards,
Patrick



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