[Asterisk-Users] [OT] Small SIP phones?

[Antony Stone]

Hi.

Does anyone know of any small SIP phones (and preferably have some experience 
of using them and happy to recommend them)?

By 'small' I mean a single-piece phone, with dial buttons in the handset, so 
that it can be carried around easily in a laptop bag.   Something like 
http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a 
standard analogue telephone).

Ideally I'd like something without a cradle, which can simply be put on a desk 
and answered by picking it up.

Thanks,

Antony.

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Re: [Asterisk-Users] [OT] Small SIP phones?

[Antony Stone]

On Sunday 12 December 2004 20:12, Clay Reiche wrote:

 I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty
 small) along with any standard analogue phone.
 http://www.worldaccxx.com I have one and carry it around in my laptop bag.
 Demensions are 6x4.5x1.25

Thanks.   In fact I already have a Grandstream ATA-486, which I'm very pleased 
with: http://www.grandstream.com/y-ht486.htm   This unit is even smaller - 
105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just wondering if there's a 
neat all-in-one solution, instead of carrying around two items?

Regards,

Antony.

 -Original Message-

 Hi.

 Does anyone know of any small SIP phones (and preferably have some
 experience of using them and happy to recommend them)?

 By 'small' I mean a single-piece phone, with dial buttons in the handset,
 so that it can be carried around easily in a laptop bag.   Something like
 http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a
 standard analogue telephone).

 Ideally I'd like something without a cradle, which can simply be put on a
 desk and answered by picking it up.

-- 
Never automate fully anything that does not have a manual override capability. 
Never design anything that cannot work under degraded conditions in emergency.

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Re: [Asterisk-Users] codec order in SIP doesn't work

[Antony Stone]

On Wednesday 15 December 2004 14:21, Roy Sigurd Karlsbakk wrote:

 hi

 using the following in sip.conf, codec preferences aren't set, and
 asterisk uses alaw whatever I do, except force it to one specific in
 the [user]

 [general]
 disallow=all
 allow=g726
 allow=g729
 allow=gsm
 allow=alaw

 then, from 'sip show peer something' it tells me

Codecs   : 0x11a (gsm|alaw|g726|g729)
Codec Order  : (none)

 can someone please explaing why?

http://www.fnords.org/~eric/asterisk/sip.conf.shtml
http://lists.digium.com/pipermail/asterisk-dev/2004-September/006406.html

 this is version

 Asterisk CVS-v1-0-10/08/04-17:29:04 built by [EMAIL PROTECTED] on a i686
 running Linux

Try an upgrade to current stable version:

http://lists.digium.com/pipermail/asterisk-users/2004-December/076772.html

Regards,

Antony.

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Re: [Asterisk-Users] TDM400p FXO module always offhook

[Antony Stone]

On Wednesday 15 December 2004 22:34, Carey Pillar wrote:

 I have a TDM400p with 3 FXS mods and 1 FXO mod.  I have all set up with
 what seems to be correct settings (according to digium and asterisk wiki).
  As soon as I plug in my POTS line into FXO mod the line goes into offhook
 state (whether I have power to the card or not).  Should this happen?
 When I try to call * box all I get is busy signal.  I've installed stable
 version, cvs version, change the phone cord from 2 wire to 4 wire. My
 config files are standard from compile and all i've added is info from
 digium's basic hardware configs.

 Any suggestions?

Are you sure you're using the correct cable to plug the FXO into the POTS?

It sounds to me like maybe you've got a crossed cable where you need a 
straight one, or vice versa - there are certainly plenty of variations in 
cables from different vendors, for different bits of equipment, in different 
countries, so maybe you just need to try another one?

There should basically only be two types - line signals on pins 1-4, or on 
2-3, so there shouldn't be too much to test...

Hope this helps,

Antony.

-- 
Bill Gates has personally assured the Spanish Academy that he will never allow 
the upside-down question mark to disappear from Microsoft word-processing 
programs, which must be reassuring for millions of Spanish-speaking people, 
though just a piddling afterthought as far as he's concerned.

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Re: [Asterisk-Users] SIP Server question / recommendations

[Antony Stone]

On Thursday 16 December 2004 01:09, Shahed wrote:

 Hello All,
 I am new to *, and this is my first post on the user list.

 I have had success with making / receiving calls to a SIP hardware Phone
 and the Console Channel Driver.

 Can anyone please suggest what would be a good SIP server to use, or is
 there a way in which I can use asterisk itself as a SIP server for my phone
 and make calls to it using the console ?? 

Yes, Asterisk is a SIP server - see /etc/asterisk/sip.conf

Antony.

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Re: [Asterisk-Users] Polycom SIP Phones

[Antony Stone]

On Thursday 16 December 2004 22:57, Jared Armstrong wrote:

 I found IP 500's for $170.

Where?

Antony

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Re: [Asterisk-Users] My Boss wants background music!!!!

[Antony Stone]

On Friday 17 December 2004 14:31, Steven Kalcevich (Lists) wrote:

 Why not just dial an extention for music when the user wants music
 from there desk.

Because then the phone will be engaged on a call and will not ring when 
someone else wants to talk to the person at the desk?

Antony.

 The requirement of the original poster was to mute the music at the desk
 when a call is in progress.

 It would be really nice if there was a hardphone capable of accepting a

 multicast high-quality stream when no call was in progress.

-- 
I own three Windows books, published by O'Reilly.   They are Windows 
Annoyances, Office 97 Annoyances and Windows 98 Annoyances.   That 
pretty much sums it up for me.

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Re: [Asterisk-Users] Asterisk and HylaFax

[Antony Stone]

On Friday 17 December 2004 16:22, Lee Howard wrote:

 On 2004.12.17 05:42 Sergio Serrano wrote:
  Hi all,
  again I try configure Hylafax with asterisk. I would like
  configure
  Asterisk in the next way:
  1)An incoming fax go into through X100P
  2)Asterisk detects Fax and redirect fax to Hylafax
 
  Is it possible?

 Yes, but it may not be as pretty as you like, and it may not function
 as well as you hope.

 Using faxdetect in your zapata.conf file will get practically all of
 the faxes coming in to the X100P routed to the fax extension.  The
 trick, then, is how to get HylaFAX at that fax extension.

Does there exist any sort of bypass box which could be used in the following 
arrangement:

POTS - X100P - Asterisk - TDM400P(FXS) - Fax machine

Hypothetical bypass box also plugs into POTS line and Fax machine, able to 
switch the X100P, Asterisk and the TDP400P out of the circuit, and just 
connect POTS to Fax directly on some command from the Asterisk PC.

Then Asterisk uses faxdetect to send ringing to the fax machine, waits for 
call to be answered, and sends (RS232?) command to bypass box, allowing fax 
machine to take the original incoming call without all the analogue - 
digital - analogue conversion going on.

If such a hypothetical bypass box could also detect remote hangup, and drop 
itself back out of circuit once the call is complete, everything returns back 
to normal ready for the next call to come in.

Electrically it seems like a very simple solution - a 2-pole 2-way relay with 
RS232 control and line-voltage detection (for the automatic switchover on 
hangup), however whether such a thing exists and has appropriate type 
approvals I have no idea

Regards,

Antony.

-- 
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Re: [Asterisk-Users] application meetme

[Antony Stone]

On Friday 17 December 2004 18:34, Geoffroy KOUMADI wrote:

 i have problem to setup application meetme. i'm using asteisk-1.0.3 and
 sjphone as client.

Thanks for letting us know.

If you want some help in solving the problem, perhaps you might tell us what 
the actual problem is?

Useful information might include:
 - what are you doing?
 - what is working?
 - what is not working?
 - how do you know it's not working?
 - what debugging information does Asterisk tell you when it's not working?

Also, tell us if things other than meetme *are* working correctly - for 
example, can you make calls between the different clients which are trying to 
join the meetme conference?

Antony.

-- 
It is also possible that putting the birds in a laboratory setting 
inadvertently renders them relatively incompetent.

 - Daniel C Dennett

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Re: [Asterisk-Users] Call on hold disconnects...

[Antony Stone]

On Friday 17 December 2004 20:24, Ferguson, Michael wrote:

 G'Day All,

 How do I fix this:

 I receive a call at the extension. Press the hold button. Music on hold
 starts. When I place the handset back on the cradle, the call gets hung
 up/disconnected. The Phone is A GrandStream Budge Tone 100.

1. What would you _expect_ to happen when you do this?
2. If this is a problem, then don't hang up the phone?
3. If you don't want this to happen, how _would_ you hang up if that was what 
you did want to happen?

Antony.

-- 
I don't know, maybe if we all waited then cosmic rays would write all our 
software for us. Of course it might take a while.

 - Ron Minnich, Los Alamos National Laboratory

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Re: [Asterisk-Users] Call on hold disconnects...

[Antony Stone]

On Friday 17 December 2004 20:43, Ferguson, Michael wrote:

 OK. I guess I was not clear. Sorry.

 The phone rings.
 The person picks up the handset and speaks to the caller.
 He then puts the call on hold by pressing the HOLD button on the GS
 100 phone.
 The caller hears music on hold.

So far, so good.

 The hand set is placed back on the cradle (as is done on a regular phone
 with a hold button)

I'm not sure I agree with this.   Some phones may allow you to hang up and not 
disconnect the call, but I don't think it's universal.   Some phones 
interpret this to mean oh, you want to hang up? Okay - I'll hang up the call 
then.

 The call is disconnected.

Well, yes, because you hung up.

What happens if you do something else, like dial another extension, or press 
the hold button again (perhaps to retreive the original caller)?

I repeat one of my original questions - if this is not what you expected to 
happen when you hang up the phone, how would you expect to hang up the call 
when you wanted to?

Antony.

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Re: [Asterisk-Users] Total newbie here looking to do a VoIP conference call?

[Antony Stone]

On Friday 17 December 2004 21:00, Patrick Campbell wrote:

 I am looking to help out my company find a more budget conscious but
 reliable way to hold conference calls between 5+ people.  4x a month we
 hold several hour long conference calls during non-business hours.  All of
 the employees have high speed internet.  Currently we dial up an ATT conf
 using regular analog phones.

 I don't have a great grasp as to what Asterisk is capable of, but my
 thoughts were that perhaps with VoIP telephone lines (either hooked up to
 the company's network or just using a 3rd party VoIP provider such as
 Packet8, which is whatI have for personal use) and an Asterisk server, that
 we could setup a VoIP conference bridge.

meetme is what you want.

 Can someone enlighten an unknowledged as to whether or not this is
 possible, and if so, how might it be done?  Would the Asterisk server need
 X number of VoIP lines?  I.e. If there's 10 participants, it'd need 10 VoIP
 lines?

There isn't really a concept of VoIP lines - each remote participant just 
comes in to the Asterisk server on your normal Internet connection - they 
each need their own SIP phone, of course, and they each need to have an 
Internet link, but as far as Asterisk is concerned, it just needs a 
connection with sufficient bandwidth to handle the total number of conference 
subscribers.

Antony.

-- 
There are two possible outcomes:

 If the result confirms the hypothesis, then you've made a measurement.
 If the result is contrary to the hypothesis, then you've made a discovery.

 - Enrico Fermi

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Re: [Asterisk-Users] Call on hold disconnects...

[Antony Stone]

On Friday 17 December 2004 21:10, Ferguson, Michael wrote:

 Antony,
 Thanks. It seems that the GS will not keep the call on hold.
 In the real world though, when you place a call on hold, it is held until
 further action.

Yes, although I might think that hanging up is a further action?

 The caller will hear messages, music, anything while you 
 are gone to look for a file, etc.

 Technically, if you place the call on hold and put the handset back on the
 cradle, you DID NOT HANG UP to end the call. If you want to hang up the
 call you will first have to take the call off hold... No.

Hm, yes, that is one reasonable way to expect things to work.   I guess it 
comes down to how this particular phone expects things to work, and if the GS 
doesn't support hangup on hold the way you want, then it's just not going to 
do things effectively for you :(

Oh well,

Antony

-- 
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but at the end of the day the size of the crowd at your funeral
will be largely dictated by the weather.

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Re: [Asterisk-Users] voicemail without prompt

[Antony Stone]

On Friday 17 December 2004 21:25, Ross Kevlin wrote:

 this would still only work if the mailbox number was the same as the caller
 id. I need some way to get the actual mailbox number of the caller.

Where / how are your mailbox numbers stored?

It shouldn't be too difficult to create a script or DB request to provide the 
CID and get the mailbox number in response?

Just out of interest, why don't you make the mailbox ID = caller ID?

Antony.

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If you want to be happy for a year, get married.
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Re: [Asterisk-Users] Asterisk Hardware

[Antony Stone]

On Friday 17 December 2004 21:42, Nihal wrote:

 Does some hardware just not work very well with Asterisk?

Yes.   (or, no, depending on how you view the question)

 I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram.

Some people have reported problems with FC3, I don't know if FC2 is the 
same...

 While listening to the demo over a softphone (over the LAN) I get a number
 of crackles and skips.

 IS THIS NORMAL FOR ASTERISK?

No.

 Or is it hardware related?

It may well be software related - try a real SIP phone instead of a softphone 
and see if the problems persist.

Softphones are not as good as hardware SIP phones.

Regards,

Antony.

-- 
In Heaven, the police are British, the chefs are Italian, the beer is Belgian, 
the mechanics are German, the lovers are French, the entertainment is 
American, and everything is organised by the Swiss.

In Hell, the police are German, the chefs are British, the beer is American, 
the mechanics are French, the lovers are Swiss, the entertainment is Belgian, 
and everything is organised by the Italians.

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Re: [Asterisk-Users] Total newbie here looking to do a VoIPconfer ence call?

[Antony Stone]

On Friday 17 December 2004 23:04, Patrick Campbell wrote:

 Come to think of it since the DTA310 uses DNS to find the SIP server, you
 could setup a DNS cache and override the DNS entry for what packet8 uses
 (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your
 own SIP server?  Kind of a hack but it should work as long as it's running
 on port 15062.  I am very new to this so I don't know if there's a port
 standard for SIP like there is for HTTP, SSH, FTP, etc.?

5060

Antony.

-- 
Abandon hope, all ye who enter here.
You'll feel much better about things once you do.

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Re: [Asterisk-Users] [OT] Small SIP phones?

[Antony Stone]

On Sunday 12 December 2004 23:08, Shoval Tomer wrote:

 I may be wrong, but if you always carry your laptop around, why don't
 purchase a USB handset?

The main reason is that (I believe) the quality of audio with a soft phone is 
generally not as good as that from a real SIP phone?

The other reason is that I want to be able to show VoIP in operation to 
clients (which is where I would be taking the phone with me), so a standalone 
phone, which is not dependent on any software installed on my laptop, is a 
much neater arrangement.

  -Original Message-
  From: Florian Overkamp [mailto:[EMAIL PROTECTED]
 
  On Sun, 2004-12-12 at 21:27, Antony Stone wrote:
   Thanks.   In fact I already have a Grandstream ATA-486, which I'm
   very pleased with: http://www.grandstream.com/y-ht486.htm   This unit is
   even smaller - 105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just
   wondering if there's a neat all-in-one solution, instead of carrying
   around two items? 
 
  Three, in fact. The powersupply also adds to the required space. This
  is one of the biggest advantages of having an all-in-one solution, because
  you don't have to generate high voltage/high power ring signalling.

True, you don't need the high voltage ringing, but with a standard SIP phone 
you still need a PSU for it.   I couldn't rely on a client's network switch 
supporting PoE for when I wanted to plug one in.

Regards,

Antony.

-- 
Wanted: telepath.   You know where to apply.

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Re: [Asterisk-Users] How expensive are the different codecs? (Regarding CPU time)

[Antony Stone]

On Wednesday 15 December 2004 21:26, Michael Vogel wrote:

 Jim Van Meggelen schrieb:
  YIKES! What kind of processor have you got there?

 Its a:
 - Pentium II (Deschutes) 333MHz
 - 128mb memory

 I'm using it as:
 - Mailserver (IMAP, SMTP)
 - Webserver (mainly for webmail)
 - Newsserver
 - Packet Radio station
 - VNC server
 - Proxy
 ...

   22:22:10 up 10 days,  1:49,  5 users,  load average: 0.01, 0.09, 0.13
 167 processes: 163 sleeping, 2 running, 2 zombie, 0 stopped
 CPU states:  12.1% user,   7.4% system,   0.0% nice,  80.5% idle
 Mem:126740K total,   124172K used, 2568K free, 4760K buffers
 Swap:   345356K total,   173684K used,   171672K free,22992K cached

 Is it a little bit too much for such a machine? What could be the
 bottleneck? CPU? Memory? Interrupts?

My advice would be to whack in a load more RAM - basically, try to get the 
poor little thing so it doesn't need to use swap.   That will make a big 
difference to performance.

Regards,

Antony.

-- 
I know I always wanted to be somebody, but I guess I should have been more 
specific.

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Re: [Asterisk-Users] Newbie setup (Hardware questions)

[Antony Stone]

On Wednesday 15 December 2004 23:24, Puddle wrote:

 Thanks, that makes a lot more sense.  Would VoIP
 phones still require FXO units or would that not
 require any special telephony hardware?

SIP phones connect by ethernet - no telephony hardware needed.

You would want an FXO port if you want to plug your Asterisk into the public 
phone system (PSTN), so your SIP phones can call normal phone numbers, and/or 
receive calls from normal phones.

However, be aware that you can get SIP - PSTN connectivity from external 
service providers (ie: they give you a phone number, when someone calls it, 
they forward the call to your Asterisk or SIP phone, or when you dial out 
from Asterisk, they will connect you to normal phone numbers (and charge you 
for the call, of course...))

Regards,

Antony.

-- 
Success is a lousy teacher.  It seduces smart people into thinking they can't 
lose.

 - William H Gates III

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Re: [Asterisk-Users] Open Ports

[Antony Stone]

On Saturday 18 December 2004 10:17, Norman Zhang wrote:

 Hi,

 May I ask what ports are necessary for SIP communication through a
 firewall? I read somewhere that UDP/5060 alone is enough. Some
 recommends more ports to be opened for RTP.

Both the above statements are correct.

SIP uses port 5060

RTP uses multiple ports, typically in the range 1-2

Remember that SIP and RTP are different - SIP is used to set up the call; RTP 
is used to carry the audio once the call has been set up.

Regards,

Antony.

-- 
Anything that improbable is effectively impossible.

 - Murray Gell-Mann, Nobel Prizewinner in Physics

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Re: [Asterisk-Users] Open Ports

[Antony Stone]

On Saturday 18 December 2004 10:58, Norman Zhang wrote:

  SIP uses port 5060
 
  RTP uses multiple ports, typically in the range 1-2
 
  Remember that SIP and RTP are different - SIP is used to set up the call;
  RTP is used to carry the audio once the call has been set up.

 Thanks. May I ask what security control can be applied to RTP besides
 reducing the opened range? Are there stateful inspection can be done on
 this?

What insecurity exists from leaving the range open?

I am not aware of any stateful helper modules (eg for netfilter) which handle 
RTP streams, and certainly not any which understand the relationship between 
SIP and RTP (eg by matching source/destination IP addresses), however I 
wouldn't have thought it should be too difficult to write a netfilter module 
to get RTP treated as related to an existing SIP connection?

But, to return to my initial question, what's the security risk in leaving 
your Asterisk server open to UDP packets from the world?

I regard it like a mail server - a firewall allowing TCP packets through to 
port 25 cannot protect against an application vulnerability in the MTA; the 
application server itself has to be secure for your system to be safe.   Same 
goes for a web server, or an Asterisk server.

Regards,

Antony.

-- 
Never automate fully anything that does not have a manual override capability. 
Never design anything that cannot work under degraded conditions in emergency.

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Re: [Asterisk-Users] Open Ports

[Antony Stone]

On Saturday 18 December 2004 11:40, Rich Adamson wrote:

  But, to return to my initial question, what's the security risk in
  leaving your Asterisk server open to UDP packets from the world?
 
  I regard it like a mail server - a firewall allowing TCP packets through
  to port 25 cannot protect against an application vulnerability in the
  MTA; the application server itself has to be secure for your system to be
  safe.   Same goes for a web server, or an Asterisk server.

 If you have a small number of remote locations passing through the
 firewall, and, you write your inbound firewall rules to allow specific
 Ip addresses, and, you forward those to a specific internal Ip address,
 then there isn't much of a security issue.

 However, if you open all udp ports (eg, 1 - 2) inbound _and_
 you happen to have other services running on that box that _might_ use
 those ports, then you're allowing access to those other services as
 well. (How many trojans, etc, happen to use ports in that range?)

I agree entirely - and I regard keeping your system free from trojans as an 
application security matter, not a network security matter (which is what 
firewalls are).

Make sure you know what applications are running on a machine (and make sure 
you trust them) before you open it to the Internet.   A firewall can't help 
against an application exploit.

Regards,

Antony.

-- 
Anyone that's normal doesn't really achieve much.

 - Mark Blair, Australian rocket engineer

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Re: [Asterisk-Users] Re: Open Ports

[Antony Stone]

On Saturday 18 December 2004 13:21, Tom Ivar Helbekkmo wrote:

 My home firewall allows my Asterisk PBX to send any UDP traffic to
 anyone, and keeps state, so they can answer.  It also specifically
 allows anyone to connect to UDP port 5060 on the PBX.

Interesting.   Does that allow other people to call you (first packets are 
inbound) as well as you calling other people (first packets are outbound)?

I guess the first few packets from them to you might get dropped because they 
don't match an established outbound connection, but as soon as you start 
sending packets to them, your firewall will allow two-way flow...

Have you done this using netfilter?

Antony.

-- 
Perfection in design is achieved not when there is nothing left to add, but 
rather when there is nothing left to take away.

 - Antoine de Saint-Exupery

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Re: [Asterisk-Users] VoIP Termination

[Antony Stone]

On Saturday 18 December 2004 18:07, Dorn Hetzel wrote:

 I wouldn't say I hate SIP, it sucks less than H.323 and
 so on by a large margin.  But, having said that, if you
 can use IAX, it sucks even much than SIP does :)

Um, are you saying IAX is good, or that it is not good?   I'm not sure I 
understand your statement above.

If you are saying that IAX is bad, why?  And what's better?

Regards,

Antony.

-- 
I don't mind that he got rich, but I do mind that he peddles himself as the 
ultimate hacker and God's own gift to technology when his track record 
suggests that he wouldn't know a decent design idea or a well-written hunk of 
code if it bit him in the face. He's made his billions selling elaborately 
sugar-coated crap that runs like a pig on [sedatives], crashes at the drop of 
an electron, and has set the computing world back by at least a decade.

 - Eric S Raymond, about Bill Gates

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Re: [Asterisk-Users] My Boss wants background music!!!!

[Antony Stone]

On Saturday 18 December 2004 20:19, Bill Seddon wrote:

 Detecting the ringing state of a specific device from, say, a desktop
 running Windows or Linux AGI is trivial.

Care to share a trivial example with us?

Sounds like a useful link for several applications...

Antony.

-- 
Software development can be quick, high quality, or low cost.

The customer gets to pick any two out of three.

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Re: [Asterisk-Users] modprobe wcfxo crashes my IBM NetFinity5000 after few seconds

[Antony Stone]

On Saturday 18 December 2004 20:27, Rodolfo Grave wrote:

 Hi and thanks once more.

 I moved the card around, and it kept the same IRQ. Then I went into
 setup and changed it. This is the output of lspci -v now:

 01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537
  Subsystem: Unknown device 8085:0003
  Flags: bus master, medium devsel, latency 144, IRQ 5
  I/O ports at 4b00 [size=256]
  Memory at c0fdf000 (32-bit, non-prefetchable) [size=4K]
  Capabilities: [40] Power Management version 2

 That's not a shared IRQ. However, the problem remains. Just after one
 min or so of executing modprobe wcfxo, the PC reboots.

 Any other ideas? This card worked great on another PC, so a hardware
 missfunctioning is not a probable choice.

Was the other PC the same architecture (CPU, m/b chipset)?

It may be that your motherboard simply doesn't do what Asterisk needs (I've 
heard that VIA chipsets in particular can be a problem, Intel ones seem 
okay).

Antony.

-- 
Bill Gates has personally assured the Spanish Academy that he will never allow 
the upside-down question mark to disappear from Microsoft word-processing 
programs, which must be reassuring for millions of Spanish-speaking people, 
though just a piddling afterthought as far as he's concerned.

 - Lynne Truss, Eats, Shoots and Leaves

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Re: [Asterisk-Users] Q about IAX (and IAXy)

[Antony Stone]

On Sunday 19 December 2004 00:26, Nabeel Jafferali wrote:

 I have heard many times that IAX is NAT-transperant. I am unsure how
 it accomplishes this.

 I do know that SIP works like this: your SIP device send a request to
 the SIP server (usually on port 5060) with whatever command. The SIP
 server respends to your device's apparent IP and port (this is decided
 depending on how that NAT is set up, STUN, etc.). The voice is then sent
 to the apparent RTP port on your device (deciding what that is, again,
 would depend on the NAT set up).

Note that in the above description, the SIP communication is one phase of the 
process, RTP (the audio channel) is a separate phase, and operates on totally 
different UDP ports from the SIP phase.   The UDP ports used by RTP vary for 
each conversation, and therefore cannot be known about by a firewall or NAT 
device in advance.

 How does IAX eliminate this problem of ports being mapped by your NAT
 router and external IPs? Does it use one port for both commands and
 voice packets? Does the remote server just use the received from IP
 address and port to respond?

Yes.   IAX uses just a single port (UDP 5036) and IAX2 uses just a single port 
(4569) to send both call setup and audio data between the endpoints.

Therefore a NAT device between two IAX systems has only a single channel, on a 
well-known port number, to deal with, and this is simple to do.

 Finally, would an IAXy work seamlessly in a configuration where it is
 plugged into a NAT router which is plugged into another NAT router  -
 double NATted? The * server is on a public IP.

Yes, so long as both NAT routers allow reply packets back through, this will 
work (and if they don't, they're not much use for anything else either).

Regards,

Antony.

-- 
The problem with television is that the people must sit and keep their eyes 
glued on a screen; the average American family hasn't time for it.

 - New York Times, following a demonstration at the 1939 World's Fair.

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Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality

[Antony Stone]

On Sunday 19 December 2004 02:00, Keith O'Brien wrote:

 Since the incoming stream is using VAD, my assumption is that it is losing
 the timing during the pauses in the speech.   Does anyone know of a way to
 just turn off VAD in *?   This would have multiple benefits (if you have
 the bandwidth).   Turning off VAD will improve voice quality by eliminating
 and front end clipping during talk spurts and I am assuming will also
 minimize the impact of not having a ZAP timing source.

 Is there a way to disable VAD in *?

It seems not:

http://www.voip-info.org/wiki-RTP+Silence+Suppression
http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html
(I'm not sure if that second one has been superseded by more recent events? - 
but the first certainyl suggests that it's the sender which decides whether 
to use VAD or not, not the receiver)

Antony.

-- 
There's no such thing as bad weather - only the wrong clothes.

 - Billy Connolly

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Re: [Asterisk-Users] Q about IAX (and IAXy)

[Antony Stone]

On Sunday 19 December 2004 01:41, Nabeel Jafferali wrote:

 Thanks for all the info so far!

  Therefore a NAT device between two IAX systems has only a
  single channel, on a well-known port number, to deal with,
  and this is simple to do.

 So then how does IAX deal with the equivalent of SIP reinvites? Or are
 all IAX calls' audio carried through the * server?

IAX stands for Inter-Asterisk eXchange :)

See if http://www.voip-info.org/wiki-IAX+versus+SIP or the links from that 
help as well.

Antony.

-- 
When you talk about Linux versus Windows, you're talking about which 
operating system is the best value for money and fit for purpose. That's a 
very basic decision customers can make if they have the information available 
to them. Quite frankly if we lose to Linux because our customers say it's 
better value for money, tough luck for us.

 - Steve Vamos, MD of Microsoft Australia

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Re: [Asterisk-Users] SMS - how to send one

[Antony Stone]

On Sunday 19 December 2004 20:18, Brian West wrote:

 The SMS in asterisk is not SMS like you're thinking... Its not for sending
 to mobile phones and not something usable in the US.

Um, sorry, but if SMS is not for sending to mobile phones, then what is it for 
(if it matters, I'm not in the US) ?

Regards,

Antony.

-- 
Linux is going to be part of the future. It's going to be like Unix was.

 - Peter Moore, Asia-Pacific general manager, Microsoft

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Re: [Asterisk-Users] SMS - how to send one

[Antony Stone]

On Sunday 19 December 2004 21:35, Antony Stone wrote:

 On Sunday 19 December 2004 20:18, Brian West wrote:
  The SMS in asterisk is not SMS like you're thinking... Its not for
  sending to mobile phones and not something usable in the US.

 Um, sorry, but if SMS is not for sending to mobile phones, then what is it
 for (if it matters, I'm not in the US) ?

Apologies for replying to my own posting, but a bit more digging has left me 
even more puzzled - I'm not using SMS yet, but I do plan to, and links such 
as http://lists.digium.com/pipermail/asterisk-cvs/2004-April/001843.html 
http://www.voip-info.org/wiki-Asterisk+cmd+Sms and  
http://www.aaisp.net.uk/aa/sms.html all seem to suggest that it can do what I 
want (and hope) - send  receive text messages to/from standard mobile 
phones.

Am I deluded in this hope?

Antony.

-- 
These clients are often infected by viruses or other malware and need to be 
fixed.  If not, the user at that client needs to be fixed...

 - Henrik Nordstrom, on Squid users' mailing list

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Re: [asterisk-users] Tesco Internet Phone

[Antony Stone]

On Friday 02 March 2007 07:46, Alan Chandler wrote:

 On Thursday 01 March 2007 20:33, bails wrote:
  plug it in a linux box and tell us what it is please,
  generic-usb-audio or what?
 
  Bails
 
  Julian Lyndon-Smith wrote:
   Yeah, that's where firefly comes from, doesn't it.
  
   I've got the base station plugged in, and the handset connected to
   it, but it always says pc unavailable.
  
   My system (xp) sees a usb phone for speakers and microphone,
   but I can't get it to work.

 Did this go any further.  I would be interested in this.

I too would really like to find or help adapt a driver for this.

Here's what I get from my USB-DECT device:

# cat /proc/bus/usb/devices
T:  Bus=01 Lev=02 Prnt=02 Port=00 Cnt=01 Dev#=  6 Spd=12  MxCh= 0
D:  Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs=  1
P:  Vendor=19af ProdID=694d Rev= 0.00
S:  Manufacturer=innoMax Technology Ltd.
S:  Product=Cordless USB Phone
C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=400mA
I:  If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=audio
I:  If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=83(I) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=02(O) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 3 Alt= 0 #EPs= 2 Cls=03(HID  ) Sub=00 Prot=00 Driver=hid
E:  Ad=81(I) Atr=03(Int.) MxPS=  32 Ivl=1ms
E:  Ad=01(O) Atr=03(Int.) MxPS=  32 Ivl=1ms

# lsusb -v -s 006

Bus 001 Device 006: ID 19af:694d
Device Descriptor:
  bLength18
  bDescriptorType 1
  bcdUSB   1.10
  bDeviceClass0 (Defined at Interface level)
  bDeviceSubClass 0
  bDeviceProtocol 0
  bMaxPacketSize064
  idVendor   0x19af
  idProduct  0x694d
  bcdDevice0.00
  iManufacturer   1 innoMax Technology Ltd.
  iProduct2 Cordless USB Phone
  iSerial 0
  bNumConfigurations  1
  Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength  214
bNumInterfaces  4
bConfigurationValue 1
iConfiguration  0
bmAttributes 0x80
MaxPower  400mA
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber0
  bAlternateSetting   0
  bNumEndpoints   0
  bInterfaceClass 1 Audio
  bInterfaceSubClass  1 Control Device
  bInterfaceProtocol  0
  iInterface  0
  AudioControl Interface Descriptor:
bLength10
bDescriptorType36
bDescriptorSubtype  1 (HEADER)
bcdADC   1.00
wTotalLength   60
bInCollection   2
baInterfaceNr( 0)   1
baInterfaceNr( 1)   2
  AudioControl Interface Descriptor:
bLength12
bDescriptorType36
bDescriptorSubtype  2 (INPUT_TERMINAL)
bTerminalID 3
wTerminalType  0x0101 USB Streaming
bAssocTerminal  4
bNrChannels 1
wChannelConfig 0x
iChannelNames   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  3 (OUTPUT_TERMINAL)
bTerminalID 4
wTerminalType  0x0301 Speaker
bAssocTerminal  3
bSourceID   5
iTerminal   0
  AudioControl Interface Descriptor:
bLength12
bDescriptorType36
bDescriptorSubtype  2 (INPUT_TERMINAL)
bTerminalID 1
wTerminalType  0x0201 Microphone
bAssocTerminal  2
bNrChannels 1
wChannelConfig 0x
iChannelNames   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  3 (OUTPUT_TERMINAL)
bTerminalID 2
wTerminalType  0x0101 USB Streaming
bAssocTerminal  1
bSourceID   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 8
bDescriptorType36
bDescriptorSubtype  6 (FEATURE_UNIT)
bUnitID 5
bSourceID   3
bControlSize1
bmaControls( 0)  0x03
  Mute
  Volume
iFeature0
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber1
 

Re: [asterisk-users] Tesco Internet Phone

[Antony Stone]

On Wednesday 21 March 2007 11:57, bails wrote:

 Antony Stone wrote:
  On Friday 02 March 2007 07:46, Alan Chandler wrote:
  On Thursday 01 March 2007 20:33, bails wrote:
  plug it in a linux box and tell us what it is please,
  generic-usb-audio or what?
 
  Bails
 
  Julian Lyndon-Smith wrote:
  Yeah, that's where firefly comes from, doesn't it.
 
  I've got the base station plugged in, and the handset connected to
  it, but it always says pc unavailable.
 
  My system (xp) sees a usb phone for speakers and microphone,
  but I can't get it to work.
 
  Did this go any further.  I would be interested in this.
 
  I too would really like to find or help adapt a driver for this.
 
  Here's what I get from my USB-DECT device:
 
  # cat /proc/bus/usb/devices
  T:  Bus=01 Lev=02 Prnt=02 Port=00 Cnt=01 Dev#=  6 Spd=12  MxCh= 0
  D:  Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs=  1
  P:  Vendor=19af ProdID=694d Rev= 0.00
  S:  Manufacturer=innoMax Technology Ltd.
  S:  Product=Cordless USB Phone
  C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=400mA
  I:  If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=audio
  I:  If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  I:  If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  E:  Ad=83(I) Atr=01(Isoc) MxPS=  64 Ivl=1ms
  I:  If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  I:  If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  E:  Ad=02(O) Atr=01(Isoc) MxPS=  64 Ivl=1ms
  I:  If#= 3 Alt= 0 #EPs= 2 Cls=03(HID  ) Sub=00 Prot=00 Driver=hid
  E:  Ad=81(I) Atr=03(Int.) MxPS=  32 Ivl=1ms
  E:  Ad=01(O) Atr=03(Int.) MxPS=  32 Ivl=1ms
 
  # lsusb -v -s 006

snipped for brevity

  Let me know if I can help with any other info.
 
 
  Antony.

 Whats the output of dmesg when you plug it in?

hub.c: new USB device 00:07.2-1.1, assigned address 7
usbaudio: device 7 audiocontrol interface 0 has 1 input and 1 output 
AudioStreaming interfaces
usbaudio: device 7 interface 1 altsetting 1 channels 1 framesize 2 configured
usbaudio: valid input sample rate 8000
usbaudio: device 7 interface 1 altsetting 1: format 0x0010 sratelo 8000 
sratehi 8000 attributes 0x00
usbaudio: device 7 interface 2 altsetting 0 does not have an endpoint
usbaudio: device 7 interface 2 altsetting 1 channels 1 framesize 2 configured
usbaudio: valid output sample rate 8000
usbaudio: device 7 interface 2 altsetting 1: format 0x0010 sratelo 8000 
sratehi 8000 attributes 0x00
usbaudio: registered dsp 14,19
usbaudio: constructing mixer for Terminal 4 type 0x0301
usbaudio: warning: found 1 of 0 logical channels.
usbaudio: assuming the channel found is the master channel (got a Philips 
camera?). Should be fine.
usbaudio: registered mixer 14,16
usbaudio: constructing mixer for Terminal 2 type 0x0101
usbaudio: unit 0 not found!
usbaudio: no mixer controls found for Terminal 2
usb_audio_parsecontrol: usb_audio_state at c11f93e0
usb_control/bulk_msg: timeout
: USB HID v1.01 Device [innoMax Technology Ltd. Cordless USB Phone] on 
usb1:7.3

-- 
It wouldn't be a good idea to talk about him behind his back in front of 
him.

 - murble

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Re: [asterisk-users] Tesco Internet Phone

[Antony Stone]

On Wednesday 21 March 2007 15:11, asterisk wrote:

 I use this driver for the SJ phone with the USB tesco internet phone:

 http://www.sjphone.org/usbphone/SJphoneDriverWebPoint.exe

Yes, but that's a corded phone which plugs into the USB socket.

# cat /proc/bus/usb/devices
P:  Vendor=19af ProdID=694d Rev= 0.00
S:  Manufacturer=innoMax Technology Ltd.
S:  Product=Cordless USB Phone

is a DECT phone where the base station plugs into the USB socket.

http://buy.tescointernetphone.com/details.asp?idProduct=669

Antony.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 [EMAIL PROTECTED]
 Sent: 21 March 2007 13:12
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: Re: [asterisk-users] Tesco Internet Phone

 I'm also interested in finding a driver for this phone.  I did find a link
 to the drivers page of the manufacturer of the phone Yamamoto.  See the
 link below.  I've also contacted them about drivers for Linux, asterisk
 etc.  I'll report back if I get a reply.

 http://www.yamamoto-group.co.uk/index.php?page=download



 Phil.

-- 
In the Beginning there was nothing, which exploded.

 - Terry Pratchett

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Re: [asterisk-users] Voicemail mailbox number passed in connection?

[Antony Stone]

On Wednesday 21 March 2007 19:54, Lutgring, Sam wrote:

 Does anyone know how to configure a SIP phone to pass the mailbox number
 to the voicemail service when dialing?  I would like to press the
 message waiting lamp and be prompted for my password instead of mailbox
 number.  Can this be passed in the set-up call or based on caller-id?

Caller ID is a simple way to do it.

Make the mailbox number the same as your phone number, then select the mailbox 
based on Caller ID.

It's in some ways more secure, too - it means only you (or at least, only your 
phone) can log in to your mailbox, instead of someone else trying from their 
phone by knowing your mailbox number and guessing your password.


Antony.

-- 
There are only 10 types of people in the world:
those who understand binary notation,
and those who don't.

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Re: [asterisk-users] "Your call is not allowed. P U A M I"

[Antony Stone]

On Thursday 20 April 2017 at 21:29:58, Steve Edwards wrote:

> Not an Asterisk question, but...
> 
> A bunch of our 8xx numbers started playing this recording when dialed. Our
> provider (Inteliquent) says it's not them.

Where are Inteliquent feeding the calls (assuming they connect instead of 
playing that message) to?

Are they a SIP trunk provider, supposedly passing calls to your PBX (in which 
case it's either them or your PBX, so there shouldn't be a lot of discussion)?

Does Inteliquent have any record of the calls being placed IN to the 8xx 
numbers (if they do, this eliminates the possibility of message being played 
by the callER's service provider)?

Does it make any difference which carrier you use to make the call?

> Does anybody know who is playing it and what it means?

I've certainly never heard (of) it.


Antony.

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Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Wednesday 19 April 2017 at 15:44:39, Atux Atux wrote:

> hello there. i am running debian 8 in my swerver and i would like to run
> asterisk as non root. i did follow the
> https://www.voip-info.org/wiki-Asterisk+non-root without any success.

Did you do the very first step:

/etc/init.d/asterisk stop   ?

> when i issue
> root@PBX: ~ $ asterisk -U asterisk -G asterisk
> Privilege escalation protection disabled!
> See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> Unable to access the running directory (Permission denied).

Did you do all the "chown" and "chmod" commands listed in those guidelines?

> Changing to '/' for compatibility.
> Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk
> -r' to connect.

Er, you can't change to running as non-root without stopping the existing 
(started by root) service first...

> root@PBX: ~ $
> 
> any ideas on how to fix that please?

Show us the output of:

# find / -name asterisk -exec ls -ld '{}' \;


Antony.

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Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:

> Hi. thanks a lot for your replies. I did stop the services and i did issued
> the  the "chown" and "chmod" commands listed in the guide.
> It is necessary to compile it, instead if using the apt-get version
> What am i missing?

Let's go back to basics for a moment - you say this is a Debian system - in my 
experience Debian already runs Asterisk as the "asterisk" user and not as 
root, so let's see what you have.

1. Start Asterisk (probably using "/etc/init.d/asterisk start", or maybe 
"service asterisk start")

2. Check who it's running as: "ps aux | grep asterisk"


Antony.


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Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote:

> Hi.
> Here is the output of the command
> 
> root@pbx: ~ $  find / -name asterisk -exec ls -ld '{}' \;
>
> drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
>
> drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
>
> -rwxr-xr-x 1 root root 9719880 Apr 19 17:27 
> /usr/src/asterisk-11.25.1/main/asterisk
>
> drwxrwxr-x 3 1013 users 4096 Apr 19 16:56 
> /usr/src/asterisk-11.25.1/include/asterisk
>
> -rwxr-xr-x 1 root root 9719880 Apr 19 17:32 /usr/sbin/asterisk

Okay, those look reasonable to me - however I'm surprised at some which are 
missing:

/var/log/asterisk
/var/spool/asterisk
/var/run/asterisk

Did you *stop* Asterisk before trying to change it to run as non-root?

I think Tzafrir Cohen's comments are very well worth following.


Antony.

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Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote:

> root@PBX: /var/www/html $ /etc/init.d/asterisk start
> [ ok ] Starting asterisk (via systemctl): asterisk.service.

I'm somewhat puzzled that your root-user prompt is "$"
instead of the more normal "#", but never mind...

> root@PBX: /var/www/html $ ps aux | grep asterisk
> asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49 
> /usr/sbin/asterisk -U asterisk -G asterisk

So, the first column of that output shows you that asterisk is
running as the user "asterisk".

On my Debian system I only have "-U asterisk" without the "-G asterisk".

> root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep asterisk

...and the grep command was run by "root"

> root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
> Privilege escalation protection disabled!
> See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk
> -r' to connect.

Who does "ls -l" show you that file /var/run/asterisk/asterisk.ctl
is owned by?

On my machine it's:

srwxrwx--- 1 asterisk asterisk 0 Apr 11 10:32 /var/run/asterisk/asterisk.ctl


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Re: [asterisk-users] IAX2 getting stuck

[Antony Stone]

On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then reconnected...
> > 
> > 1. What's the name of the network device for this adapter?  Is it the
> > same name as it previously had?
> > 
> > 2. What does 'ifconfig' say the IP address is for this adapter?
> > 
> > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
> > 'bindport'?
> > 
> > 4. Do you have SIP connections on the same network interface, and are
> > those working as normal?
> > 
> > 
> > Antony.
> 
> 1- No changes to device names.  eth0 is the main link to the network,
> eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
> goes to another SIP provider.  All IAX trunks use eth0
> 
> 2- ifconfig gives the proper IP and netmask for all interfaces
> 
> 3- We do not specify bindaddr or bindport in the config file as the
> default is to bind to 0.0.0.0
> 
> 4- We had to make new SIP trunks to replace the IAX2 trunks to all
> servers.  The SIP trunk is working with no problems.  Except for two SIP
> links to PSTN all internal extensions use the same network interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful suggestion.


Antony.

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Re: [asterisk-users] IAX2 getting stuck

[Antony Stone]

On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote:

> On 4/19/17 4:09 PM, Antony Stone wrote:
> > On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:
> >>I have a server that had been operating for a few years now with
> >> 
> >> IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
> >> now say UNREACHABLE.
> > 
> > ...snip...
> > 
> >> So far the only thing different is that the receive queue for port 4569
> >> is not zero like all the other servers:
> >> 
> >> udp   128760  0 0.0.0.0:45690.0.0.0:*
> >> 
> >>Basically all packets for IAX2 are getting stuck in the queue.
> >>Any
> >> 
> >> suggestions?
> > 
> > Have you tried rebooting the router which connects this machine to the
> > Internet?
> > 
> > It sounds like a stale connection-tracking table entry to me.
> > 
> > 
> > Antony.
> 
>  We have already tried that.  One of the servers that has an IAX
> trunk to this server is on the same local network so that eliminates any
> firewall/router in the way.  We disabled iptables just in case too.

Hm :(

You say the USB ethernet adapter got unplugged and then reconnected...

1. What's the name of the network device for this adapter?  Is it the same 
name as it previously had?

2. What does 'ifconfig' say the IP address is for this adapter?

3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and 'bindport'?

4. Do you have SIP connections on the same network interface, and are those 
working as normal?


Antony.

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Re: [asterisk-users] IAX2 getting stuck

[Antony Stone]

On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:

>   I have a server that had been operating for a few years now with
> IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
> now say UNREACHABLE.

...snip...

> So far the only thing different is that the receive queue for port 4569 is
> not zero like all the other servers:
> 
> udp   128760  0 0.0.0.0:45690.0.0.0:*
> 
>   Basically all packets for IAX2 are getting stuck in the queue. Any
> suggestions?

Have you tried rebooting the router which connects this machine to the 
Internet?

It sounds like a stale connection-tracking table entry to me.


Antony.

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Re: [asterisk-users] asterisk name in mysql

[Antony Stone]

On Saturday 22 April 2017 at 22:25:52, Atux Atux wrote:

> Thanks a lot for the reply.
> I did follow that already, but i do have a problem. Here is my
> extensions.conf part for that particular number
> exten => 6912345678,1,Answer()
> exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd
> asterisk)

...snip...

> and here is the error i am getting
> [Apr 22 23:20:29] WARNING[9725][C-0002]: pbx.c:4991
> pbx_extension_helper: No application 'MYSQL' for extension (IncomingDial,
> 6951921078, 2)
>   == Spawn extension (DialIn, 6912345678, 2) exited non-zero on
> 'Dongle/dongle0-010002'
> 
> 
> Any ideas please?

What have you put into func_odbc.conf?

ie: what's the definition of MYSQL?


Antony.

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[asterisk-users] Dial() using full SIP account details

[Antony Stone]

Hi.

I'm having problems with the Dial() application when I use full SIP account 
details in it.

I'm looking at the O'Reilly book https://www.amazon.co.uk/dp/1449332420 on 
page 135, where it says "The Dial() application also allows you to connect to 
a remote VoIP endpoint not previously defined in one of the channel 
configuration files.  The full syntax is:

Dial(technology/user[:password]@remotehost[:port][/remote_extension])

As an example, you can dial into a demonstration server at Digium using the 
IAX2 protocol by using the following extension:

exten => 500,1,Dial(IAX2/gu...@misery.digium.com/s)"


I'm using Asterisk 11.13.1 under Debian 7.

I am trying to dial from Asterisk to another SIP server using an account on 
that server, for which I know the username and password.

Just to confirm, if I put the account credentials into a telephone and register 
to the remote server, I can place calls as expected.

When I try to do the same thing using Asterisk, however:

1. The password I have been assigned on the remote server contains a ! symbol, 
and it seems that Asterisk is ignoring this symbol and everything after it:

The account name (slightly obfuscated for security in this email) is 832+ios
The password (ditto) is 31oNPMLQ!9d_XuQu
I wish to dial through that account to the number 0203 (which works 
from a telephone).

In my dialplan I have (all on one line of course):

exten => 936,1,Dial(SIP/832+ios:31oNPMLQ!
9d_x...@remote.server.com/0203yyy)

Dialling extension 936 results in:

-
-- Executing [936@outbound:1] Dial("SIP/1000-00db", 
"SIP/832+ios:31oNPMLQ!9d_x...@remote.server.com/0203yyy") in new stack

  == Using SIP RTP CoS mark 5

[2017-02-28 11:38:16] ERROR[1005][C-0d21]: netsock2.c:269 
ast_sockaddr_resolve: getaddrinfo("832+ios", "31oNPMLQ", ...): Servname not 
supported for ai_socktype

[2017-02-28 11:38:16] WARNING[1005][C-0d21]: chan_sip.c:6057 create_addr: 
No such host: 832+ios:31oNPMLQ

[2017-02-28 11:38:16] WARNING[1005][C-0d21]: app_dial.c:2437 
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber 
absent)
-

I've tried:
 - escaping the ! by prefixing it with a \
 - enclosing the entire password within '
 - enclosing the entire username / password within '
but Asterisk still simply stops reading at the ! and ignores everything which 
follows.

So, how can I get it to use this password which happens to contain a ! ?


2. If I get my remote provider to change the password so that it does not 
contain the ! symbol, Asterisk's behaviour changes:

exten => 
936,1,Dial(SIP/832+ios:31onpmlq_9d_x...@remote.server.com/0203yyy)

This now results in:

-
-- Executing [936@outbound:1] Dial("SIP/1000-00dc", 
"SIP/832+ios:31onpmlq_9d_x...@remote.server.com/0203yyy") in new stack

[2017-02-28 11:43:47] NOTICE[1011][C-0d22]: chan_sip.c:29848 
sip_request_call: Conflicting extension values given. Using '832+ios' and not 
'0203yyy'

  == Using SIP RTP CoS mark 5

-- Called SIP/832+ios:31onpmlq_9d_x...@remote.server.com/0203yyy

[2017-02-28 11:43:47] NOTICE[11692][C-0d22]: chan_sip.c:23010 
handle_response_invite: Failed to authenticate on INVITE to '"Antony Stone" 
<sip:1000@4x.4x.1x.2x>;tag=as6ef135a8'
-

So, I appear to have given the parameters in the correct form:

Dial(technology/user[:password]@remotehost[:port][/remote_extension])

and I get told that the username does not match the remote_extension (ie: the 
number I want to dial) - well, of course it doesn't - the username is part of 
my authentication to the server, nothing to do with who I want to call?


Incidentally, I do know I can put a Register statement into sip.conf, and then 
be able to use the Dial() application just using the username (and this 
works), however I need a solution which can support two or more accounts at 
different remote providers having the same username.

Therefore the username alone will not be unique, but the combination of 
username + password + server name will be, hence the reason why I would need 
to use this in the dialplan.


If anyone can offer suggestions on how to use the full SIP credentials in a 
Dial() statement, and also how to escape special characters such as ! I would 
be very grateful.


Thanks,


Antony.

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Re: [asterisk-users] [asterisk13] Multiple transport objects of same protocol in pjsip.conf

[Antony Stone]

On Saturday 29 July 2017 at 19:03:55, Joshua Colp wrote:

> On Sat, Jul 29, 2017, at 02:55 PM, O. Hartmann wrote:
> > Scenario:
> > 
> > Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to
> > 192.168.254.1:5060) is behind
> > a NAT, acting as a client to our ITSPs SIP server. But also, this
> > Asterisk is server for
> > various VoIP telephones.
> > 
> > Acoording to Asterisk's wiki, the transport section of pjsip.conf is
> > configured as
> > follows:
> > 
> > ; Transport via UDP
> > [transport-nat-udp]
> > type=   transport
> > protocol=   udp
> > local_net=  192.168.254.0/24
> > local_net=  127.0.0.1/32
> > bind=   192.168.254.1:5060
> > external_media_address= ddns.gdr
> > external_signaling_address= ddns.gdr
> 
> 
> 
> You should only need this single transport as it will get used by
> everything. Only when contacting external things will the external
> values be used instead. This is determined based on the "local_net"
> values you've provided.

Also, setting a transport to expect NAT, when in fact there isn't any, won't 
cause problems.

Telling Asterisk to expect NAT simply means that it pays attention to where 
the packets come from, not what addresses they contain inside them.

If you have no NAT, these two addresses are the same, so no harm done.


Antony.

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[asterisk-users] AMI column widths

[Antony Stone]

Hi.

I'm trying to get a list of the channels currently in use on an Asterisk server 
(1.8.32.1 if it matters) over AMI.

I send the command "sip show channels", and I get back a response along the 
lines of (* used to protect the innocent...):

Peer User/ANR Call ID  Format   Hold 
Last MessageExpiry Peer  
*8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No   Tx: 
ACKSIPtrunkNu
*.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No   Rx: 
OPTIONS   
*.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No   Rx: 
ACK200101
*.1*.19.70 (None)   149831567021051  0x0 (nothing)No   Rx: 
REGISTER  

So, firstly, the "Call ID" column is clearly truncated, because it should show 
more than is indicated above,
but more importantly for me, the "Peer" column is truncated, and what should 
show as "SIPtrunkNumber8"
is only shown as "SIPtrunkNu".

How can I get the full column widths of these items shown in the output?

Note that it is not a solution just to say "don't call it 'SIPtrunkNumber8'; 
call it 'SIPtrunk8' instead", because
this name has also been modified slightly to conceal the real name of the 
trunk, which is actually longer
than "SIPtrunkNo8", but still with the most important information at the end.

What I'm looking for is how to get the *full* details of all the channels shown.

I have checked, and there is no "verbose" option to the "sip show channels" 
command.


Thanks,


Antony.

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Re: [asterisk-users] AMI column widths

[Antony Stone]

On Saturday 08 July 2017 at 10:16:19, Antony Stone wrote:

> On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:
> > There are no sip show channels on AMI. Also, the output that you sent is
> > not a AMI output. Are u using AMI ou running commands on console?
> 
> I'm using AMI.
> 
> I have a connection to the Asterisk server on port 5038, initated with:
> 
>   Action: Login
>   Username: x
>   Secret: y
>   Events: off
> 
> I receive back:
> 
>   Response: Success
>   Message: Authentication accepted
> 
> I then issue:
> 
>   Action: Command
>   Command: SIP show channels
> 
> and I get back:
> 
>   Response: Follows
>   Privilege: Command
>   Peer User/ANR Call ID  Format   Hold
> Last MessageExpiry Peer
> 
> plus the data I quoted previously.
> 
> > Running commands on console and parsing the output is the worst way to
> > obtain data, first because it is not easily parseable.
> 
> And also because it is very inefficient with connection setups, I believe.
> 
> > Second, it doesn't show you all data.
> > 
> > Third, you can have these truncate problems, because that's not intention
> > of CLI.
> > 
> > Using proper AMI Actions you will probably achieve your goals
> > 
> > https://wiki.asterisk.org/wiki/display/AST/AMI+Actions
> 
> Hm, I don't see anything there which will give me a list of the SIP
> channels currently in use - what command should I be using for that?

Hm, Action: CoreShowChannels looks like it can be made to work - it's not 
specifically SIP, but I can parse that out of the channel name.  Strange that 
there is a DAHDIShowChannels command, and a CoreShowChannels, but no 
SIPShowChannels..

If anyone has a better idea, please let me know...


> Thanks,
> 
> 
> Antony.
> 
> > On 7 Jul 2017 10:32 pm, Antony Stone wrote:
> > 
> > Hi.
> > 
> > I'm trying to get a list of the channels currently in use on an Asterisk
> > server (1.8.32.1 if it matters) over AMI.
> > 
> > I send the command "sip show channels", and I get back a response along
> > the lines of (* used to protect the innocent...):
> > 
> > Peer User/ANR Call ID  Format   Hold
> >  Last MessageExpiry Peer
> > *8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No
> >  Tx: ACKSIPtrunkNu
> > *.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No
> >  Rx: OPTIONS
> > *.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No
> >  Rx: ACK200101
> > *.1*.19.70 (None)   149831567021051  0x0 (nothing)No
> >  Rx: REGISTER   
> > 
> > So, firstly, the "Call ID" column is clearly truncated, because it should
> > show more than is indicated above,
> > but more importantly for me, the "Peer" column is truncated, and what
> > should show as "SIPtrunkNumber8"
> > is only shown as "SIPtrunkNu".
> > 
> > How can I get the full column widths of these items shown in the output?
> > 
> > Note that it is not a solution just to say "don't call it
> > 'SIPtrunkNumber8'; call it 'SIPtrunk8' instead", because
> > this name has also been modified slightly to conceal the real name of the
> > trunk, which is actually longer
> > than "SIPtrunkNo8", but still with the most important information at the
> > end.
> > 
> > What I'm looking for is how to get the *full* details of all the channels
> > shown.
> > 
> > I have checked, and there is no "verbose" option to the "sip show
> > channels" command.
> > 
> > 
> > Thanks,
> > 
> > 
> > Antony.

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Re: [asterisk-users] AMI column widths

[Antony Stone]

On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:

> There are no sip show channels on AMI. Also, the output that you sent is
> not a AMI output. Are u using AMI ou running commands on console?

I'm using AMI.

I have a connection to the Asterisk server on port 5038, initated with:

Action: Login
Username: x
Secret: y
Events: off

I receive back:

Response: Success
Message: Authentication accepted

I then issue:

Action: Command
Command: SIP show channels

and I get back:

Response: Follows
Privilege: Command
Peer User/ANR Call ID  Format   
Hold 
Last MessageExpiry Peer

plus the data I quoted previously.

> Running commands on console and parsing the output is the worst way to
> obtain data, first because it is not easily parseable.

And also because it is very inefficient with connection setups, I believe.

> Second, it doesn't show you all data.
> 
> Third, you can have these truncate problems, because that's not intention
> of CLI.
> 
> Using proper AMI Actions you will probably achieve your goals
> 
> https://wiki.asterisk.org/wiki/display/AST/AMI+Actions

Hm, I don't see anything there which will give me a list of the SIP channels 
currently in use - what command should I be using for that?


Thanks,


Antony.

> On 7 Jul 2017 10:32 pm, Antony Stone wrote:
> 
> Hi.
> 
> I'm trying to get a list of the channels currently in use on an Asterisk
> server (1.8.32.1 if it matters) over AMI.
> 
> I send the command "sip show channels", and I get back a response along the
> lines of (* used to protect the innocent...):
> 
> Peer User/ANR Call ID  Format   Hold
>  Last MessageExpiry Peer
> *8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No
>  Tx: ACKSIPtrunkNu
> *.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No
>  Rx: OPTIONS
> *.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No
>  Rx: ACK200101
> *.1*.19.70 (None)   149831567021051  0x0 (nothing)No
>  Rx: REGISTER   
> 
> So, firstly, the "Call ID" column is clearly truncated, because it should
> show more than is indicated above,
> but more importantly for me, the "Peer" column is truncated, and what
> should show as "SIPtrunkNumber8"
> is only shown as "SIPtrunkNu".
> 
> How can I get the full column widths of these items shown in the output?
> 
> Note that it is not a solution just to say "don't call it
> 'SIPtrunkNumber8'; call it 'SIPtrunk8' instead", because
> this name has also been modified slightly to conceal the real name of the
> trunk, which is actually longer
> than "SIPtrunkNo8", but still with the most important information at the
> end.
> 
> What I'm looking for is how to get the *full* details of all the channels
> shown.
> 
> I have checked, and there is no "verbose" option to the "sip show channels"
> command.
> 
> 
> Thanks,
> 
> 
> Antony.

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Re: [asterisk-users] choppy calls version Asterisk 13.17 on CentOS 7

[Antony Stone]

On Friday 14 July 2017 at 23:34:37, Motty Cruz wrote:

> Since the upgrade our remote users' conversions are choppy.

> Monitoring using CLI, I noticed the device always select ulaw
> for codec.

What's the device?

What are its codec settings?

What's your available & used bandwidth on the server's connection?

Are all users affected, or only some?

How many concurrent calls do you have going through the Asterisk server when 
they notice the problems?

What's the load average on the server while it's handling these calls?

Try turning on call recording on the server and see whether the recording is 
choppy as well as what the users hear.


Antony.

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Re: [asterisk-users] Asterisk 13.16.0 segfault

[Antony Stone]

On Thursday 20 July 2017 at 20:46:30, Marcelo Terres wrote:

> I don't have much knowledge about freepbx, but if some day I had to use it,
> I would  prefer to use the Asterisk compiled from source, unless it comes
> with an Asterisk package (rpm, supposing it is running CentOS).

FreePBX (as a distro) is based on CentOS and comes with its own compiled 
version of Asterisk - you install FreePBX and you get everything - CentOS, 
Asterisk, Apache, PBX scripts, web frontend - the lot.

FreePBX as a package can be installed on CentOS (or Debian for that matter) 
but the FreePBX project's documentation recommends compiling Asterisk from 
source in this instance.

> On 20 Jul 2017 5:08 pm, "Carlos Chavez"  wrote:
> > On 7/20/17 8:47 AM, Marcelo Terres wrote:
> > 
> > Which version of Asterisk are you using? Are you compiling it with the
> > bundle pjproject ?
> > 
> > --with-pjproject-bundled
> > 
> > Regards,
> > 
> > Marcelo H. Terres
> > 
> > On 19 July 2017 at 17:03, Carlos Chavez  wrote:
> >> On 7/19/17 2:37 AM, Marcelo Terres wrote:
> >> 
> >> This is the pjsip library.
> >> 
> >> Is it possible to you to update pjsip for the latest version to test if
> >> it solves the problem?
> >> 
> >> On 18 Jul 2017 3:52 pm, "Carlos Chavez"  wrote:
> >>> I am getting frequent segfaults on a new Asterisk installation.  So far
> >>> the only message I see is:
> >>> 
> >>> Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip
> >>> 7fb2d535723f sp 7fb25a11b5c0 error 4 in
> >>> libasteriskpj.so.2[7fb2d52e5000+18]
> >>> Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip
> >>> 7f4afea0c23f sp 7f4a7f7e35c0 error 4 in
> >>> libasteriskpj.so.2[7f4afe99a000+18]
> >>> Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188 ip
> >>> 7f2eb611923f sp 7f2e3aec25c0 error 4 in
> >>> libasteriskpj.so.2[7f2eb60a7000+18]
> >>> Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188 ip
> >>> 7fc5758dd23f sp 7fc4fa6245c0 error 4 in
> >>> libasteriskpj.so.2[7fc57586b000+18]
> >>> Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188 ip
> >>> 7f5f29abb23f sp 7f5eae8285c0 error 4 in
> >>> libasteriskpj.so.2[7f5f29a49000+18]
> >>> 
> >>> Since this is a Freepbx distro does could the problem be related to
> >>> their flavor of Asterisk?  I have several other plain Asterisk servers
> >>> running on this version without any problems.  Any recommendations on
> >>> how to debug this?
> >>> 
> >>> My solution to this is going to be compiling Asterisk manually
> >> 
> >> instead of using their pre packaged version as debugging will take a lot
> >> more time.
> >> 
> > The Freepbx distro still uses a separate pjproject as far as I know.
> > 
> > When I compile my own I always use the bundled version now.

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Re: [asterisk-users] Simplest way of executing a non-blocking (async) python AGI script?

[Antony Stone]

On Friday 30 June 2017 at 19:11:08, Jonathan H wrote:

> I use a python AGI which pulls some info from a web service, which should
> take half a second.
> 
> Sometimes, it takes 5-10 seconds which blocks the dialplan execution, but
> the dialplan should continue immediately as it's not dependent on the
> AGI/web service data.
> 
> What's the simplest, easiest quickest least-code way of firing off an AGI
> with some variable, and then returning to the dialplan?

Write your python code to fork() the lookup to a child process, and let the 
parent return immediately to Asterisk.

> I've seen people talking about fastAGI, stasis, python ASYNC... all seems
> rather complicated. I feel I must be missing something embarrassingly
> obvious - isn't there just the equivalent of the unix shell's "&"?!

Not inside Asterisk, no.


Antony.

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[asterisk-users] PJSIP list of peers online/offline?

[Antony Stone]

Hi.

I have some Nagios / Icinga monitoring plugins I've created for Asterisk, and 
one of them checks the percentage of SIP accounts which are currently 
registered on an Asterisk server.

It does this by running "sip show peers" via AMI and analysing the summary 
line at the end:

1066 sip peers [Monitored: 747 online, 310 offline Unmonitored: 3 online, 6 
offline]

I then calculate 747 divided by (747+310) and report the % online (because I 
know I'm not interested in the unmonitored ones).


However, a customer has upgraded one of their servers from Asterisk 11 to 
Asterisk 13, and "sip show peers" no longer works.


I can see a whole list of commands starting with "pjsip" but there's no "pjsip 
show peers", so what's the new command which will tell me how many online and 
how many offline SIP peers there are?


Thanks in advance,


Antony.

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Re: [asterisk-users] Writing CDR's to two database servers

[Antony Stone]

On Monday 19 June 2017 at 18:12:35, Sebastian Gutierrez wrote:

> use replication

1. Agreed - use replication.

2. If you want an HA (High Availability, not dependent on a single Master DB 
server replicating to a slave) solution, consider setting up Master-Master 
replication, with an LVS (Linux Virtual Server) HA machine in front of the 
two, so that writes can go to either server using only a single IP address 
configured in Asterisk.

Then, if one fails, you can still write to (and read from) the other, repair 
the failed one, and restore replication.


Antony

> > On Jun 19, 2017, at 17:47, Tech Support  wrote:
> > 
> > All;
> > 
> > I know that there are probably several solutions to this problem, but
> > what I am trying to do is provide some redundancy for my customers
> > CDR data. I know that doing simple backups of MySQL is probably the
> > easiest way to go, but I’m thinking that there may be some benefit
> > to simultaneously writing the CDR data to multiple servers at once.
> > However, I’m drawing a blank on this one. Has anyone else done this
> > before? Any insight at all would be greatly appreciated.
> > 
> > Thanks Much;
> > John V.

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Re: [asterisk-users] Autodialer - call simultaneously to both ends

[Antony Stone]

On Monday 26 June 2017 at 14:06:10, Harel wrote:

> Hello List,
> I'm working on an autodialer project.
> At the moment I use the Originate application then I "throw" it to an
> extension where I Dial() the other party and then both legs are bridged.
>
> The problem is that the Dial() will only run after the Originate finish
> its bit and I have lots of wasted time or even worse, the remote party
> hanging the call because instead of a human speaking to him the call setup
> to the 2nd leg only starts when remote answers.

Sounds like you're dialling the legs the wrong way round.

> Is there a way to start calling both parties at the same time and bridge
> them when one answers (which will then hear the ringback tone until 2nd
> party answers)?

You should dial the extension of the person who wants the autodial function 
first (ie: the person who knows about this system).

They answer their phone (which should be quick, because they're expecting it 
to ring after they've initiated the autodial), and they then wait for the 
remote party (who doesn't know there's an autodialler involved) to answer.

Dialling both numbers simultaneously always runs the risk that the remote 
party (who doesn't know about the autodialler) will answer the call first, so 
unless you have some recorded announcement "please wait while we connect your 
call" (which if I heard it would make me hang up immediately, because I'd know 
it was some automated dialler, probably a cold-calling sales organisation), 
they answer the phone, hear ringing, think "what the hell?" or even "oh, one 
of them again" and hang up.

Always start from the "local" end - ie: the person who knows about the auto-
dialler.



Antony.

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Re: [asterisk-users] Autodialer - call simultaneously to both ends

[Antony Stone]

On Monday 26 June 2017 at 18:01:22, J Montoya or A J Stiles wrote:

> On Monday 26 Jun 2017, Harel wrote:
> > Hello List,
> > I'm working on an autodialer project.
> > At the moment I use the Originate application then I "throw" it to an
> > extension where I Dial() the other party and then both legs are bridged.
> > The problem is that the Dial() will only run after the Originate finish
> > its bit and I have lots of wasted time or even worse, the remote party
> > hanging the call because instead of a human speaking to him the call
> > setup to the 2nd leg only starts when remote answers. Is there a way to
> > start calling both parties at the same time and bridge them when one
> > answers (which will then hear the ringback tone until 2nd party
> > answers)? Thank you
> 
> Our auto-dialler works as follows;
> 
> * Agent clicks number on screen in their web browser
> * Agent's phone rings
> * Agent picks up phone
> * Far end party's phone rings
> * Far end party answers
> * Agent and far end party are bridged.
> 
> and is implemented using the truly ancient technology of callfiles.

These work well and are implementable using any language capable of producing 
a text file.

It's also extremely simple (so long as you can write a network client 
application) to achieve the same thing using an AMI Originate request.

> All you need then is a Perl or PHP script, which accepts the destination
> number as a query parameter.  Your script then needs to identify the
> workstation by means of its IP address and determine the number of the
> nearest phone  (this does require proper configuration of DHCP server, but
> is worth it),  then write out a callfile.
> 
> 
> Note:  There exists a race condition in Asterisk  (at least, when using the
> common Linux file systems, which update a folder's directory as soon as the
> *first* block of a file is written)  which means that if a callfile exceeds
> one block, Asterisk could end up reading only the first block and ignoring
> the rest.  If there is any danger that a callfile could exceed one block
> on your filesystem, you must write the callfile to a different folder, and
> then use the `mv` command to move it to /var/spool/asterisk/outgoing/ . 
> This sidesteps the race condition due to the behaviour of the mv command. 
> When moving *within* a filesystem, the whole file was already on the disk
> anyway when the directory is updated; when moving from one filesystem to
> another, it does not update the directory of the destination folder until
> the *last* block is written.

Yes, that is a very important point.  Always use mv with callfiles :)


However, to get back to the original poster's question, I believe it's the 
logic of which way round the calls are being made that's the problem (I agree 
toally with your 6-step summary above), rather than the mechanism for being 
able to make calls.


Antony.

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Re: [asterisk-users] Automatically dial a number, then an extension

[Antony Stone]

On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote:

> All;
> 
> What I did was add a line in the dialplan that used the SendDTMF()
> application and that worked great. The problem that I’ve run into now is
> that dialing the extension screwed up the answering machine detection. The
> sample context looks something like this.
> 
> [play-audiomsg]
> exten => s,1,AMD
> exten => s,n,ExecIf($["${EXT}" != ""]?SendDTMF(${EXTEN}))
> exten => s,n,Background(${AUDIOMSG})
> exten => s,n,Hangup
> 
> As you can see, it's very simple. Modifying the amd.conf configuration
> wasn’t the answer since I don’t know how long it will take for the
> extension to pick up.

Isn't it safe to assume that if you've been given an extension number to dial 
after the initial call is answered, then it wasn't answered by an answering 
machine?

The extension might be answered by an answering machine, I suppose, but that's 
not the problem you're talking about (I think).

I would create two contexts:

1. Does AMD and gets called when there is no follow-on extension to dial

2. Dials a follow-on extension and doesn't do AMD (or at least, not at the 
start)

Then you choose which context to place the call through depending on whether a 
follow-on extension has been supplied for that customer's number or not.

> Simply placing the AMD command after the SendDTMF() wasn’t the answer

Why wasn't it the answer?  What happens or doesn't happen when you try this?


Antony

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Re: [asterisk-users] Automatically dial a number, then an extension

[Antony Stone]

On Tuesday 23 May 2017 at 20:01:14, Tech Support wrote:

> Ok, the purpose of the answering machine detection (AMD) is to
> determine when the audio file should start playing *after* the call has
> been picked up. Typically, if a call has been picked up by a person, they
> say a short greeting, for example "Hello, this is John, how can I help
> you?" or simply "Hello?" or something similar. If a call has been picked
> up by an answering machine, usually the message is somewhat longer, maybe
> 10 seconds or so, maybe longer. Ideally, the AMD tries to make sure that
> the audio file starts right after the greeting is over. It's not exact,
> but my experience is that it works fairly well. The problem that I am
> having is that when I also have to dial an extension, the call has already
> picked up and the AMD will start working immediately after the SendDTMF()
> even if dialing the extension means that it may ring anywhere from 5 - 20
> seconds plus the greeting on the far end. There doesn’t appear to be a way
> for the AMD to wait until extension gets picked up, either by a human or a
> machine. So what happens is that the AMD gets confused and the audio file
> starts playing while the extension is still ringing. I hope this helps.

Okay, so my suggestion still stands:

Create two contexts:

 - one which does AMD and gets called when there is no follow-on extension to 
dial

 - another which dials a follow-on extension and doesn't do AMD (or at least, 
not at the start)

Then you choose which context to place the call through depending on whether a 
follow-on extension has been supplied for that customer's number or not - if 
there's no follow-on extenstion, use the first context; if there is, use the 
second one.


Antony.

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Re: [asterisk-users] Upgraded server crashes on voicemail storage

[Antony Stone]

On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote:

> Hi all,
> 
> I'm upgrading to Asterisk 13.14.0 x86_64.  During my beta testing, I've
> discovered that my server crashes as soon as I leave a voicemail message. 
> I'm using odbc voicemail storage as well as mysql dynamic configuration.
> 
> I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1
> 
> I suspect that the odbc drivers are the problem.  Is ther an alternative
> drive that I should be using?
> 
> Failing that, any other ideas?

Give us more details of what you mean by "crashes".

What happens, what do you get in the Asterisk logs, what do you get in syslog, 
what state is the machine in afterwards, is there a kernel panic, what 
information leads you to suspect the ODBC drivers...?

Also, what have you upgraded from, what machine specs are you running on, 
what's the dialplan section dealing with leaving voicemail...?


The more info you give us, the more likely it is we can suggest something 
useful.


Antony.

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Re: [asterisk-users] asterisk server - no sound

[Antony Stone]

On Tuesday 06 June 2017 15:18:32 andre castro wrote:

> I just installed asterisk in a debian server.
> All seems to be running fine, but the audio sent by the server.

> But I hear nothing at the peer's end.
> 
> When one peer calls another, sound comes through just fine.

Tell us about your networking arrangement - are both phones and the Asterisk 
machine on the same network?

Is there a router in between any of them?

Is there any NAT involved?

> Do I need to have alsa installed??

No.


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Re: [asterisk-users] asterisk server - no sound

[Antony Stone]

On Tuesday 06 June 2017 16:57:07 andre castro wrote:

> On 06/06/2017 04:36 PM, Antony Stone wrote:
> > 
> > Tell us about your networking arrangement - are both phones and the
> > Asterisk machine on the same network?
> 
> Nop. They are in 2 different networks. The phones in one and the
> Asterisk machine in another.

Okay, that is why you have audio between the two phones, then - they can see 
each other directly, on the same network, and nothing is interfering with the 
traffic between them.

> > Is there a router in between any of them?
> 
> Yes. In the phones network.
> 
> > Is there any NAT involved?
> 
> Yes in the phones' network. They both have different private IP address
> and one public IP.

Okay, I suspect that this NATting router is not passing the UDP packets from 
the server back to the phones correctly, based on the SIP connection (when the 
phone makes the call).

SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.

If it's a Linux router, you need to make sure you are allowing FORWARDed 
traffic 
which matches ESTABLISHED, RELATED.

If it's not a Linux router, you need to find out how to get it to support SIP 
and RTSP.


Good luck,


Antony.

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Re: [asterisk-users] CM for menuselect choices

[Antony Stone]

On Friday 05 May 2017 at 16:21:20, Richard Kenner wrote:

> I'd like to be able to save the choices made in menuselect in a way
> that they can be tracked in a CM system and applied to a later release
> of Asterisk using an automated tool like Ansible.  What's the best
> way to do that?

menuselect should create a file containing your choices called 
menuselect.makeopts - that forms the input to the 'make' process which builds 
the binaries from the source tree.

All you should need to do is copy menuselect.makeopts onto your target system 
and then run 'make && make install' etc in the usual way.

Of course, you might run into problems if the later release introduces new 
options (or deprecates old ones) which then aren't going to be in your 
makeopts file, but at least it's a good place to start.


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Re: [asterisk-users] CM for menuselect choices

[Antony Stone]

On Friday 05 May 2017 at 16:52:39, Richard Kenner wrote:

> > Of course, you might run into problems if the later release introduces
> > new options (or deprecates old ones) which then aren't going to be in
> > your makeopts file
> 
> That's my question: how do I reflect the changes that I made to the
> defaults in a way that's not dependent on the exact set of options
> that each release has?

I cannot think of a possible answer to that, because you are trying to guard 
against features in a future release which may not even have been considered 
by the developers yet.

Maybe your best bet would be to take the default options file for the "current 
release" (whatever you regard that as), create a 'diff' between that and the 
file with your selections in, and then use that to 'patch' future options 
files, 
on the basis that any new options will then keep their (future) default 
values, and any still-existing options will be changed to your choices.  The 
only problem I can immediately see is if an option stays, but its default gets 
changed, the patch file will no longer match - but at least you'll get an error 
message when you try to do the patching, and can investigate the problem.


Regards,


Antony.

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Re: [asterisk-users] SwitchVox and Asterisk

[Antony Stone]

On Monday 08 May 2017 at 16:43:24, Luca Pradovera wrote:

> Hello,
> I need to have an extension on a SwitchVox server dial out to one on an
> Asterisk (FreePBX actually) box which will host a voice directory.

What's a voice directory?

> The Asterisk server will then need to dial one of the SwitchVox extensions
> if it gets a voice match.

You mean, listen to the caller speaking and identify who they are?

Sounds "non-trivial" to me...

> Anyone has done that, and could let me know how? So far it looks like IAX
> peering (what SW calls "SwitchVox peering") could work?

IAX will connect two Asterisk servers and allow them to communicate (it stands 
for Inter Asterisk eXchange) - think of it in the same way as a SIP trunk - 
you can have multiple calls to/from multiple numbers going over the link.

However, are you saying that you've already got the "voice directory" and 
"voice match" parts working in Asterisk, and you just need to know how to dial 
between that and SwitchVox?

Or is the "voice" part of the challenge also something you're looking for help 
with?


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Re: [asterisk-users] Dial an extension to modify dialplan

[Antony Stone]

On Monday 08 May 2017 at 15:44:47, Marcelo Terres wrote:

> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_Dialpl
> anExtensionAdd
> 
> Is it enough?

Is there a similar call to delete an extension, or to modify an existing one?

On the basis that the OP already has extension 2000 defined, he would need to 
delete this and replace it with a new definition, or alter the current 
definition, to get the required results.

Simply being able to add a new extension to an existing dialplan isn't quite 
enough.


Antony.

> On 8 May 2017 at 15:35, Frank Vanoni  wrote:
> > Hello
> > 
> > I have the following scenario:
> > 
> > [mynicecontext]
> > exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC)
> > 
> > As expected, by dialing 2000, all three devices will ring. And that's
> > fine.
> > However, there are situations where I only want "deviceA" and "deviceB"
> > to ring. I would like to have an extension to dial in order to modify
> > the dialplan.
> > 
> > Here is what I did...
> > 
> > In extensions.conf:
> > 
> > -- snip -
> > [mynicecontext]
> > #include "ringdevice.conf
> > 
> > exten => 2000,1,GoTo(ringdevice,ring,1)
> > 
> > exten => 4000,1,System(/bin/cat /etc/asterisk/twodevices.txt
> > 
> >> /etc/asterisk/ringdevice.conf)
> > 
> > exten => 4000,2,Wait(3)
> > exten => 4000,3,System(/usr/sbin/asterisk -rx "dialplan reload")
> > exten => 4000,4,Playback(service)
> > 
> > exten => 4001,1,System(/bin/cat /etc/asterisk/alldevices.txt
> > 
> >> /etc/asterisk/ringdevice.conf)
> > 
> > exten => 4001,2,Wait(3)
> > exten => 4001,3,System(/usr/sbin/asterisk -rx "dialplan reload")
> > exten => 4001,4,Playback(service)
> > -- end snip -
> > 
> > twodevices.txt contains
> > exten => ring,1,Dial(SIP/deviceA)
> > 
> > alldevices.txt contains
> > exten => ring,1,Dial(SIP/deviceA/deviceC)
> > 
> > By dialing 4000 or 4001, the dialplan is modified and reloaded
> > accordingly.
> > 
> > Is there a better solution?
> > 
> > Frank

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Re: [asterisk-users] Need to restart Asterisk if remote server not working?

[Antony Stone]

On Saturday 06 May 2017 at 09:21:16, Luca Bertoncello wrote:

> Antony Stone schrieb:
> 
> > 4. Did the IP address of Telekom's end of the connection change?
> 
> I really don't know, but I suppose not

I suspect this may in fact have been the cause of your problem.

Firstly, I notice that tel.t-online.de has a non-trivial DNS entry:

$ host tel.t-online.de
tel.t-online.de is an alias for ims.voip.t-ipnet.de.
ims.voip.t-ipnet.de is an alias for ims001.voip.t-ipnet.de.
ims001.voip.t-ipnet.de is an alias for b-epp-001.isp.t-ipnet.de.
b-epp-001.isp.t-ipnet.de has address 217.0.18.36

Secondly I find forum entries from people observing that either the IP address 
changes from time to time, or even that Telekom's DNS servers do not give the 
same result as root name servers:

https://telekomhilft.telekom.de/t5/Telefonie-Internet/IP-Adressbereich-tel-t-
online-de/td-p/2325114

https://telekomhilft.telekom.de/t5/Festnetz-Internet/VoIP-Telefonie-DNS-
Aufloesung-von-tel-t-online-de/td-p/1563089

Certainly, with allth eother information you gave, if the IP addresses at both 
ends stayed the same, I wouldn't expect an Asterisk restart to be necessary.


Regards,


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Re: [asterisk-users] Need to restart Asterisk if remote server not working?

[Antony Stone]

On Saturday 06 May 2017 at 08:37:39, Luca Bertoncello wrote:

> Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't
> connect to the remote Server (by Telekom) until today about 7:30.
> 
> Well, it could happen...
> What I find really annoying was that I needed to restart Asterisk as I
> checked with sipsak that the Telekom-Server works...

What was Asterisk doing until you restarted it?  What happened when it tried 
to use the (stale, but now restored) connection?

> I think, this should not be normal... Can someone explain me why it happens
> and what I have to change in the configuration to avoid this problem?

1. How is your Asterisk server connecting to Deutsche Telekom (SIP, IAX2, 
other...)?

2. How do you authenticate on that connection (password, certificate, IP 
address...)?

3. Do you connect to an IP address at Telekom, or to a hostname?

4. Did the IP address of Telekom's end of the connection change?

5. Did the IP address of your end of the connection change?



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Re: [asterisk-users] A bit OT - Configure GoIP for Asterisk

[Antony Stone]

On Monday 02 October 2017 at 20:58:33, Steve Edwards wrote:

> I recently received a GoIP-32 for a client project -- primarily outbound
> calling.
> 
> How should a GoIP be configured for Asterisk?

Have you tried http://www.hybertone.com/en/solutionsClass.asp?Id=78


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[asterisk-users] Discovering ring time immediately after call is answered

[Antony Stone]

Hi.

Does anyone know of a way to find out the ring time of a call as soon as it has 
been answered (ie: without waiting for the call to be completed, when it's 
part of the standard CDR record)?

I'm looking for a way to place a call, wait for it to be answered, and then 
perform different actions (eg: bridge the call to another number) depending on 
how long it took for the call to be answered (eg: less than X seconds or more 
than Y seconds).

Anyone got any ideas (for any reasonable version of Asterisk)?


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Re: [asterisk-users] Asterisk bugs make a right mess of RTP

[Antony Stone]

On Friday 01 September 2017 at 16:48:17, Dovid Bender wrote:

> On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp  wrote:
> > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote:
> > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/

> As Josh mentioned this is an issue with RTP and the SDP and when customers
> use NAT you need a way to figure out what their external RTP IP is. One
> option is to use IPv6 so the IP in the SDP is the one and only IP the media
> should be coming from. Another option is to increase the range of RTP ports
> in use. By default asterisk uses ports 10,000 to 20,000. You can change
> that to say use 20,000 to 30, or better yet use 10,000 to 20,
> widening the range of ports being used.

I'm not quite sure what numbers you're trying to quote here.  I agree that 
Asterisk uses 10,000 to 20,000 by default.

What are you suggesting this can be changed to in order to increase the range?


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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Antony Stone]

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote:

> I was hoping Asterisk would handle more than 4k simultaneous calls.

I know from experience that Asterisk can handle more than 4k simultaneous 
calls, however it's an extreme case to have all of them playing music on hold.

I think that if you tested 4k simultaneous calls with standard media streams 
on the majority of them, you would not experience the problem.

Is this a real problem for you - that Asterisk can't manage 4k MoH sessions 
simultaneously, even though it can manage 4k standard phone calls?


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[asterisk-users] Adding custom commands to AMI

[Antony Stone]

Hi.

https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There are 
a finite (but extendable) set of actions available to the client, determined by 
the modules presently loaded in the Asterisk engine."

Can anyone point me at some appropriate documentation for adding custom 
commands to the AMI to extend the available actions?


Thanks,


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Re: [asterisk-users] Adding custom commands to AMI

[Antony Stone]

On Sunday 12 November 2017 at 18:27:56, Tzafrir Cohen wrote:

> On Sun, Nov 12, 2017 at 04:45:45PM +0000, Antony Stone wrote:
> > Hi.
> > 
> > https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There
> > are a finite (but extendable) set of actions available to the client,
> > determined by the modules presently loaded in the Asterisk engine."
> > 
> > Can anyone point me at some appropriate documentation for adding custom
> > commands to the AMI to extend the available actions?
> 
> Generally: write your own asterisk module (in C), build and install it.

Okay, I guess that was implied from the "determined by the modules presently 
loaded" in the wiki article.

Any good online docs on writing Asterisk modules?

I'm comfortable enough writing C, but where do I start for library calls etc?


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Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

[Antony Stone]

On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote:

> Hello!
> 
> I'm facing the following scenario:
> 
> - Initial call opened to asterisk: SDP g722,alaw,ulaw
> 
> - Outgoing call to provider started with Invite / SDP alaw, g726 and
>   g729.

So, you're claiming to the provider that you can support all those codecs.

> - Provider sends 183 Session progress SDP: g729, alaw
> 
> - Provider sends g729 rtp packages
> 
> 
> But: there is no license to transcode g729.

So, you shouldn't be offering it.

> 
> What is asterisk doing?
> Asterisk decides to stop the call at all:
> - Sends cancel to provider and 603 decline to internal caller.
> 
> What would have been correct?
> It would have been correctly to switch to alaw as provider offered it, too.

Once the codec's been agreed on, between what the two sides offer to each 
other, you can't change it later.  Only offer what you're prepared to accept.

> Workaround:
> My workaround is to disable g729 and things are working fine again for
> me (for this special case).

That's not a workaround - that's correct configuation.

If you don't have a G.729 licence, don't offer G.729 to the peer.


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Re: [asterisk-users] Looking for the carrier that owns a particular DID

[Antony Stone]

On Thursday 02 November 2017 at 16:33:04, Tech Support wrote:

> I have a customer who is looking for a particular DID. (I dialed it and
> it is not in service). I searched through my preferred upstream provider's
> list but I came up empty. I wrote them, and this is their reply.
> 
> "We currently do not have that specific number in stock as this number is
> owned by another carrier that we do not have a business relationship with."

This suggests that they either:

 - identified who owned it, and hence established that they had no business 
relationship with the owner, or

 - identified that it wasn't owned by any of their business partners, in which 
case they may genuinely not know who does own it.

Either way, it's probably worth asking them, in case they have no objection to 
telling you, but simply didn't provide the information as it wasn't an answer 
to your question.

> So my question is this. How do I find out which carrier owns the DID in
> question?

Failing the above, I would start with the relevant country's telecoms 
regulator.


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Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

[Antony Stone]

On Wednesday 01 November 2017 at 12:15:08, Michael Maier wrote:

> On 11/01/2017 at 10:14 AM Antony Stone wrote:
> > On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote:
> >> 
> >> I'm facing the following scenario:
> >> 
> >> - Initial call opened to asterisk: SDP g722,alaw,ulaw
> >> 
> >> - Outgoing call to provider started with Invite / SDP alaw, g726 and
> >>   g729.
> > 
> > So, you're claiming to the provider that you can support all those
> > codecs.
> > 
> >> - Provider sends 183 Session progress SDP: g729, alaw
> >> 
> >> - Provider sends g729 rtp packages
> >> 
> >> But: there is no license to transcode g729.
> > 
> > So, you shouldn't be offering it.
> 
> Why? Asterisk lists this codec as supported - it just cannot transcode
> it (but it could be passed through). And it wouldn't be necessary to
> transcode at all, because provider offered alaw, too.

I don't think it's possible to tell Asterisk either:

 - only to offer a codec for pass-through without also offering it for 
transcoding

 - to select a codec based on pass-through in preference to another which 
needs transcoding

> BTW: here is a g729 library to transcode:
> https://gist.github.com/worldadventurer/c80e4d051937db887b40f3ab0084ce06
> 
> >> What is asterisk doing?
> >> Asterisk decides to stop the call at all:
> >> - Sends cancel to provider and 603 decline to internal caller.
> >> 
> >> What would have been correct?
> >> It would have been correctly to switch to alaw as provider offered it,
> >> too.
> > 
> > Once the codec's been agreed on,
> 
> Asterisk didn't agree! There has been no 200 ok sdp. Therefore Asterisk
> would have the chance to pick the other codec. But it didn't try it at
> all. It just canceled the call.

It cancelled the call because it couldn't bridge the two legs.  It offered 
G.729 and it was accepted by the peer, so that's what this leg was going to 
use.

> > between what the two sides offer to each other, you can't change it later. 
> > Only offer what you're prepared to accept.
> > 
> >> Workaround:
> >> My workaround is to disable g729 and things are working fine again for
> >> me (for this special case).
> > 
> > That's not a workaround - that's correct configuation.
> > 
> > If you don't have a G.729 licence, don't offer G.729 to the peer.
> 
> Passthrough would work if there would be a phone on the other side
> supporting g729.

Agreed.

> Therefore it's ok to offer it.

Only if you can offer it for pass-through but not for transcoding.  I don't 
think Asterisk supports this.

I can see why you think alaw would have been a good choice for this call, but 
I can't think of way to explain that to Asterisk without simply removing G.729 
from what it offers to handle.


Antony.

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[asterisk-users] Measuring total end-to-end latency

[Antony Stone]

Hi.

Does anyone have some recommendations for measuring total end-to-end latency 
(by which I mean: the time between person A saying something and person B 
hearing it) when there are both SIP and PSTN/analogue/mobile legs in the call 
path?

Examples:

Person A has a SIP phone registered to Asterisk, which has a SIP trunk to a 
connectivity provider, who has connections to PSTN (analogue landline) 
connectivity providers and to mobile network (Vodafone, Orange, etc) 
providers.

Person B might answer the call on an analogue landline telephone.

Person C might answer the call on a mobile phone (perhaps on its home network, 
perhaps roaming on a foreign network).


Is there any way to measure total latency of calls between A and B or A and C?


Thanks in advance for any ideas / suggestions.


Antony.

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Re: [asterisk-users] Using ControlPlayback with AWS S3

[Antony Stone]

On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote:

> Hi,
> 
> I have tested ControlPlayback and grabbed files via an apache server with
> no issue.

ControlPlayback is an Asterisk dialplan function.

How have you integrated this with Apache?

> I want to be able to grab files via aws S3 which would require me to add some
> headers to authenticate.

Presumably you mean you need to add some headers to an HTTP reuqest?

> Is there any way to have Asterisk add headers or would I need a http proxy
> in the middle?

Where and how is Asterisk making an HTTP request at all?

I don't really understand the connection between Apache/HTTP and 
ControlPlayback.  They're two quite separate things to me.



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Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

[Antony Stone]

On Tuesday 05 June 2018 at 08:33:26, David P wrote:

> We're using Asterisk 14.7.6 and I have a dialplan that ends like this:
> 
>  same => n,Dial(SIP/${EXTEN:0:4}@peer1)
>  same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH})
>  same => n,Hangup()
> 
> When peer1 hangsup, the priorities after the Dial are executed fine. But
> when the caller hangsup during the Dial, the cleanup steps aren't done.
> Why?
> 
> I did read "Note that on a successful connection, in the absence of the g
> and G modifiers (below), the Dial command does not return to allow
> execution of further commands for that extension in that context." at
> https://www.voip-info.org/asterisk-cmd-dial/ But it seems not to apply
> because I'm seeing the 'g' behavior without specifying that option, and the
> 'G' option seems intended for a far more complicated scenario.

If you're getting "g" functionality without specifying it, congratulations.

If you want something similar when the callER hangs up, you want to use the F 
option.

Regards,


Antony.

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Re: [asterisk-users] Using ControlPlayback with AWS S3

[Antony Stone]

On Wednesday 06 June 2018 at 16:30:08, Dovid Bender wrote:

> On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone wrote:
> > On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote:
> > > Hi,
> > > 
> > > I have tested ControlPlayback and grabbed files via an apache server
> > > with no issue.
> > 
> > ControlPlayback is an Asterisk dialplan function.
> > 
> > How have you integrated this with Apache?
> 
> By apache I mean
> ControlPlayBack(http://voice1.mydomain.net:8090/1.wav,3,6,4,0,5,1)

Aha.

> > > I want to be able to grab files via aws S3 which would require me to
> > > add some headers to authenticate.
> > 
> > Presumably you mean you need to add some headers to an HTTP request?
> 
> Correct
> 
> > > Is there any way to have Asterisk add headers or would I need a http
> > > proxy in the middle?
> > 
> > Where and how is Asterisk making an HTTP request at all?
> 
> Asterisk is using  URI Media Playback.

Ah.  So the problem (or challenge) is with URI Media Playback rather than 
ControlPlayback, I think.

> Please see:
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Project+-+URI+Media+
> Playback

ControlPlayback will play anything you give it; the challenge here seems to be 
fetching the media with URI Media Playback, and that's somethign I have no 
familiatiry with, so I'll let someone else step in with any ideas.


Regards,


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Re: [asterisk-users] AMI manager logins - omitting from logging output?

[Antony Stone]

On Thursday 07 June 2018 at 10:44:15, Tony Mountifield wrote:

> In article <201806070119.51560>, Antony Stone wrote:
> > 
> > Is there any way to tell AMI that I don't want it to log login attempts -
> > or, to put it another way, is there any way to tell the logger module to
> > ignore AMI?
> 
> Look in /etc/asterisk/manager.conf for the option "displayconnects =
> yes/no".
> 
> It can be set globally in [general] or individually in [ServiceCheck] (for
> example).

Lovely - thank you.

I've never seen that option in any manager documentation before, and it's not 
in the default file from Debian (Stretch).

Oddly, though, changing it and then doing a "manager reload" had no effect; I 
had to restart Asterisk for the setting to work.


Cheers,


Antony.

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[asterisk-users] AMI manager logins - omitting from logging output?

[Antony Stone]

Hi.

Is there any way to eliminate AMI manager logins from the logging output 
(without just turning the log level down and thereby losing lots of other stuff 
as well)?

I'm running Asterisk 13.14.1 as a backend service to LVS/IPVS, and using the 
AMI login as the "service alive" check to see which backend servers are 
available to take new commands.

This results in lots of

[Jun  7 00:15:19]   == Manager 'ServiceCheck' logged on from 10.100.42.254
[Jun  7 00:15:19]   == Manager 'ServiceCheck' logged off from 10.100.42.254

entries appearing in the console whenever I'm doing something else on the 
machine, which is pretty distracting.

Is there any way to tell AMI that I don't want it to log login attempts - or, 
to put it another way, is there any way to tell the logger module to ignore 
AMI?


Thanks,


Antony.

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Re: [asterisk-users] doing dnsmgr_lookup for

[Antony Stone]

On Thursday 31 May 2018 at 15:52:53, Jonas Kellens wrote:

> Hello list
> 
> is there a way to limit the number of dns lookups for 1 and the same host?
> 
> I see on Asterisk CLI a flood of :
> 
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> 
> I have several sip peer definitions (sip trunks) pointing at this same
> host.

Does it matter?

So long as you have a local caching DNS server (for highest performance, on 
the Asterisk server itself, with /etc/resolv.conf pointing to 127.0.0.1 or 
::1) the effect should not be noticeable.


Antony.

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Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

[Antony Stone]

On Sunday 31 December 2017 at 00:49:17, sean darcy wrote:

> I've been getting a lot of timeouts on non-critical invite transactions.

> So how is someone on a Dutch ISP using my server to mess with a US DoD
> ip address ?

What's your setting for "allowguest" (under [general]) in 
/etc/asterisk/sip.conf ?

What are your firewall rules for UDP 5060?


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Re: [asterisk-users] sip trunk with social media

[Antony Stone]

On Thursday 04 January 2018 at 01:27:59, bilal ghayyad wrote:

> Hello
> It will be amazing if possible to do sip trunk with any of social media
> providers like: whatsapp, facebook, imo, viber, ... etc

To the best of my knowledge none of the services you mention either operate 
over SIP or provide SIP connectivity to their systems.

Therefore I agree with you; it would be amazing if this were possible.


Antony.

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Re: [asterisk-users] remote Asterisk console

[Antony Stone]

On Tuesday 16 January 2018 at 18:19:30, Paul Neuwirth wrote:

> On Tue, 16 Jan 2018 18:18:18 +0200 Tzafrir Cohen wrote:
>
> > Anyway, as mentioned before: you should probably use AMI.
> 
> Thank you both. That was (most likely) what I was looking for - but
> still some worries about sending plaintext passwords...

AMI can operate over TLS.


Antony.

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Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

[Antony Stone]

On Saturday 20 January 2018 at 18:45:49, Jonathan H wrote:

> Oh, what a good idea! That's exactly the kind of lateral thinking I
> was hoping someone would come up with.
> 
> I thought it was called MixMonitor, and tried to wrap my head around
> it but couldn't.

MixMonitor is related, but different (and as the name suggests, automatically 
mixes the two channels, so I think Tim's suggestion to use Monitor is much 
better.

Note that you may well need to use the 'b' option with Monitor, to make sure 
you can record when there's no bridge between two channels.

> I'll give this a go tomorrow and let you know what I come up with!

Please do report back - this is a useful feature.


Antony.

> On 20 January 2018 at 17:03, Tim S wrote:
> > Just a quick and dirty thought, try the MONITOR application.
> > 
> > 
> > Pseudo-code:
> > 
> > Anchor-point
> > PLAYBACK ("press or say")
> > MONITOR (use the split audio files mode, not the mixed - this way you can
> > roughly separate which side did the "talking")
> > READ (audio file "1 to 5", try to grab one digit)
> > STOPMONITOR
> > IF (READ variable timed-out, send the incoming half of the monitor file
> > to Google Speech)
> > 
> >  Playback (some sound effect to indicate "thinking" on the Asterisk
> >  side
> > 
> > - user feedback is good)
> > 
> >  Check Google Speech result against a white-list
> >  IF filtered result was not a valid option
> >  
> >  PLAYBACK "I didn't understand that"
> >  GOTO to Anchor-point
> >  
> >  ELSE
> >  
> >  Goto next step using valid decoded speech data
> > 
> > ELSE
> > 
> >  Check DTMF result against a white-list
> >  IF filtered DTMFresult was not a valid option
> >  
> >  PLAYBACK "I didn't understand that"
> >  GOTO to Anchor-point
> >  
> >  ELSE
> >  
> >  Goto next step using valid decoded DTMF data
> > 
> > Catch-all, should never get here.
> > 
> > /Pseudo-code
> > 
> > 
> > Don't forget to filter your user sourced data against your white-list,
> > always assume users are hostile, this is part of the total picture of
> > defence-in-depth.
> > 
> > -Tim
> > 
> > On Sat, Jan 20, 2018 at 12:42 AM, Jonathan H wrote:
> >> Hello,
> >> 
> >> I want to start recording with a prompt of "press or say 1 to 5". If
> >> no DMTF is pressed, I want to send the recording to Google Speech to
> >> get the number back (got that part working already).
> >> 
> >> If any dtmf key is pressed while Application_Record  is running with
> >> option y, then the recording terminates and sends
> >> RECORD_STATUS of "DTMF" (A terminating DTMF was received).
> >> 
> >> But I need to know **what** number that DTMF was, and I can't see a
> >> way of grabbing it after the fact.
> >> 
> >> I can see in the code where the right variables are..
> >> 
> >> https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L140
> >> dtmf_response
> >> 
> >> https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L166
> >> * \param dtmf_integer the integer value of the DTMF key received
> >> 
> >> So,3 questions I guess:
> >> 
> >> 1: Am I going about this the right way? (unimrcp is not an option here)
> >> 2: Can someone explain in layman's terms how a simpleton like me could
> >> copy, hack about with and make a new module, like, for example,
> >> app_record_alt.c, that would stick around each time I updated Asterisk
> >> from source?
> >> 3: Or, is anyone willing to make the simple code change to the file to
> >> improve it to send back the DTMF to the dialplan? For free to improve
> >> core code? If not, and I posted on the commercial list, how much would
> >> I be looking at to modify about 6 lines of code and return an extra
> >> variable?
> >> 
> >> So, ultimately, I'm hoping for something like:
> >> 
> >> Currently:
> >> option "y" returns a RECORD_STATUS of "DTMF" if a key was press
> >> 
> >> Hopefully:
> >> option "z" returns a RECORD_STATUS of  showing which key
> >> was pressed.
> >> Or possibly even DTMF_VALUE (if an app can return two variables to the
> >> dialplan?)
> >> 
> >> I'm sure this would benefit a lot of people.
> >> 
> >> I posted this a few days ago in the forum at
> >> 
> >> https://community.asterisk.org/t/can-anyone-help-with-a-quick-app-record
> >> -c-module-improvement-and-can-explain-over-riding-modules/73221 but
> >> no-one bit, so, I'm hoping this list can help.
> >> 
> >> Many thanks!

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Re: [asterisk-users] how to make International calls from asterisk PBX

[Antony Stone]

On Monday 12 February 2018 at 12:25:00, Uzma Anjum wrote:

> Hello...
> 
> I'm running asterisk-13 and international calls are not working in it.How
> can I make it work.Can anyone please tell me.

We are sorry, but all our telepaths and clairvoyants are busy dealing with 
other queries right now.

Please supply us with more information about how you are currently trying to 
place international calls, and what error messages you get in response, and we 
may be able to help you.

Alternatively you may wait for someone to obtain the magical inspiration which 
enables them to diagnose your problem without any details to work from.


Regards,


Antony.

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Re: [asterisk-users] Sip cause and response codes in dialplan

[Antony Stone]

On Tuesday 20 February 2018 at 14:09:05, Marcus Kvarsell wrote:

> Hi,
> 
> I am experimenting with getting hold of the sip cause and sip response from
> outgoing call. How could i make a userevent printing the sip cause and/or
> sip response. I have tried using hangupcause, sip_cause and such , but i
> am not getting any data.

You don't say which version of Asterisk you're using, so I can't guarantee 
that the following will work for you, but I got this to work using Asterisk 
11.13.1:

In sip.conf, under the [general] stanza, define:
storesipcause=yes

You will get a warning to use hangupcause instead, but I haven't got that to 
do the same thing, so it's no substitute, I think.

Then, in your Dial() command, use M() to call a macro when the call gets 
answered.  https://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

In the macro definition, you can use ${HASH(SIP_CAUSE,${CDR(channel)})} to get 
the SIP response code.  It returns values such as "SIP 200 OK".


Hope that helps,


Antony.

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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Antony Stone]

On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote:

> On 2/22/18 1:07 PM, Antony Stone wrote:
> > On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:
> >>   Usually phone companies set the outgoing CallerID for you but
> >> 
> >> recently we got control over that and are now setting the outgoing
> >> Calleir ID ourselves.  My problem now is that the CDR will put the
> >> outgoing CID in the CDR instead of the extension that dialed and that
> >> causes problems for reports.  What is the proper way to set outgoing CID
> >> and keeping the original extension number in the CDR?
> > 
> > Surely the CDR field "clid" is your Caller ID, whereas the CDR field
> > "src" is the originating extension?
> > 
> > 
> > Antony.
> 
> No, the src field contains the external number and the clid field has
> the extension name but also the external number.

Okay, then; what do you get in the "channel" field?


Antony.

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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Antony Stone]

On Thursday 22 February 2018 at 20:07:47, Antony Stone wrote:

> On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:
> >  Usually phone companies set the outgoing CallerID for you but
> > 
> > recently we got control over that and are now setting the outgoing
> > Calleir ID ourselves.  My problem now is that the CDR will put the
> > outgoing CID in the CDR instead of the extension that dialed and that
> > causes problems for reports.  What is the proper way to set outgoing CID
> > and keeping the original extension number in the CDR?
> 
> Surely the CDR field "clid" is your Caller ID, whereas the CDR field "src"
> is the originating extension?

Another thought - if that doesn't automatically work for you (probably depends 
on your dialplan / Asterisk setup), then how about setting:
CDR(accountcode)=${CALLERID(number)}
in your dialplan sometime before you set the outbound Caller ID to whatever 
your PSTN number is?

Then you have the internal extension number in accountcode and the external 
CallerID in clid.


Antony.

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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Antony Stone]

On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:

>  Usually phone companies set the outgoing CallerID for you but
> recently we got control over that and are now setting the outgoing
> Calleir ID ourselves.  My problem now is that the CDR will put the
> outgoing CID in the CDR instead of the extension that dialed and that
> causes problems for reports.  What is the proper way to set outgoing CID
> and keeping the original extension number in the CDR?

Surely the CDR field "clid" is your Caller ID, whereas the CDR field "src" is 
the originating extension?


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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Antony Stone]

On Thursday 22 February 2018 at 23:44:43, Carlos Chavez wrote:

> On 2/22/18 4:40 PM, Carlos Chavez wrote:
> > On 2/22/18 3:54 PM, Carlos Chavez wrote:
> >> On 2/22/18 3:46 PM, Antony Stone wrote:
> >>> On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote:
> >>>> On 2/22/18 1:07 PM, Antony Stone wrote:
> >>>>> On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:
> >>>>>>Usually phone companies set the outgoing CallerID for you but
> >>>>>> 
> >>>>>> recently we got control over that and are now setting the outgoing
> >>>>>> Calleir ID ourselves.  My problem now is that the CDR will put the
> >>>>>> outgoing CID in the CDR instead of the extension that dialed and
> >>>>>> that
> >>>>>> causes problems for reports.  What is the proper way to set
> >>>>>> outgoing CID
> >>>>>> and keeping the original extension number in the CDR?
> >>>>> 
> >>>>> Surely the CDR field "clid" is your Caller ID, whereas the CDR field
> >>>>> "src" is the originating extension?
> >>>>> 
> >>>>> 
> >>>>> Antony.
> >>>> 
> >>>> No, the src field contains the external number and the clid field has
> >>>> the extension name but also the external number.
> >>> 
> >>> Okay, then; what do you get in the "channel" field?
> >> 
> >> Channels contains PJSIP/-(id)
> >> 
> >> Like I mentioned, the problem really lies in that the software
> >> used for call reports is coded to the "src" field. Than is why I need
> >> src to hace the extension number.
> > 
> > The solution to this problem is to set CDR(ani) to the original
> > extension number before changing the outgoing callerid.  With this src
> > will remain as the extension number.
> 
> Sorry, I meant CALLERID(ani).
> Set(CALLERID(ani)=${CALLERID(num)})

Aha, thank you :)

Now I understand why our systems do this by default - as per my first reply to 
you (I thought that src should contain the extension and clid the external 
number) - I just didn't realise there was a line in the dialplan responsible 
for ensuring this.

I keep learning something new every day.


Antony.

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Re: [asterisk-users] Increasing timeout before ending call from AMI

[Antony Stone]

On Tuesday 31 July 2018 at 12:38:04, Raimundo Pérez Nieves wrote:

> Hi guys, I sent a dial to asterisk

Which verson?

> with a specific timeout, I want to increase it for some users if it is
> approaching to the end, but when I send AbsoluteTimeout action

Show us what command you are sending?

> and change it timeout I get success but hangup at initial timeout, other
> words, it doesn’t increase timeout. I am doing this from AMI using telnet.
> There is any solution for this?
> Thanks for your help

Regards,


Antony.

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Re: [asterisk-users] How to know the IP of "manager show connected" in dialplan

[Antony Stone]

On Wednesday 25 July 2018 at 19:53:47, Saint Michael wrote:

> ​I need to launch a remote process at the machine that has the dialer. I
> could hard-code the IP address in a global variable, but It would be much
> more elegant if the dialplan would have a "manager" object where I could
> read "manager-->connected". ​

If the dialer is connected to Asterisk using AMI, how about issuing a 
UserEvent in the dialplan, which will then be seen by the logged-in dialer 
process (assuming it's looking at the event stream) and can be acted upon to 
launch the (now local) process?


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Re: [asterisk-users] Disable asterisk ssl how to

[Antony Stone]

On Wednesday 08 August 2018 at 22:30:52, Saint Michael wrote:

> I am trying to install Asterisk 11

Why?

> on debian 9

Have you tried installing https://packages.debian.org/jessie/asterisk from 
Debian 8 to see if it'll go onto Debian 9?


Antony.

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Re: [asterisk-users] Issues with install DAHDI

[Antony Stone]

On Wednesday 15 August 2018 at 23:10:21, Dovid Bender wrote:

> Hi,
> 
> I am trying to install wanpipe

Tell us how you are trying to install things.

> with dahdi on a CentOS7 box and I am running in to a few issues. My setup.
> 
> CentOS 7
> asterisk-15.5.0
> libpri-1.6.0
> dahdi linux and dahdi tools - 2.11.1
> 
> There are two issues.
> 
> 1) For some reason dahdi_tools isnt being built.

So, you're building from source?

What things are you building?  How are you building them?

> 2) When I try to load chan_dahdi and I get "Unknown signalling method
> 'pri_cpe' at line 35"
> 
> Based on what I found online it seems that it's an issue with libpri not
> being installed but I have it on the box. Any ideas?

How did it get installed on the box?

Try to give us enough information to reproduce your problem.


Antony.

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Re: [asterisk-users] change dialing process on live call

[Antony Stone]

On Sunday 19 August 2018 at 14:20:35, Khalil Khamlichi wrote:

> Thanks for your response, this works but we cannot hardcode this in the
> dialplan, we need this to be done from an external application connected
> either via manager or stasis.

Have you considered using Asterisk Realtime to store (part of) your dial plan 
in a database?  That can be updated dynamically and takes effect without a 
reload.

Obviously, if you have a Dial() command in the dial plan, you can't change 
that command *while* it's being executed, but you can change it for the next 
time that context gets executed.


Antony.

> On Sun, Aug 19, 2018, 11:14 AM Doug Lytle wrote:
> > On 08/19/2018 05:57 AM, Khalil Khamlichi wrote:
> > 
> > Is there a way to add another extension to a live dial, for example
> > 
> > Dial(PJSIP/1000,,)
> > 
> > and after 20 secondes change it to
> > 
> > Dial(PJSIP/1000/1001,,)
> > 
> > 
> > This is a simple one.
> > 
> > exten => s,1,Dial(SIP/1000,20)
> > exten => s,n,Dial(SIP/1000/1001,20)
> > exten => s,n,Hangup()
> > 
> > The first dial will ring with a 20 second timeout and proceed to the next
> > dial and ring both extensions for 20 seconds and finally hangup
> > 
> > Doug

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Re: [asterisk-users] Merging 2 conference bridges

[Antony Stone]

On Wednesday 22 August 2018 at 23:49:29, Ahmed Chohan wrote:

> Hi,
> 
> I would like to know how can I achieve merge 2 conference rooms in same
> asterisk server. For example 10 users joined bridge A and max user limit is
> set to 10. If more than 10 users try to join this bridge A, 11th user
> should join to the dynamically created bridge B and merge with bridge A. So
> that all eleven participants should be able to talk to each other.

My first question upon seeing this is:

 - if you want all 11 people to be able to talk to each other, why do you set 
a 10-participant limit on the original conference?


Antony.

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Re: [asterisk-users] STUN re-evalutation every 2 minutes ??

[Antony Stone]

On Saturday 01 September 2018 at 22:12:50, sean darcy wrote:

> 13.21.0
> 
> Every 2-3 minutes:

Does it really vary, or is it more like "every 150 seconds"?

> Sep  1 16:00:57] WARNING[150257]: res_stun_monitor.c:140
> stun_monitor_request: STUN poll got no response. Re-evaluating STUN
> server address.

What relevant firewall rules have you got?

> [Sep  1 16:02:18] NOTICE[150257]: res_stun_monitor.c:151
> stun_monitor_request: Old external address/port :42562 now
> seen as :33904.

What sort of Internet connectivity router are you using?

>   IAX, got a network change message, renewing all IAX registrations.
>   SIP, got a network change message, renewing all SIP registrations.
> 
> Always just for a different port number.

Sounds to me like your router has got a very short term connection tracking 
table, or else your firewall rules aren't allowing the required replies.

> I've tried a number of STUN servers with the same result. Now using
> counterpath :
> 
> /etc/asterisk/res_stun_monitor.conf:stunaddr = stun.counterpath.net
> 
> Probably harmless, but odd.
> 
> sean


Regards,


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