Re: [asterisk-users] asterisk admin interface

[Telium Technical Support]

I think only PJSIP and MWI support Sorcery – so that likely won’t do what’s 
being asked for…

 

And reading/writing a flat file should be even easier than learning the ARI

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Re: [asterisk-users] asterisk admin interface

[John Kiniston]

You could explore using ARI with it's Push configuration.

https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration

On Mon, May 16, 2016 at 11:33 AM, Goke Aruna  wrote:

> hi all,
> can anyone give me a guide on any asterisk admin solution / interface for
> config management, and monitoring?
> No database use is intended and I prefer open source.
>
> Thanks for support.
>
> Regards
>
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Re: [asterisk-users] asterisk admin interface

[Telium Technical Support]

>Thanks Raj
>You are correct. Is there any open source application in that? 

Not that I know of – I think it’s getting too simplistic J  We created some C++ 
functions for our High Availability for Asterisk product (HAAst) which modify 
config files and extensions files, but it’s more work to adapt them than just 
write your own.  In a few hours of coding you should be able to have it done…

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Re: [asterisk-users] asterisk admin interface

[Steve Edwards]

On May 16, 2016 22:15, "Telium Technical Support"  wrote:



> In this case a very simple solution is to modify the Asterisk config files
> to add/remove users, then tell Asterisk to reload from the CLI/AMI.  And
> that's it!



  On 17/05/2016, at 9:55 am, Goke Aruna  wrote:
  You are correct. Is there any open source application in that?


On Tue, 17 May 2016, Pete Mundy wrote:


According to WikiPedia, there are open-source implementations of vi available:
https://en.wikipedia.org/wiki/Vi


But that only takes care of the configuration. Is there an open source 
application to execute "sudo asterisk -r -x 'reload'"?


However, if we use emacs instead of vi, we could write an 'asterisk' mode 
and hook the 'reload' into the 'save-buffer' function...


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Re: [asterisk-users] asterisk admin interface

[Brian Wilson]

You are correct. Is there any open source application in that?


> According to WikiPedia, there are open-source implementations of vi
> available:
>

All my instincts say "No no! Use emacs not vi!"
but I think the OP might not know saying "vi" is intended as a joke?

For small systems using a text editor is okay... watch out for typos that
will disable your entire pbx... do "dialplan reload" when you change
extensions.conf and then scroll back looking for ERROR messages. I find the
color coding to be extremely helpful.

Likewise open "asterisk -r" immediately after starting asterisk and watch
for error messages. Be warned that sometimes the errors will lead you far
far astray. Usually they are useful.


Brian
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Re: [asterisk-users] asterisk admin interface

[Pete Mundy]

> On 17/05/2016, at 9:55 am, Goke Aruna  wrote:
> On May 16, 2016 22:15, "Telium Technical Support"  > wrote:
> > 
> >
> > In this case a very simple solution is to modify the Asterisk config files
> > to add/remove users, then tell Asterisk to reload from the CLI/AMI.  And
> > that's it!
> 
> Thanks Raj
> 
> You are correct. Is there any open source application in that?
> 


According to WikiPedia, there are open-source implementations of vi available:

https://en.wikipedia.org/wiki/Vi


Pete




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Re: [asterisk-users] asterisk admin interface

[Goke Aruna]

On May 16, 2016 22:15, "Telium Technical Support"  wrote:
>
> You don't mention a configuration generator (like Elastix/FreePBX) so I'll
> assume you are using a plain old vanilla Asterisk installation.  In which
> case all user/endpoint information is kept in config (ini) files, and no
> user/endpoint manipulation is done through the CLI or AMI.
>
> In this case a very simple solution is to modify the Asterisk config files
> to add/remove users, then tell Asterisk to reload from the CLI/AMI.  And
> that's it!
>
> -Raj-
>
>
> --
>

Thanks Raj
You are correct. Is there any open source application in that?
regards_
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Re: [asterisk-users] asterisk admin interface

[Telium Technical Support]

You don't mention a configuration generator (like Elastix/FreePBX) so I'll
assume you are using a plain old vanilla Asterisk installation.  In which
case all user/endpoint information is kept in config (ini) files, and no
user/endpoint manipulation is done through the CLI or AMI.

In this case a very simple solution is to modify the Asterisk config files
to add/remove users, then tell Asterisk to reload from the CLI/AMI.  And
that's it!

-Raj-


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[asterisk-users] JABBER_RECEIVE timeout don't work

[Annus Fictus]

Hello,

I'm trying to use JABBER_RECEIVE function on my dialplan but the timeout 
function don't work.


This is my dialplan:

[google-in]
exten => s,1,NoOp( Call from Gtalk )
same => n,SendText(Hola,Como te llamas?)
same => n,Set(nombre=${JABBER_RECEIVE(google,${CALLERID(name)},30)})
same => n,SendText(Hola ${nombre}, bienvenido en XYZ)
same => n,Set(CALLERID(name)=${nombre})
same => n,Wait(2)
same => n,SendText(Espera un momento mientras te comunicamos)
same => n,Dial(PJSIP/1000,30)
same => n,Hangup()

In theory the function have to wait a answer for 30 seconds but don't.  
Asterisk execute next dialplan line immediately.




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Re: [asterisk-users] Asterisk PJSIP Multi-tenant

[Annus Fictus]

Done.

ASTERISK-26026

El 16/05/2016 a las 14:40, George Joseph escribió:



On Sun, May 15, 2016 at 10:17 PM, Annus Fictus > wrote:


Hello,

with qualify_frequency=0 I can't receive calls from others endpoints.

Other strange think is if I set mailboxes parameter on the
console, when the endpoint registering, i can see:

ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable
to create outbound NOTIFY request to endpoint 1...@sip.domain.com

WARNING[2208]: res_pjsip_mwi.c:379
send_unsolicited_mwi_notify_to_contact: Unable to create
unsolicited NOTIFY request to endpoint 1...@sip.domain.com
 URI
sip:1001@95.250.29.3:50673;rinstance=1af959e7c0059fc4

Unsolicited NOTIFY and OPTIONS both use the out-of-dialog path so I'm 
guessing we have an issue there.  Open an issue at 
https://issues.asterisk.org




Regards

El 16/05/2016 a las 02:52, George Joseph escribió:



On Sun, May 15, 2016 at 12:00 PM, Annus Fictus
> wrote:

Hello List,

following this thread:


http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains

I tried to configure on the pjsip.conf the same endpoint with
different domains like:

[1...@sip.domain.com ]
type=endpoint

[1...@sip1.domain.com ]
type=endpoint

I can register the two 1000 endpoints using different domain
but on the Asterisk console:

ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request:
Unable to create outbound OPTIONS request to endpoint
1...@sip.domain.com 

ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact:
Unable to create request to qualify contact
sip:1000@95.250.29.3:53570
;rinstance=d90827763e4353c0

in the aor section I'm using:

qualify_frequency=30


If you set qualify_frequency=0, can that endpoint make and
receive calls?  Not suggesting this as a solution, just asking to
narrow possibilities down.


Any hint?

Regards


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www.asterisk.org 






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www.asterisk.org 






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Re: [asterisk-users] asterisk admin interface

[Steve Edwards]

Please don't top post.


On Mon, 16 May 2016, Goke Aruna wrote:


can anyone give me a guide on any asterisk admin solution / interface 
for config management, and monitoring? No database use is intended and I 
prefer open source.


On May 16, 2016 20:50, "Steve Edwards"  
wrote:


Can you be a bit more specific about what you want to accomplish?


On Mon, 16 May 2016, Goke Aruna wrote:


I want to be able edit my dialplan and add /edit / delete sip accounts.

Then be able to see active calls though I have done that AMI in the past 
but i would not mind learning the best way it is done today.


I think most people use FreePBX for this kind of task, but that may be 
historical inertia.


You may want to take a look at nerdvittles.com. I think their Incredible 
PBX package uses FreePBX or some new GUI whose name escapes me.


They also have a couple of articles on XiVO which they seem to be excited 
about.


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Re: [asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message

[Brian Wilson]

Did you build from source on one machine and install on another? I ran into
something like that, have to turn off optimizations in the build
environment if you do that and the machine architecture is different.

On Mon, May 16, 2016 at 1:03 PM, asterisk 
wrote:

> Hi folks,
>
> I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7
> LTS), I've just configured the voicemail function, and it's mostly working
> fine... except when I try to leave a voicemail! This crashes asterisk with
> no entries in the messages log.
>
> The system is running on Centos 6 (or maybe 6.5, I'm not sure how to check
> this). uname -a returns:
>
> Linux asterisk.sjssolutions.local 3.10.0-327.13.1.el7.x86_64 #1 SMP
> Thu Mar 31 16:04:38 UTC 2016 x86_64 x86_64 x86_64 GNU/Linux
>
> On the CLI, I get this:
>
>   == Using SIP RTP CoS mark 5
>   == Extension Changed 5103[hints] new state InUse for Notify User 5104
>   == Extension Changed 5103[hints] new state InUse for Notify User 5103
> -- Executing [5106@internal:1] NoOp("SIP/5103-", "-- Calling
> SJS extension 5106 from SIP/5103-, transferring context") in new
> stack
> -- Executing [5106@internal:2] Goto("SIP/5103-",
> "sjs_extensions,5106,1") in new stack
> -- Goto (sjs_extensions,5106,1)
> -- Executing [5106@sjs_extensions:1] Dial("SIP/5103-",
> "IAX2/remoteAsterisk/5106,10") in new stack
> -- Called IAX2/remoteAsterisk/5106
> -- Call accepted by  (format ulaw)
> -- Format for call is (ulaw)
> -- IAX2/remoteAsterisk-17114 is ringing
> -- IAX2/remoteAsterisk-17114 is ringing
> -- Nobody picked up in 1 ms
> -- Hungup 'IAX2/remoteAsterisk-17114'
> -- Executing [5106@sjs_extensions:2] VoiceMail("SIP/5103-",
> "5103,u") in new stack
> [May 16 20:37:58] WARNING[14514][C-]: res_rtp_asterisk.c:4264
> ast_rtp_read: RTP Read too short
> [May 16 20:37:58] WARNING[14514][C-]: res_rtp_asterisk.c:4264
> ast_rtp_read: RTP Read too short
> [May 16 20:37:58] WARNING[14514][C-]: res_rtp_asterisk.c:4264
> ast_rtp_read: RTP Read too short
>> 0x7f61e008b750 -- Probation passed - setting RTP source address
> to 10.0.0.190:5004
> --  Playing 'vm-theperson.ulaw' (language 'en_GB')
> --  Playing 'digits/5.ulaw' (language 'en_GB')
> --  Playing 'digits/1.ulaw' (language 'en_GB')
> --  Playing 'digits/0.ulaw' (language 'en_GB')
> --  Playing 'digits/3.ulaw' (language 'en_GB')
> --  Playing 'vm-isunavail.ulaw' (language 'en_GB')
> --  Playing 'vm-intro.ulaw' (language 'en_GB')
> --  Playing 'beep.ulaw' (language 'en_GB')
> -- Recording the message
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/5103/tmp/nzuoKd format: wav,
> 0x7f621800bba8
> asterisk*CLI>
> Disconnected from Asterisk server
> Asterisk cleanly ending (0).
> Executing last minute cleanups
>
> (note: Yes, it's deliberate that it's going to a different extension VM...
> the call goes via another asterisk server to a remote phone; then comes
> back if unanswered to record the VM.)
>
> The system starts to create the file, and sometimes even records some
> bytes, before dying:
>
> [root@asterisk tmp]# ls -l
> total 4
> -rw-r--r-- 1 root root  0 May 16 20:38 nzuoKd
> -rw-r--r-- 1 root root 44 May 16 20:38 nzuoKd.wav
>
> Note: I've since changed the safe_asterisk script to start up Asterisk as
> asterisk:asterisk, it seems to still work; apart from VM which crashes the
> same way.
>
> I tried setting the file format to ulaw, this had the same problem (except
> the temp file ended with .ulaw). I saw a similar problem had been solved in
> version 1.6.1, except that didn't seem to show the "x=0, open writing:"
> message.
>
> System has plenty of available disk space (40G or 179G depending on which
> bit of the filesystem you look at).
>
> I've never seen this on any of the Asterisk servers I've run (many, since
> v1.4), but I mostly run it on Ubuntu variants, this is my first Centos...
>
> Addendum: I modified safe_asterisk & got the following when it quit:
>
> /usr/sbin/safe_asterisk: line 163: 18115 Illegal instruction (core
> dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS}
> > /dev/${TTY} 2>&1 < /dev/${TTY}
>
> Any ideas gratefully received. I'm going to try installing a
> compiled-from-source version of 13.7 at the weekend, can't do it before
> then as it's our production office system... Everything apart from VM seems
> to work (although if anyone can shed any light on the frequent
> "res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short" warnings I'm
> seeing, that'd also be appreciated.
>
> Oh - one more thing, I had to disable 2 codecs (lpc10 and ilbc) because
> they used an instruction that doesn't exist on the server (it's an oldish
> HP mini-server). I'm guessing from the above message that VM might be
> afflicted by the same issue. Presumably compiling from source will 

[asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message

[asterisk]

Hi folks,

I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 
LTS), I've just configured the voicemail function, and it's mostly 
working fine... except when I try to leave a voicemail! This crashes 
asterisk with no entries in the messages log.


The system is running on Centos 6 (or maybe 6.5, I'm not sure how to 
check this). uname -a returns:


Linux asterisk.sjssolutions.local 3.10.0-327.13.1.el7.x86_64 #1 SMP 
Thu Mar 31 16:04:38 UTC 2016 x86_64 x86_64 x86_64 GNU/Linux


On the CLI, I get this:

  == Using SIP RTP CoS mark 5
  == Extension Changed 5103[hints] new state InUse for Notify User
   5104
  == Extension Changed 5103[hints] new state InUse for Notify User
   5103
-- Executing [5106@internal:1] NoOp("SIP/5103-", "--
   Calling SJS extension 5106 from SIP/5103-, transferring
   context") in new stack
-- Executing [5106@internal:2] Goto("SIP/5103-",
   "sjs_extensions,5106,1") in new stack
-- Goto (sjs_extensions,5106,1)
-- Executing [5106@sjs_extensions:1] Dial("SIP/5103-",
   "IAX2/remoteAsterisk/5106,10") in new stack
-- Called IAX2/remoteAsterisk/5106
-- Call accepted by  (format ulaw)
-- Format for call is (ulaw)
-- IAX2/remoteAsterisk-17114 is ringing
-- IAX2/remoteAsterisk-17114 is ringing
-- Nobody picked up in 1 ms
-- Hungup 'IAX2/remoteAsterisk-17114'
-- Executing [5106@sjs_extensions:2]
   VoiceMail("SIP/5103-", "5103,u") in new stack
   [May 16 20:37:58] WARNING[14514][C-]:
   res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
   [May 16 20:37:58] WARNING[14514][C-]:
   res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
   [May 16 20:37:58] WARNING[14514][C-]:
   res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
   > 0x7f61e008b750 -- Probation passed - setting RTP source
   address to 10.0.0.190:5004
--  Playing 'vm-theperson.ulaw' (language
   'en_GB')
--  Playing 'digits/5.ulaw' (language 'en_GB')
--  Playing 'digits/1.ulaw' (language 'en_GB')
--  Playing 'digits/0.ulaw' (language 'en_GB')
--  Playing 'digits/3.ulaw' (language 'en_GB')
--  Playing 'vm-isunavail.ulaw' (language
   'en_GB')
--  Playing 'vm-intro.ulaw' (language 'en_GB')
--  Playing 'beep.ulaw' (language 'en_GB')
-- Recording the message
-- x=0, open writing:
   /var/spool/asterisk/voicemail/default/5103/tmp/nzuoKd format: wav,
   0x7f621800bba8
   asterisk*CLI>
   Disconnected from Asterisk server
   Asterisk cleanly ending (0).
   Executing last minute cleanups

(note: Yes, it's deliberate that it's going to a different extension 
VM... the call goes via another asterisk server to a remote phone; then 
comes back if unanswered to record the VM.)


The system starts to create the file, and sometimes even records some 
bytes, before dying:


   [root@asterisk tmp]# ls -l
   total 4
   -rw-r--r-- 1 root root  0 May 16 20:38 nzuoKd
   -rw-r--r-- 1 root root 44 May 16 20:38 nzuoKd.wav

Note: I've since changed the safe_asterisk script to start up Asterisk 
as asterisk:asterisk, it seems to still work; apart from VM which 
crashes the same way.


I tried setting the file format to ulaw, this had the same problem 
(except the temp file ended with .ulaw). I saw a similar problem had 
been solved in version 1.6.1, except that didn't seem to show the "x=0, 
open writing:" message.


System has plenty of available disk space (40G or 179G depending on 
which bit of the filesystem you look at).


I've never seen this on any of the Asterisk servers I've run (many, 
since v1.4), but I mostly run it on Ubuntu variants, this is my first 
Centos...


Addendum: I modified safe_asterisk & got the following when it quit:

/usr/sbin/safe_asterisk: line 163: 18115 Illegal instruction 
(core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} 
${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}


Any ideas gratefully received. I'm going to try installing a 
compiled-from-source version of 13.7 at the weekend, can't do it before 
then as it's our production office system... Everything apart from VM 
seems to work (although if anyone can shed any light on the frequent 
"res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short" warnings I'm 
seeing, that'd also be appreciated.


Oh - one more thing, I had to disable 2 codecs (lpc10 and ilbc) because 
they used an instruction that doesn't exist on the server (it's an 
oldish HP mini-server). I'm guessing from the above message that VM 
might be afflicted by the same issue. Presumably compiling from source 
will solve this? (I've compiled 13.7, no errors reported, but I've not 
tried running it yet)


Cheers!
Ade.

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Re: [asterisk-users] asterisk admin interface

[Goke Aruna]

Thanks Steve
I want to be able edit my dialplan and add /edit / delete sip accounts.

Then be able to see active calls though I have done that AMI in the past
but i would not mind learning the best way it is done today.
Regards
On May 16, 2016 20:50, "Steve Edwards"  wrote:

On Mon, 16 May 2016, Goke Aruna wrote:

can anyone give me a guide on any asterisk admin solution / interface for
> config management, and monitoring? No database use is intended and I prefer
> open source.
>

Based upon the above requirements, 'use the CLI' would fit the bill.

Can you be a bit more specific about what you want to accomplish?

I suspect at some point, most solutions will require a database of some
sorts.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] asterisk admin interface

[Steve Edwards]

On Mon, 16 May 2016, Goke Aruna wrote:

can anyone give me a guide on any asterisk admin solution / interface 
for config management, and monitoring? No database use is intended and I 
prefer open source.


Based upon the above requirements, 'use the CLI' would fit the bill.

Can you be a bit more specific about what you want to accomplish?

I suspect at some point, most solutions will require a database of some 
sorts.


--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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[asterisk-users] asterisk admin interface

[Goke Aruna]

hi all,
can anyone give me a guide on any asterisk admin solution / interface for
config management, and monitoring?
No database use is intended and I prefer open source.

Thanks for support.

Regards
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Re: [asterisk-users] DAHDI press button get fast busy

[Greg Woods]

On Mon, May 16, 2016 at 9:06 AM, Tzafrir Cohen 
wrote:

> asterisk -rx "dialplan show $context"


There is no existence of 'from-internal' context

OK, so now I know what to work on; thank you very much for that.

I can see that, in fact, the dahdi-channels.conf file sets the context as
"from-internal". I found an old version of this file, and it has the
context set to "internal", which is most likely what it should be. I will
test this when I get home tonight. Thanks again for pointing me in the
right direction.

--Greg
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Re: [asterisk-users] Russian and French sounds

[Dovid Bender]

Tzafrir,

Thanks. I poke around but was never able to find it.


On Mon, May 16, 2016 at 11:25 AM, Tzafrir Cohen 
wrote:

> On Wed, May 11, 2016 at 06:05:49AM -0400, Dovid Bender wrote:
> > Hi,
> >
> > Does anyone know who did the prompts for French and Russian for
> Asterisk? I
> > need some custom prompts.
>
> You can find CREDIT files inside the tarballs (the trick is: ls | grep
> -v '$'   or: ls [A-Z]*)
>
> $ cat asterisk-core-sounds-ru-gsm/CREDITS-asterisk-core-ru-1.4.27
> core-sounds-ru v1.1
>
> Provided by: http://www.ivrvoice.ru
> To order new files recorded with the same voice please sent your request
> to i...@pbxware.ru
>
> Acknowledgments:
> Max Litnitskiy - idea and team building,
> Andrew Roman - director of recording,
> Alex Litnitskiy - testing and bug fixing,
> Denis Gaidamak - assistant of producer  of initial release.
>
>
> $ cat asterisk-core-sounds-fr-gsm/CREDITS-asterisk-core-fr-1.4.27
> Recorded by:
> June Wallack (http://www.junewallack.com)
>
> Translated into French by:
> Clod Patry 
> Kristopher Lalletti 
> June Wallack
>
> Financial Contributions by:
> Digium, Inc. (http://www.digium.com)
> Unlimitel (http://www.unlimitel.ca)
> BGM Informatique (www.bgm.qc.ca)
>
> --
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
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Re: [asterisk-users] Russian and French sounds

[Tzafrir Cohen]

On Wed, May 11, 2016 at 06:05:49AM -0400, Dovid Bender wrote:
> Hi,
> 
> Does anyone know who did the prompts for French and Russian for Asterisk? I
> need some custom prompts.

You can find CREDIT files inside the tarballs (the trick is: ls | grep
-v '$'   or: ls [A-Z]*)

$ cat asterisk-core-sounds-ru-gsm/CREDITS-asterisk-core-ru-1.4.27
core-sounds-ru v1.1

Provided by: http://www.ivrvoice.ru
To order new files recorded with the same voice please sent your request
to i...@pbxware.ru

Acknowledgments:
Max Litnitskiy - idea and team building,
Andrew Roman - director of recording,
Alex Litnitskiy - testing and bug fixing,
Denis Gaidamak - assistant of producer  of initial release.


$ cat asterisk-core-sounds-fr-gsm/CREDITS-asterisk-core-fr-1.4.27 
Recorded by:
June Wallack (http://www.junewallack.com)

Translated into French by:
Clod Patry 
Kristopher Lalletti 
June Wallack

Financial Contributions by:
Digium, Inc. (http://www.digium.com)
Unlimitel (http://www.unlimitel.ca)
BGM Informatique (www.bgm.qc.ca)

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

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Re: [asterisk-users] DAHDI press button get fast busy

[Tzafrir Cohen]

On Thu, May 12, 2016 at 03:04:02PM -0600, Greg Woods wrote:
> My DAHDI phones were broken since a recent power outage (which required a
> reboot). For some reason the Asterisk or DAHDI configuration is messed up
> somewhere (probably from an update that was applied before the reboot?).
> 
> I am using a Digium TDM410 card.
> 
> At first, nothing worked at all, but I discovered that if I loaded things
> manually (e.g. modprobe dahdi; modprobe wctdm24xxp, then go into asterisk
> -r and "module unload chan_dahdi; module load chan_dahdi") then at least
> the lights on the card come on and the phones half work. Inbound calls work
> fine, but if I try to dial out, as soon as I press a numeric button on the
> phone, I get an immediate fast busy, so outbound calls don't work. Outbound
> calls from a SIP phone off the same server work fine (the outbound line is
> an IAX link to a VOIP provider).
> 
> I have been Googling this for nearly an hour without finding any reference
> to this problem. Anybody have any idea where I can look to debug this, or a
> guess as to what might be wrong with the configuration?

Sounds like you did not set a proper dialplan for them. I assume the
phone is on dahdi channel 1. If so, please provide the output of:

  context=`asterisk -rx 'dahdi show channel 1' | awk '/Context: / {print $2}'
  asterisk -rx "dialplan show $context"

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

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Re: [asterisk-users] Asterisk PJSIP Multi-tenant

[George Joseph]

On Sun, May 15, 2016 at 10:17 PM, Annus Fictus 
wrote:

> Hello,
>
> with qualify_frequency=0 I can't receive calls from others endpoints.
>
> Other strange think is if I set mailboxes parameter on the console, when
> the endpoint registering, i can see:
>
> ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to
> create outbound NOTIFY request to endpoint 1...@sip.domain.com
> WARNING[2208]: res_pjsip_mwi.c:379 send_unsolicited_mwi_notify_to_contact:
> Unable to create unsolicited NOTIFY request to endpoint
> 1...@sip.domain.com URI
> sip:1001@95.250.29.3:50673;rinstance=1af959e7c0059fc4
>
Unsolicited NOTIFY and OPTIONS both use the out-of-dialog path so I'm
guessing we have an issue there.  Open an issue at
https://issues.asterisk.org





> Regards
> El 16/05/2016 a las 02:52, George Joseph escribió:
>
>
>
> On Sun, May 15, 2016 at 12:00 PM, Annus Fictus 
> wrote:
>
>> Hello List,
>>
>> following this thread:
>>
>>
>> http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains
>>
>> I tried to configure on the pjsip.conf the same endpoint with different
>> domains like:
>>
>> [1...@sip.domain.com]
>> type=endpoint
>>
>> [1...@sip1.domain.com]
>> type=endpoint
>>
>> I can register the two 1000 endpoints using different domain but on the
>> Asterisk console:
>>
>> ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to
>> create outbound OPTIONS request to endpoint 1...@sip.domain.com
>>
>> ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable to
>> create request to qualify contact sip:1000@95.250.29.3:53570
>> ;rinstance=d90827763e4353c0
>>
>> in the aor section I'm using:
>>
>> qualify_frequency=30
>>
>
> If you set qualify_frequency=0, can that endpoint make and receive
> calls?  Not suggesting this as a solution, just asking to narrow
> possibilities down.
>
>
>
>>
>> Any hint?
>>
>> Regards
>>
>>
>> --
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at:  www.digium.com &
> www.asterisk.org
>
>
>
>
>
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-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Questions... connecting Asterisk to the World

[A J Stiles]

On Saturday 14 May 2016, Stefan Becker wrote:
> Greetings,
> 
> asterisk list and community,
> 
> I have a problem in how our telefon switch (Siemens HiCOM)
> "talks" with my new configured Asterisk server (V.11.18.0)
> 
> without my Asterisks server in the middle
> 
>  <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom
> 
> A phone connected to the switch requests an "Outgoing" line
> by dialing "0". The party is connected via ISDN to
> the carrier (deutsche Telekom) where the party preceeds
> to dial numbers... and the call is connected
> 
> What I can see while I am dialing is that with every
> digit I press it is being displayed on my phone.
> Further more, these digits are being processed by the
> carrier. The call goes through, rings, immediately on
> completion on the number or is rejected if busy.
> 
> 
> WITH my Asterisks server in the middle of the exchange...
> 
> A phone connected to the switch requests an "Outgoing" line
> by dialing "0". -->  Asterisks recieves incoming call on "s".
> The dialed digits are collected. The dial plan is
> executed accordingly but the "caller" recieves no
> more information about the dialed number. The number is
> not placed in the "dialed" numbers simple functions like
> "redial" do not work anymore.
> 
> Does anybody know what I am doing wrong here. Is there a
> way to teach asterisk to behave exactly as if it were the
> PBX (deutsche Telekom).
> So, as to say, act in a way that NO ONE will rightly know
> the differance between having asterisk taking over the
> function of the ISDN PBX.
> 
> What do I need?  A better dial plan to somehow better simulate
> the way the switch normaly behaves?
> Is hardware the problem?
> 
> 
> My ISDN card in the server is:
> "QuadBRI ISDN Digium Wildcard b410P"
> 
> Most everything else functionly works. incoming and outgoing calls
> from and to ISDN, VoIP and other equipment work fine.
> 
> Just that the phones and switch don't recieve the "collected"
> number sequence the was dialed.
> 
> Any help or ideas anyone might have would be greatly appreciated.


Your problem is that you are still thinking in terms of old-fashioned, clicky-
clicky mechanical telephone exchanges.  Instead of "dialling 0 to request an 
outside line", you need to let Asterisk accept all the digits and then 
determine for itself whether the call is going to be an inside or outside one.

- If the user dials 3 digits  (or however long your internal numbers are),  
treat it as an internal number.
- If the user dials 6 digits  (or however long numbers are on your local 
exchange),  treat it as an external, local number.
- If the user dials 11 digits starting with 0  (or however long a number is in 
your country, including the STD code),  treat it as an external, STD number.
- If the number dials 9 or more digits starting with 00, treat it as an 
external, IDD number.


It will make your dialplan a little more complicated; but if it is too simple, 
you won't be taking full advantage of the power of Asterisk.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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