gmane.comp.telephony.freeswitch.user

Re: Cannot ring extension from DID

Mike Hendrie — 2013-05-22T13:15:50

I do.  I can also ring extension 1000.
 On May 22, 2013 7:52 AM, "Philippe Le Toquin" <philippe-9OQXqz8nkhQ< at >public.gmane.org> wrote:

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Re: Cannot ring extension from DID

Philippe Le Toquin — 2013-05-22T12:49:43

This is why it goes to voicemail I guess

   1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/
   5555555555-rf9+wGdtdM2ZQwJf0e2p3w< at >public.gmane.org EXPORTING[export_vars]
[dialed_extension]=[1001]to event
   2. 2013-05-21 22:28:54.107400 [DEBUG]
switch_ivr_originate.c:2044Parsing global variables
   3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing
   variable [sip_invite_domain]=[GothamCity.xom]
   4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing
   variable [presence_id]=[1001-tZ9VlegwiPmU7knFcz9Dig< at >public.gmane.org]
   5. 2013-05-21 22:28:54.107400 [NOTICE]
switch_ivr_originate.c:2639Cannot create outgoing channel of type
   [error] cause: [USER_NOT_REGISTERED]
   6. 2013-05-21 22:28:54.107400 [DEBUG]
switch_ivr_originate.c:3605Originate Resulted in Error Cause:
   606 [USER_NOT_REGISTERED]
   7. 2013-05-21 22:28:54.107400 [NOTICE]
switch_ivr_originate.c:2639Cannot create outgoing channel of type
   [user] cause: [USER_NOT_REGISTERED]
   8. 2013-0

Re: Cannot ring extension from DID

Mike Hendrie — 2013-05-22T11:55:11

Thank you.

Here is the log URL.
http://pastebin.freeswitch.org/20960



On Wed, May 22, 2013 at 1:39 AM, Michael Collins <msc-YF8E+gPBBv73h3GqohbjpQ< at >public.gmane.org> wrote:

_________________________________________________________________________
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Re: Node.JS ESL libraries

Dmitry Sytchev — 2013-05-22T09:17:30

https://github.com/englercj/node-esl  is modeled after standard FS esl libs
and seems to implement full spec
I faced some strange troubles with malformed UTF8 symbols in messages, but
it works well and is fully async.


2013/5/22 Thomas Lee <thomas.lee-PmgLHhJpKdFeoWH0uzbU5w< at >public.gmane.org>

Re: DTMF outbound call

2013-05-22T08:29:33

So i'm the only man with this problem ?
nobody use dtmf ?

:'(


Le 2013-05-21 14:03, Brian Foster a écrit :


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Re: OpenVZ tuning tips

Zenny — 2013-05-22T07:56:06

Tony, do you mean "very new kernel" means 3.2.xx kernel?

Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not
possible. And that is what CentOS6 offers, too.

However, I installed FS as openvz guest, it works fine for outgoing,
but not DNAT works for incoming connections even after throroughly
following http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT.

Just my two cents.




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Re: One Way Audio

Michael Collins — 2013-05-22T07:11:01

You need to show this pcap to your carrier and ask them what's up. You are
definitely sending RTP.

-MC


On Tue, May 21, 2013 at 6:12 PM, Sean Devoy <sdevoy-4IvgkAfB424yY3YROqfsYA< at >public.gmane.org> wrote:

Re: Switch on/off call recording

Michael Collins — 2013-05-22T06:44:45

You could have your web app set a channel variable and then only turn on
recording if that channel variable was set to true:

<condition field="${recording_enabled}" expression="^true$">
  <action application="record_session" data="/path/to/file.wav"/>
</condition>

Just be sure to use continue="true" on that extension. ;)

-MC



On Tue, May 21, 2013 at 11:29 PM, Ashish gautam <ashish-kMxHSFh002n/PtFMR13I2A< at >public.gmane.org> wrote:

Re: DTMF outbound call

Hermouet Erwan — 2013-05-22T06:44:16

Brian did you find something on log file ?
Maybe i miss config option.

Tks


Brian Foster <bdfoster-v1fKxmSWr8YAvxtiuMwx3w< at >public.gmane.org> a écrit :


Hermouet Erwan
Responsable technique
Bluetel_________________________________________________________________________
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Re: Cannot ring extension from DID

Michael Collins — 2013-05-22T06:39:39

Post a FreeSWITCH debug log of the incoming call. Use
pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax
highlighting. Paste the URL in this email thread and we'll take a look.
-MC


On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie <mike-0JVlv975Uqdt1OO0OYaSVA< at >public.gmane.org> wrote:

Re: Switch on/off call recording

Ashish gautam — 2013-05-22T06:29:21

I am doing it through dialplan app record_session. And it records for all
the calls. Now I want it for selected calls only.


On Wed, May 22, 2013 at 11:50 AM, Michael Collins <msc-YF8E+gPBBv73h3GqohbjpQ< at >public.gmane.org>wrote:

_________________________________________________________________________
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Re: Switch on/off call recording

Michael Collins — 2013-05-22T06:20:02

Are you currently recording calls now? If so, how are you doing that?
-MC


On Tue, May 21, 2013 at 10:04 PM, Ashish gautam <ashish-kMxHSFh002n/PtFMR13I2A< at >public.gmane.org> wrote:

Re: Cannot ring extension from DID

Mike Hendrie — 2013-05-22T02:27:52

Thank you for your assistance. I made the suggested modification below,
however, when calling the number it goes directly to voicemail.

  /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml

<include>

  <extension name="vitel-inbound">
    <condition field="destination_number" expression="^1?(262xxxxxxx)$">
       <action application="transfer" data="1000 XML default"/>
    </condition>
  </extension>

</include>



On Tue, May 21, 2013 at 4:37 PM, Michael Collins <msc-YF8E+gPBBv73h3GqohbjpQ< at >public.gmane.org> wrote:

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting-YF8E+gPBBv73h3GqohbjpQ< at >public.gmane.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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Re: Cannot ring extension from DID

Mike Hendrie — 2013-05-22T04:35:15

Correction:
I had a second dialplan in the public folder that was causing confusion.
Below is the dialplan I am using.
If I change the extension in the dialplan from 1000 to 1001 I get
the appropriate voice mail extension, however, the phones never ring.

I have the fs configured as a multi-tenant solution.

Could the dialplan be using the default extensions (1000 and 1001) under
/conf/directory/default and not reference the
/conf/directory/GothamCity.xom domain? That would explain why I get to the
voicemail for the correct extension when the phone never rings.

Thanks

=====
 /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml

<include>

  <extension name="GothamCity.xom 1001">
    <condition field="destination_number" expression="^1?(2624481175)$">
       <action application="set" data="domain_name=GothamCity.xom"/>
       <action application="transfer" data="1001 XML default"/>
    </condition>
  </extension>

</include>

=====
/conf/directory/GothamCity.xom.xml


  <domain name="GothamCity.xom

Switch on/off call recording

Ashish gautam — 2013-05-22T05:04:57

Hi,

I have setup a little call center application using mod_callcenter. Now
what I want is to record all the calls based on the click of an ON?OFF
button on the web app. Is there any variable or something like that which I
can set like True or False to switch the call recording on or off?

Also I want to know if the recording is possible per agent or not i.e. the
call session to be recorded for a particular agent.

Please throw some light on this.

Thanks in advance.

Regards
-Ashish
_________________________________________________________________________
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Re: Node.JS ESL libraries

Thomas Lee — 2013-05-22T03:35:14

Hi, 

Please take a look. 

I'm using node.js and esl (https://github.com/shimaore/esl) 

For example, 
-------------------------------------------------- 
// 
// esl-test1.js 
// 

var esl = require('esl'); 
var util = require('util'); 

var audio_file = []; 
// 
// play and get digits 
// <min> <max> <tries> <timeout> <terminators> <file> <invalid_file>
<var_name> <regexp> <digit_timeout>
// 
audio_file[0]='1 1 3 5000 # sounds/mywav/0.wav'; 
audio_file[1]='1 1 3 5000 # sounds/mywav/1.wav'; 
audio_file[2]='1 1 3 5000 # sounds/mywav/2.wav'; 
audio_file[3]='1 1 3 5000 # sounds/mywav/3.wav'; 
audio_file[4]='1 1 3 5000 # sounds/mywav/4.wav'; 
audio_file[5]='1 1 3 5000 # sounds/mywav/5.wav'; 
audio_file[6]='1 1 3 5000 # sounds/mywav/6.wav'; 
audio_file[7]='1 1 3 5000 # sounds/mywav/7.wav'; 
audio_file[8]='1 1 3 5000 # sounds/mywav/8.wav'; 
audio_file[9]='1 1 3 5000 # sounds/mywav/9.wav'; 
audio_file[10]='1 1 3 5000 # sounds/mywav/10.wav'; 
audio_file[11]='1 1 3 5000 # sounds/mywav/11.wav'; 
audio_file[12]='1 1 3

Re: No audio RTP ports available! & I/O Error

Brian Foster — 2013-05-22T03:12:48

Please use http://pastebin.freeswitch.org to copy
/usr/local/freeswitch/conf/switch.conf.xml (if you're on a Linux machine).

- BDF

Thank you,

Brian Foster
Project Manager/Owner's Representative
Davri Investments, Incorporated
P: +1-317-787-2686
M: +1-317-600-9753
Indianapolis, Indiana


On Tue, May 21, 2013 at 7:43 PM, Jagadish Thoutam <jaganthoutam-Re5JQEeQqe8AvxtiuMwx3w< at >public.gmane.org>wrote:

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

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Re: OpenVZ tuning tips

Anthony Minessale — 2013-05-22T03:06:18

You should consider centos6 or debian stable.  Make sure the host kernel is
very new to get maximum results.


On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky <jalsot-Re5JQEeQqe8AvxtiuMwx3w< at >public.gmane.org> wrote:

Re: Cannot ring extension from DID

Philippe Le Toquin — 2013-05-22T01:46:41

your inbound is properly configure (well it is registered at least) so 
it should be able to send any inbound call to your FS server.

As MC explained it much better than I did  I am not going to try to 
confuse you but the magic is in the dialplan!

I have been using FS for some time now but only using it like you to 
make and receive calls so I am not exactly an expert.

One advice I can give you is to read the Wiki 
(http://wiki.freeswitch.org/wiki/Main_Page)
It contains a lot of example and you will most likely find inspiration there

and of course the Freeswitch book is a must to get even more detailed 
information. You can buy it form the FS website(http://www.freeswitch.org/)



On 13-05-21 01:47 PM, Mike Hendrie wrote:

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting-YF8E+gPBBv73h3GqohbjpQ< at >public.gmane.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.

Re: One Way Audio

Sean Devoy — 2013-05-22T01:12:55

Hey Michael,

 

The problem is back and seems to all the time now.

 

Pcapsipdump for a single call is here
http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz 

 

I would like to know why calling FROM FS to a SPRINT phone results in audio
FROM SPRINT, but not to SPRINT.  Reversing the call works every time.

 

Also, any tips on getting started with wireshark to investigate myself next
time would be appreciated.

 

Thanks,

Sean

 

From: freeswitch-users-bounces-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b< at >public.gmane.org
[mailto:freeswitch-users-bounces-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b< at >public.gmane.org] On Behalf Of Michael
Collins
Sent: Friday, May 17, 2013 7:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] One Way Audio

 

It wouldn't be the first time that a computer decided to behave because it
knew Daddy was watching...

-MC

 

On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <sdevoy-4IvgkAfB424yY3YROqfsYA< at >public.gmane.org> wrote:

Thanks MC.  Had to load the pcapdev-lib, but got pcapsi

No audio RTP ports available! & I/O Error

Jagadish Thoutam — 2013-05-21T23:43:43

Hi All,

FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d
2013-04-23 19:49:07Z)

Error :   while error its not accepting calls



o=Sonus_UAC 902944909 319279515 IN IP4 YY.YY.YY.142
s=SIP Media Capabilities
c=IN IP4 YY.YY.YY.131
t=0 0
m=audio 28390 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

2013-05-21 11:09:13.488398 [NOTICE] sofia.c:5799 Pre-Answer
sofia/external/17863565651!
2013-05-21 11:09:13.488398 [DEBUG] switch_channel.c:3273 Send signal
sofia/external/XXXXXXXXXX-Q0ErXNX1RuY0CR83fZKIog< at >public.gmane.org [BREAK]
2013-05-21 11:09:13.488398 [DEBUG] switch_channel.c:3277
(sofia/external/17863565651) Callstate Change DOWN -> EARLY
2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:1720 Set Codec
sofia/external/17863565651 PROXY/8000 20 ms 160 samples 0 bits
2013-05-21 11:09:13.488398 [DEBUG] switch_core_codec.c:111
sofia/external/17863565651 Original read codec set to PROXY:0
2013-05-21 11:09:13.488398 [DEBUG] switch_core_med