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cursor at telecomabmex...
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 Posted: Wed Sep 09, 2015 4:11 pm Post subject: [asterisk-users] No ring sound when calling SIP extensions o |
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I am having a small problem that is driving me nuts. I can make
calls over my Webrtc client without any problems and audio sounds fine.
The only problem I have is that when I call an internal SIP extension on
my PBX I do not hear the ring while I wait for the call to be answered.
My dial command does include the rR options. If I make an external call
to a land line or a mobile phone I do hear the ring sounds, only
internal extensions have this problem. Why would the webrtc client
ignore the ringing when calling another SIP extension? Any ideas?
We are using Asterisk 13.4 on CentOS 7 with SIP.js
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
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darcy at Vex.Net
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| Posted: Wed Sep 09, 2015 4:22 pm Post subject: [asterisk-users] No ring sound when calling SIP extensions o |
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On Wed, 9 Sep 2015 16:11:03 -0500
Carlos Chavez <cursor@telecomabmex.com> wrote:
| Quote: | I am having a small problem that is driving me nuts. I can make
calls over my Webrtc client without any problems and audio sounds
fine. The only problem I have is that when I call an internal SIP
extension on my PBX I do not hear the ring while I wait for the call
to be answered. My dial command does include the rR options. If I
make an external call to a land line or a mobile phone I do hear the
ring sounds, only internal extensions have this problem. Why would
the webrtc client ignore the ringing when calling another SIP
extension? Any ideas?
|
I had a similar problem. Turned out that my indications.conf file was
empty. Check that out.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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cursor at telecomabmex...
Guest
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| Posted: Wed Sep 09, 2015 4:27 pm Post subject: [asterisk-users] No ring sound when calling SIP extensions o |
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On 9/9/15 4:22 PM, D'Arcy J.M. Cain wrote:
| Quote: | On Wed, 9 Sep 2015 16:11:03 -0500
Carlos Chavez <cursor@telecomabmex.com> wrote:
| Quote: | I am having a small problem that is driving me nuts. I can make
calls over my Webrtc client without any problems and audio sounds
fine. The only problem I have is that when I call an internal SIP
extension on my PBX I do not hear the ring while I wait for the call
to be answered. My dial command does include the rR options. If I
make an external call to a land line or a mobile phone I do hear the
ring sounds, only internal extensions have this problem. Why would
the webrtc client ignore the ringing when calling another SIP
extension? Any ideas?
|
I had a similar problem. Turned out that my indications.conf file was
empty. Check that out.
|
The file is full of definitions for many countries. It specifically has
one for Mexico but we usually use the same one as the USA.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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asandovalros at gmail.com
Guest
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| Posted: Wed Sep 09, 2015 4:31 pm Post subject: [asterisk-users] No ring sound when calling SIP extensions o |
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Could you please share your Asterisk files configuration? I'm running on problems with audio when calling from/to a webrtc script (jsSIP) and Asterisk 12.
I'm using an extension attached to a SIP trunk and my calls do fine but there is no audio on any side.
--
Enviado desde la aplicación myMail para Android miércoles, 09 septiembre 2015, 04:27p.m. -05:00 de Carlos Chavez :
| Quote: | On 9/9/15 4:22 PM, D'Arcy J.M. Cain wrote:
| Quote: | On Wed, 9 Sep 2015 16:11:03 -0500
Carlos Chavez <[url=/compose?To=cursor@telecomabmex.com]cursor@telecomabmex.com[/url]> wrote:
| Quote: | I am having a small problem that is driving me nuts. I can make
calls over my Webrtc client without any problems and audio sounds
fine. The only problem I have is that when I call an internal SIP
extension on my PBX I do not hear the ring while I wait for the call
to be answered. My dial command does include the rR options. If I
make an external call to a land line or a mobile phone I do hear the
ring sounds, only internal extensions have this problem. Why would
the webrtc client ignore the ringing when calling another SIP
extension? Any ideas?
|
I had a similar problem. Turned out that my indications.conf file was
empty. Check that out.
|
The file is full of definitions for many countries. It specifically has
one for Mexico but we usually use the same one as the USA.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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darcy at Vex.Net
Guest
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| Posted: Wed Sep 09, 2015 6:13 pm Post subject: [asterisk-users] No ring sound when calling SIP extensions o |
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On Wed, 9 Sep 2015 16:27:19 -0500
Carlos Chavez <cursor@telecomabmex.com> wrote:
| Quote: | The file is full of definitions for many countries. It specifically
has one for Mexico but we usually use the same one as the USA.
|
I simplified mine:
[general]
country=us ; default location
[us]
description = United States / North America
ringcadence = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/10000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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