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darcy at Vex.Net
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 Posted: Tue May 13, 2014 6:02 am Post subject: [asterisk-users] Ringing issue |
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I have an issue with ringing. Some users who call my switch hear
ringing and others don't. I have researched this and understand the
issue of firewalling and RTP. My switch has UDP ports 10000 to 20000
open. In any case I think that blocked RTP would block all ringing,
not just some.
I have one origination provider. As far as I can tell the issue is
related to the remote user's provider. My sister does not hear ringing
when she calls from her Roger's cell phone but she does from her Vonage
phone. I hear ringing when calling in from my Koodo cell phone. Some
land lines work and others do not.
The server is not behind a NAT and neither is the origination
provider. There is a firewall but port 5060 is open (UDP and, just in
case, TCP) as well as the RTP ports mentioned above.
I am not sure where to look next. I assume that there is some sort of
signaling that I am not doing but I can't figure out where. Can anyone
suggest what area I should be looking?
Thanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
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mailinglist+asterisk a...
Guest
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| Posted: Tue May 13, 2014 9:28 am Post subject: [asterisk-users] Ringing issue |
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You would need to provide more information. Mobiles and landlines are
not SIP and yet you say calls are coming into your asterisk over SIP. So
what or who is doing the translation?
Initial thoughts are that it could be you are sending back SIP/180 with
no session progress and indicating ringing but the other end is
misconfiguration and not generating its own ring tone. This is possible
if you have multiple providers sending you calls or one provider using
different kit for different geographic areas.
On 13/05/14 12:01, D'Arcy J.M. Cain wrote:
| Quote: | I have an issue with ringing. Some users who call my switch hear
ringing and others don't. I have researched this and understand the
issue of firewalling and RTP. My switch has UDP ports 10000 to 20000
open. In any case I think that blocked RTP would block all ringing,
not just some.
I have one origination provider. As far as I can tell the issue is
related to the remote user's provider. My sister does not hear ringing
when she calls from her Roger's cell phone but she does from her Vonage
phone. I hear ringing when calling in from my Koodo cell phone. Some
land lines work and others do not.
The server is not behind a NAT and neither is the origination
provider. There is a firewall but port 5060 is open (UDP and, just in
case, TCP) as well as the RTP ports mentioned above.
I am not sure where to look next. I assume that there is some sort of
signaling that I am not doing but I can't figure out where. Can anyone
suggest what area I should be looking?
Thanks.
|
--
_____________________________________________________________________
-- 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 May 14, 2014 7:02 am Post subject: [asterisk-users] Ringing issue |
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On Tue, 13 May 2014 15:28:26 +0100
Gareth Blades <mailinglist+asterisk@dns99.co.uk> wrote:
| Quote: | You would need to provide more information. Mobiles and landlines are
not SIP and yet you say calls are coming into your asterisk over SIP.
So what or who is doing the translation?
|
My origination provider. While I do have a SIP address, no one is
calling it and other than local sets (which don't seem to have this
issue) all calls are coming through my single origination provider.
This is why I am confused. Virtually all calls are coming from the
PSTN through one connection. If all callers had the problem it would
almost make more sense.
| Quote: | Initial thoughts are that it could be you are sending back SIP/180
with no session progress and indicating ringing but the other end is
misconfiguration and not generating its own ring tone. This is
possible if you have multiple providers sending you calls or one
provider using different kit for different geographic areas.
|
Geographic doesn't seem to be the issue. Most calls are coming from
Toronto, Canada where I am. They come from major carriers. Rogers is
the largest cell carrier here and that appears to be one place where it
fails. I am on Koodo which uses the Telus network, the second largest,
and mine works fine.
--
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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darcy at Vex.Net
Guest
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| Posted: Wed May 14, 2014 10:15 am Post subject: [asterisk-users] Ringing issue |
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|
On Tue, 13 May 2014 15:28:26 +0100
Gareth Blades <mailinglist+asterisk@dns99.co.uk> wrote:
| Quote: | Initial thoughts are that it could be you are sending back SIP/180
with no session progress and indicating ringing but the other end is
misconfiguration and not generating its own ring tone. This is
possible if you have multiple providers sending you calls or one
provider using different kit for different geographic areas.
|
I seem to have solved this, sorta. My Provider, Thinktel in Canada,
normally sets "PBX plays ringback" to false meaning that they generate
the ring tone in all cases. By mistake it was set to true on my
trunk. They changed that and now the callers are hearing a ring tone.
It's still an interesting question I think. What if I wanted to do
something with early media? That is not possible with this setup.
Anyway, here it is for future searchers. Talk to your origination
provider if you have this problem.
--
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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