Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] How to read RTP ports from CLI ?


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
oza.4h07 at gmail.com
Guest





PostPosted: Tue Aug 12, 2014 9:20 am    Post subject: [asterisk-users] How to read RTP ports from CLI ? Reply with quote

Hello,

How can I read RTP ports from CLI (to double check what could be
included in rtp.conf file) ?
"sip show settings" do not provide the answer.

Regards

--
_____________________________________________________________________
-- 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
Back to top
mikael at wiraya.com
Guest





PostPosted: Wed Aug 13, 2014 8:39 am    Post subject: [asterisk-users] How to read RTP ports from CLI ? Reply with quote

On 12 August 2014 16:19, Olivier <oza.4h07@gmail.com (oza.4h07@gmail.com)> wrote: 
Quote:
How can I read RTP ports from CLI (to double check what could be
included in rtp.conf file) ?
"sip show settings" do not provide the answer.


On way would be to activate SIP debugging:

sip set debug on


Then check the INVITE body/SDP for port on a row that looks like this:
m=audio 11374 RTP/AVP 8 101


Where 11374 would be the port.


/Mikael Fredin
Back to top
oza.4h07 at gmail.com
Guest





PostPosted: Wed Aug 13, 2014 9:30 am    Post subject: [asterisk-users] How to read RTP ports from CLI ? Reply with quote

2014-08-13 15:38 GMT+02:00 Mikael Fredin <mikael@wiraya.com>:
Quote:
On 12 August 2014 16:19, Olivier <oza.4h07@gmail.com> wrote:
Quote:

How can I read RTP ports from CLI (to double check what could be
included in rtp.conf file) ?
"sip show settings" do not provide the answer.


On way would be to activate SIP debugging:
sip set debug on

Then check the INVITE body/SDP for port on a row that looks like this:
m=audio 11374 RTP/AVP 8 101

Where 11374 would be the port.

/Mikael Fredin

Sure but what I'm looking for is to:
- type something like "rtp show settings"
- and read something like : Port range 10000-20000


Quote:

--
_____________________________________________________________________
-- 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

--
_____________________________________________________________________
-- 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
Back to top
mjordan at digium.com
Guest





PostPosted: Wed Aug 13, 2014 12:08 pm    Post subject: [asterisk-users] How to read RTP ports from CLI ? Reply with quote

On Wed, Aug 13, 2014 at 9:29 AM, Olivier <oza.4h07@gmail.com> wrote:
Quote:
2014-08-13 15:38 GMT+02:00 Mikael Fredin <mikael@wiraya.com>:
Quote:
On 12 August 2014 16:19, Olivier <oza.4h07@gmail.com> wrote:
Quote:

How can I read RTP ports from CLI (to double check what could be
included in rtp.conf file) ?
"sip show settings" do not provide the answer.


On way would be to activate SIP debugging:
sip set debug on

Then check the INVITE body/SDP for port on a row that looks like this:
m=audio 11374 RTP/AVP 8 101

Where 11374 would be the port.

/Mikael Fredin

Sure but what I'm looking for is to:
- type something like "rtp show settings"
- and read something like : Port range 10000-20000

That information is not available via a CLI command.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- 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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services