Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

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

[asterisk-users] PJSIP question


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





PostPosted: Wed Jun 18, 2014 6:06 am    Post subject: [asterisk-users] PJSIP question Reply with quote

A few months ago I started using and had to abandon PJSIP because my
dialplan could not read the inbound signalling IP address, which I can
read now in Asterisk11 using CHANNEL(recvip). My app relies on this
information. The
question is, is it possible now access the signalling IP of an
incoming SIP call using PJSIP?
Philip

--
_____________________________________________________________________
-- 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 Jun 18, 2014 8:03 am    Post subject: [asterisk-users] PJSIP question Reply with quote

On Wed, Jun 18, 2014 at 6:05 AM, CDR <venefax@gmail.com (venefax@gmail.com)> wrote:
Quote:
A few months ago I started using and had to abandon PJSIP because my
dialplan could not read the inbound signalling IP address, which I can
read now in Asterisk11 using CHANNEL(recvip). My app relies on this
information. The
question is, is it possible now access the signalling IP of an
incoming SIP call using PJSIP?
Philip



The CHANNEL function [1] was integrated with chan_pjsip in the first official release, 12.0.0. You can obtain the address of the remote party using CHANNEL(pjsip,remote_addr).



[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_CHANNEL

--
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
Back to top
venefax at gmail.com
Guest





PostPosted: Thu Jun 26, 2014 11:31 am    Post subject: [asterisk-users] PJSIP question Reply with quote

In a PJSIP endpoint, how do I set all no-named settings so they get
inherited from another place and I don't need to mention them again
and again for all my endpoints?
In regular sip you could specify those options and they remained valid
if not redefined by a peer. A case would be the codecs allowed.
I tried to include those global options in a section called
[global]
disallow=all
allow=ulaw

but the endpoints do not have knowledge of any such global options.

--
_____________________________________________________________________
-- 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
chad at mothersell.net
Guest





PostPosted: Thu Jun 26, 2014 11:49 am    Post subject: [asterisk-users] PJSIP question Reply with quote

You can use templates.
Templates are defines by putting a “(!)” next to the context name.
[template-name](!)
disallow=all
allow=ulaw

Then define the template next to the endpoint in parenthesis.
[endpoint-name](template-name)

Chad

On Jun 26, 2014, at 12:30 PM, CDR <venefax@gmail.com> wrote:

Quote:
In a PJSIP endpoint, how do I set all no-named settings so they get
inherited from another place and I don't need to mention them again
and again for all my endpoints?
In regular sip you could specify those options and they remained valid
if not redefined by a peer. A case would be the codecs allowed.
I tried to include those global options in a section called
[global]
disallow=all
allow=ulaw

but the endpoints do not have knowledge of any such global options.

--
_____________________________________________________________________
-- 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
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