To sort out RTP problems, I would recommend:
1) on all endpoints use codec of allow=!all,ulaw -- this is or should be
supported by all endpoints and eliminates any issues of mismatch,
translation, etc., and can be adjusted later once everything is working
2) add an Echo() application to your di
] Asterisk 13.1.0/PJSIP peer IP address issue [Spam
score:10%]
Well, I thought it worked, but it actually doesn't--I am able to get the caller
pick up the phone, but for some reason, I cannot hear anything on either side
no matter who does the calling. Again, my two SIP phones a
Solved!
The issue was that RTP flows were being established to the wrong IP address.
I figured out this issue--I had to disable STUN in both SIP phones for this
to work correctly.
Still, I wish a working configuration for Asterisk, and two SIP phones in
the same 192.168.1.0/24 network would have
Well, I thought it worked, but it actually doesn't--I am able to get the
caller pick up the phone, but for some reason, I cannot hear anything on
either side no matter who does the calling. Again, my two SIP phones are on
the local 192.168.1.0/24 network (do not go over the Internet) and the
Asteri
Thank you for your note, Scott.
I set rewrite_contact=yes for both contacts, and I also had to do
remove_existing=yes because I had to remove the existing contact
information (max_contacts = 1 was preventing new contact information)
using pjsip
qualify demo-alice etc., after which the right IP add
I am following the instructions in
https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am
trying to make a call from extension Alice (6001) to extension for Bob
(6002). When I make the call, I can hear the ringing on Alice's phone
(caller), but Bob's phone (callee) doesn't ring
It would appear that you have the Asterisk server on a public IP address,
your two endpoints are behind a NAT, and you have rewrite_contact enabled
in pjsip.conf.
In which case, what you are seeing is correct. In order to be able to send
a call to an extension where it is behind NAT, Asterisk mus
