Displaying 20 results from an estimated 62 matches for "rtp_symmetric".
2016 Mar 03
3
RTP / NAT question ( pjsip )
Thank you for the response Joshua .
I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP.
tcpdump on asterisk server showing UDP packet bound for my remote endpo...
2015 Sep 22
2
How to set the global setting for each pjsip endpoint
...xample
I have two endpoints and I repeat the same thing,
[100]
type=endpoint
aors=100
auth=100-auth
allow=ulaw,alaw,gsm,g726
context=from-internal
callerid=device <100>
dtmf_mode=rfc4733
use_avpf=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
message_context=astsms
[200]
type=endpoint
aors=200
auth=200-auth
allow=ulaw,alaw,gsm,g726
context=from-internal
callerid=device <200>
dtmf_mode=rfc4733
use_avpf=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
send_pai=yes
rtp_sy...
2019 Jan 15
2
(NAT) direct media to host on local net when registering from external address
On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer.
> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?
That would require using chan_pjsip wouldn't it? Not that I am opposed
to trying that. I need to use chan_pjsip at some point to be able to
authenticate to my SIP provider for SIP SIMPLE a...
2015 Mar 05
2
PJSIP configuration for AWS/EC2 based Asterisk 13.1.0
Hello All,
I have an Asterisk server v13.1.0 running on EC2 and I am able to connect
and register SIP devices and "see" them on the asterisk CLI. I am also able
to place calls, but I am not able to hear any audio on either end after the
call is picked up.
I was wondering if you can tell me what a minimal configuration for
Asterisk on EC2 looks like. My current pjsip.conf configuration
2016 Sep 08
3
PJSIP Weirdness, or just my weirdness?
...auth_type=userpass
username=murftest12
password=SjU3
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:57969
[murftest12] ; Cisco SPA514G mac=A4:93:4C:FE:1D:A2
type=endpoint
auth=murftest12
transport=transport-udp
aors=murftest12
moh_suggest=default
force_rport=yes
rewrite_contact=yes
rtp_symmetric=yes
dtmf_mode=rfc4733
disallow=all
allow=ulaw ; from phonetype
allow=g722 ; from phonetype
allow=alaw ; from phonetype
allow=alaw ; from phonetype (G.729 replaced with alaw)
direct_media=no
context=phone
rtp_timeout=120
set_var=__phoneid=12
set_var=__contacttypeid=4
set_var=__phonelineid=78
calleri...
2014 Dec 16
3
PJSIP configuration question
...gnaling_address=<your public address>*
[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93
[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no
*from_user=<your main vitelity account name> ; Not subaccount*
[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93
-------------- next pa...
2015 Sep 22
2
How to set the global setting for each pjsip endpoint
...xt=from-internal
>>
>> callerid=device <100>
>>
>> dtmf_mode=rfc4733
>>
>> use_avpf=no
>>
>> ice_support=no
>>
>> media_use_received_transport=no
>>
>> trust_id_inbound=yes
>>
>> send_pai=yes
>>
>> rtp_symmetric=yes
>>
>> rewrite_contact=yes
>>
>> message_context=astsms
>>
>>
>> [200]
>>
>> type=endpoint
>>
>> aors=200
>>
>> auth=200-auth
>>
>> allow=ulaw,alaw,gsm,g726
>>
>> context=from-internal
>>...
2019 Jan 15
2
(NAT) direct media to host on local net when registering from external address
This is going to be a bit of an odd situation, but perhaps might become
more and more common (as mobile phone SIP clients utilize PUSH proxies
instead of the battery draining direct registering with SIP servers).
I have a SIP client which can be on the same RFC-1918 LAN as my
Asterisk server. Even though it's on the same LAN as the Asterisk
server, it's registration comes from an IP
2015 Jul 08
6
tls on asterisk 13
...y config
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
method=tlsv1
[XXXX]
type=endpoint
context=XX-Xip
disallow=all
allow=ulaw
allow=alaw
transport=transport-tls
direct_media=no
force_rport=yes
rtp_symmetric=yes
mailboxes=XXXX at default
auth=XXXX
aors=XXXX
media_encryption=sdes
dtmfmode=rfc4733
regardss
--
rickygm
http://gnuforever.homelinux.com
2017 Oct 09
6
PJSIP, NAT and STUN/ICE
...k, I had to deal with NAT. The Asterisk config
object of type=transport knows about essential entries:
local_net= 192.168.254.1/24
bind= 192.168.254.1:5060
external_media_address= dyndns FQDN
external_signaling_address= dyndns FQDN
direct_media= no
rtp_symmetric= yes
force_rport= yes
dyndns FQDN is the FQDN of my broadband access point provided by some dynamical DNS
provider.
This setup is not working properly with when external_media_address= and
external_signaling_address= are set that way, but commenting out both makes all of the
ITSP which provid...
2018 Feb 08
3
pjsip trunking configuration issue
...le=cert_file
priv_key_file=key_file
method=tlsv1
external_media_address=X.Y.Z.D
external_signaling_address=X.Y.Z.D
verify_client=no
verify_server=no
allow_reload=yes
[twilio](!)
type=endpoint
transport=transport-tls
context=from-twilio
disallow=all
allow=ulaw
dtmf_mode=inband
media_encryption=sdes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
canreinvite=no
tlsdontverifyserver=yes
[auth-out](!)
type=auth
auth_type=userpass
[twilio]
aors=twilio-aors
[twilio-aors]
type=aor
contact=sips:trunkname.pstn.twilio.com:5061 ;tried with sip: also
[twilio]
type=identify
endpoint=twilio
match=54.172.60.0...
2015 Mar 04
1
PJSIP works on UDP but not TCP
...My transport looks like this. My box is not behind NAT.
[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5061
My endpoint looks like this:
[user1]
type=endpoint
transport=transport-tcp
context=local_out
disallow=all
allow=alaw
allow=ulaw
allow=g722
auth=user1
aors=user1
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
[user1]
type=auth
auth_type=userpass
password=123456
username=user1
[user1]
type=aor
remove_existing=yes
max_contacts=1
I have two endpoints user1 and user 2. Both are able to register fine.
With both endpoints I can call into asterisk and do an echo test...
2015 Mar 06
0
PJSIP configuration for AWS/EC2 based Asterisk 13.1.0
OK. I think I found the issue.
The key is to add
rtp_symmetric=yes
Here's what my final configuration looks like:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
;; for within EC2
local_net=172.31.32.0/20
;; For softphones within EC2
local_net=192.168.1.0/24
external_media_address=<publicIPOfEC2Instance>
external_signaling_address...
2015 Mar 09
1
PJSIP and Kamailio without registration
...bit stumped, I've tried everything I could think of, even configuring
everything to work on the public IP, but no luck.
Here's my PJSIP conf:
[kamailio]
type=endpoint
transport=transport-udp
context=from_kamailio
disallow=all
allow=alaw
allow=g722
allow=ulaw
aors=kamailio
direct_media=no
rtp_symmetric=no
force_rport=no
rewrite_contact=no
[kamailio]
type=identify
endpoint=kamailio
match=xxx.xxx.xxx.xxx (removed kamailios private IP)
[kamailio-mars]
type=aor
contact=sip:xxx.xxx.xxx.xxx:5060 (removed kamailios private IP).
My end goal is for all my phones to register to Kamailio. Kamailio shoul...
2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
type=endpoint
rewrite_contact=yes
force_rport=yes
rtp_symmetric=yes
On 6/21/23 14:36, TTT wrote:
> I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:
>
> From: "MYNAME" <sip:16667778888 a...
2014 Dec 16
1
PJSIP configuration question
...ort
bind = 0.0.0.0
protocol = udp
[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:outbound.vitelity.net
[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no
[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93
Have a great day!
Dan
2016 Sep 09
2
Asterisk 13 PJSIP with Snom 710
...on the phone and
asterisk CLI also does not show anything. my config is. please advice.
[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm
[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes
[2001]
type=auth...
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
...0-auth
>
> allow=ulaw,alaw,gsm,g726
>
> context=from-internal
>
> callerid=device <100>
>
> dtmf_mode=rfc4733
>
> use_avpf=no
>
> ice_support=no
>
> media_use_received_transport=no
>
> trust_id_inbound=yes
>
> media_encryption=no
>
> rtp_symmetric=yes
>
> rewrite_contact=yes
>
> *message_context=astsms*
>
>
>
> On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> Hello,
>>
>> I am looking for documentation support for enabling instant messagin...
2014 Dec 15
2
PJSIP configuration question
...t; qualify_frequency = 60
>
> contact = sip:64.2.142.93
>
>
>
> [outbound.vitelity.net]
>
> type = endpoint
>
> context = TestApp
>
> transport = transport1
>
> aors = outbound.vitelity.net
>
> dtmf_mode = rfc4733
>
> force_rport = yes
>
> rtp_symmetric = yes
>
> rewrite_contact = yes
>
> send_rpid = yes
>
> trust_id_inbound = yes
>
> disallow = all
>
> allow = ulaw
>
> direct_media = no
>
>
>
> [outbound.vitelity.net]
>
> type = identify
>
> endpoint = outbound.vitelity.net
>
> mat...
2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
...ne 21, 2023 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; TTT <lists at telium.io>
Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE
type=endpoint
rewrite_contact=yes
force_rport=yes
rtp_symmetric=yes
On 6/21/23 14:36, TTT wrote:
> I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:
>
> From: "MYNAME" <sip:16667778888 a...