VoIP Mailing List Archives :: View topic - [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem

Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community

[asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem

VoIP Mailing List Archives Forum Index -> Asterisk Users

View previous topic :: View next topic

Author

Message

rainer.piper at soho-p...
Guest

Posted: Wed May 07, 2014 12:11 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

PS.

if I configure both extension 7000 and 7001 to,

disallow=all
allow=alaw
or
disallow=all
allow=g722

everything is fine. as long as the allowed codec is equal in both extensions.

Am 07.05.2014 07:00, schrieb Rainer Piper:

Quote:

Hi!

my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides.

I compiled pjsip with no resample in pjsip.

Quote:

./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr

is there a way to force asterisk back to do the codec translation?

Attachment:
sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw)

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
www.soho-piper.de

rainer.piper at soho-p...
Guest

Posted: Wed May 07, 2014 12:35 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

that's funny

I recompiled asterisk without bridge_native_rtp.so
to force asterisk to go to simple_bridge and not to native_bridge...

!!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu

Am 07.05.2014 07:11, schrieb Rainer Piper:

Quote:

Hi!

my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides.

I compiled pjsip with no resample in pjsip.

Quote:

./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
www.soho-piper.de

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
www.soho-piper.de NOC +49 228 97167161 - sip.soho-piper.de
NOC +882 990111550 via e164.org International Network NOC +49 2247 9064188 - sip.tefonix.de - D293
NOC +882 990045450 via e164.org International Network

rainer.piper at soho-p...
Guest

Posted: Wed May 07, 2014 12:58 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

perhaps a silly question ...

if a channel switches from simple_bridge to native_bridge ... is the channel switching to direct_media between the endpoints ?

if so, why doesn't turn direct_media = no and disable_direct_media_on_nat = yes switching to native_bridge off ?

my pjsip.conf endpoint 7000 and 7001

[7000]
type=endpoint
context=outgoing
disallow=all
allow=alaw,ulaw,g722
transport=transport-udp
auth=auth7000
aors=7000
direct_media = no
disable_direct_media_on_nat = yes

[auth7000]
type=auth
auth_type=userpass
password=xxxxxxxxxxxxxxxxxxxxxxxxx
username=7000

[7000]
type=aor
max_contacts=10
qualify_frequency=60

[7001]
type=endpoint
context=outgoing
disallow=all
allow=g722,alaw,ulaw
transport=transport-udp
auth=auth7001
aors=7001
direct_media = no
disable_direct_media_on_nat = yes

[auth7001]
type=auth
auth_type=userpass
password=xxxxxxxxxxxxxxxxxxxxxxxxx
username=7001

[7001]
type=aor
max_contacts=10
qualify_frequency=60

Am 07.05.2014 07:35, schrieb Rainer Piper:

Quote:

Hi!

my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides.

I compiled pjsip with no resample in pjsip.

Quote:

./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
www.soho-piper.de

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161

jcolp at digium.com
Guest

Posted: Wed May 07, 2014 5:36 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

Rainer Piper wrote:

Quote:

The bridge_native_rtp module can actually "native bridge" in two ways:

1. Media directly between both sides
2. Media within the RTP stack

Even with NAT #2 can still operate fine as media still goes through
Asterisk, just not as much.

As for your issue I would suggest you get a complete console log output
with debug and create an issue[1] as this sounds like a bug.

Cheers,

[1] https://issues.asterisk.org/jira

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.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

rainer.piper at soho-p...
Guest

Posted: Wed May 07, 2014 6:08 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

Hi Joshua,

I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. Wink

wow ... early bird it is 03:36 (PDT) in the morning at your place

Thanks!

Rainer

Am 07.05.2014 12:36, schrieb Joshua Colp:

Quote:

Rainer Piper wrote:

Quote:

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161

jcolp at digium.com
Guest

Posted: Wed May 07, 2014 6:10 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

Rainer Piper wrote:

Quote:

Hi Joshua,

I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. Wink

wow ... early bird it is 03:36 (PDT) in the morning at your place

The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada
though where it is 8:09AM. Not toooo early.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.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

rainer.piper at soho-p...
Guest

Posted: Wed May 07, 2014 6:14 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

and I get ready for launch in germany at 13:15 Wink

Am 07.05.2014 13:09, schrieb Joshua Colp:

Quote:

Rainer Piper wrote:

Quote:

Hi Joshua,

I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. Wink

wow ... early bird it is 03:36 (PDT) in the morning at your place

The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada though where it is 8:09AM. Not toooo early.

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161

rainer.piper at soho-p...
Guest

Posted: Wed May 07, 2014 6:18 am Post subject: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation

upps ... off topic

and typo lunch not launch Wink

Am 07.05.2014 13:14, schrieb Rainer Piper:

Quote:

and I get ready for launch in germany at 13:15 Wink

Am 07.05.2014 13:09, schrieb Joshua Colp:

Quote:

Rainer Piper wrote:

Quote:

Hi Joshua,

I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. Wink

wow ... early bird it is 03:36 (PDT) in the morning at your place

The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada though where it is 8:09AM. Not toooo early.

--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161

Display posts from previous:

	VoIP Mailing List Archives Forum Index -> Asterisk Users	All times are GMT - 5 Hours
Page 1 of 1

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