Re: [asterisk-users] dreaded one-way audio with nat=yes
On 03/10/2012 12:13 AM, Vladimir Mikhelson wrote:
Sean,
I do not have experience with the Amazon service. Cannot advise how to
implement it in their environment.
You need to have a route from your public IP(s) to your Asterisk
instance for all incoming connections on RTP ports.
Absence of this routing explains why SIP connection to your home
(egress) worked whereas incoming SIP connection from your SIP provider
(ingress) has a packed drop issue. The egress connection is initiated
from the LAN and firewall happily NATs in this case. On the ingress
connection firewall drops all RTP traffic originated by your provider
while happily NATing the traffic originated by your Asterisk.
It is also a good idea to have qualify=yes in your SIP peers' settings
to keep these NAT tables on the firewall updated for incoming SIP traffic.
-Vladimir
On 3/9/2012 9:15 PM, sean darcy wrote:
On 03/09/2012 09:42 PM, Arstan Jusupov wrote:
Udp port 5060, udp port range 1-2 open? Those are for sip.
For iax2 udp port 4569
Make sure they are open.
Also can you register two ext from the same instance and see if you
can hear both ways
What kind of trunk do you have to the other side you calling?
Arstan
Sent from my iPhone
On Mar 10, 2012, at 10:20 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on
amazon ec2? The instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcyseandar...@gmail.comwrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten =_j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten =_j.,n,Set(3digitexten=${EXTEN:12:3}
exten =_j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten =_j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten =123,1,NoOp()
exten =123,n,Answer()
exten =123,n,Dial(SIP/jnctn/1212xxx)
exten =123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to
SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But
now I get no audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945,
) in new stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
Well that's interesting. I hadn't realized that iptables was set up
on the instance, as well as the firewall from the security group on
the control panel.
Flushed the instance iptables, which fixed a problem I was having
with a phone registering.
But I still have my one-way audio. The calling party hears nothing
from the called party.
sean
The instance firewall is flushed. The security group allows udp
Re: [asterisk-users] dreaded one-way audio with nat=yes
On 03/09/2012 04:16 PM, sean darcy wrote: I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Have you given some thought towards the RTP setup on the Amazon servers ? Whenever I had issues with one way sound, it was RTP ports that were blocked on one side. GG -- _ -- 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
Re: [asterisk-users] dreaded one-way audio with nat=yes
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten = _j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten = _j.,n,Set(3digitexten=${EXTEN:12:3}
exten = _j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten = _j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten = 123,1,NoOp()
exten = 123,n,Answer()
exten = 123,n,Dial(SIP/jnctn/1212xxx)
exten = 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But now I
get no audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945, ) in
new stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
--
_
-- 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
Re: [asterisk-users] dreaded one-way audio with nat=yes
It may sound silly but did you configure/open firewall ports on amazon ec2? The
instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcy seandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten = _j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten = _j.,n,Set(3digitexten=${EXTEN:12:3}
exten = _j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten = _j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten = 123,1,NoOp()
exten = 123,n,Answer()
exten = 123,n,Dial(SIP/jnctn/1212xxx)
exten = 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But now I get
no audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945, ) in new
stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
--
_
-- 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
Re: [asterisk-users] dreaded one-way audio with nat=yes
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on amazon ec2? The
instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten = _j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten = _j.,n,Set(3digitexten=${EXTEN:12:3}
exten = _j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten = _j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten = 123,1,NoOp()
exten = 123,n,Answer()
exten = 123,n,Dial(SIP/jnctn/1212xxx)
exten = 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But now I get no
audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945, ) in new stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
Well that's interesting. I hadn't realized that iptables was set up on
the instance, as well as the firewall from the security group on the
control panel.
Flushed the instance iptables, which fixed a problem I was having with a
phone registering.
But I still have my one-way audio. The calling party hears nothing from
the called party.
sean
--
_
-- 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
Re: [asterisk-users] dreaded one-way audio with nat=yes
Have you looked at rtp debug? Is it possible reinvites are enabled?
On Mar 9, 2012 9:20 PM, sean darcy seandar...@gmail.com wrote:
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on amazon
ec2? The instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten = _j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten = _j.,n,Set(3digitexten=${EXTEN:**12:3}
exten = _j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten = _j.,n,GoTo(from-outside,${**3digitexten},1)
[from-outside]
exten = 123,1,NoOp()
exten = 123,n,Answer()
exten = 123,n,Dial(SIP/jnctn/**1212xxx)
exten = 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to
SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But now I
get no audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945, ) in
new stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
Well that's interesting. I hadn't realized that iptables was set up on the
instance, as well as the firewall from the security group on the control
panel.
Flushed the instance iptables, which fixed a problem I was having with a
phone registering.
But I still have my one-way audio. The calling party hears nothing from
the called party.
sean
--
__**__**_
-- 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-**usershttp://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
Re: [asterisk-users] dreaded one-way audio with nat=yes
Udp port 5060, udp port range 1-2 open? Those are for sip.
For iax2 udp port 4569
Make sure they are open.
Also can you register two ext from the same instance and see if you can hear
both ways
What kind of trunk do you have to the other side you calling?
Arstan
Sent from my iPhone
On Mar 10, 2012, at 10:20 AM, sean darcy seandar...@gmail.com wrote:
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on amazon ec2?
The instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten = _j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten = _j.,n,Set(3digitexten=${EXTEN:12:3}
exten = _j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten = _j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten = 123,1,NoOp()
exten = 123,n,Answer()
exten = 123,n,Dial(SIP/jnctn/1212xxx)
exten = 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But now I
get no audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945, ) in new
stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
Well that's interesting. I hadn't realized that iptables was set up on the
instance, as well as the firewall from the security group on the control
panel.
Flushed the instance iptables, which fixed a problem I was having with a
phone registering.
But I still have my one-way audio. The calling party hears nothing from the
called party.
sean
--
_
-- 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
Re: [asterisk-users] dreaded one-way audio with nat=yes
On 03/09/2012 09:42 PM, Arstan Jusupov wrote:
Udp port 5060, udp port range 1-2 open? Those are for sip.
For iax2 udp port 4569
Make sure they are open.
Also can you register two ext from the same instance and see if you can hear
both ways
What kind of trunk do you have to the other side you calling?
Arstan
Sent from my iPhone
On Mar 10, 2012, at 10:20 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on amazon ec2? The
instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten = _j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten = _j.,n,Set(3digitexten=${EXTEN:12:3}
exten = _j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten = _j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten = 123,1,NoOp()
exten = 123,n,Answer()
exten = 123,n,Dial(SIP/jnctn/1212xxx)
exten = 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But now I get no
audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945, ) in new stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
Well that's interesting. I hadn't realized that iptables was set up on the
instance, as well as the firewall from the security group on the control panel.
Flushed the instance iptables, which fixed a problem I was having with a phone
registering.
But I still have my one-way audio. The calling party hears nothing from the
called party.
sean
The instance firewall is flushed. The security group allows udp
1-2 , 5060 and 4569.
Well it gets stranger:
I set up a sip link to my home. Dialed the teliax number from my cell.
Asterisk used the sip link to my home - and that worked!
Dial(IAX2/iaxtest-584, SIP/sip-to-home)
Which seems to mean that the teliax - asterisk link is fine.
But if I use a SIP/PSTN provider , I get one-way audio:
Dial(IAX2/iaxtest-515, SIP/jnctn/home-pstn)
Completely baffled.
sean
--
_
-- 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
Re: [asterisk-users] dreaded one-way audio with nat=yes
Sean,
I do not have experience with the Amazon service. Cannot advise how to
implement it in their environment.
You need to have a route from your public IP(s) to your Asterisk
instance for all incoming connections on RTP ports.
Absence of this routing explains why SIP connection to your home
(egress) worked whereas incoming SIP connection from your SIP provider
(ingress) has a packed drop issue. The egress connection is initiated
from the LAN and firewall happily NATs in this case. On the ingress
connection firewall drops all RTP traffic originated by your provider
while happily NATing the traffic originated by your Asterisk.
It is also a good idea to have qualify=yes in your SIP peers' settings
to keep these NAT tables on the firewall updated for incoming SIP traffic.
-Vladimir
On 3/9/2012 9:15 PM, sean darcy wrote:
On 03/09/2012 09:42 PM, Arstan Jusupov wrote:
Udp port 5060, udp port range 1-2 open? Those are for sip.
For iax2 udp port 4569
Make sure they are open.
Also can you register two ext from the same instance and see if you
can hear both ways
What kind of trunk do you have to the other side you calling?
Arstan
Sent from my iPhone
On Mar 10, 2012, at 10:20 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on
amazon ec2? The instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten = _j.,1,NoOp(From teliax sip with exten ${EXTEN})
exten = _j.,n,Set(3digitexten=${EXTEN:12:3}
exten = _j.,n,NoOp(Callerid is ${CALLERID(all)} )
exten = _j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten = 123,1,NoOp()
exten = 123,n,Answer()
exten = 123,n,Dial(SIP/jnctn/1212xxx)
exten = 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
DirectMedia : No
And the cli doesn't show any problems:
NoOp(SIP/teliax-0022, From teliax sip with exten
somename12lg(123)) in new stack
Set(SIP/teliax-0022, 3digitexten=123) in new stack
NoOp(SIP/teliax-0022, Callerid is ) in new stack
Goto(SIP/teliax-0022, from-outside,123,1) in new stack
-- Goto (from-outside,123,1)
NoOp(SIP/teliax-0022, ) in new stack
Answer(SIP/teliax-0022, ) in new stack
Dial(SIP/teliax-0022, SIP/jnctn/1212aaa) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to
SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
So I tried having teliax connect to the asterisk box with iax. But
now I get no audio both ways!
Answer(IAX2/iaxtest-1945, ) in new stack
GotoIf(IAX2/iaxtest-1945, 1?123,1) in new stack
-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp(IAX2/iaxtest-1945,
) in new stack
-- Executing [123@from-outside:2] Dial(IAX2/iaxtest-1945,
SIP/jnctn/1aaabbb) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945
Really puzzled.
sean
Well that's interesting. I hadn't realized that iptables was set up
on the instance, as well as the firewall from the security group on
the control panel.
Flushed the instance iptables, which fixed a problem I was having
with a phone registering.
But I still have my one-way audio. The calling party hears nothing
from the called party.
sean
The instance firewall is flushed. The security group
