Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Jerry Geis
Thanks for the information
This is now working...

externip=EC2 public IP
localnet=EC2 local range
nat=force_rport,comedia

I got audio, Fantastic

Jerry

>
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Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Antony Stone
On Thursday 06 October 2022 at 15:24:22, Jerry Geis wrote:

> I added:
> 
> externip=xxx
> nat=force_rport,comedia
> 
> to the general section of sip.conf
> 
> its still sending to the local IP.

Does your local router (the one connecting Linphone to the Internet) have a 
"SIP helper" or "SIP ALG" feature?

If so, ensure that it is turned off.

Antony.

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meaning Christmas tree decorations, and is not a quote from Linus Torvalds.

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Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Joshua C. Colp
On Thu, Oct 6, 2022 at 10:24 AM Jerry Geis  wrote:

> >The sample configuration file outlines how things work, and the options for
> >it:
> >https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874
> >in general localnet and externip (or externaddr, or externhost)
>
> I added:
>
> externip=xxx
> nat=force_rport,comedia
>
> to the general section of sip.conf
>
> its still sending to the local IP.
>

Look at the actual SIP/SDP signaling to see what is being sent and what IP
addresses are being used. If they're correct, then see if you are receiving
traffic using "rtp set debug on". If you aren't then it's something outside
of Asterisk preventing incoming traffic. Until Asterisk receives traffic it
can't know the IP address+port to send outgoing to, beyond what was given
in the SIP/SDP.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Jerry Geis
>The sample configuration file outlines how things work, and the options for
>it:
>https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874
>in general localnet and externip (or externaddr, or externhost)

I added:

externip=xxx
nat=force_rport,comedia

to the general section of sip.conf

its still sending to the local IP.

Jerry
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Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Joshua C. Colp
On Thu, Oct 6, 2022 at 10:17 AM Jerry Geis  wrote:

>
>
> On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis  wrote:
>
>>
>>
>> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis  wrote:
>>
>>> I am trying to get audio to work on AWS using asterisk 18.14.0
>>>
>>> I have enabled the firewall to allow ALL UDP on AWS
>>>
>>> My SIP extension has
>>> nat=force_rport,comedia
>>> qualify=yes
>>> allow=ulaw
>>> allow=alaw
>>> allow=gsm
>>> canreinvite=yes
>>>
>>> I enable "rtp set debug on" and the console is printing info.
>>>
>>> The call comes into my linphone softphone - but I get no audio on my
>>> linphone softphone.
>>> What might I be missing to allow the audio ?
>>> Volume is up.
>>>
>>> Thanks
>>>
>>> Jerry
>>>
>>
>>
>> I just noticed the RTP log is sending to 192.168.2.0 which is my local
>> lan address of the linphone - it should be sending to the NAT address and
>> is not.
>> What did I not set correctly ?
>> I am not using pjsip - but the older asterisk.
>>
>> Thanks
>>
>> Jerry
>>
>
>  >Have you configured chan_sip to know it is behind NAT itself and what its
> >public IP address is? If not, then you'll get no audio.
>
> I'm thinking I have not. What did I miss ?
>

The sample configuration file outlines how things work, and the options for
it:
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874
in general localnet and externip (or externaddr, or externhost)

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Jerry Geis
On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis  wrote:

>
>
> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis  wrote:
>
>> I am trying to get audio to work on AWS using asterisk 18.14.0
>>
>> I have enabled the firewall to allow ALL UDP on AWS
>>
>> My SIP extension has
>> nat=force_rport,comedia
>> qualify=yes
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>> canreinvite=yes
>>
>> I enable "rtp set debug on" and the console is printing info.
>>
>> The call comes into my linphone softphone - but I get no audio on my
>> linphone softphone.
>> What might I be missing to allow the audio ?
>> Volume is up.
>>
>> Thanks
>>
>> Jerry
>>
>
>
> I just noticed the RTP log is sending to 192.168.2.0 which is my local lan
> address of the linphone - it should be sending to the NAT address and is
> not.
> What did I not set correctly ?
> I am not using pjsip - but the older asterisk.
>
> Thanks
>
> Jerry
>

 >Have you configured chan_sip to know it is behind NAT itself and what its
>public IP address is? If not, then you'll get no audio.

I'm thinking I have not. What did I miss ?

Thanks,

Jerry
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Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Joshua C. Colp
On Thu, Oct 6, 2022 at 10:03 AM Jerry Geis  wrote:

>
>
> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis  wrote:
>
>> I am trying to get audio to work on AWS using asterisk 18.14.0
>>
>> I have enabled the firewall to allow ALL UDP on AWS
>>
>> My SIP extension has
>> nat=force_rport,comedia
>> qualify=yes
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>> canreinvite=yes
>>
>> I enable "rtp set debug on" and the console is printing info.
>>
>> The call comes into my linphone softphone - but I get no audio on my
>> linphone softphone.
>> What might I be missing to allow the audio ?
>> Volume is up.
>>
>> Thanks
>>
>> Jerry
>>
>
>
> I just noticed the RTP log is sending to 192.168.2.0 which is my local lan
> address of the linphone - it should be sending to the NAT address and is
> not.
> What did I not set correctly ?
> I am not using pjsip - but the older asterisk.
>

Have you configured chan_sip to know it is behind NAT itself and what its
public IP address is? If not, then you'll get no audio.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Jerry Geis
On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis  wrote:

> I am trying to get audio to work on AWS using asterisk 18.14.0
>
> I have enabled the firewall to allow ALL UDP on AWS
>
> My SIP extension has
> nat=force_rport,comedia
> qualify=yes
> allow=ulaw
> allow=alaw
> allow=gsm
> canreinvite=yes
>
> I enable "rtp set debug on" and the console is printing info.
>
> The call comes into my linphone softphone - but I get no audio on my
> linphone softphone.
> What might I be missing to allow the audio ?
> Volume is up.
>
> Thanks
>
> Jerry
>


I just noticed the RTP log is sending to 192.168.2.0 which is my local lan
address of the linphone - it should be sending to the NAT address and is
not.
What did I not set correctly ?
I am not using pjsip - but the older asterisk.

Thanks

Jerry
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[asterisk-users] Trying asterisk on AWS

2022-10-06 Thread Jerry Geis
I am trying to get audio to work on AWS using asterisk 18.14.0

I have enabled the firewall to allow ALL UDP on AWS

My SIP extension has
nat=force_rport,comedia
qualify=yes
allow=ulaw
allow=alaw
allow=gsm
canreinvite=yes

I enable "rtp set debug on" and the console is printing info.

The call comes into my linphone softphone - but I get no audio on my
linphone softphone.
What might I be missing to allow the audio ?
Volume is up.

Thanks

Jerry
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[asterisk-users] asterisk 8.14.0 and multicast sometimes not hear anythign

2022-10-06 Thread Jerry Geis
I am just doing a basic call in.

exten => 140,1,Answer
exten => 140,n,Playback(beep)
exten => 140,n,Dial(MulticastRTP/basic/239.168.4.90:3040//t(15))
exten => 140,n,Hangup

this works - but "sometimes" I get reports that "nothing" was heard.
Is there anything special to do for multicast ?
Any thoughts on why once in a great while nothing would be heard ?

Thanks

Jerry
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