Re: [AFMUG] OT asterisk

[chuck]

PIAF normally keeps the config files pretty empty.  

I don’t even see a sip.conf.  

There is a sip_custom.conf and a sip_nat.conf

From: Adam Moffett 
Sent: Thursday, April 12, 2018 10:48 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

Can you share the nat=, insecure=, and canreinvite= settings being used for the 
phones' peers?
I'm sorry I don't know where those are in FreePBX.  In vanilla asterisk they'd 
be in sip.conf.


-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 12:28:12 PM
Subject: Re: [AFMUG] OT asterisk

  I don’t think it is natted.  My son set it up, but he is away at college/work 
and I have limited access to him.






  From: Adam Moffett 
  Sent: Thursday, April 12, 2018 9:57 AM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk

  It has to be related to NAT and/or reinvite. 
  Look for what IP is being presented to the outside world.  There are 
definitely relevant settings in Asterisk, could be in the phone too.

  Assuming the phones have private IPs, I'd set canreinvite=no for the SIP peer 
used by the phones, and make sure the phones are not set to override the WAN IP.

  Asterisk has a setting showing what IP it uses to talk to the worldI'm 
sorry I can't recall the name of the option.  If it has its own public IP, then 
use that.  If it's private and behind NAT then it may need to present to the 
world that it has the router's WAN IP.  The NAT router could have SIP helper 
features.  You may need to turn that either on or off depending on what else 
you have going on.

  You'll get better clues from a packet capture.  You can use tcpdump on the 
Asterisk Box, and/or do it on the router.  You can verify whether the incoming 
RTP stream is or isn't getting to you.  You can check the outbound SIP packets 
to see what source IP is being used.  The other end doesn't reply to the source 
from the IP header, he replies to the source IP used in the SIP messages.  If 
Asterisk is identifying itself as peer@192.168.1.10 then your carrier may be 
sending RTP to the private address which obviously can't cross the internet.



  -- Original Message --
  From: ch...@wbmfg.com
  To: af@afmug.com
  Sent: 4/12/2018 11:41:25 AM
  Subject: Re: [AFMUG] OT asterisk

I am grateful for any bones that can be tossed this way.  Just stating 
“hey, I am having this problem” which implies that any suggestions to fix the 
problem are most welcome.  

Re: [AFMUG] OT asterisk

[Adam Moffett]

I guess the front end wasn't in sync with the back end.


-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 12:49:13 PM
Subject: Re: [AFMUG] OT asterisk

Arrgh, I turned on nat and turned it back off and now it is 
working!  I hate it when I fix something like this and I do not 
know how I fixed it...


From:ch...@wbmfg.com
Sent: Thursday, April 12, 2018 10:28 AM
To:af@afmug.com
Subject: Re: [AFMUG] OT asterisk

I don’t think it is natted.  My son set it up, but he is away at 
college/work and I have limited access to him.






From:Adam Moffett
Sent: Thursday, April 12, 2018 9:57 AM
To:af@afmug.com
Subject: Re: [AFMUG] OT asterisk

It has to be related to NAT and/or reinvite.
Look for what IP is being presented to the outside world.  There are 
definitely relevant settings in Asterisk, could be in the phone too.


Assuming the phones have private IPs, I'd set canreinvite=no for the 
SIP peer used by the phones, and make sure the phones are not set to 
override the WAN IP.


Asterisk has a setting showing what IP it uses to talk to the 
worldI'm sorry I can't recall the name of the option.  If it has 
its own public IP, then use that.  If it's private and behind NAT then 
it may need to present to the world that it has the router's WAN IP.  
The NAT router could have SIP helper features.  You may need to turn 
that either on or off depending on what else you have going on.


You'll get better clues from a packet capture.  You can use tcpdump on 
the Asterisk Box, and/or do it on the router.  You can verify whether 
the incoming RTP stream is or isn't getting to you.  You can check the 
outbound SIP packets to see what source IP is being used.  The other 
end doesn't reply to the source from the IP header, he replies to the 
source IP used in the SIP messages.  If Asterisk is identifying itself 
as peer@192.168.1.10 then your carrier may be sending RTP to the 
private address which obviously can't cross the internet.




-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 11:41:25 AM
Subject: Re: [AFMUG] OT asterisk

I am grateful for any bones that can be tossed this way.  Just stating 
“hey, I am having this problem” which implies that any suggestions to 
fix the problem are most welcome.

Re: [AFMUG] OT asterisk

[chuck]

Arrgh, I turned on nat and turned it back off and now it is working!  I 
hate it when I fix something like this and I do not know how I fixed it...

From: ch...@wbmfg.com 
Sent: Thursday, April 12, 2018 10:28 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

I don’t think it is natted.  My son set it up, but he is away at college/work 
and I have limited access to him.






From: Adam Moffett 
Sent: Thursday, April 12, 2018 9:57 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

It has to be related to NAT and/or reinvite. 
Look for what IP is being presented to the outside world.  There are definitely 
relevant settings in Asterisk, could be in the phone too.

Assuming the phones have private IPs, I'd set canreinvite=no for the SIP peer 
used by the phones, and make sure the phones are not set to override the WAN IP.

Asterisk has a setting showing what IP it uses to talk to the worldI'm 
sorry I can't recall the name of the option.  If it has its own public IP, then 
use that.  If it's private and behind NAT then it may need to present to the 
world that it has the router's WAN IP.  The NAT router could have SIP helper 
features.  You may need to turn that either on or off depending on what else 
you have going on.

You'll get better clues from a packet capture.  You can use tcpdump on the 
Asterisk Box, and/or do it on the router.  You can verify whether the incoming 
RTP stream is or isn't getting to you.  You can check the outbound SIP packets 
to see what source IP is being used.  The other end doesn't reply to the source 
from the IP header, he replies to the source IP used in the SIP messages.  If 
Asterisk is identifying itself as peer@192.168.1.10 then your carrier may be 
sending RTP to the private address which obviously can't cross the internet.



-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 11:41:25 AM
Subject: Re: [AFMUG] OT asterisk

  I am grateful for any bones that can be tossed this way.  Just stating “hey, 
I am having this problem” which implies that any suggestions to fix the problem 
are most welcome.  

Re: [AFMUG] OT asterisk

[Adam Moffett]

Can you share the nat=, insecure=, and canreinvite= settings being used 
for the phones' peers?
I'm sorry I don't know where those are in FreePBX.  In vanilla asterisk 
they'd be in sip.conf.



-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 12:28:12 PM
Subject: Re: [AFMUG] OT asterisk

I don’t think it is natted.  My son set it up, but he is away at 
college/work and I have limited access to him.






From:Adam Moffett
Sent: Thursday, April 12, 2018 9:57 AM
To:af@afmug.com
Subject: Re: [AFMUG] OT asterisk

It has to be related to NAT and/or reinvite.
Look for what IP is being presented to the outside world.  There are 
definitely relevant settings in Asterisk, could be in the phone too.


Assuming the phones have private IPs, I'd set canreinvite=no for the 
SIP peer used by the phones, and make sure the phones are not set to 
override the WAN IP.


Asterisk has a setting showing what IP it uses to talk to the 
worldI'm sorry I can't recall the name of the option.  If it has 
its own public IP, then use that.  If it's private and behind NAT then 
it may need to present to the world that it has the router's WAN IP.  
The NAT router could have SIP helper features.  You may need to turn 
that either on or off depending on what else you have going on.


You'll get better clues from a packet capture.  You can use tcpdump on 
the Asterisk Box, and/or do it on the router.  You can verify whether 
the incoming RTP stream is or isn't getting to you.  You can check the 
outbound SIP packets to see what source IP is being used.  The other 
end doesn't reply to the source from the IP header, he replies to the 
source IP used in the SIP messages.  If Asterisk is identifying itself 
as peer@192.168.1.10 then your carrier may be sending RTP to the 
private address which obviously can't cross the internet.




-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 11:41:25 AM
Subject: Re: [AFMUG] OT asterisk

I am grateful for any bones that can be tossed this way.  Just stating 
“hey, I am having this problem” which implies that any suggestions to 
fix the problem are most welcome.

Re: [AFMUG] OT asterisk

[chuck]

I don’t think it is natted.  My son set it up, but he is away at college/work 
and I have limited access to him.






From: Adam Moffett 
Sent: Thursday, April 12, 2018 9:57 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

It has to be related to NAT and/or reinvite. 
Look for what IP is being presented to the outside world.  There are definitely 
relevant settings in Asterisk, could be in the phone too.

Assuming the phones have private IPs, I'd set canreinvite=no for the SIP peer 
used by the phones, and make sure the phones are not set to override the WAN IP.

Asterisk has a setting showing what IP it uses to talk to the worldI'm 
sorry I can't recall the name of the option.  If it has its own public IP, then 
use that.  If it's private and behind NAT then it may need to present to the 
world that it has the router's WAN IP.  The NAT router could have SIP helper 
features.  You may need to turn that either on or off depending on what else 
you have going on.

You'll get better clues from a packet capture.  You can use tcpdump on the 
Asterisk Box, and/or do it on the router.  You can verify whether the incoming 
RTP stream is or isn't getting to you.  You can check the outbound SIP packets 
to see what source IP is being used.  The other end doesn't reply to the source 
from the IP header, he replies to the source IP used in the SIP messages.  If 
Asterisk is identifying itself as peer@192.168.1.10 then your carrier may be 
sending RTP to the private address which obviously can't cross the internet.



-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 11:41:25 AM
Subject: Re: [AFMUG] OT asterisk

  I am grateful for any bones that can be tossed this way.  Just stating “hey, 
I am having this problem” which implies that any suggestions to fix the problem 
are most welcome.  

Re: [AFMUG] OT asterisk

[Faisal Imtiaz]

>>>that any suggestions to fix the problem are most welcome. 
Tough to offer suggestion without knowing the details of your settings and 
setup. 
A lot easier to review and fix / troubleshoot. 
This is an offer to help with no strings attached. 

Your generic request for help reminds of the college days.. when we were coding 
in fortran. 

My program is not working.. I am not sure what is wrong. 
Can you help me ? 
Yes, but only if you show me your code so that I can check for syntax. 
(dang those periods and commas ! ) 

The issue is obvious, you are not able to open the audio channel... 
the causes for such a condition are few most of them depend on the 
specifics of the configuration. 

Regards. 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: ch...@wbmfg.com
> To: af@afmug.com
> Sent: Thursday, April 12, 2018 11:41:24 AM
> Subject: Re: [AFMUG] OT asterisk

> I am grateful for any bones that can be tossed this way. Just stating “hey, I 
> am
> having this problem” which implies that any suggestions to fix the problem are
> most welcome.
> From: Faisal Imtiaz
> Sent: Thursday, April 12, 2018 9:33 AM
> To: af@afmug.com
> Subject: Re: [AFMUG] OT asterisk
> Yep.. I know..
> Not sure if Chuck is venting or asking for assistance ...
> Being a technical person I have lot of trouble understanding a generic 
> request /
> comment from another technical person..
> ' I have screw that is not working to keep the two parts together !'. or 
> as
> a lot of other computer people will relate to this one.. " My computer does 
> not
> work ! " ...
> ;)
> Regardless, the offer to help and solve this problem quickly still stands.. 
> but
> of course in the infamous words of Number Nine "Need More Input" !
> :)
> Regards.
> Faisal Imtiaz
> Snappy Internet & Telecom
> http://www.snappytelecom.net

> Tel: 305 663 5518 x 232

> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>> From: "Adam Moffett" <dmmoff...@gmail.com>
>> To: af@afmug.com
>> Sent: Thursday, April 12, 2018 11:06:39 AM
>> Subject: Re: [AFMUG] OT asterisk

>> He means PBX in a Flash. It is FreePBX along with some other things.

>> -- Original Message --
>> From: "Faisal Imtiaz" < fai...@snappytelecom.net >
>> To: af@afmug.com
>> Sent: 4/12/2018 11:01:43 AM
>> Subject: Re: [AFMUG] OT asterisk

>>> It is too painful to read your emails which are lacking a lot of 
>>> details..
>>> Anyhow.. if you need assistance if fixing this, feel free to call me.
>>> Regards.
>>> Faisal Imtiaz
>>> Snappy Internet & Telecom
>>> http://www.snappytelecom.net

>>> Tel: 305 663 5518 x 232

>>> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>>>> From: ch...@wbmfg.com
>>>> To: af@afmug.com
>>>> Sent: Thursday, April 12, 2018 10:53:15 AM
>>>> Subject: Re: [AFMUG] OT asterisk

>>>> PIAF
>>>> From: Faisal Imtiaz
>>>> Sent: Thursday, April 12, 2018 8:51 AM
>>>> To: af@afmug.com
>>>> Subject: Re: [AFMUG] OT asterisk
>>>> are you using asterisk or some variant of a PBX implementation e.g. 
>>>> freepbx or
>>>> elastix ?
>>>> Faisal Imtiaz
>>>> Snappy Internet & Telecom
>>>> http://www.snappytelecom.net

>>>> Tel: 305 663 5518 x 232

>>>> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>>>>> From: ch...@wbmfg.com
>>>>> To: af@afmug.com
>>>>> Sent: Thursday, April 12, 2018 10:28:05 AM
>>>>> Subject: Re: [AFMUG] OT asterisk

>>>>> It is a carrier that we have used for years. Someone hacked something a 
>>>>> month or
>>>>> two ago and was making thousands of calls from our asterisk system. So we
>>>>> rebuilt everything trying to up the security. Now, trying to get it to go
>>>>> again.
>>>>> From: Lewis Bergman
>>>>> Sent: Thursday, April 12, 2018 6:22 AM
>>>>> To: af@afmug.com
>>>>> Subject: Re: [AFMUG] OT asterisk
>>>>> What carrier are you using? Some carriers try to get out of the media 
>>>>> path and
>>>>> hand the endpoint IP's off to each side. Depending on how your system is
>>>>> configured and what IP is handed during the ReInvite it can cause one way
>>>>> audio.
>>>>> On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman < 
>>>>> j...@imaginenetworksllc.com >
>>>>> wrote:

>>>>>> Between phones or through origination/termination?

>>>>>> Josh Luthman
>>>>>> Office: 937-552-2340
>>>>>> Direct: 937-552-2343
>>>>>> 1100 Wayne St
>>>>>> Suite 1337
>>>>>> Troy, OH 45373
>>>>>> On Wed, Apr 11, 2018, 5:44 PM < ch...@wbmfg.com > wrote:

>>>>>>> I have extension to extension audio working now but one way outbound 
>>>>>>> audio on
>>>>>>> outbound calls arrgh

Re: [AFMUG] OT asterisk

[Adam Moffett]

It has to be related to NAT and/or reinvite.
Look for what IP is being presented to the outside world.  There are 
definitely relevant settings in Asterisk, could be in the phone too.


Assuming the phones have private IPs, I'd set canreinvite=no for the SIP 
peer used by the phones, and make sure the phones are not set to 
override the WAN IP.


Asterisk has a setting showing what IP it uses to talk to the 
worldI'm sorry I can't recall the name of the option.  If it has its 
own public IP, then use that.  If it's private and behind NAT then it 
may need to present to the world that it has the router's WAN IP.  The 
NAT router could have SIP helper features.  You may need to turn that 
either on or off depending on what else you have going on.


You'll get better clues from a packet capture.  You can use tcpdump on 
the Asterisk Box, and/or do it on the router.  You can verify whether 
the incoming RTP stream is or isn't getting to you.  You can check the 
outbound SIP packets to see what source IP is being used.  The other end 
doesn't reply to the source from the IP header, he replies to the source 
IP used in the SIP messages.  If Asterisk is identifying itself as 
peer@192.168.1.10 then your carrier may be sending RTP to the private 
address which obviously can't cross the internet.




-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/12/2018 11:41:25 AM
Subject: Re: [AFMUG] OT asterisk

I am grateful for any bones that can be tossed this way.  Just stating 
“hey, I am having this problem” which implies that any suggestions to 
fix the problem are most welcome.

Re: [AFMUG] OT asterisk

[chuck]

I am grateful for any bones that can be tossed this way.  Just stating “hey, I 
am having this problem” which implies that any suggestions to fix the problem 
are most welcome.  

From: Faisal Imtiaz 
Sent: Thursday, April 12, 2018 9:33 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

Yep.. I  know..


Not sure if Chuck is venting or asking for assistance ...


Being a technical person I have lot of trouble understanding a generic request 
/ comment from another technical person..


' I have screw that is not working to keep the two parts together !'. or as 
a lot of other computer people will relate to this one.. " My computer does not 
work ! " ...


;)


Regardless, the offer to help and solve this problem quickly still stands.. but 
of course in the infamous words of Number Nine  "Need More Input" !


:)


Regards.


Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net




  From: "Adam Moffett" <dmmoff...@gmail.com>
  To: af@afmug.com
  Sent: Thursday, April 12, 2018 11:06:39 AM
  Subject: Re: [AFMUG] OT asterisk

  He means PBX in a Flash.  It is FreePBX along with some other things.



  -- Original Message --
  From: "Faisal Imtiaz" <fai...@snappytelecom.net>
  To: af@afmug.com

  Sent: 4/12/2018 11:01:43 AM
  Subject: Re: [AFMUG] OT asterisk

It is too painful to read your emails which are lacking a lot of 
details..


Anyhow.. if you need assistance if fixing this, feel free to call me.


Regards.


Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net





  From: ch...@wbmfg.com
  To: af@afmug.com
  Sent: Thursday, April 12, 2018 10:53:15 AM
  Subject: Re: [AFMUG] OT asterisk

  PIAF

  From: Faisal Imtiaz

  Sent: Thursday, April 12, 2018 8:51 AM
  To: af@afmug.com

  Subject: Re: [AFMUG] OT asterisk

  are you using asterisk or some variant of a PBX implementation e.g. 
freepbx or elastix ?


  Faisal Imtiaz
  Snappy Internet & Telecom
  http://www.snappytelecom.net

  Tel: 305 663 5518 x 232

  Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net



--

From: ch...@wbmfg.com
To: af@afmug.com
Sent: Thursday, April 12, 2018 10:28:05 AM
Subject: Re: [AFMUG] OT asterisk

It is a carrier that we have used for years.  Someone hacked something 
a month or two ago and was making thousands of calls from our asterisk system.  
So we rebuilt everything trying to up the security.  Now, trying to get it to 
go again.

From: Lewis Bergman

Sent: Thursday, April 12, 2018 6:22 AM
    To: af@afmug.com

Subject: Re: [AFMUG] OT asterisk

What carrier are you using? Some carriers try to get out of the media 
path and hand the endpoint IP's off to each side. Depending on how your system 
is configured and what IP is handed during the ReInvite it can cause one way 
audio.

On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman 
<j...@imaginenetworksllc.com> wrote:

  Between phones or through origination/termination?



  Josh Luthman
  Office: 937-552-2340
  Direct: 937-552-2343
  1100 Wayne St
  Suite 1337
  Troy, OH 45373


  On Wed, Apr 11, 2018, 5:44 PM <ch...@wbmfg.com> wrote:

I have extension to extension audio working now but one way 
outbound audio on outbound calls  arrgh

Re: [AFMUG] OT asterisk

[Faisal Imtiaz]

Yep.. I know.. 

Not sure if Chuck is venting or asking for assistance ... 

Being a technical person I have lot of trouble understanding a generic request 
/ comment from another technical person.. 

' I have screw that is not working to keep the two parts together !'. or as 
a lot of other computer people will relate to this one.. " My computer does not 
work ! " ... 

;) 

Regardless, the offer to help and solve this problem quickly still stands.. but 
of course in the infamous words of Number Nine "Need More Input" ! 

:) 

Regards. 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: "Adam Moffett" <dmmoff...@gmail.com>
> To: af@afmug.com
> Sent: Thursday, April 12, 2018 11:06:39 AM
> Subject: Re: [AFMUG] OT asterisk

> He means PBX in a Flash. It is FreePBX along with some other things.

> -- Original Message --
> From: "Faisal Imtiaz" < fai...@snappytelecom.net >
> To: af@afmug.com
> Sent: 4/12/2018 11:01:43 AM
> Subject: Re: [AFMUG] OT asterisk

>> It is too painful to read your emails which are lacking a lot of 
>> details..

>> Anyhow.. if you need assistance if fixing this, feel free to call me.

>> Regards.

>> Faisal Imtiaz
>> Snappy Internet & Telecom
>> http://www.snappytelecom.net

>> Tel: 305 663 5518 x 232

>> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>>> From: ch...@wbmfg.com
>>> To: af@afmug.com
>>> Sent: Thursday, April 12, 2018 10:53:15 AM
>>> Subject: Re: [AFMUG] OT asterisk

>>> PIAF
>>> From: Faisal Imtiaz
>>> Sent: Thursday, April 12, 2018 8:51 AM
>>> To: af@afmug.com
>>> Subject: Re: [AFMUG] OT asterisk
>>> are you using asterisk or some variant of a PBX implementation e.g. freepbx 
>>> or
>>> elastix ?
>>> Faisal Imtiaz
>>> Snappy Internet & Telecom
>>> http://www.snappytelecom.net

>>> Tel: 305 663 5518 x 232

>>> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>>>> From: ch...@wbmfg.com
>>>> To: af@afmug.com
>>>> Sent: Thursday, April 12, 2018 10:28:05 AM
>>>> Subject: Re: [AFMUG] OT asterisk

>>>> It is a carrier that we have used for years. Someone hacked something a 
>>>> month or
>>>> two ago and was making thousands of calls from our asterisk system. So we
>>>> rebuilt everything trying to up the security. Now, trying to get it to go
>>>> again.
>>>> From: Lewis Bergman
>>>> Sent: Thursday, April 12, 2018 6:22 AM
>>>> To: af@afmug.com
>>>> Subject: Re: [AFMUG] OT asterisk
>>>> What carrier are you using? Some carriers try to get out of the media path 
>>>> and
>>>> hand the endpoint IP's off to each side. Depending on how your system is
>>>> configured and what IP is handed during the ReInvite it can cause one way
>>>> audio.
>>>> On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman < 
>>>> j...@imaginenetworksllc.com >
>>>> wrote:

>>>>> Between phones or through origination/termination?

>>>>> Josh Luthman
>>>>> Office: 937-552-2340
>>>>> Direct: 937-552-2343
>>>>> 1100 Wayne St
>>>>> Suite 1337
>>>>> Troy, OH 45373
>>>>> On Wed, Apr 11, 2018, 5:44 PM < ch...@wbmfg.com > wrote:

>>>>>> I have extension to extension audio working now but one way outbound 
>>>>>> audio on
>>>>>> outbound calls arrgh

Re: [AFMUG] OT asterisk

[Adam Moffett]

He means PBX in a Flash.  It is FreePBX along with some other things.


-- Original Message --
From: "Faisal Imtiaz" <fai...@snappytelecom.net>
To: af@afmug.com
Sent: 4/12/2018 11:01:43 AM
Subject: Re: [AFMUG] OT asterisk

It is too painful to read your emails which are lacking a lot of 
details..


Anyhow.. if you need assistance if fixing this, feel free to call me.

Regards.

Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net



From: ch...@wbmfg.com
To: af@afmug.com
Sent: Thursday, April 12, 2018 10:53:15 AM
Subject: Re: [AFMUG] OT asterisk
PIAF

From:Faisal Imtiaz
Sent: Thursday, April 12, 2018 8:51 AM
To:af@afmug.com
Subject: Re: [AFMUG] OT asterisk

are you using asterisk or some variant of a PBX implementation e.g. 
freepbx or elastix ?


Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net



From: ch...@wbmfg.com
To: af@afmug.com
Sent: Thursday, April 12, 2018 10:28:05 AM
Subject: Re: [AFMUG] OT asterisk
It is a carrier that we have used for years.  Someone hacked 
something a month or two ago and was making thousands of calls from 
our asterisk system.  So we rebuilt everything trying to up the 
security.  Now, trying to get it to go again.


From:Lewis Bergman
Sent: Thursday, April 12, 2018 6:22 AM
To:af@afmug.com
Subject: Re: [AFMUG] OT asterisk

What carrier are you using? Some carriers try to get out of the media 
path and hand the endpoint IP's off to each side. Depending on how 
your system is configured and what IP is handed during the ReInvite 
it can cause one way audio.


On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman 
<j...@imaginenetworksllc.com> wrote:

Between phones or through origination/termination?


Josh Luthman
Office: 937-552-2340 <tel:(937)%20552-2340>
Direct: 937-552-2343 <tel:(937)%20552-2343>
1100 Wayne St 
<https://maps.google.com/?q=1100+Wayne+St+Suite+1337+Troy,+OH+45373=gmail=g>
Suite 1337 
<https://maps.google.com/?q=1100+Wayne+St+Suite+1337+Troy,+OH+45373=gmail=g>
Troy, OH 45373 
<https://maps.google.com/?q=1100+Wayne+St+Suite+1337+Troy,+OH+45373=gmail=g>


On Wed, Apr 11, 2018, 5:44 PM <ch...@wbmfg.com> wrote:
I have extension to extension audio working now but one way 
outbound audio on outbound calls  arrgh

Re: [AFMUG] OT asterisk

[Faisal Imtiaz]

It is too painful to read your emails which are lacking a lot of details.. 

Anyhow.. if you need assistance if fixing this, feel free to call me. 

Regards. 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: ch...@wbmfg.com
> To: af@afmug.com
> Sent: Thursday, April 12, 2018 10:53:15 AM
> Subject: Re: [AFMUG] OT asterisk

> PIAF
> From: Faisal Imtiaz
> Sent: Thursday, April 12, 2018 8:51 AM
> To: af@afmug.com
> Subject: Re: [AFMUG] OT asterisk
> are you using asterisk or some variant of a PBX implementation e.g. freepbx or
> elastix ?
> Faisal Imtiaz
> Snappy Internet & Telecom
> http://www.snappytelecom.net

> Tel: 305 663 5518 x 232

> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>> From: ch...@wbmfg.com
>> To: af@afmug.com
>> Sent: Thursday, April 12, 2018 10:28:05 AM
>> Subject: Re: [AFMUG] OT asterisk

>> It is a carrier that we have used for years. Someone hacked something a 
>> month or
>> two ago and was making thousands of calls from our asterisk system. So we
>> rebuilt everything trying to up the security. Now, trying to get it to go
>> again.
>> From: Lewis Bergman
>> Sent: Thursday, April 12, 2018 6:22 AM
>> To: af@afmug.com
>> Subject: Re: [AFMUG] OT asterisk
>> What carrier are you using? Some carriers try to get out of the media path 
>> and
>> hand the endpoint IP's off to each side. Depending on how your system is
>> configured and what IP is handed during the ReInvite it can cause one way
>> audio.
>> On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman < j...@imaginenetworksllc.com >
>> wrote:

>>> Between phones or through origination/termination?

>>> Josh Luthman
>>> Office: 937-552-2340
>>> Direct: 937-552-2343
>>> 1100 Wayne St
>>> Suite 1337
>>> Troy, OH 45373
>>> On Wed, Apr 11, 2018, 5:44 PM < ch...@wbmfg.com > wrote:

>>>> I have extension to extension audio working now but one way outbound audio 
>>>> on
>>>> outbound calls arrgh

Re: [AFMUG] OT asterisk

[chuck]

PIAF

From: Faisal Imtiaz 
Sent: Thursday, April 12, 2018 8:51 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

are you using asterisk or some variant of a PBX implementation e.g. freepbx or 
elastix ?


Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net




  From: ch...@wbmfg.com
  To: af@afmug.com
  Sent: Thursday, April 12, 2018 10:28:05 AM
  Subject: Re: [AFMUG] OT asterisk

  It is a carrier that we have used for years.  Someone hacked something a 
month or two ago and was making thousands of calls from our asterisk system.  
So we rebuilt everything trying to up the security.  Now, trying to get it to 
go again. 

  From: Lewis Bergman

  Sent: Thursday, April 12, 2018 6:22 AM
  To: af@afmug.com

  Subject: Re: [AFMUG] OT asterisk

  What carrier are you using? Some carriers try to get out of the media path 
and hand the endpoint IP's off to each side. Depending on how your system is 
configured and what IP is handed during the ReInvite it can cause one way audio.

  On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman <j...@imaginenetworksllc.com> 
wrote:

Between phones or through origination/termination?



Josh Luthman
Office: 937-552-2340
Direct: 937-552-2343
1100 Wayne St
Suite 1337
Troy, OH 45373


On Wed, Apr 11, 2018, 5:44 PM <ch...@wbmfg.com> wrote:

  I have extension to extension audio working now but one way outbound 
audio on outbound calls  arrgh

Re: [AFMUG] OT asterisk

[Faisal Imtiaz]

are you using asterisk or some variant of a PBX implementation e.g. freepbx or 
elastix ? 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: ch...@wbmfg.com
> To: af@afmug.com
> Sent: Thursday, April 12, 2018 10:28:05 AM
> Subject: Re: [AFMUG] OT asterisk

> It is a carrier that we have used for years. Someone hacked something a month 
> or
> two ago and was making thousands of calls from our asterisk system. So we
> rebuilt everything trying to up the security. Now, trying to get it to go
> again.
> From: Lewis Bergman
> Sent: Thursday, April 12, 2018 6:22 AM
> To: af@afmug.com
> Subject: Re: [AFMUG] OT asterisk
> What carrier are you using? Some carriers try to get out of the media path and
> hand the endpoint IP's off to each side. Depending on how your system is
> configured and what IP is handed during the ReInvite it can cause one way
> audio.
> On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman < j...@imaginenetworksllc.com >
> wrote:

>> Between phones or through origination/termination?

>> Josh Luthman
>> Office: 937-552-2340
>> Direct: 937-552-2343
>> 1100 Wayne St
>> Suite 1337
>> Troy, OH 45373
>> On Wed, Apr 11, 2018, 5:44 PM < ch...@wbmfg.com > wrote:

>>> I have extension to extension audio working now but one way outbound audio 
>>> on
>>> outbound calls arrgh

Re: [AFMUG] OT asterisk

[chuck]

It is a carrier that we have used for years.  Someone hacked something a month 
or two ago and was making thousands of calls from our asterisk system.  So we 
rebuilt everything trying to up the security.  Now, trying to get it to go 
again.  

From: Lewis Bergman 
Sent: Thursday, April 12, 2018 6:22 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

What carrier are you using? Some carriers try to get out of the media path and 
hand the endpoint IP's off to each side. Depending on how your system is 
configured and what IP is handed during the ReInvite it can cause one way audio.

On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman <j...@imaginenetworksllc.com> 
wrote:

  Between phones or through origination/termination?



  Josh Luthman
  Office: 937-552-2340
  Direct: 937-552-2343
  1100 Wayne St
  Suite 1337
  Troy, OH 45373

  On Wed, Apr 11, 2018, 5:44 PM <ch...@wbmfg.com> wrote:

I have extension to extension audio working now but one way outbound audio 
on outbound calls  arrgh

Re: [AFMUG] OT asterisk

[chuck]

Phone to phone works.

Phone to outside sends audio out but no audio in from the external connection.  

From: Josh Luthman 
Sent: Wednesday, April 11, 2018 9:17 PM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

Between phones or through origination/termination?


Josh Luthman
Office: 937-552-2340
Direct: 937-552-2343
1100 Wayne St
Suite 1337
Troy, OH 45373

On Wed, Apr 11, 2018, 5:44 PM <ch...@wbmfg.com> wrote:

  I have extension to extension audio working now but one way outbound audio on 
outbound calls  arrgh

Re: [AFMUG] OT asterisk

[Lewis Bergman]

What carrier are you using? Some carriers try to get out of the media path
and hand the endpoint IP's off to each side. Depending on how your system
is configured and what IP is handed during the ReInvite it can cause one
way audio.

On Wed, Apr 11, 2018 at 10:17 PM Josh Luthman 
wrote:

> Between phones or through origination/termination?
>
>
> Josh Luthman
> Office: 937-552-2340 <(937)%20552-2340>
> Direct: 937-552-2343 <(937)%20552-2343>
> 1100 Wayne St
> 
> Suite 1337
> 
> Troy, OH 45373
> 
>
> On Wed, Apr 11, 2018, 5:44 PM  wrote:
>
>> I have extension to extension audio working now but one way outbound
>> audio on outbound calls  arrgh
>>
>

Re: [AFMUG] OT asterisk

[Josh Luthman]

Between phones or through origination/termination?

Josh Luthman
Office: 937-552-2340
Direct: 937-552-2343
1100 Wayne St
Suite 1337
Troy, OH 45373

On Wed, Apr 11, 2018, 5:44 PM  wrote:

> I have extension to extension audio working now but one way outbound audio
> on outbound calls  arrgh
>

[AFMUG] OT asterisk

[chuck]

I have extension to extension audio working now but one way outbound audio on 
outbound calls  arrgh

Re: [AFMUG] OT Asterisk

[Faisal Imtiaz]

>>I would just use one interface and put a VLAN on it for private phone 
>>traffic. 

LOL ! .. you know what "VLAN" actually stands for ? 

Virtual Local Area Network 

TWO NIC vs One NIC + VLAN = Michelob in a bottle Michelob served in a Beer 
Glass ;) 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: "can...@believewireless.net" <p...@believewireless.net>
> To: af@afmug.com
> Sent: Thursday, April 5, 2018 8:52:15 AM
> Subject: Re: [AFMUG] OT Asterisk

> I would just use one interface and put a VLAN on it for private phone traffic.

> On Thu, Apr 5, 2018 at 8:19 AM, Faisal Imtiaz < fai...@snappytelecom.net >
> wrote:

>> Yes.

>> Faisal Imtiaz
>> Snappy Internet & Telecom
>> http://www.snappytelecom.net

>> Tel: 305 663 5518 x 232

>> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>>> From: ch...@wbmfg.com
>>> To: af@afmug.com
>>> Sent: Wednesday, April 4, 2018 3:38:46 PM
>>> Subject: [AFMUG] OT Asterisk

>>> How can you make asterisk use two eth interfaces so that all the phones are 
>>> on a
>>> private internal interface and the world is on the other?

Re: [AFMUG] OT Asterisk

[Faisal Imtiaz]

Which part ? 

it is like setting up any dual homed server, 
Two NIC's with two IP ranges the service (asterisk) has to have service 
bindings on each interface accordingly. 

Regards. 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: "Tim Reichhart" <timreichh...@hometowncable.net>
> To: af@afmug.com
> Sent: Thursday, April 5, 2018 10:11:34 AM
> Subject: Re: [AFMUG] OT Asterisk

> Faisal
> how can you set that up?

> Tim

>> -Original Message-
>> From: "Faisal Imtiaz" < fai...@snappytelecom.net >
>> To: af@afmug.com
>> Date: 04/05/18 08:19
>> Subject: Re: [AFMUG] OT Asterisk

>> Yes.

>> Faisal Imtiaz
>> Snappy Internet & Telecom
>> http://www.snappytelecom.net

>> Tel: 305 663 5518 x 232

>> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

>>> From: ch...@wbmfg.com
>>> To: af@afmug.com
>>> Sent: Wednesday, April 4, 2018 3:38:46 PM
>>> Subject: [AFMUG] OT Asterisk

>>> How can you make asterisk use two eth interfaces so that all the phones are 
>>> on a
>>> private internal interface and the world is on the other?

Re: [AFMUG] OT Asterisk

[Tim Reichhart]

Faisal
how can you set that up?

Tim



 

-Original Message-
From: "Faisal Imtiaz" <fai...@snappytelecom.net>
To: af@afmug.com
Date: 04/05/18 08:19
Subject: Re: [AFMUG] OT Asterisk


Yes.


Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net


From: ch...@wbmfg.com
To: af@afmug.com
Sent: Wednesday, April 4, 2018 3:38:46 PM
Subject: [AFMUG] OT Asterisk


How can you make asterisk use two eth interfaces so that all the phones are on 
a private internal interface and the world is on the other?

Re: [AFMUG] OT Asterisk

[can...@believewireless.net]

I would just use one interface and put a VLAN on it for private phone
traffic.

On Thu, Apr 5, 2018 at 8:19 AM, Faisal Imtiaz <fai...@snappytelecom.net>
wrote:

> Yes.
>
> Faisal Imtiaz
> Snappy Internet & Telecom
> http://www.snappytelecom.net
>
> Tel: 305 663 5518 x 232
>
> Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net
>
> --
>
> *From: *ch...@wbmfg.com
> *To: *af@afmug.com
> *Sent: *Wednesday, April 4, 2018 3:38:46 PM
> *Subject: *[AFMUG] OT Asterisk
>
> How can you make asterisk use two eth interfaces so that all the phones
> are on a private internal interface and the world is on the other?
>
>

Re: [AFMUG] OT Asterisk

[Faisal Imtiaz]

Yes. 

Faisal Imtiaz 
Snappy Internet & Telecom 
http://www.snappytelecom.net 

Tel: 305 663 5518 x 232 

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net 

> From: ch...@wbmfg.com
> To: af@afmug.com
> Sent: Wednesday, April 4, 2018 3:38:46 PM
> Subject: [AFMUG] OT Asterisk

> How can you make asterisk use two eth interfaces so that all the phones are 
> on a
> private internal interface and the world is on the other?

Re: [AFMUG] OT Asterisk

[Josh Reynolds]

What Adam said.

On Wed, Apr 4, 2018, 3:08 PM Adam Moffett <dmmoff...@gmail.com> wrote:

> Put a private IP on the second interface and have the phones register to
> that one.
> I've done that before, and I don't think there was much more to it than
> that.  I think there's a listen= or bindaddress= line in one of the config
> files you may have to adjust.
>
>
>
> -- Original Message --
> From: ch...@wbmfg.com
> To: af@afmug.com
> Sent: 4/4/2018 3:38:46 PM
> Subject: [AFMUG] OT Asterisk
>
> How can you make asterisk use two eth interfaces so that all the phones
> are on a private internal interface and the world is on the other?
>
>

Re: [AFMUG] OT Asterisk

[Adam Moffett]

Put a private IP on the second interface and have the phones register to 
that one.
I've done that before, and I don't think there was much more to it than 
that.  I think there's a listen= or bindaddress= line in one of the 
config files you may have to adjust.




-- Original Message --
From: ch...@wbmfg.com
To: af@afmug.com
Sent: 4/4/2018 3:38:46 PM
Subject: [AFMUG] OT Asterisk

How can you make asterisk use two eth interfaces so that all the phones 
are on a private internal interface and the world is on the other?

[AFMUG] OT Asterisk

[chuck]

How can you make asterisk use two eth interfaces so that all the phones are on 
a private internal interface and the world is on the other?

Re: [AFMUG] OT asterisk

[Chuck McCown via Af]

It is getting to the asterisk box.  It shows up in the logfile.  But it looks 
like it is treating it like a phone number to me.  Especially if I add the 
extension and group number.  

From: Nate Burke via Af 
Sent: Wednesday, October 15, 2014 8:45 PM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

Is the *45 getting to the server, or is the dialplan in the voip appliance 
trying to interpret it locally?  

On 10/15/2014 6:25 PM, Chuck McCown via Af wrote: 
  Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.  
Extension does not appear in queue.  

  Static agents work just fine.  Everything else works just fine.  Voice 
prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic agents.

  Any suggestions.

Re: [AFMUG] OT asterisk

[Adam Moffett via Af]

I've never used that featurebut I wonder if you have to include a 
special context that knows how to handle the code, like you do with call 
parking.


It is getting to the asterisk box.  It shows up in the logfile.  But 
it looks like it is treating it like a phone number to me.  Especially 
if I add the extension and group number.

*From:* Nate Burke via Af mailto:af@afmug.com
*Sent:* Wednesday, October 15, 2014 8:45 PM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
Is the *45 getting to the server, or is the dialplan in the voip 
appliance trying to interpret it locally?


On 10/15/2014 6:25 PM, Chuck McCown via Af wrote:
Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.  
Extension does not appear in queue.
Static agents work just fine.  Everything else works just fine.  
Voice prompts, IVR, voicemail etc etc.  Just having zero luck with 
dynamic agents.

Any suggestions.

Re: [AFMUG] OT asterisk

[Chuck McCown via Af]

It is one of the built in feature codes.  Others, like announce the current 
time, work.  I cannot imagine why it would ignore a feature code.  

From: Adam Moffett via Af 
Sent: Thursday, October 16, 2014 9:23 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk


I've never used that featurebut I wonder if you have to include a special 
context that knows how to handle the code, like you do with call parking.


  It is getting to the asterisk box.  It shows up in the logfile.  But it looks 
like it is treating it like a phone number to me.  Especially if I add the 
extension and group number.  

  From: Nate Burke via Af 
  Sent: Wednesday, October 15, 2014 8:45 PM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk

  Is the *45 getting to the server, or is the dialplan in the voip appliance 
trying to interpret it locally?  

  On 10/15/2014 6:25 PM, Chuck McCown via Af wrote: 
Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.  
Extension does not appear in queue.  

Static agents work just fine.  Everything else works just fine.  Voice 
prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic agents.

Any suggestions.

Re: [AFMUG] OT asterisk

[Adam Moffett via Af]

So there's a *45 defined in features.conf?
It is one of the built in feature codes.  Others, like announce the 
current time, work.  I cannot imagine why it would ignore a feature code.

*From:* Adam Moffett via Af mailto:af@afmug.com
*Sent:* Thursday, October 16, 2014 9:23 AM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
I've never used that featurebut I wonder if you have to include a 
special context that knows how to handle the code, like you do with 
call parking.


It is getting to the asterisk box.  It shows up in the logfile.  But 
it looks like it is treating it like a phone number to me.  
Especially if I add the extension and group number.

*From:* Nate Burke via Af mailto:af@afmug.com
*Sent:* Wednesday, October 15, 2014 8:45 PM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
Is the *45 getting to the server, or is the dialplan in the voip 
appliance trying to interpret it locally?


On 10/15/2014 6:25 PM, Chuck McCown via Af wrote:
Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.  
Extension does not appear in queue.
Static agents work just fine.  Everything else works just fine.  
Voice prompts, IVR, voicemail etc etc.  Just having zero luck with 
dynamic agents.

Any suggestions.

Re: [AFMUG] OT asterisk

[Chuck McCown via Af]

I didn’t check the file, it is in the list in the freepbx gui.  I guess I 
better peek at the file.

From: Adam Moffett via Af 
Sent: Thursday, October 16, 2014 9:29 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk


So there's a *45 defined in features.conf? 

  It is one of the built in feature codes.  Others, like announce the current 
time, work.  I cannot imagine why it would ignore a feature code.  

  From: Adam Moffett via Af 
  Sent: Thursday, October 16, 2014 9:23 AM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk


  I've never used that featurebut I wonder if you have to include a special 
context that knows how to handle the code, like you do with call parking.


It is getting to the asterisk box.  It shows up in the logfile.  But it 
looks like it is treating it like a phone number to me.  Especially if I add 
the extension and group number.  

From: Nate Burke via Af 
Sent: Wednesday, October 15, 2014 8:45 PM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

Is the *45 getting to the server, or is the dialplan in the voip appliance 
trying to interpret it locally?  

On 10/15/2014 6:25 PM, Chuck McCown via Af wrote: 
  Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.  
Extension does not appear in queue.  

  Static agents work just fine.  Everything else works just fine.  Voice 
prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic agents.

  Any suggestions.

Re: [AFMUG] OT asterisk

[Adam Moffett via Af]

You've probably thought of this already, but I would make sure the web 
page is not accessible from the internet at large.they've had remote 
hacks in the past. Then you'll get the bill for the phone calls to 
Uzbekistan.



So far I am loving it.  Everything else is working perfectly.
*From:* Adam Moffett via Af mailto:af@afmug.com
*Sent:* Thursday, October 16, 2014 12:08 PM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
Wish I could help.  FreePBX makes me mad every time I use itso I 
don't use it anymore.

Never tried that particular feature either.
I joined the FreePBX forum.  Hopefully I can get some help there once 
the hazing of the newbe is over.

*From:* Chuck McCown via Af mailto:af@afmug.com
*Sent:* Thursday, October 16, 2014 11:36 AM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
No actually.  Features.conf has include files of other features files.
Went through all of them and are not finding many of the feature 
codes in the GUI.

They must be hiding somewhere else.
*From:* Adam Moffett via Af mailto:af@afmug.com
*Sent:* Thursday, October 16, 2014 9:29 AM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
So there's a *45 defined in features.conf?
It is one of the built in feature codes.  Others, like announce the 
current time, work.  I cannot imagine why it would ignore a feature 
code.

*From:* Adam Moffett via Af mailto:af@afmug.com
*Sent:* Thursday, October 16, 2014 9:23 AM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
I've never used that featurebut I wonder if you have to include 
a special context that knows how to handle the code, like you do 
with call parking.


It is getting to the asterisk box.  It shows up in the logfile.  
But it looks like it is treating it like a phone number to me. 
Especially if I add the extension and group number.

*From:* Nate Burke via Af mailto:af@afmug.com
*Sent:* Wednesday, October 15, 2014 8:45 PM
*To:* af@afmug.com mailto:af@afmug.com
*Subject:* Re: [AFMUG] OT asterisk
Is the *45 getting to the server, or is the dialplan in the voip 
appliance trying to interpret it locally?


On 10/15/2014 6:25 PM, Chuck McCown via Af wrote:
Arrgh, cannot get *45 toggle queue login to work.  No voice 
prompt.  Extension does not appear in queue.
Static agents work just fine.  Everything else works just fine. 
Voice prompts, IVR, voicemail etc etc. Just having zero luck with 
dynamic agents.

Any suggestions.

Re: [AFMUG] OT asterisk

[Mike Delp via Af]

Chuck, did you add the extensions that you want to toggle in the dynamic
members of the queue?
Mike

On Thu, Oct 16, 2014 at 1:12 PM, Adam Moffett via Af af@afmug.com wrote:


 You've probably thought of this already, but I would make sure the web
 page is not accessible from the internet at large.they've had remote
 hacks in the past.  Then you'll get the bill for the phone calls to
 Uzbekistan.

   So far I am loving it.  Everything else is working perfectly.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 12:08 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 Wish I could help.  FreePBX makes me mad every time I use itso I don't
 use it anymore.
 Never tried that particular feature either.

  I joined the FreePBX forum.  Hopefully I can get some help there once
 the hazing of the newbe is over.

  *From:* Chuck McCown via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 11:36 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

   No actually.  Features.conf has include files of other features files.
 Went through all of them and are not finding many of the feature codes in
 the GUI.
 They must be hiding somewhere else.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 9:29 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 So there's a *45 defined in features.conf?

  It is one of the built in feature codes.  Others, like announce the
 current time, work.  I cannot imagine why it would ignore a feature code.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 9:23 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 I've never used that featurebut I wonder if you have to include a
 special context that knows how to handle the code, like you do with call
 parking.

   It is getting to the asterisk box.  It shows up in the logfile.  But it
 looks like it is treating it like a phone number to me.  Especially if I
 add the extension and group number.

  *From:* Nate Burke via Af af@afmug.com
 *Sent:* Wednesday, October 15, 2014 8:45 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

  Is the *45 getting to the server, or is the dialplan in the voip
 appliance trying to interpret it locally?

 On 10/15/2014 6:25 PM, Chuck McCown via Af wrote:

  Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.
 Extension does not appear in queue.

 Static agents work just fine.  Everything else works just fine.  Voice
 prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic agents.

 Any suggestions.

Re: [AFMUG] OT asterisk

[Chuck McCown via Af]

Yep.  At least I filled in the box.  For example, the phone on my desk is 200
So in the box I have 200,0

We have some static extensions in the box above too and they work as expected.
But when I try to toggle it seems like it is ignoring the feature code 
completely and trying to dial out.
No announcement, no nothing.  The queue is 700.  So dialing *45200700 gives me 
this in the log:

*45200700' rejected because extension not found in context 'from-internal'.

That is a dialplan type of problem I would think.  I have been reading through 
the extensions.conf files and not really seeing anything that points me to the 
problem.  


From: Mike Delp via Af 
Sent: Thursday, October 16, 2014 12:18 PM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

Chuck, did you add the extensions that you want to toggle in the dynamic 
members of the queue? 
Mike

On Thu, Oct 16, 2014 at 1:12 PM, Adam Moffett via Af af@afmug.com wrote:


  You've probably thought of this already, but I would make sure the web page 
is not accessible from the internet at large.they've had remote hacks in 
the past.  Then you'll get the bill for the phone calls to Uzbekistan.


So far I am loving it.  Everything else is working perfectly.  

From: Adam Moffett via Af 
Sent: Thursday, October 16, 2014 12:08 PM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk


Wish I could help.  FreePBX makes me mad every time I use itso I don't 
use it anymore.
Never tried that particular feature either.

  I joined the FreePBX forum.  Hopefully I can get some help there once the 
hazing of the newbe is over.

  From: Chuck McCown via Af 
  Sent: Thursday, October 16, 2014 11:36 AM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk

  No actually.  Features.conf has include files of other features files.
  Went through all of them and are not finding many of the feature codes in 
the GUI.
  They must be hiding somewhere else.  

  From: Adam Moffett via Af 
  Sent: Thursday, October 16, 2014 9:29 AM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk


  So there's a *45 defined in features.conf? 

It is one of the built in feature codes.  Others, like announce the 
current time, work.  I cannot imagine why it would ignore a feature code.  

From: Adam Moffett via Af 
Sent: Thursday, October 16, 2014 9:23 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk


I've never used that featurebut I wonder if you have to include a 
special context that knows how to handle the code, like you do with call 
parking.


  It is getting to the asterisk box.  It shows up in the logfile.  But 
it looks like it is treating it like a phone number to me.  Especially if I add 
the extension and group number.  

  From: Nate Burke via Af 
  Sent: Wednesday, October 15, 2014 8:45 PM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk

  Is the *45 getting to the server, or is the dialplan in the voip 
appliance trying to interpret it locally?  

  On 10/15/2014 6:25 PM, Chuck McCown via Af wrote: 
Arrgh, cannot get *45 toggle queue login to work.  No voice prompt. 
 Extension does not appear in queue.  

Static agents work just fine.  Everything else works just fine.  
Voice prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic 
agents.

Any suggestions.

Re: [AFMUG] OT asterisk

[Mike Delp via Af]

You should only have to dial *45 to toggle that extension inot the queue,
and again to toggle out of the queue.  I have 125, and 126 (cordless
phones) as dynamic members.

On Thu, Oct 16, 2014 at 1:23 PM, Chuck McCown via Af af@afmug.com wrote:

   Yep.  At least I filled in the box.  For example, the phone on my desk
 is 200
 So in the box I have 200,0

 We have some static extensions in the box above too and they work as
 expected.
 But when I try to toggle it seems like it is ignoring the feature code
 completely and trying to dial out.
 No announcement, no nothing.  The queue is 700.  So dialing *45200700
 gives me this in the log:

 *45200700' rejected because extension not found in context 'from-internal'.

 That is a dialplan type of problem I would think.  I have been reading
 through the extensions.conf files and not really seeing anything that
 points me to the problem.


  *From:* Mike Delp via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 12:18 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

  Chuck, did you add the extensions that you want to toggle in the dynamic
 members of the queue?
 Mike

 On Thu, Oct 16, 2014 at 1:12 PM, Adam Moffett via Af af@afmug.com wrote:


 You've probably thought of this already, but I would make sure the web
 page is not accessible from the internet at large.they've had remote
 hacks in the past.  Then you'll get the bill for the phone calls to
 Uzbekistan.

So far I am loving it.  Everything else is working perfectly.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 12:08 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 Wish I could help.  FreePBX makes me mad every time I use itso I
 don't use it anymore.
 Never tried that particular feature either.

  I joined the FreePBX forum.  Hopefully I can get some help there once
 the hazing of the newbe is over.

  *From:* Chuck McCown via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 11:36 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

   No actually.  Features.conf has include files of other features files.
 Went through all of them and are not finding many of the feature codes in
 the GUI.
 They must be hiding somewhere else.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 9:29 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 So there's a *45 defined in features.conf?

  It is one of the built in feature codes.  Others, like announce the
 current time, work.  I cannot imagine why it would ignore a feature code.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 9:23 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 I've never used that featurebut I wonder if you have to include a
 special context that knows how to handle the code, like you do with call
 parking.

   It is getting to the asterisk box.  It shows up in the logfile.  But
 it looks like it is treating it like a phone number to me.  Especially if I
 add the extension and group number.

  *From:* Nate Burke via Af af@afmug.com
 *Sent:* Wednesday, October 15, 2014 8:45 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

 Is the *45 getting to the server, or is the dialplan in the voip
 appliance trying to interpret it locally?

 On 10/15/2014 6:25 PM, Chuck McCown via Af wrote:

  Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.
 Extension does not appear in queue.

 Static agents work just fine.  Everything else works just fine.  Voice
 prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic agents.

 Any suggestions.

Re: [AFMUG] OT asterisk

[Chuck McCown via Af]

Yep, that is what I think too.  Just not getting anywhere with *45

From: Mike Delp via Af 
Sent: Thursday, October 16, 2014 12:34 PM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

You should only have to dial *45 to toggle that extension inot the queue, and 
again to toggle out of the queue.  I have 125, and 126 (cordless phones) as 
dynamic members.

On Thu, Oct 16, 2014 at 1:23 PM, Chuck McCown via Af af@afmug.com wrote:

  Yep.  At least I filled in the box.  For example, the phone on my desk is 200
  So in the box I have 200,0

  We have some static extensions in the box above too and they work as expected.
  But when I try to toggle it seems like it is ignoring the feature code 
completely and trying to dial out.
  No announcement, no nothing.  The queue is 700.  So dialing *45200700 gives 
me this in the log:

  *45200700' rejected because extension not found in context 'from-internal'.

  That is a dialplan type of problem I would think.  I have been reading 
through the extensions.conf files and not really seeing anything that points me 
to the problem.  


  From: Mike Delp via Af 
  Sent: Thursday, October 16, 2014 12:18 PM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk

  Chuck, did you add the extensions that you want to toggle in the dynamic 
members of the queue? 
  Mike

  On Thu, Oct 16, 2014 at 1:12 PM, Adam Moffett via Af af@afmug.com wrote:


You've probably thought of this already, but I would make sure the web page 
is not accessible from the internet at large.they've had remote hacks in 
the past.  Then you'll get the bill for the phone calls to Uzbekistan.


  So far I am loving it.  Everything else is working perfectly.  

  From: Adam Moffett via Af 
  Sent: Thursday, October 16, 2014 12:08 PM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk


  Wish I could help.  FreePBX makes me mad every time I use itso I 
don't use it anymore.
  Never tried that particular feature either.

I joined the FreePBX forum.  Hopefully I can get some help there once 
the hazing of the newbe is over.

From: Chuck McCown via Af 
Sent: Thursday, October 16, 2014 11:36 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

No actually.  Features.conf has include files of other features files.
Went through all of them and are not finding many of the feature codes 
in the GUI.
They must be hiding somewhere else.  

From: Adam Moffett via Af 
Sent: Thursday, October 16, 2014 9:29 AM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk


So there's a *45 defined in features.conf? 

  It is one of the built in feature codes.  Others, like announce the 
current time, work.  I cannot imagine why it would ignore a feature code.  

  From: Adam Moffett via Af 
  Sent: Thursday, October 16, 2014 9:23 AM
  To: af@afmug.com 
  Subject: Re: [AFMUG] OT asterisk


  I've never used that featurebut I wonder if you have to include a 
special context that knows how to handle the code, like you do with call 
parking.


It is getting to the asterisk box.  It shows up in the logfile.  
But it looks like it is treating it like a phone number to me.  Especially if I 
add the extension and group number.  

From: Nate Burke via Af 
Sent: Wednesday, October 15, 2014 8:45 PM
To: af@afmug.com 
Subject: Re: [AFMUG] OT asterisk

Is the *45 getting to the server, or is the dialplan in the voip 
appliance trying to interpret it locally?  

On 10/15/2014 6:25 PM, Chuck McCown via Af wrote: 
  Arrgh, cannot get *45 toggle queue login to work.  No voice 
prompt.  Extension does not appear in queue.  

  Static agents work just fine.  Everything else works just fine.  
Voice prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic 
agents.

  Any suggestions.

Re: [AFMUG] OT asterisk

[Mike Delp via Af]

Chuck, give me a call if you want.  I can do a remote session and look into
it.  It is working for me.  314-735-0270

On Thu, Oct 16, 2014 at 2:21 PM, Chuck McCown via Af af@afmug.com wrote:

   Yep, that is what I think too.  Just not getting anywhere with *45

  *From:* Mike Delp via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 12:34 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

  You should only have to dial *45 to toggle that extension inot the
 queue, and again to toggle out of the queue.  I have 125, and 126 (cordless
 phones) as dynamic members.

 On Thu, Oct 16, 2014 at 1:23 PM, Chuck McCown via Af af@afmug.com wrote:

   Yep.  At least I filled in the box.  For example, the phone on my desk
 is 200
 So in the box I have 200,0

 We have some static extensions in the box above too and they work as
 expected.
 But when I try to toggle it seems like it is ignoring the feature code
 completely and trying to dial out.
 No announcement, no nothing.  The queue is 700.  So dialing *45200700
 gives me this in the log:

 *45200700' rejected because extension not found in context
 'from-internal'.

 That is a dialplan type of problem I would think.  I have been reading
 through the extensions.conf files and not really seeing anything that
 points me to the problem.


  *From:* Mike Delp via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 12:18 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

  Chuck, did you add the extensions that you want to toggle in the
 dynamic members of the queue?
 Mike

 On Thu, Oct 16, 2014 at 1:12 PM, Adam Moffett via Af af@afmug.com
 wrote:


 You've probably thought of this already, but I would make sure the web
 page is not accessible from the internet at large.they've had remote
 hacks in the past.  Then you'll get the bill for the phone calls to
 Uzbekistan.

So far I am loving it.  Everything else is working perfectly.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 12:08 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 Wish I could help.  FreePBX makes me mad every time I use itso I
 don't use it anymore.
 Never tried that particular feature either.

  I joined the FreePBX forum.  Hopefully I can get some help there once
 the hazing of the newbe is over.

  *From:* Chuck McCown via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 11:36 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

   No actually.  Features.conf has include files of other features files.
 Went through all of them and are not finding many of the feature codes
 in the GUI.
 They must be hiding somewhere else.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 9:29 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 So there's a *45 defined in features.conf?

  It is one of the built in feature codes.  Others, like announce the
 current time, work.  I cannot imagine why it would ignore a feature code.

  *From:* Adam Moffett via Af af@afmug.com
 *Sent:* Thursday, October 16, 2014 9:23 AM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk


 I've never used that featurebut I wonder if you have to include a
 special context that knows how to handle the code, like you do with call
 parking.

   It is getting to the asterisk box.  It shows up in the logfile.  But
 it looks like it is treating it like a phone number to me.  Especially if I
 add the extension and group number.

  *From:* Nate Burke via Af af@afmug.com
 *Sent:* Wednesday, October 15, 2014 8:45 PM
 *To:* af@afmug.com
 *Subject:* Re: [AFMUG] OT asterisk

 Is the *45 getting to the server, or is the dialplan in the voip
 appliance trying to interpret it locally?

 On 10/15/2014 6:25 PM, Chuck McCown via Af wrote:

  Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.
 Extension does not appear in queue.

 Static agents work just fine.  Everything else works just fine.  Voice
 prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic agents.

 Any suggestions.

[AFMUG] OT asterisk

[Chuck McCown via Af]

Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.  Extension 
does not appear in queue.  

Static agents work just fine.  Everything else works just fine.  Voice prompts, 
IVR, voicemail etc etc.  Just having zero luck with dynamic agents.

Any suggestions.

Re: [AFMUG] OT asterisk

[Nate Burke via Af]

Is the *45 getting to the server, or is the dialplan in the voip 
appliance trying to interpret it locally?


On 10/15/2014 6:25 PM, Chuck McCown via Af wrote:
Arrgh, cannot get *45 toggle queue login to work.  No voice prompt.  
Extension does not appear in queue.
Static agents work just fine.  Everything else works just fine.  Voice 
prompts, IVR, voicemail etc etc.  Just having zero luck with dynamic 
agents.

Any suggestions.

Re: [AFMUG] OT Asterisk question

[George Skorup (Cyber Broadcasting) via Af]

Maybe this helps?

Type=peer is used for outbound trunks. Type=user is used for inbound 
trunks. Type=friend allows both inbound and outbound to work on one trunk.


On 9/17/2014 5:43 PM, Chuck McCown via Af wrote:
No NAT involved.  But there is a router and the Asterisk box is on a 
private IP.


-Original Message- From: George Skorup (Cyber Broadcasting) 
via Af

Sent: Wednesday, September 17, 2014 4:40 PM
To: af@afmug.com
Subject: Re: [AFMUG] OT Asterisk question

Isn't unregistered SIP called peer-SIP? That's what VoipInnovations'
trunks are. We have put them into Asterisk boxes and it works fine. If
you're dealing with NAT, your NAT/router/firewall box probably has a SIP
helper function that rewrites the SIP messages. And/or the switch might
have NAT traversal that can figure everything out for you.

On 9/17/2014 5:25 PM, Chuck McCown via Af wrote:

Does a register string always have to be populated in trunks?

I am doing a new system connected to a Genband/Nortel switch. And 
even though it is our switch, the switch techs here have not done 
this before.


Genband has not been much help either.

Calls will actually come in, ring an extension but no audio cut 
through and the caller will receive fast busy after about 10 
seconds.  Outbound calls on that same trunk go to busy.


Debug shows:
chan_sip.c: Unable to create/find SIP channel for this INVITE on 
some of the calls.


I am thinking this may be an IP problem on the outbound calls. Do not 
recall seeing that on the inbound calls, just no audio cut through.

Re: [AFMUG] OT Asterisk question

[Chuck McCown via Af]

Did not know about friend.  Will give that a try.  

-Original Message- 
From: George Skorup (Cyber Broadcasting) via Af 
Sent: Wednesday, September 17, 2014 4:51 PM 
To: af@afmug.com 
Subject: Re: [AFMUG] OT Asterisk question 


Maybe this helps?

Type=peer is used for outbound trunks. Type=user is used for inbound 
trunks. Type=friend allows both inbound and outbound to work on one trunk.


On 9/17/2014 5:43 PM, Chuck McCown via Af wrote:
No NAT involved.  But there is a router and the Asterisk box is on a 
private IP.


-Original Message- From: George Skorup (Cyber Broadcasting) 
via Af

Sent: Wednesday, September 17, 2014 4:40 PM
To: af@afmug.com
Subject: Re: [AFMUG] OT Asterisk question

Isn't unregistered SIP called peer-SIP? That's what VoipInnovations'
trunks are. We have put them into Asterisk boxes and it works fine. If
you're dealing with NAT, your NAT/router/firewall box probably has a SIP
helper function that rewrites the SIP messages. And/or the switch might
have NAT traversal that can figure everything out for you.

On 9/17/2014 5:25 PM, Chuck McCown via Af wrote:

Does a register string always have to be populated in trunks?

I am doing a new system connected to a Genband/Nortel switch. And 
even though it is our switch, the switch techs here have not done 
this before.


Genband has not been much help either.

Calls will actually come in, ring an extension but no audio cut 
through and the caller will receive fast busy after about 10 
seconds.  Outbound calls on that same trunk go to busy.


Debug shows:
chan_sip.c: Unable to create/find SIP channel for this INVITE on 
some of the calls.


I am thinking this may be an IP problem on the outbound calls. Do not 
recall seeing that on the inbound calls, just no audio cut through.

Re: [AFMUG] OT Asterisk question

[Chuck McCown via Af]

Still no joy but it changed things.
Now, on outbound calls, no busy.  Just nothing.
Inbound still rings the extension, no audio, caller hears fast busy after 
about 10 seconds.


Will work on IP routing issues tomorrow.  Thanks for the suggestions.

-Original Message- 
From: George Skorup (Cyber Broadcasting) via Af

Sent: Wednesday, September 17, 2014 4:51 PM
To: af@afmug.com
Subject: Re: [AFMUG] OT Asterisk question

Maybe this helps?

Type=peer is used for outbound trunks. Type=user is used for inbound
trunks. Type=friend allows both inbound and outbound to work on one trunk.

On 9/17/2014 5:43 PM, Chuck McCown via Af wrote:
No NAT involved.  But there is a router and the Asterisk box is on a 
private IP.


-Original Message- From: George Skorup (Cyber Broadcasting) via Af
Sent: Wednesday, September 17, 2014 4:40 PM
To: af@afmug.com
Subject: Re: [AFMUG] OT Asterisk question

Isn't unregistered SIP called peer-SIP? That's what VoipInnovations'
trunks are. We have put them into Asterisk boxes and it works fine. If
you're dealing with NAT, your NAT/router/firewall box probably has a SIP
helper function that rewrites the SIP messages. And/or the switch might
have NAT traversal that can figure everything out for you.

On 9/17/2014 5:25 PM, Chuck McCown via Af wrote:

Does a register string always have to be populated in trunks?

I am doing a new system connected to a Genband/Nortel switch. And even 
though it is our switch, the switch techs here have not done this before.


Genband has not been much help either.

Calls will actually come in, ring an extension but no audio cut through 
and the caller will receive fast busy after about 10 seconds.  Outbound 
calls on that same trunk go to busy.


Debug shows:
chan_sip.c: Unable to create/find SIP channel for this INVITE on some 
of the calls.


I am thinking this may be an IP problem on the outbound calls. Do not 
recall seeing that on the inbound calls, just no audio cut through.