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Robert Withrow

2004-Jun-15 11:35 UTC

[Asterisk-Users] Grandstreams randomly go busy with Asterisk?

I've searched the lists but I didn't find anything exactly like this.

I have two Grandstream BT101 phones connected to an Asterisk. 
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy.  Dialing that
phone gives:

    -- Executing Macro("SIP/24567-7856",
"dialphone|SIP/27654") in new stack
    -- Executing Dial("SIP/24567-7856", "SIP/27654|10|tr")
in new stack
Jun 15 14:23:41 NOTICE[1343506]: app_dial.c:536 dial_exec: Unable to create
channel of type 'SIP'
  == Everyone is busy at this time

But dialing in the other direction (from the busy phone out) gives
normal (good) results:
 
    -- Executing Macro("SIP/27654-6e2b",
"dialphone|SIP/24567") in new stack
    -- Executing Dial("SIP/27654-6e2b", "SIP/24567|10|tr")
in new stack
    -- Called 24567

I have noticed that when the problem is happening I see this:

CLI> sip show peers
Name/username    Host                 Mask             Port     Status
24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
27654/27654      (Unspecified)   (D)  255.255.255.255  0        Unmonitored

Rebooting the offending phone always fixes the problem for a while. 
After rebooting I see:

CLI> sip show peers
Name/username    Host                 Mask             Port     Status
24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
27654/27654      192.168.2.254   (D)  255.255.255.255  5060     Unmonitored

The BT101s are running 1.0.4.55.  Asterisk is 0.9.0.

Any suggestions?

-- 
Robert Withrow, BWithrow@NortelNetworks.com, +1 978 288 8256, ESN 248

Kyle Hagan

2004-Jun-15 11:53 UTC

head link

[Asterisk-Users] Grandstreams randomly go busy with Asterisk?

I have one of 2 doing the same thing.

Kyle

Robert Withrow wrote:
>I've searched the lists but I didn't find anything exactly like
this.
>
>I have two Grandstream BT101 phones connected to an Asterisk. 
>Periodically, for reasons that I can't determine, one or the other (or
>both) of the BT101s decide(s) to go on permanent busy.  Dialing that
>phone gives:
>
>    -- Executing Macro("SIP/24567-7856",
"dialphone|SIP/27654") in new stack
>    -- Executing Dial("SIP/24567-7856",
"SIP/27654|10|tr") in new stack
>Jun 15 14:23:41 NOTICE[1343506]: app_dial.c:536 dial_exec: Unable to create
channel of type 'SIP'
>  == Everyone is busy at this time
>
>But dialing in the other direction (from the busy phone out) gives
>normal (good) results:
> 
>    -- Executing Macro("SIP/27654-6e2b",
"dialphone|SIP/24567") in new stack
>    -- Executing Dial("SIP/27654-6e2b",
"SIP/24567|10|tr") in new stack
>    -- Called 24567
>
>I have noticed that when the problem is happening I see this:
>
>CLI> sip show peers
>Name/username    Host                 Mask             Port     Status
>24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
>27654/27654      (Unspecified)   (D)  255.255.255.255  0        Unmonitored
>
>Rebooting the offending phone always fixes the problem for a while. 
>After rebooting I see:
>
>CLI> sip show peers
>Name/username    Host                 Mask             Port     Status
>24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
>27654/27654      192.168.2.254   (D)  255.255.255.255  5060     Unmonitored
>
>The BT101s are running 1.0.4.55.  Asterisk is 0.9.0.
>
>Any suggestions?
>
>  
>

Brian Buhrow

2004-Jun-15 13:09 UTC

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[Asterisk-Users] Re: Grandstreams randomly go busy with Asterisk?

Hello.  I've seen this behavior.  What happens is that the
Grandstreams forget to continue registering with Asterisk after a while.  I
bet when you find this happening, that sip show peers doesn't show ext/ext
ip address for the one that isn't working.
	You can work around the problem by explicitly telling Asterisk how to
dial the GS by giving it an explicit IP address in its sip.conf extension
entry.  Alternatively, you can upgrade the Grandstream to a newer load of
firmware.  I'm running 1.0.4.68 on my HT286, and it seems to behave much
better.
	I got my firmware load from:
http://www.voiptalk.org/products/download/
They seem to have 1.0.4.63, and 1.0.5.0, but not 1.0.4.68 anymore.  

Hope that helps.
-Brian


	
I've searched the lists but I didn't find anything exactly like this.

I have two Grandstream BT101 phones connected to an Asterisk. 
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy.  Dialing that
phone gives:

    -- Executing Macro("SIP/24567-7856",
"dialphone|SIP/27654") in new stack
    -- Executing Dial("SIP/24567-7856", "SIP/27654|10|tr")
in new stack
Jun 15 14:23:41 NOTICE[1343506]: app_dial.c:536 dial_exec: Unable to create
channel of type 'SIP'
  == Everyone is busy at this time

But dialing in the other direction (from the busy phone out) gives
normal (good) results:
 
    -- Executing Macro("SIP/27654-6e2b",
"dialphone|SIP/24567") in new stack
    -- Executing Dial("SIP/27654-6e2b", "SIP/24567|10|tr")
in new stack
    -- Called 24567

I have noticed that when the problem is happening I see this:

CLI> sip show peers
Name/username    Host                 Mask             Port     Status
24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
27654/27654      (Unspecified)   (D)  255.255.255.255  0        Unmonitored

Rebooting the offending phone always fixes the problem for a while. 
After rebooting I see:

CLI> sip show peers
Name/username    Host                 Mask             Port     Status
24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
27654/27654      192.168.2.254   (D)  255.255.255.255  5060     Unmonitored

The BT101s are running 1.0.4.55.  Asterisk is 0.9.0.

Any suggestions?

Adam Goryachev

2004-Jun-15 16:50 UTC

head link

[Asterisk-Users] Grandstreams randomly go busy with Asterisk?

On Wed, 2004-06-16 at 04:35, Robert Withrow wrote:> I've searched the lists but I didn't find anything exactly like
this.
> 
> I have two Grandstream BT101 phones connected to an Asterisk. 
> Periodically, for reasons that I can't determine, one or the other (or
> both) of the BT101s decide(s) to go on permanent busy.  Dialing that
> phone gives:
> 
>     -- Executing Macro("SIP/24567-7856",
"dialphone|SIP/27654") in new stack
>     -- Executing Dial("SIP/24567-7856",
"SIP/27654|10|tr") in new stack
> Jun 15 14:23:41 NOTICE[1343506]: app_dial.c:536 dial_exec: Unable to create
channel of type 'SIP'
>   == Everyone is busy at this time
> But dialing in the other direction (from the busy phone out) gives
> normal (good) results:
>  
>     -- Executing Macro("SIP/27654-6e2b",
"dialphone|SIP/24567") in new stack
>     -- Executing Dial("SIP/27654-6e2b",
"SIP/24567|10|tr") in new stack
>     -- Called 24567
> 
> I have noticed that when the problem is happening I see this:
> 
> CLI> sip show peers
> Name/username    Host                 Mask             Port     Status
> 24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
> 27654/27654      (Unspecified)   (D)  255.255.255.255  0        Unmonitored
I had the same problem when I first got my phone, I can't be sure it is
solved, but it seemed to be.
> Rebooting the offending phone always fixes the problem for a while. 
> After rebooting I see:
> 
> CLI> sip show peers
> Name/username    Host                 Mask             Port     Status
> 24567/24567      192.168.2.253   (D)  255.255.255.255  5060     Unmonitored
> 27654/27654      192.168.2.254   (D)  255.255.255.255  5060     Unmonitored
Sure, it re-registers on re-boot.
> The BT101s are running 1.0.4.55.  Asterisk is 0.9.0.
Dunno what version I'm using, it is the one that came with the phone. I
am using Asterisk CVS HEAD.
> Any suggestions?
What 'seemed' to fix the problem was to change the expiry time from 1 (I
wanted it to re-register every minute. This was when I had the problem.
I changed it to some higher number (maybe 5 or 10) and this seemed to
solve the problem.

Regards,
Adam

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