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nunne
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cant dial more than 5 sip phones at once

Hi,

I have added a "group" for example GROUP=SIP/1&SIP/2&SIP/3 ... to 9.
So 9 phones.

But it only dials 4-5 of them.
The others get:

__sip_xmit: sip_xmit of 0x1fb5be0 (len 835) to 192.168.100.254:5060 returned -1: Operation not permitted

And it seems like the phones are ringing, but no one can pick up the call.

Im using stable build 424 for bri4.
Anyone have any ideas what this might be?!

I have tried with different sip-options like "canreinvite" yes/no. dialing with dial option "r" to disable early media.

but nothing seems work sad


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nunne
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Re: cant dial more than 5 sip phones at once

Can anyone please confirm if this is a limitation in the blackfin/switchfin or just in the version of asterisk im using?

or perhaps its possible to "spawn" more asterisk-processes?

Because from what I can tell if I have alot of phones. Say 20 that subscribes to each others extensions. I only seem to be able to dial 3-4 of them at the same time.

When Im remove my subscriptions I can dial 7 phones. But nothing more.
If i first Dial 4 phones. Then try to Dial a group with an additional 4 only 3 of them will ring.

And this is a very weird limitation.. And I hope its only in Rev.424 (which im using)..

A month or two ago I used the trunk version. But it froze from time to time. Thats why Im using rev424.


Funny is that I have tried for example to make 28 concurrent sip calls (sip to outisde via sip-trunk) without any problem. So I know the hardware is capable of this sad


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admin
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Re: cant dial more than 5 sip phones at once

Hi Nunne,

The performance of BR4 Appliance should be well above 10 concurrent calls so most probably this is related with the Asterisk settings.
You can measure the memory/CPU load of your box.
Is there a sign of overloading?
 

If you suspect the exact version of Linux kernel/Asterisk used in Rev 424 please send me backup of your configuration so I test it on the trunk.

Best Regards
Dimitar


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nunne
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Re: cant dial more than 5 sip phones at once

Dimitar:

This seems to be an issue with asterisk 1.4.32. So it does not occur in trunk. (I tried smile).
And it's only a problem with the new line of sip phones from panasonic UT133/136 113/123.
Which are really nice, function wise, if you havent checked them out wink Since they fill a gap that has been missing when comparing to other traditional pbx:es (like panasonic, avaya and what not). Namely 24 blf buttons smile

But upgrading to trunk solves the sip xmit problem and all phones now dial.
There is a delay between the "dial" actually commits.. and same delay before and answer goes through. But i think this is directly related to the phone, and not asterisk.

The phones are quite new.. So one might expect a few quirks are still there tongue

But I have a problem with the trunk version (and is why i havent used it at customers).
Is that I get SEGV in asterisk.. So I cant do "asterisk -r". I can still access it via SSH for a little while. Then it freezes (and can give a error message within a second trying to SSH into it) or it reboots.

You have noticed this behaviour.. Because I have a customer with 20 panasonic phones. So I have a dilema here big_smile haha..


Best Regards
Anders Wallin


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