We are testing our Asterisk server prior to deployment. The server has a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and one PRI for local calls. We are using sipp from two different stations routing a test number out the LD lines and another test number out the PRI line. We can not get more then 23 total active calls to connect to the test numbers, the test numbers terminate to another PBX that we can monitor. We have dialed out using cell phones to this other PBX while the test is happening and it connects, meaning it has more then 23 active calls on it. If we place more then 23 calls then it seems to 'queue' the extra calls, though not all of the extra calls complete after we stop adding new calls. They seem to get stuck in a queue or lost. We will send 200 calls through the Asterisk server and all but about 20 do eventually complete. Those 20 or so are stuck as Asterisk thinks the channels are busy with the calls when in fact there are no 'real' calls on the server. We can send 30 calls through the LD or PRI and only 23 are actually connected at a time. We can send 30 calls to both LD and PRI at the same time and still only a mixture of 23 calls are actually active at one time. So our issue seems to be located in our Asterisk server. Is there a way to limit or throttle an Asterisk server so that it will not place more then 'x' calls? We need to be able to support 48 calls. Any ideas?
On Tue, 2005-08-16 at 23:53 -0700, Pudenz, Duane wrote:> We are testing our Asterisk server prior to deployment. The server has > a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and > one PRI for local calls. > > We are using sipp from two different stations routing a test number out > the LD lines and another test number out the PRI line. > > We can not get more then 23 total active calls to connect to the test > numbers, the test numbers terminate to another PBX that we can monitor. > We have dialed out using cell phones to this other PBX while the test is > happening and it connects, meaning it has more then 23 active calls on > it.Include more info like your extensions.conf zaptel.conf and zapata.conf (not the comments) and we might be able to offer some hints... Also, some output from the CLI might be helpful.... that would probably tell you why the calls are failing already.... Regards, Adam
It looks like you are sending calls out over one port. To help you out, we will
need to look at your extensions.conf and zapata.conf. My hunch is that you are
dialing out using something like
Dial(zap/g3/${EXTEN},20,) where the group of channels you're using is on one
port of your Digium card.
If my math is right, you should be able to send 69 calls long distance, and 23
local calls at a time with no failover.
Louie Tarpo
Adam Aircraft
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Pudenz,
Duane
Sent: Wednesday, August 17, 2005 12:53 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can not dial more then 23 calls
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number out
the LD lines and another test number out the PRI line.
We can not get more then 23 total active calls to connect to the test
numbers, the test numbers terminate to another PBX that we can monitor.
We have dialed out using cell phones to this other PBX while the test is
happening and it connects, meaning it has more then 23 active calls on
it.
If we place more then 23 calls then it seems to 'queue' the extra calls,
though not all of the extra calls complete after we stop adding new
calls. They seem to get stuck in a queue or lost. We will send 200
calls through the Asterisk server and all but about 20 do eventually
complete. Those 20 or so are stuck as Asterisk thinks the channels are
busy with the calls when in fact there are no 'real' calls on the
server.
We can send 30 calls through the LD or PRI and only 23 are actually
connected at a time. We can send 30 calls to both LD and PRI at the
same time and still only a mixture of 23 calls are actually active at
one time.
So our issue seems to be located in our Asterisk server. Is there a way
to limit or throttle an Asterisk server so that it will not place more
then 'x' calls?
We need to be able to support 48 calls.
Any ideas?
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Eric Wieling aka ManxPower
2005-Aug-18 07:23 UTC
[Asterisk-Users] Can not dial more then 23 calls
Pudenz, Duane wrote:> We are testing our Asterisk server prior to deployment. The server has > a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and > one PRI for local calls. > > We are using sipp from two different stations routing a test number out > the LD lines and another test number out the PRI line. > > We can not get more then 23 total active calls to connect to the test > numbers, the test numbers terminate to another PBX that we can monitor. > We have dialed out using cell phones to this other PBX while the test is > happening and it connects, meaning it has more then 23 active calls on > it. > > If we place more then 23 calls then it seems to 'queue' the extra calls, > though not all of the extra calls complete after we stop adding new > calls. They seem to get stuck in a queue or lost. We will send 200 > calls through the Asterisk server and all but about 20 do eventually > complete. Those 20 or so are stuck as Asterisk thinks the channels are > busy with the calls when in fact there are no 'real' calls on the > server. > > We can send 30 calls through the LD or PRI and only 23 are actually > connected at a time. We can send 30 calls to both LD and PRI at the > same time and still only a mixture of 23 calls are actually active at > one time. > > So our issue seems to be located in our Asterisk server. Is there a way > to limit or throttle an Asterisk server so that it will not place more > then 'x' calls?That is the value of ${DIALSTATUS} or ${HANGUPCAUSE} for the failed calls?