VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
benoit at plessis.info
Guest
|
 Posted: Thu Jun 05, 2008 1:55 am Post subject: [asterisk-users] handling SIP trunk with limited concurent c |
 |
|
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization 
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a "call-limit=2" in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
--
Benoit |
|
| Back to top |
|
 |
gordon+asterisk at dro...
Guest
|
| Posted: Thu Jun 05, 2008 10:52 am Post subject: [asterisk-users] handling SIP trunk with limited concurent c |
|
|
On Thu, 5 Jun 2008, benoit plessis wrote:
| Quote: | Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a "call-limit=2" in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
|
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon |
|
| Back to top |
|
|
benoit at plessis.info
Guest
|
| Posted: Thu Jun 05, 2008 11:10 am Post subject: [asterisk-users] handling SIP trunk with limited concurent c |
|
|
Gordon Henderson a ?crit :
| Quote: | On Thu, 5 Jun 2008, benoit plessis wrote:
| Quote: | Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a "call-limit=2" in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
|
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
|
Isn't there a risk of getting a CONGESTION message from the other party ?
benoit |
|
| Back to top |
|
|
sherwood.mcgowan at gm...
Guest
|
| Posted: Fri Jun 06, 2008 7:57 am Post subject: [asterisk-users] handling SIP trunk with limited concurent c |
|
|
Gordon Henderson wrote:
| Quote: | On Fri, 6 Jun 2008, Benoit Plessis wrote:
| Quote: | Benoit Plessis a ?crit :
| Quote: | Gordon Henderson a ?crit :
| Quote: | On Thu, 5 Jun 2008, benoit plessis wrote:
| Quote: | Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a "call-limit=2" in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the
end.
So, is there a way with the DIALSTATUS variable to detect a 'full'
peer
?
|
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION
... (You may not need the call-limit=2, if they check for you, then
if at a later date, they increase the limit, then you don't need to
change anything)
Gordon
|
Isn't there a risk of getting a CONGESTION message from the other
party ?
|
|
Isn't CONGESTION what you want? And if the remote SIP provider returns
CONGESTION, then it ought to return it almost instantly too, so
"scanning" a list of SIP providers in-turn, before ending up with a
PSTN interface ought to take fractions of a second..
Just don't confuse CONGESTION with BUSY.
| Quote: | Another problem i foresee is long delay in dialing sequence when
asterisk will have to dial using 4/5 account
before having a working channel
|
See above - the SIP channels ought to return CONGESTION immediately if
they're "full".. (I can't think what else they might return though?)
| Quote: | i think i should look after another sip provider
|
I currently use this in 2 applications - one is to a SIP -> GSM box
with 2 ports, when each port is busy with a call, it returns
CONGESTION, so I try port 1, then port 2, then fall-back to PSTN, (and
I had to tell the box to give me CONGESTION in this case rather than
BUSY!), and in another application where I do it the other way round -
I dial out via 3 analogue lines, but when they're full, Zap/G1 returns
CONGESTION and I then dial out via the Internet and a VoIP service.
Gordon
------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
He's right, you should get congestion in less than a second (unless your
provider is slow anyway in which case you should switch providers anyway).
--
Sherwood McGowan
VoIP / Telecom Solutions
sherwood.mcgowan at gmail.com |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
