[asterisk-users] Failover SIP logic
I need some extensions logic assistance, I'm trying to dial out one of multiple
SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call
per trunk) and roll over to a second or third depending on that busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the 1st
trunk is busy)
extensions.conf:
[globals]
trunk_1 => SIP/trunk1
trunk_2 => SIP/trunk2
trunk_3 => SIP/trunk3
[macro-trunkdial]
exten => s,1,Dial(${trunk_1}/${ARG1})
exten => s,2,Hangup()
exten => s,102,Dial(${trunk_2}/${ARG1})
exten => s,103,Hangup()
exten => s,203,Dial(${trunk_3}/${ARG1})
exten => s,204,Hangup()
[from-internal]
exten => _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
exten => _1NXXNXX,1,Macro(trunkdial,+${EXTEN})
sip.conf:
[trunk1]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk2]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk3]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
Here's asterisk output when someone dials out:
Executing [EMAIL PROTECTED]:1] Macro("SIP/6001-007e2840",
"trunkdial|+1xx") in new stack
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/6001-007e2840",
"SIP/trunk1/+1xx") in new stack
[Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to
peer 'trunk1' rejected due to usage limit of 1
-- Couldn't call trunk1/+1xx
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("SIP/6001-007e2840", "") in new
stack
I don't want the dialplan to cascade like:
exten => 1,dial...
exten => 2,dial...
Because if the remote end hangs up I don't want it going to priority 2 to dial
out again(in case my user doesn't hit hang-up on their end) so I need logic to
detect a busy channel and jump to the next section..
Thanks for any help.
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Re: [asterisk-users] Failover SIP logic
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Jeremy Mann wrote:
> I need some extensions logic assistance, I'm trying to dial out one of
> multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
> allow 1 call per trunk) and roll over to a second or third depending on that
> busy status
>
> Here's what I've got for a macro thusfar, but it's not working(fails if the
> 1st trunk is busy)
> extensions.conf:
>
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
>
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()
>
> [from-internal]
> exten => _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
> exten => _1NXXNXX,1,Macro(trunkdial,+${EXTEN})
>
> sip.conf:
>
> [trunk1]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxx
> call-limit=1
>
> [trunk2]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxx
> call-limit=1
>
> [trunk3]
> host=xxx.xxx.xxx.xxx
> port=5060
> type=peer
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> nat=no
> fromuser=+xxx
> call-limit=1
>
> Here's asterisk output when someone dials out:
> Executing [EMAIL PROTECTED]:1] Macro("SIP/6001-007e2840",
> "trunkdial|+1xx") in new stack
> -- Executing [EMAIL PROTECTED]:1] Dial("SIP/6001-007e2840",
> "SIP/trunk1/+1xx") in new stack
> [Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to
> peer 'trunk1' rejected due to usage limit of 1
> -- Couldn't call trunk1/+1xx
> == Everyone is busy/congested at this time (0:0/0/0)
> -- Executing [EMAIL PROTECTED]:2] Hangup("SIP/6001-007e2840", "") in new
> stack
>
> I don't want the dialplan to cascade like:
>
> exten => 1,dial...
> exten => 2,dial...
>
> Because if the remote end hangs up I don't want it going to priority 2 to
> dial out again(in case my user doesn't hit hang-up on their end) so I need
> logic to detect a busy channel and jump to the next section..
If you have this:
exten => _X.,1,Dial(SIP/trunk1)
exten => _X.,2,Dial(SIP/trunk2)
exten => _X.,3,Dial(SIP/trunk3)
then, only if trunk is busy, will it go to trunk2, if thats busy, it will go to
trunk 3.
Reason is, is that control wont return to the dial plan(except h) if the call
was
successfull. SO if the call went through on trunk 1, then it will exit, not
dial trunk2
or trunk3. So this dial plan will work. But its very sequential, i.e. will
try trunk1,
then trunk2, then trunk3. If you want to replicate round-robin, r, then do
this:
[globals]
IPt=trunk1-trunk2-trunk3
COUNTt=0
NoOfChannels=3
[just-an-idea]
exten => _X.,1,Gotoif($["${COUNTt}" = "${NoOfChannels}"] ? 2:3)
exten => _X.,2,SetGlobalVar(COUNTt=0])
exten => _X.,3,SetGlobalVar(COUNTt=$[${COUNTt}+1])
exten => _X.,4,Set(tr=${CUT(IPt,-,${COUNTt})})
exten => _X.,5,Dial(SIP/tr/${EXTEN})
modify at your leisure. So if you get a few more trunks, you just change
"NoOfChannels"
--
thanks,
Yusuf
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Re: [asterisk-users] Failover SIP logic
Ciao Jeremy,
> I need some extensions logic assistance, I'm trying to dial out one of
> multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
> allow 1 call per trunk) and roll over to a second or third depending on that
> busy status
>
> Here's what I've got for a macro thusfar, but it's not working(fails if the
> 1st trunk is busy) extensions.conf:
>
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
>
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()
Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.
HTH,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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Re: [asterisk-users] Failover SIP logic
Asterisk 1.4.11
Sorry, meant to include that
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Spadaccini
Sent: Monday, September 10, 2007 10:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Failover SIP logic
Ciao Jeremy,
> I need some extensions logic assistance, I'm trying to dial out one of
> multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
> allow 1 call per trunk) and roll over to a second or third depending on that
> busy status
>
> Here's what I've got for a macro thusfar, but it's not working(fails if the
> 1st trunk is busy) extensions.conf:
>
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
>
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()
Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.
HTH,
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
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