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Home > Archive > Voice Over IP in UK > October 2005 > Asterisk is very very slow...
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Asterisk is very very slow...
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| Hi,
When we receive an incoming call it takes asterisk upto 8 seconds to
decide where to route it. This applies to pstn and voip calls.
Fax detection is off.
Is there anything that can be done to improve this, otherwise I'll be
under pressure to dump asterisk....
Any ideas?
Thanks
Matthew
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Matt wrote:
|| Hi,
||
|| When we receive an incoming call it takes asterisk upto 8 seconds to
|| decide where to route it. This applies to pstn and voip calls.
||
|| Fax detection is off.
||
|| Is there anything that can be done to improve this, otherwise I'll be
|| under pressure to dump asterisk....
||
|| Any ideas?
||
|| Thanks
||
|| Matthew
Which flavour of asterisk are you using?
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Jono wrote:
> Matt wrote:
> || Hi,
> ||
> || When we receive an incoming call it takes asterisk upto 8 seconds to
> || decide where to route it. This applies to pstn and voip calls.
> ||
> || Fax detection is off.
> ||
> || Is there anything that can be done to improve this, otherwise I'll be
> || under pressure to dump asterisk....
> ||
> || Any ideas?
> ||
> || Thanks
> ||
> || Matthew
>
> Which flavour of asterisk are you using?
Asterisk at Home 1.5.
Thanks for the reply
Matthew
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Matt wrote:
|| Jono wrote:
||| Matt wrote:
||||| Hi,
|||||
||||| When we receive an incoming call it takes asterisk upto 8 seconds
||||| to decide where to route it. This applies to pstn and voip calls.
|||||
||||| Fax detection is off.
|||||
||||| Is there anything that can be done to improve this, otherwise
||||| I'll be under pressure to dump asterisk....
|||||
||||| Any ideas?
|||||
||||| Thanks
|||||
||||| Matthew
|||
||| Which flavour of asterisk are you using?
||
||
|| Asterisk at Home 1.5.
||
|| Thanks for the reply
||
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|| Matthew
I'm using 1.3 with a sipura SPA3000 to provide the *@H with a connection to
PSTN. I set them up using the wizards at http:\\voxilla.com (fantastic
resource)
Give us a bit more info - SIP provider; Method of connection to PSTN;
Router; Ports forwarded etc.
Do internal calls ring immediately?
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| ||| Matt wrote:
> ||||| Hi,
> |||||
> ||||| When we receive an incoming call it takes asterisk upto 8 seconds
> ||||| to decide where to route it. This applies to pstn and voip calls.
> |||||
> ||||| Fax detection is off.
> |||||
> ||||| Is there anything that can be done to improve this, otherwise
> ||||| I'll be under pressure to dump asterisk....
> |||||
i am not an expert but is that anything to do with CLID?
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| Matt <matthew.humphreys@gmail.com> wrote:
> When we receive an incoming call it takes asterisk upto 8 seconds to
> decide where to route it. This applies to pstn and voip calls.
> Is there anything that can be done to improve this, otherwise I'll be
> under pressure to dump asterisk....
This sort of thing is usually because your X100P, ATA, or whatever is
doing the PSTN interfacing is waiting for Bellcore CLID, and it takes
about six seconds for it to realise it's not going to get any.
Try disabling caller ID. Without knowing what hardware you're using, I
can't really give any further help.
--
I have heard it said that the reason Microsoft is choosing to work with the
government of Nigeria in stopping 419 scammers is that it's easier to rebuild
the Nigerian government and economy than to fix the bugs in Microsoft code.
- Mike Andrews in the Monastery
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| Cheers.
I'm using sipgate and voiptalk and pstn for incoming calls. The
connection to the pstn is via a xp100 clone sourced from ebay.
The same delay occurs on all external calls. Internal call ring
immediately.
Hope this helps.
Matthew
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| Cheers Peter,
The PSTN interface is via an xp100 clone - so how do I tell it to stop
looking for the bellcore ID?
How do I disable caller-id in asterisk?
Thanks for your help,
Matthew
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Matt wrote:
|| Cheers.
||
|| I'm using sipgate and voiptalk and pstn for incoming calls. The
|| connection to the pstn is via a xp100 clone sourced from ebay.
||
|| The same delay occurs on all external calls. Internal call ring
|| immediately.
||
|| Hope this helps.
||
|| Matthew
Have you set up any ring groups? and set up your Incoming Calls page in AMP
(which only affects your clone card) to send any external calls to a
specific extension or ring group?
have you set your Sipgate DID to point to a ring group/ extension?
Gradwell's instructions work very well for Sipgate
http://esupport.gradwell.net/index...._j=subcat&_i=29 -
obviously substituting Gradwell's server details with Sipgate's
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| I've got a couple of ring groups, the incoming calls page points to
extension 200 - it all works, just takes a while to decide what to do!
The sipgate number points are Ring Group 301 - again this works, just
slowly.
Cheers
Matthew
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Matt wrote:
|| I've got a couple of ring groups, the incoming calls page points to
|| extension 200 - it all works, just takes a while to decide what to
|| do!
||
|| The sipgate number points are Ring Group 301 - again this works, just
|| slowly.
||
|| Cheers
||
||
|| Matthew
Well let's hope an expert comes along at some point!
Have you set up your VOIP exactly as described in Gradwell's support?
http://esupport.gradwell.net/index...._j=subcat&_i=29
I've been through all my settings & cannot find anything that would cause
what you describe.
It will be something simple, no doubt.
Have a browse around http://voxilla.com & do a google for "Nerd Vittles"
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> Cheers Peter,
>
> The PSTN interface is via an xp100 clone - so how do I tell it to stop
> looking for the bellcore ID?
>
> How do I disable caller-id in asterisk?
>
> Thanks for your help,
>
> Matthew
>
it is in zapata.conf look for usecallerid=no
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| >From the CLI
Verbosity was 3 and is now 999999
-- Starting simple switch on 'Zap/1-1'
-- Detected ring pattern: 368,248,218
-- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours|s|1:") in new
stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours-nofax|s|1:2")
in new stack
-- Goto (from-pstn-reghours-nofax,s,1)
-- Executing SetVar("Zap/1-1", "intype=GRP-300") in new stack
-- Executing Cut("Zap/1-1", "intype=intype|-|1") in new stack
-- Executing GotoIf("Zap/1-1", "0?4:5") in new stack
-- Goto (from-pstn-reghours-nofax,s,5)
-- Executing GotoIf("Zap/1-1", "1?6:7") in new stack
-- Goto (from-pstn-reghours-nofax,s,6)
-- Executing Goto("Zap/1-1", "ext-group|300|1") in new stack
-- Goto (ext-group,300,1)
-- Executing Macro("Zap/1-1", "rg-group|45||200-205") in new stack
-- Executing GotoIf("Zap/1-1", "0?3:2") in new stack
-- Goto (macro-rg-group,s,2)
-- Executing SetCIDName("Zap/1-1", "") in new stack
-- Executing SetVar("Zap/1-1", "RGPREFIX=") in new stack
-- Executing SetCIDName("Zap/1-1", "") in new stack
-- Executing SetVar("Zap/1-1", "RecordMethod=Group") in new stack
-- Executing Macro("Zap/1-1", "record-enable|300|Group") in new
stack
-- Executing GotoIf("Zap/1-1", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("Zap/1-1", "0?5:8") in new stack
-- Goto (macro-record-enable,s,8)
-- Executing GotoIf("Zap/1-1", "1?9:12") in new stack
-- Goto (macro-record-enable,s,9)
-- Executing AGI("Zap/1-1", "recordingcheck") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing SetVar("Zap/1-1",
"CALLFILENAME=g300-20051020-104025-1129801219.187") in new stack
-- Executing Goto("Zap/1-1", "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("Zap/1-1", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("Zap/1-1", "NO RECORDING NEEDED") in new stack
-- Executing Macro("Zap/1-1", "dial|45|tr|200-205") in new stack
-- Executing GotoIf("Zap/1-1", "1?4:2") in new stack
-- Goto (macro-dial,s,4)
-- Executing AGI("Zap/1-1", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 4
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1129801219.187
-- dialparties.agi: channel = Zap/1-1
-- dialparties.agi: callerid = unknown
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = Zap
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID is not set
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Added extension 205 to extension map
-- dialparties.agi: Extension 205 cf is disabled
-- dialparties.agi: Extension 200 cf is disabled
-- dialparties.agi: Extension 205 do not disturb is disabled
-- dialparties.agi: Extension 200 do not disturb is disabled
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 205 has call waiting disabled
dialparties.agi: Extension 200 has call waiting disabled
-- dialparties.agi: DbDel CALLTRACE/205 - Caller ID is not defined
-- dialparties.agi: DbDel CALLTRACE/200 - Caller ID is not defined
dialparties.agi: Dial string is SIP/205&SIP/200|45|tr
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("Zap/1-1", "SIP/205&SIP/200|45|tr") in new stack
-- Called 205
-- Called 200
-- SIP/200-504b is ringing
-- SIP/205-82d6 is ringing
Many thanks
Matthew
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