asterisk users - Jun 2005 - timeout on incoming PRI call

hello,

i've an asterisk box which is connected to an E1/PRI via a TE110P card.

incoming calls from mobile phones where the number is transfered as a
whole block work fine, but when dialing from an analog or ISDN line to
the asterisk box there is a timeout of about 3-5 seconds.

originally my incoming context looked like:
exten => _X.,1,Dial(SIP/${EXTEN}@domain.tld)

so i assumed that the timeout was caused because asterisk didn't know if
the number is complete or if further digits are sent, so i now replaced
this config with a realtime config which lists each number individually.

even when using this realtimeconfig (which includes only 'full' numbers
- no wildcards, etc.) it seems that asterisk does the db-lookup after
the timeout - so the delay is still there, although the dialed number is
distinct.

any suggestions about the cause of this problem / how to solve it?

cu
/gst

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I am not sure about E1 but it _should_ be the same. The Dialed Number is usually
transferred in 'a whole block' as the Telco passing the call to you has
already routed that call to you.  What type of switch are you connected to?? 
Could your switch be expecting a ACK of some sort from *?? Have you turned on
debugging? (pri debug span 1).

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> G?nther Starnberger
> Sent: Wednesday, June 29, 2005 10:13 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] timeout on incoming PRI call
> 
> hello,
> 
> i've an asterisk box which is connected to an E1/PRI via a 
> TE110P card.
> 
> incoming calls from mobile phones where the number is 
> transfered as a whole block work fine, but when dialing from 
> an analog or ISDN line to the asterisk box there is a timeout 
> of about 3-5 seconds.
> 
> originally my incoming context looked like:
> exten => _X.,1,Dial(SIP/${EXTEN}@domain.tld)
> 
> so i assumed that the timeout was caused because asterisk 
> didn't know if the number is complete or if further digits 
> are sent, so i now replaced this config with a realtime 
> config which lists each number individually.
> 
> even when using this realtimeconfig (which includes only 
> 'full' numbers
> - no wildcards, etc.) it seems that asterisk does the 
> db-lookup after the timeout - so the delay is still there, 
> although the dialed number is distinct.
> 
> any suggestions about the cause of this problem / how to solve it?
> 
> cu
> /gst
> 
>

Alexander Lopez wrote:

Hello,
> I am not sure about E1 but it _should_ be the same. The Dialed Number
> is usually transferred in 'a whole block' as the Telco passing the
> call to you has already routed that call to you.  What type of switch
> are you connected to??  Could your switch be expecting a ACK of some
> sort from *?? Have you turned on debugging? (pri debug span 1). 
when i do a "pri debug" i only see the incoming call after the delay.
in the debugging output i already see the "complete" number - so
either
our 'upstream' isn't sending the call to us until some timeout
happens
or the zaptel driver doesn't log before the timeoout.

on the other hand i just did some further tests: when i call from an
analog time the call-setup needs 10 seconds when i have a wildcard
entry in my extensions.conf, but only 5 seconds when i list each entry
individually (with the realtime config). at first i didn't notice this
as 5 seconds for the callsetup are still rather long when compared to
about 2 seconds for a call to another landline.

there is a 3 seconds delay hardcoded in the chan_zap.c source, but this
is only called if the number is ambiguous (which isn't the case here).
i'll check with our telco first how long they need for the forwarding of
the call before i continue blaming asterisk :)

cu
/gst

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