thr3ads.net - asterisk users - [asterisk-users] All calls want to go out only on interface ZAP/g0 [Sep 2008]

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Mark Best

2008-Sep-04 00:12 UTC

[asterisk-users] All calls want to go out only on interface ZAP/g0

I have a legacy PBX that I want to slowly move off of. Below is a
diagram of what I want my setup to look-like after testing.

Old Mitel---24 Channels---Asterisk---PSTN
|                              |       |
Ext. 3060              SIP. 2054      Cellular
 

No matter my dial-plan logic; all calls seem to default to ZAP/g0. I
can't seem to get any calls to go directly to ZAP/g2.

NOTE: For testing 11# is added to the front of all calls coming from the
PSTN.

PSTN to Asterisk (g0) from-pstn

Asterisk to LegacyPBX (g2) from-internal

-------------
-Deleted all Outbound routes.
-Re-writing Zaptel to only include Port 1 & Port 3 (No 'red alarms'
in
zttool)
AMI, D4, E & M and Wink - Master Timing on Port 3 (source from Port 1).
-Added 'To_PSTN' on port g0.
-Added 'To_LegacyPBX' on port g2.
-Added New 'Catch all Route' to PSTN and to LegacyPBX (.)

Test Performed: SIP to Cellular = Worked

- Executing [s at macro-dialout-trunk:20] Dial("SIP/2054-b7801d70",
"ZAP/g0/2085553870|300|") in new stack
-- Called g0/2085553870
-- Zap/1-1 answered SIP/2054-b7801d70

Test Performed: SIP to 3060 = Failed
SIP to 3060 seems to go out g0 then came back in from g0 

-- Goto (macro-dialout-trunk,s,17)
-- Executing [s at macro-dialout-trunk:17] Macro("SIP/2054-b7801d70",
"dialout-trunk-predial-hook|") in new stack
-- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/2054-b7801d70",
"0?bypass|1") in new stack
-- Executing [s at macro-dialout-trunk:19] GotoIf("SIP/2054-b7801d70",
"0?customtrunk") in new stack
-- Executing [s at macro-dialout-trunk:20] Dial("SIP/2054-b7801d70",
"ZAP/g0/3060|300|") in new stack
-- Called g0/3060
-- Starting simple switch on 'Zap/24-1'
-- Zap/1-1 answered SIP/2054-b7801d70
== Unknown extension '11#3060' in context 'from-pstn' requested
-- <Zap/24-1> Playing 'ss-noservice' (language 'en')

Added 11#3060 to both PSTN and LegacyPBX dialplan
Test Performed: SIP to 3060 = Failed
-Goes out g0 and comes back unknown.

-- Executing [s at macro-dialout-trunk:13] Set("SIP/2054-b7802098",
"OUTNUM=3060") in new stack
-- Executing [s at macro-dialout-trunk:14] Set("SIP/2054-b7802098",
"custom=ZAP/g0") in new stack
-- Executing [s at macro-dialout-trunk:15] GotoIf("SIP/2054-b7802098",
"1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s at macro-dialout-trunk:17] Macro("SIP/2054-b7802098",
"dialout-trunk-predial-hook|") in new stack
-- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/2054-b7802098",
"0?bypass|1") in new stack
-- Executing [s at macro-dialout-trunk:19] GotoIf("SIP/2054-b7802098",
"0?customtrunk") in new stack
-- Executing [s at macro-dialout-trunk:20] Dial("SIP/2054-b7802098",
"ZAP/g0/3060|300|") in new stack
-- Called g0/3060
-- Starting simple switch on 'Zap/24-1'
-- Zap/1-1 answered SIP/2054-b7802098
== Unknown extension '11#3060' in context 'from-pstn' requested
-- <Zap/24-1> Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/1-1'

NOTE: For testing 11# is added to the front of all calls comming from
the PSTN.

Trying a Misc. Destination & Inbound route combination:
Added Misc Destination 811#3060
Changed DialPLan on LegacyPBX

.
11#3060
8|11#3060
8|11.
8|.
8|1NXXNXXXXXX
8|NXXXXXX

Added 'inbound route' of 11#3060 - to go to 'Misc dest 811#3060'
Test Performed: SIP to 3060 = Failed

-- Zap/1-1 answered SIP/2054-b7801bf0
== Unknown extension '11#30603060' in context 'from-pstn'
requested
-- <Zap/24-1> Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/24-1'

Added only 8|. to dial plan
Test Performed: SIP to 3060 = Failed
-Fast Busy

-- Executing [s at macro-dialout-trunk:20] Dial("Zap/24-1",
"ZAP/g0/811#|300|") in new stack
-- Called g0/811#

What a mess! What else can I try?

 

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Paul Hales

2008-Sep-04 00:50 UTC

head link

[asterisk-users] All calls want to go out only on interface ZAP/g0

Slightly confused - this isn't to hard to do (I have done it quite a few 
times before )

The dialplan to do this should only be several lines long. Can you 
provide a copy of your dialplan?

PaulH


Mark Best wrote:>
> I have a legacy PBX that I want to slowly move off of. Below is a 
> diagram of what I want my setup to look-like after testing.
>
> Old Mitel---24 Channels---Asterisk---PSTN
> |                              |       |
> Ext. 3060              SIP. 2054      Cellular
>  
>
> No matter my dial-plan logic; all calls seem to default to ZAP/g0. I 
> can't seem to get any calls to go directly to ZAP/g2.
>
> NOTE: For testing 11# is added to the front of all calls coming from 
> the PSTN.
>
> PSTN to Asterisk (g0) from-pstn
>
> Asterisk to LegacyPBX (g2) from-internal
>
> -------------
> -Deleted all Outbound routes.
> -Re-writing Zaptel to only include Port 1 & Port 3 (No 'red
alarms' in
> zttool)
> AMI, D4, E & M and Wink - Master Timing on Port 3 (source from Port 1).
> -Added 'To_PSTN' on port g0.
> -Added 'To_LegacyPBX' on port g2.
> -Added New 'Catch all Route' to PSTN and to LegacyPBX (.)
>
> *Test Performed: SIP to Cellular = Worked*
>
> - Executing [s at macro-dialout-trunk:20]
Dial("SIP/2054-b7801d70", "ZAP/g0/2085553870|300|") in new
stack
> -- Called g0/2085553870
> -- Zap/1-1 answered SIP/2054-b7801d70
>
> *Test Performed: SIP to 3060 = Failed*
> SIP to 3060 seems to go out g0 then came back in from g0
>
> -- Goto (macro-dialout-trunk,s,17)
> -- Executing [s at macro-dialout-trunk:17]
Macro("SIP/2054-b7801d70", "dialout-trunk-predial-hook|") in
new stack
> -- Executing [s at macro-dialout-trunk:18]
GotoIf("SIP/2054-b7801d70", "0?bypass|1") in new stack
> -- Executing [s at macro-dialout-trunk:19]
GotoIf("SIP/2054-b7801d70", "0?customtrunk") in new stack
> -- Executing [s at macro-dialout-trunk:20]
Dial("SIP/2054-b7801d70", "ZAP/g0/3060|300|") in new stack
> -- Called g0/3060
> -- Starting simple switch on 'Zap/24-1'
> -- Zap/1-1 answered SIP/2054-b7801d70
> == Unknown extension '11#3060' in context 'from-pstn'
requested
> -- <Zap/24-1> Playing 'ss-noservice' (language 'en')
>
> Added 11#3060 to both PSTN and LegacyPBX dialplan
> *Test Performed: SIP to 3060 = Failed*
> -Goes out g0 and comes back unknown.
>
> -- Executing [s at macro-dialout-trunk:13]
Set("SIP/2054-b7802098", "OUTNUM=3060") in new stack
> -- Executing [s at macro-dialout-trunk:14]
Set("SIP/2054-b7802098", "custom=ZAP/g0") in new stack
> -- Executing [s at macro-dialout-trunk:15]
GotoIf("SIP/2054-b7802098", "1?gocall") in new stack
> -- Goto (macro-dialout-trunk,s,17)
> -- Executing [s at macro-dialout-trunk:17]
Macro("SIP/2054-b7802098", "dialout-trunk-predial-hook|") in
new stack
> -- Executing [s at macro-dialout-trunk:18]
GotoIf("SIP/2054-b7802098", "0?bypass|1") in new stack
> -- Executing [s at macro-dialout-trunk:19]
GotoIf("SIP/2054-b7802098", "0?customtrunk") in new stack
> -- Executing [s at macro-dialout-trunk:20]
Dial("SIP/2054-b7802098", "ZAP/g0/3060|300|") in new stack
> -- Called g0/3060
> -- Starting simple switch on 'Zap/24-1'
> -- Zap/1-1 answered SIP/2054-b7802098
> == Unknown extension '11#3060' in context 'from-pstn'
requested
> -- <Zap/24-1> Playing 'ss-noservice' (language 'en')
> -- Hungup 'Zap/1-1'
>
> /NOTE: For testing 11# is added to the front of all calls comming from 
> the PSTN./
>
> *Trying a Misc. Destination & Inbound route combination:*
> Added Misc Destination 811#3060
> Changed DialPLan on LegacyPBX
>
> .
> 11#3060
> 8|11#3060
> 8|11.
> 8|.
> 8|1NXXNXXXXXX
> 8|NXXXXXX
>
> Added 'inbound route' of 11#3060 - to go to 'Misc dest
811#3060'
> *Test Performed: SIP to 3060 = Failed*
>
> -- Zap/1-1 answered SIP/2054-b7801bf0
> == Unknown extension '11#30603060' in context 'from-pstn'
requested
> -- <Zap/24-1> Playing 'ss-noservice' (language 'en')
> -- Hungup 'Zap/24-1'
>
> Added only 8|. to dial plan
> *Test Performed: SIP to 3060 = Failed*
> -Fast Busy
>
> -- Executing [s at macro-dialout-trunk:20] Dial("Zap/24-1",
"ZAP/g0/811#|300|") in new stack
> -- Called g0/811#
>
> What a mess! What else can I try?
>
>  
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Paul Hales

2008-Sep-04 00:55 UTC

head link

[asterisk-users] All calls want to go out only on interface ZAP/g0

To provide a better example:
(this is untested hack work - as I usually provide to this list)

exten => _2XXX,1,Dial(SIP/${EXTEN})

exten => _3XXX,1,Dial(ZAP/G2/${EXTEN})

exten =>  _XXXXX.,1,Dial(ZAP/G0/{EXTEN})

Clean up and test as appropriate. :)

PaulH



Mark Best wrote:>
> I have a legacy PBX that I want to slowly move off of. Below is a 
> diagram of what I want my setup to look-like after testing.
>
> Old Mitel---24 Channels---Asterisk---PSTN
> |                              |       |
> Ext. 3060              SIP. 2054      Cellular
>  
>
> No matter my dial-plan logic; all calls seem to default to ZAP/g0. I 
> can't seem to get any calls to go directly to ZAP/g2.
>
> NOTE: For testing 11# is added to the front of all calls coming from 
> the PSTN.
>
> PSTN to Asterisk (g0) from-pstn
>
> Asterisk to LegacyPBX (g2) from-internal
>
> -------------
> -Deleted all Outbound routes.
> -Re-writing Zaptel to only include Port 1 & Port 3 (No 'red
alarms' in
> zttool)
> AMI, D4, E & M and Wink - Master Timing on Port 3 (source from Port 1).
> -Added 'To_PSTN' on port g0.
> -Added 'To_LegacyPBX' on port g2.
> -Added New 'Catch all Route' to PSTN and to LegacyPBX (.)
>
> *Test Performed: SIP to Cellular = Worked*
>
> - Executing [s at macro-dialout-trunk:20]
Dial("SIP/2054-b7801d70", "ZAP/g0/2085553870|300|") in new
stack
> -- Called g0/2085553870
> -- Zap/1-1 answered SIP/2054-b7801d70
>
> *Test Performed: SIP to 3060 = Failed*
> SIP to 3060 seems to go out g0 then came back in from g0
>
> -- Goto (macro-dialout-trunk,s,17)
> -- Executing [s at macro-dialout-trunk:17]
Macro("SIP/2054-b7801d70", "dialout-trunk-predial-hook|") in
new stack
> -- Executing [s at macro-dialout-trunk:18]
GotoIf("SIP/2054-b7801d70", "0?bypass|1") in new stack
> -- Executing [s at macro-dialout-trunk:19]
GotoIf("SIP/2054-b7801d70", "0?customtrunk") in new stack
> -- Executing [s at macro-dialout-trunk:20]
Dial("SIP/2054-b7801d70", "ZAP/g0/3060|300|") in new stack
> -- Called g0/3060
> -- Starting simple switch on 'Zap/24-1'
> -- Zap/1-1 answered SIP/2054-b7801d70
> == Unknown extension '11#3060' in context 'from-pstn'
requested
> -- <Zap/24-1> Playing 'ss-noservice' (language 'en')
>
> Added 11#3060 to both PSTN and LegacyPBX dialplan
> *Test Performed: SIP to 3060 = Failed*
> -Goes out g0 and comes back unknown.
>
> -- Executing [s at macro-dialout-trunk:13]
Set("SIP/2054-b7802098", "OUTNUM=3060") in new stack
> -- Executing [s at macro-dialout-trunk:14]
Set("SIP/2054-b7802098", "custom=ZAP/g0") in new stack
> -- Executing [s at macro-dialout-trunk:15]
GotoIf("SIP/2054-b7802098", "1?gocall") in new stack
> -- Goto (macro-dialout-trunk,s,17)
> -- Executing [s at macro-dialout-trunk:17]
Macro("SIP/2054-b7802098", "dialout-trunk-predial-hook|") in
new stack
> -- Executing [s at macro-dialout-trunk:18]
GotoIf("SIP/2054-b7802098", "0?bypass|1") in new stack
> -- Executing [s at macro-dialout-trunk:19]
GotoIf("SIP/2054-b7802098", "0?customtrunk") in new stack
> -- Executing [s at macro-dialout-trunk:20]
Dial("SIP/2054-b7802098", "ZAP/g0/3060|300|") in new stack
> -- Called g0/3060
> -- Starting simple switch on 'Zap/24-1'
> -- Zap/1-1 answered SIP/2054-b7802098
> == Unknown extension '11#3060' in context 'from-pstn'
requested
> -- <Zap/24-1> Playing 'ss-noservice' (language 'en')
> -- Hungup 'Zap/1-1'
>
> /NOTE: For testing 11# is added to the front of all calls comming from 
> the PSTN./
>
> *Trying a Misc. Destination & Inbound route combination:*
> Added Misc Destination 811#3060
> Changed DialPLan on LegacyPBX
>
> .
> 11#3060
> 8|11#3060
> 8|11.
> 8|.
> 8|1NXXNXXXXXX
> 8|NXXXXXX
>
> Added 'inbound route' of 11#3060 - to go to 'Misc dest
811#3060'
> *Test Performed: SIP to 3060 = Failed*
>
> -- Zap/1-1 answered SIP/2054-b7801bf0
> == Unknown extension '11#30603060' in context 'from-pstn'
requested
> -- <Zap/24-1> Playing 'ss-noservice' (language 'en')
> -- Hungup 'Zap/24-1'
>
> Added only 8|. to dial plan
> *Test Performed: SIP to 3060 = Failed*
> -Fast Busy
>
> -- Executing [s at macro-dialout-trunk:20] Dial("Zap/24-1",
"ZAP/g0/811#|300|") in new stack
> -- Called g0/811#
>
> What a mess! What else can I try?
>
>  
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

asterisk users - Sep 2008 - All calls want to go out only on interface ZAP/g0

[asterisk-users] All calls want to go out only on interface ZAP/g0

[asterisk-users] All calls want to go out only on interface ZAP/g0

[asterisk-users] All calls want to go out only on interface ZAP/g0