Re: [asterisk-users] Merlin Legend integration not routing calls back to PSTN.

2010-01-06 Thread Shane Brath 
I fixed it, the problem was that the 2nd T1 didn't have a Switch Identifyer
set.
I set the Switch Identifyer and now I can route calls to the PSTN. Merlin
has a default that a trunk with a Null Switch Identifyer  is considered a CO
trunk. So the Merlin was getting confused, and routing it to the unknown
extension because it didn't want to create a routing loop

I sourced 2 documents that lead me to this conslusion, and after much
research. So I post this for the good of Humanity!

*
http://marketingtools.avaya.com/knowledgebase/ipoffice/mergedProjects/bulletins/techtips/134_techtip.htm

* http://www.tek-tips.com/viewthread.cfm?qid=1486737page=9


Now my only issue is that the Merlin is stripping the CID from the Tandem
connection, and only sending the area code to the PSTN.. Any sugestions
:)


On Tue, Jan 5, 2010 at 10:00 PM, C F shma...@gmail.com wrote:

 On Tue, Jan 5, 2010 at 9:40 PM, Shane Brath sh...@brath.net wrote:
 
  Folks,
 
  I have a Merlin Legend R7 V10.0 with a 2 100D cards.

 Sorry, I feel your pain.

 
  I have 1 card in slot 4 going to CenturyTel, and the card in slot 10
 going
  to a flip cable to a TE110P card in a Asterisk 1.6.x box.
 
  I have routing setup on the Merlin to send a block of numbers to the
  Asterisk.
 
  Currently the PSTN can dial the Asterisk Extensions.
  The Asterisk can dial Merlin Extensions.
  The Merlin can Dial Asterisk extensions.
 
  But the Asterisk can't dial out to the PSTN?
 
  I have tried everything, and I'm hoping someone else can shed some light
 on
  this. I'm open to ideas.
  I've already removed the barrier codes, and disable access code
 requirements
  on Tie and Non-Tie lines, with no effect.
  I made sure that the Asterisk is dialing 9XXX when sending the call
 over
  the DAHDI trunk to the Merlin.
 
  Whenever you call from the Asterisk to the Merlin you are redirected to
 the
  Unassigned Extension extension, and dropped to the Operator. I have a
  suspicion that this might have something to do with the NetwkService on
 the
  Slot 4 100D card ( out to PSTN ).
 
  Here are some relavant files for comment:
 
  Merlin PRIINFO:
  A PRI INFORMATION
 
 
 
  A Slot 4 Switch: 5ESS
 
  A Slot 10 Switch: Legend-Ntwk
 
  A System: By line
 
  A BchnlGrp #: Slot: TestTelNum: NtwkServ: Incoming Routing:
  A 1 4 CallbyCall By Dial Plan
 
  A Channel ID: 23 22 21 20 19 18 17 16 15 14
  A 13 12 11 10 9 8 7 6 5 4
  A 3 2 1
 
  A Line PhoneNumber NumberToSend
  A 801 NPANXX
  A 802 NPANXX
  A 803 NPANXX
  A 804 NPANXX
  A 805 NPANXX
  A 806 NPANXX
  A 807 NPANXX
  A 808 NPANXX
  A 809 NPANXX
  A 810 NPANXX
  A 811 NPANXX
  A 812 NPANXX
  A 813 NPANXX
  A 814 NPANXX
  A 815 NPANXX
  A 816 NPANXX
  A 817 NPANXX
  A 818 NPANXX
  A 819 NPANXX
  A 820 NPANXX
  A 821 NPANXX
  A 822 NPANXX
  A 823 NPANXX
 
  A BchnlGrp #: Slot: TestTelNum: NtwkServ: Incoming Routing:
  A 80 10 ElecTandNtwk Route Directly to UDP
 
  A Channel ID: 23 22 21 20 19 18 17 16 15 14
  A 13 12 11 10 9 8 7 6 5 4
  A 3 2 1
 
  A Line PhoneNumber NumberToSend
  A 829 NPANXX
  A 830 NPANXX
  A 831 NPANXX
  A 832 NPANXX
  A 833 NPANXX
  A 834 NPANXX
  A 835 NPANXX
  A 836 NPANXX
  A PRI INFORMATION
 
 
  A 837 NPANXX
  A 838 NPANXX
  A 839 NPANXX
  A 840 NPANXX
  A 841 NPANXX
  A 842 NPANXX
  A 843 NPANXX
  A 844 NPANXX
  A 845 NPANXX
  A 846 NPANXX
  A 847 NPANXX
  A 848 NPANXX
  A 849 NPANXX
  A 850 NPANXX
  A 851 NPANXX
 
  A Network Selection Table
 
  A Entry Number: 0 1 2 3
  A Pattern to Match: 101 10***
 
  A Special Service Table
 
  A Entry Number: 0 1 2 3 4 5 6 7
  A Pattern to Match: 011 010 01 00 1
  A Operator: none OP OP OP/P OP none none none
  A Type of Number: I I I N N N N N
  A Digits to Delete: 3 0 0 0 0 0 0 0
 
  A Call-By-Call Service Table
 
  A Entry Number: 0 1 2 3 4
  A Pattern 0: 0
  A Pattern 1: 1
  A Pattern 2: 2
  A Pattern 3: 3
  A Pattern 4: 4
  A Pattern 5: 5
  A Pattern 6: 6
  A Pattern 7: 7
  A Pattern 8: 8
  A Pattern 9: 9
  A Call Type: BOTH BOTH BOTH BOTH BOTH
  A NtwkServ: No Service
  A DeleteDigits: 0 0 0 0 0
 
  A Entry Number: 5 6 7 8 9
  A Call Type: BOTH BOTH BOTH BOTH BOTH
  A NtwkServ:
  A DeleteDigits: 0 0 0 0 0
 
  A Dial Plan Routing Table
 
  A Entry Number: 0 1 2 3
  A NtwkServ: Any service Any service
  A PRI INFORMATION
 
 
  A Expected Digits: 4 4 0 0
  A Pattern to Match: 
  A Digits to Delete: 0 4 0 0
  A Digits to Add: 
 
  A Entry Number: 4 5 6 7
  A NtwkServ:
  A Expected Digits: 0 0 0 0
  A Pattern to Match:
  A Digits to Delete: 0 0 0 0
  A Digits to Add:
 
  A Entry Number: 8 9 10 11
  A NtwkServ:
  A Expected Digits: 0 0 0 0
  A Pattern to Match:
  A Digits to Delete: 0 0 0 0
  A Digits to Add:
 
  A Entry Number: 12 13 14 15
  A NtwkServ:
  A Expected Digits: 0 0 0 0
  A Pattern to Match:
  A Digits to Delete: 0 0 0 0

[asterisk-users] Merlin Legend integration not routing calls back to PSTN.

2010-01-05 Thread Shane Brath 
) --  - - --:-- A

A Pool Absorb Other Digits FRL Call type Start Pattern
A 1)70-- 00  2 BOTH --:-- B
A 2) --  - - --:-- B
A 3) --  - - --:-- B
A 4) --  - - --:-- B
A 5) --  - - --:-- B
A 6) --  - - --:-- B
A AUTOMATIC ROUTE SELECTION



A TABLE 19: Dial 0 Output Table

A Pool Absorb Other Digits FRL Call type Start Pattern
A 1)70-- 00  3 BOTH --:-- A
A AUTOMATIC ROUTE SELECTION



A TABLE 20: N11 Output Table
A 01)411 02)611 03)811 04)911

A Pool Absorb Other Digits FRL Call type Start Pattern
A 1)70-- 00  3 BOTH --:-- A

-
NON-LOCAL DIALPLAN

A NON-LOCAL DIALPLAN



A Range Ptn Dgt Range Ptn Dgt Range Ptn Dgt
A 1) 9340-9350 01 04 18) - 35) -
A 2) - 19) - 36) -
A 3) - 20) - 37) -
A 4) - 21) - 38) -
A 5) - 22) - 39) -
A 6) - 23) - 40) -
A 7) - 24) - 41) -
A 8) - 25) - 42) -
A 9) - 26) - 43) -
A 10) - 27) - 44) -
A 11) - 28) - 45) -
A 12) - 29) - 46) -
A 13) - 30) - 47) -
A 14) - 31) - 48) -
A 15) - 32) - 49) -
A 16) - 33) - 50) -
A 17) - 34) -

A Pattern 1:

A Pool Absorb Other Digits FRL Call type
A 1)890- 00  3 VOICE
A 2) --  - -
A 3) --  - -
A 4) --  - -



---

Asterisk configs:
/etc/dahdi/system.conf
---
span = 1,1,0,esf,b8zs
bchan = 1-23
dchan = 24
echocanceller = mg2,1-240
loadzone = us
defaultzone = us


-- other blocks:

[span_1]
group = 1
usecallingpres=no
hasexten = no
switchtype = 5ess
signalling = pri_net
trunkname = Span 1
trunkstyle = digital
hassip = no
hasiax = no
context = DID_span_1
dahdichan = 1-23


in extensions.conf
exten = _XXX,n,Dial(DAHDI/g1/9${EXTEN})


dahdi-channels.com
; Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) B8ZS/ESF
group=1
context=from-pstn
;switchtype = national
switchtype = 5ess
signalling = pri_net
channel = 1-23



 debug of the call setup from asterisk 


== Using SIP RTP CoS mark 5
-- Executing [9number removed@shane:1] Dial(SIP/4342-0004,
DAHDI/g1/9number removed) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/9number removed
-- DAHDI/1-1 is proceeding passing it to SIP/4342-0004
-- DAHDI/1-1 is ringing
-- Hungup 'DAHDI/1-1'
== Spawn extension (shane, 9number removed, 1) exited non-zero on
'SIP/4342-0004'




I know I have the span_1 and the dadhi-channels kind of duplicated, and I'm
not sure why the gui did that, but I can remove it if someone thinks that is
confusing things..




Shane Brath
sh...@brath.net
Skype: shane.brath
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