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Sending call from sip trunk to provider (to analog line)

Sending call from sip trunk to provider (to analog line)

(OP)
Hi!
How to send number that coming from SIP trunk to another trunk in AVAYA?

Tac trace:
11:02:19 SIP<INVITE sip:989801216477@10.19.0.22:5060 SIP/2.0
11:02:19 active trunk-group 11 member 63 cid 0x3309
11:02:19 dial 989801216 route:ARS
11:02:19 term trunk-group 1 cid 0x3309
11:02:19 dial 989801216477 route:ARS
11:02:19 route-pattern 1 preference 1 cid 0x3309
11:02:19 seize trunk-group 1 member 9 cid 0x3309
11:02:19 Setup digits 89801216477
11:02:19 Calling Number & Name 7706 7706
11:02:19 Proceed trunk-group 1 member 9 cid 0x3309
11:02:19 denial event 1166: Unassigned number D1=0x83000b D2=0xf01
11:02:19 SIP>SIP/2.0 404 Not Found
11:02:19 idle trunk-group 1 member 9 cid 0x3309
11:02:23 TRACE COMPLETE trunk-group 11 cid 0x0

What rule (table) in AVAYA decides where to send incoming call from trunk?

CODE -->

ARS DIGIT ANALYSIS REPORT

                            Location:  all

               Dialed            Total        Route    Call      Node
               String          Min    Max    Pattern   Type     Number

         1                      7      7      p3       pubu
         2                      7      7      p3       pubu
         3                      7      7      p3       pubu
         4                      7      7      p3       pubu
         5                      7      7      p3       pubu
         6                      7      7      p3       pubu
         7                      7      7      p3       pubu
         8                      7      7      p3       pubu
         8                      11     11     p11      pubu
	 89                     11     11     p11      pubu
         9                      7      7      p3       pubu 

CODE -->

DIAL PLAN ANALYSIS TABLE
                                   Location:  all           Percent Full:    0

       Dialed   Total  Call    Dialed   Total  Call     Dialed   Total  Call
       String   Length Type    String   Length Type     String   Length Type
     0            3    dac    4           3    udp
     0            5    ext    4           4    ext
     1            4    ext    5           3    udp
     11           4    ext    5           4    ext
     12           4    ext    6           4    aar
     13           4    ext    7           4    ext
     14           4    ext    8           1    fac
     15           4    ext    9           1    fac
     16           4    ext    *           3    fac
     17           4    ext    #           3    fac
     2            3    udp
     2            4    ext
     25           4    ars
     3            3    udp
     3            4    ext 

RE: Sending call from sip trunk to provider (to analog line)

See what your incoming call handling treatment says to do:
disp inc-call-handling-trtmt trunk-group X

RE: Sending call from sip trunk to provider (to analog line)

(OP)

AvayaCiscoAdmin, thank you for your answer.
Inside incoming call handling table there are DID numbers that we translate to our extensions or vdn's. We know what DID's we have and we know where to route them.
But in this case we don't know what number will be. We want a number (by number i mean any external number to another city, country or mobile phone) dialed from cisco sip phone to route from avaya. (sip phone->asterisk->avaya->outside world)

RE: Sending call from sip trunk to provider (to analog line)

(OP)
Also we can dial all internal avaya extension from cisco phone and avaya phone is ringing. Is there a way to tell away to look not only in inbound routes but in outbound too?
For example in asterisk depending on trunk context option, number that came from trunk can be managed either by inbound routes rules or by outbound routes rules. Is there such a thing in avaya?

RE: Sending call from sip trunk to provider (to analog line)

yes ARS tables

RE: Sending call from sip trunk to provider (to analog line)

(OP)
Then, based on tac trace and ARS table from my first post i don't understand how avaya chose trunk group 1?

RE: Sending call from sip trunk to provider (to analog line)

you would have to display partition route table 11 as it matches closest to 89, this will then tell you what route pattern the call is using

RE: Sending call from sip trunk to provider (to analog line)

(OP)
li partition-route-table 11

CODE -->

PARTITION ROUTING TABLE

                             Routing Patterns
           PGN 1   PGN 2   PGN 3   PGN 4   PGN 5   PGN 6   PGN 7   PGN 8
           PGN 9   PGN 10  PGN 11  PGN 12  PGN 13  PGN 14  PGN 15  PGN 16
   Route   PGN 17  PGN 18  PGN 19  PGN 20  PGN 21  PGN 22  PGN 23  PGN 24
   Index   PGN 25  PGN 26  PGN 27  PGN 28  PGN 29  PGN 30  PGN 31  PGN 32
   -----   ------  ------  ------  ------  ------  ------  ------  ------
   11      1       11      -       1       -       11      1       1
   12      -       -       -       -       -       -       -       -
   13      -       -       -       -       -       -       -       -
   14      -       -       -       -       -       -       -       -
   15      -       -       -       -       -       -       -       -
   16      -       -       -       -       -       -       -       -
   17      -       -       -       -       -       -       -       -
   18      -       -       -       -       -       -       -       -
   19      -       -       -       -       -       -       -       -
   20      -       -       -       20      -       -       -       -
   21      -       -       -       -       -       -       -       - 

RE: Sending call from sip trunk to provider (to analog line)

look at route pattern 1

RE: Sending call from sip trunk to provider (to analog line)

(OP)

CODE -->

Pattern Number: 1   Pattern Name: TrunkName1
                             SCCAN? n     Secure SIP? n
    Grp FRL NPA Pfx Hop Toll No.  Inserted                             DCS/ IXC
    No          Mrk Lmt List Del  Digits                               QSIG
                             Dgts                                      Intw
 1: 1    0                                                              n   user
 2: 2    0                                                              n   user
 3:                                                                     n   user
 4:                                                                     n   user
 5:                                                                     n   user
 6:                                                                     n   user

     BCC VALUE  TSC CA-TSC    ITC BCIE Service/Feature PARM  No. Numbering LAR
    0 1 2 M 4 W     Request                                 Dgts Format
                                                         Subaddress
 1: y y y y y n  n            rest                               pub-unk   none
 2: y y y y y n  n            rest                                         none
 3: y y y y y n  n            rest                                         none
 4: y y y y y n  n            rest                                         none
 5: y y y y y n  n            rest                                         none
 6: y y y y y n  n            rest                                         none 

RE: Sending call from sip trunk to provider (to analog line)

There you go it's using trunk group 1 as it's 1st choice, is this a long distance call? if so change it in ars to use a ld route pattern

RE: Sending call from sip trunk to provider (to analog line)

(OP)
I've thought p11 in ARS is a route pattern and it turned out to be a partition-route-table.
What do you mean by long distance call? Call in trace is mobile call so i assume it can be a long distance call.

I don't understand what should i change in ARS table. (I have very little experience and knowledge in avaya system)

RE: Sending call from sip trunk to provider (to analog line)

I only know how to route calls in the usa, this looks like you are in another country and as I do not understand your local dialing rules then all I can say is look how you treat another mobile telephone number and give it the same route pattern and retest.

RE: Sending call from sip trunk to provider (to analog line)

(OP)
I've made a test.
when i dial mobile number 989003331960 from avaya phone it goes like this and all is ok:

CODE -->

13:59:48     dial 989003331960 route:ARS
13:59:48     route-pattern  11 preference 1  cid 0x15e
13:59:48     seize trunk-group 11 member 97  cid 0x15e
13:59:48     Calling Number & Name 2466 user1
13:59:48 SIP>INVITE sip:89003331960@10.19.0.22 SIP/2.0
13:59:48     Setup digits 89003331960
13:59:48     Calling Number & Name 2466 user1
13:59:48 SIP<SIP/2.0 100 Trying
13:59:48     Proceed trunk-group 11 member 97  cid 0x15e
13:59:48 SIP<SIP/2.0 183 Session Progress 

When i dialed from cisco phone and dialed number goes in trunk to avaya

CODE -->

13:02:26 SIP<INVITE sip:989003331960@10.9.0.2:5060  SIP/2.0
13:02:26     active trunk-group 11 member 2  cid 0x153d
13:02:26     dial 989003331 route:ARS
13:02:26     term trunk-group 1    cid 0x153d
13:02:26     dial 989003331960 route:ARS
13:02:26     route-pattern  1 preference 1  cid 0x153d
13:02:26     seize trunk-group 1 member 5  cid 0x153d
13:02:26     Setup digits 89003331960
13:02:26     Calling Number & Name 7706  7706 
13:02:26     Proceed trunk-group 1 member 5  cid 0x153d
13:02:26     denial event 1166: Unassigned number D1=0x83000b D2=0xf01
13:02:26 SIP>SIP/2.0 404 Not Found
13:02:26     idle trunk-group 1 member 5  cid 0x153d 

I've thought that the problem is in wrong number in sip invite (SIP<INVITE sip:989003331960@10.9.0.2:5060 SIP/2.0), so i changed the number that goes to trunk from asterisk to avaya to look like this SIP<INVITE sip:89003331960@10.9.0.2:5060 SIP/2.0

But when i try to call on 989003331960 from cisco phone, i get this on avaya:

CODE -->

18:52:58 SIP<INVITE sip:89003331960@10.9.0.2:5060  SIP/2.0
18:52:58     active trunk-group 11 member 1  cid 0x129a
18:52:58 SIP>SIP/2.0 404 Not Found
18:52:58     dial 8
18:52:58     term trunk-group 11    cid 0x129a
18:52:58     idle trunk-group 11    cid 0x129a
18:53:07 TRACE COMPLETE trunk-group  11 cid 0x0 

Why does it dial 8?
I thought this because of 8 fac in dialplan, but even after removing 8 fac drom dialplan, avaya still dials 8.

RE: Sending call from sip trunk to provider (to analog line)

Forget about 8 for now

Post the first page of display features

Can you do a full trace on the tac with a /s on the trace so list trace tac xxx/s

Post a working and non working call and explain where you are diallig from and too

POst your trunk group pages of the link between the cm and asterik

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

RE: Sending call from sip trunk to provider (to analog line)

(OP)
feature-access-codes:

CODE -->

FEATURE ACCESS CODE (FAC)
         Abbreviated Dialing List1 Access Code: *01
         Abbreviated Dialing List2 Access Code:
         Abbreviated Dialing List3 Access Code:
Abbreviated Dial - Prgm Group List Access Code:
                      Announcement Access Code: *99
                       Answer Back Access Code: #10
                         Attendant Access Code:
      Auto Alternate Routing (AAR) Access Code: *91
    Auto Route Selection (ARS) - Access Code 1: 9     Access Code 2:
                 Automatic Callback Activation: *09    Deactivation: #09
Call Forwarding Activation Busy/DA:        All: *00    Deactivation: #00
   Call Forwarding Enhanced Status:        Act:        Deactivation:
                         Call Park Access Code: *10
                       Call Pickup Access Code: *11
CAS Remote Hold/Answer Hold-Unhold Access Code:
                  CDR Account Code Access Code:
                        Change COR Access Code:
                   Change Coverage Access Code:
            Conditional Call Extend Activation:        Deactivation:
                   Contact Closure   Open Code:          Close Code: 

li trace tac 011/s (not working call from cisco phone to mobile)

CODE -->

14:26:45 SIP<INVITE sip:89003331960@10.19.0.22:5060  SIP/2.0
14:26:45 SIP|From: "7706" sip:7706@10.19.0.251;Tag=as60c8b6a7
14:26:45 SIP|To:  sip:89003331960@10.19.0.22
14:26:45 SIP|Call-ID: 5f77fddb603f628743bc525e7a7f944b@10.19.0.251
14:26:45 SIP|1:5060
14:26:45 SIP|CSeq: 102 INVITE
14:26:45 SIP|Content-Length: 254
14:26:45     active trunk-group 11 member 32  cid 0x1657
14:26:45 SIP>SIP/2.0 404 Not Found
14:26:45 SIP|From: "7706" <sip:7706@10.19.0.251>;tag=as60c8b6a7
14:26:45 SIP|To: <sip:89003331960@10.19.0.22>;tag=802a75b8c6a2e71a
14:26:45 SIP|33a5798b11500
14:26:45 SIP|Call-ID: 5f77fddb603f628743bc525e7a7f944b@10.19.0.251
14:26:45 SIP|1:5060
14:26:45 SIP|CSeq: 102 INVITE
14:26:45     dial 8
14:26:45     term trunk-group 11    cid 0x1657
14:26:45     idle trunk-group 11    cid 0x1657 


li trace tac 011/s (call from avaya phone to mobile)

CODE -->

14:33:32     dial 989003331960 route:ARS
14:33:32     route-pattern  11 preference 1  cid 0x1668
14:33:32     seize trunk-group 11 member 99  cid 0x1668
14:33:32     Calling Number & Name 2466 user1
14:33:32 SIP>INVITE sip:89003331960@10.19.0.251 SIP/2.0
14:33:32 SIP|From: "user1" <sip:2466@xx.yyy.zz>;t
14:33:32 SIP|ag=06a5abc7a2e71e73a5798b11500
14:33:32 SIP|To: "89003331960" <sip:89003331960@10.19.0.251>
14:33:32 SIP|Call-ID: 06a5abc7a2e71e83a5798b11500
14:33:32 SIP|CSeq: 1 INVITE
14:33:32 SIP|Content-Length: 239
14:33:32     Setup digits 89003331960
14:33:32     Calling Number & Name 2466 user1
14:33:32 SIP<SIP/2.0 100 Trying
14:33:32 SIP|From: "user1" sip:2466@xx.yyy.zz;Tag
14:33:32 SIP|=06a5abc7a2e71e73a5798b11500
14:33:32 SIP|To: "89003331960" sip:89003331960@10.19.0.251
14:33:32 SIP|Call-ID: 06a5abc7a2e71e83a5798b11500
14:33:32 SIP|CSeq: 1 INVITE
14:33:32 SIP|Content-Length: 0
14:33:32     Proceed trunk-group 11 member 99  cid 0x1668
14:33:32 SIP<SIP/2.0 183 Session Progress
14:33:32 SIP|From: "user1" sip:2466@xx.yyy.zz;Tag
14:33:32 SIP|=06a5abc7a2e71e73a5798b11500
14:33:32 SIP|To: "89003331960" sip:89003331960@10.19.0.251;Tag=as
14:33:32 SIP|1222a457
14:33:32 SIP|Call-ID: 06a5abc7a2e71e83a5798b11500
14:33:32 SIP|CSeq: 1 INVITE
14:33:32 SIP|Require: timer
14:33:32 SIP|Content-Length: 254
14:33:32     G711MU ss:off ps:20
             rgn:1 [10.19.0.251]:17984
             rgn:1 [10.9.0.33]:2078
14:33:32     xoip options: fax:Relay modem:off tty:US  uid:0x50081
             xoip ip: [10.9.0.33]:2078
             VOIP data from: [10.9.0.33]:2078
14:33:43     Jitter:0 0 0 0 0 0 0 0 0 0: Buff:33 WC:8 Avg:0
14:33:43     Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
             VOIP data from: [10.9.0.33]:2124
14:33:44     Jitter:0 0 0 0 0 0 0 0 0 0: Buff:14 WC:0 Avg:0
14:33:44     Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
14:33:45 SIP>CANCEL sip:89003331960@10.19.0.251 SIP/2.0
14:33:45 SIP|From: "user1" <sip:2466@xx.yyy.zz>;t
14:33:45 SIP|ag=06a5abc7a2e71e73a5798b11500
14:33:45 SIP|To: "89003331960" <sip:89003331960@10.19.0.251>
14:33:45 SIP|Call-ID: 06a5abc7a2e71e83a5798b11500
14:33:45 SIP|CSeq: 1 CANCEL
14:33:45     idle station    2466 cid 0x1668 

trunk-group

CODE -->

display trunk-group 11                                          Page   1 of  21
                                TRUNK GROUP

Group Number: 11                   Group Type: sip           CDR Reports: y
  Group Name: Elastix                     COR: 99       TN: 1        TAC: 011
   Direction: two-way        Outgoing Display? n
 Dial Access? n                                   Night Service:
Queue Length: 0
Service Type: tie                   Auth Code? n

                                                       Signaling Group: 11
                                                     Number of Members: 160
													 
display trunk-group 11                                          Page   2 of  21
      Group Type: sip

TRUNK PARAMETERS

     Unicode Name: auto

                                            Redirect On OPTIM Failure: 5000

            SCCAN? n                               Digital Loss Group: 18
                      Preferred Minimum Session Refresh Interval(sec): 600
					  
display trunk-group 11                                          Page   3 of  21
TRUNK FEATURES
          ACA Assignment? n            Measured: none
                                                          Maintenance Tests? y



                     Numbering Format: public
                                                UUI Treatment: service-provider

                                                 Replace Restricted Numbers? n
                                                Replace Unavailable Numbers? n

display trunk-group 11                                          Page   4 of  21
                              PROTOCOL VARIATIONS

                      Mark Users as Phone? n
            Prepend '+' to Calling Number? n
      Send Transferring Party Information? n
                 Network Call Redirection? n
                    Send Diversion Header? n
                  Support Request History? n
             Telephone Event Payload Type:

 Show ANSWERED BY on Display? y		

display trunk-group 11                                          Page   5 of  21
                                 TRUNK GROUP
                                      Administered Members (min/max):   1/160
GROUP MEMBER ASSIGNMENTS                  Total Administered Members: 160

       Port             Name
  1: T00341             Elastix
  2: T00342             Elastix
  3: T00343             Elastix
  4: T00344             Elastix
  5: T00345             Elastix
  6: T00371             Elastix
  7: T00372             Elastix
  8: T00373             Elastix
  9: T00374             Elastix
 10: T00375             Elastix
 11: T00081             Elastix
 12: T00082             Elastix
 13: T00083             Elastix
 14: T00084             Elastix
 15: T00085             Elastix
 ...
 and it goes to 160 

RE: Sending call from sip trunk to provider (to analog line)

So hang on have you got a Cisco pbx trunked into the avaya via sip ?

What system is actually utilsed to dial to PSTN

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

RE: Sending call from sip trunk to provider (to analog line)

so the problem is on the cisco side, you could have saved lots of time in telling us this right up front

RE: Sending call from sip trunk to provider (to analog line)

(OP)
Yes, asterisk pbx connected to avaya pbx via sip trunk. We are using asterisk to dial to PTSN, and also all incoming calls from outside world are going through asterisk to avaya.

I need co clear what i'm trying to achieve, so you don't get mislead.
We have 2 offices and 1 avaya nodes in each of them, they are connected by VPN and replication is configured beetween those two nodes. In our second office there is analog (copper) line that coming straight to avaya. We want to record all calls that are coming to or outside second office. There is no such capability in avaya, so we decided to route all calls from PTSN by SIP trunk to asterisk (to another asterisk, not in our first office) to record call there and then back to office to cisco sip phones (PTSN->avaya->asterisk->cisco phones). The same with calls to PTSN: cisco phone->asterisk->back to avaya->PTSN (by analog line).

I don't know why, but i cannot do trace on avaya in our second office, it shows nothing. That's why i decided to test this scheme on avaya in our first office. And because of replication configs are identical between two nodes. The difference is we, in our first office, don't use analog lines, we have sip trunks between asterisk and telephone service providers. And also the call from PTSN in second office goes straight to avaya and to avaya phones, while in our first office the call from PTSN goes to asterisk and then to avaya and to avaya phones.


RE: Sending call from sip trunk to provider (to analog line)

So you are sending your call from Avaya to Asterisk and you get a denial?
If that's the case, show us an Asterisk log.

RE: Sending call from sip trunk to provider (to analog line)

(OP)
To send call (in first office) from avaya to asterisk, the call should be send to trunk-group 11, but from my first post tracing shows that before i get the "denial event 1166: Unassigned number" avaya sends it to trunk 1 which is isdn type trunk.
That's why there is no any log messages in asterisk on path from avaya to astersik.

I don't even know should it work at all, because the test call comes from asterisk to avaya by trunk-group 11, and then avaya should send it back to trunk-group 11 to astersik, so astersik can route it to PTSN.

RE: Sending call from sip trunk to provider (to analog line)

try adding that number to you private numbering plan, I am guessing that the trunk to the other pbx is private

RE: Sending call from sip trunk to provider (to analog line)

(OP)
What is private numbering plan? Where is it?

RE: Sending call from sip trunk to provider (to analog line)

1st off what is the trunk type for the trunk from avaya to the other pbx? command would be change private numbering x

RE: Sending call from sip trunk to provider (to analog line)

(OP)
I entered change private-numbering from 0 to 12, and the are all empty

RE: Sending call from sip trunk to provider (to analog line)

sounds like you need to bring in some qualified help on this

RE: Sending call from sip trunk to provider (to analog line)

That fist denial means the avaya does not find a valid number to dial, resolve that

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

RE: Sending call from sip trunk to provider (to analog line)

(OP)
That's what I didn't understand.
From tac trace i see thet call went to ARS table, ARS means outside, to PSTN. So what number number AVAYA couldn't find?

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