Solved

Avaya IP Office in tandem

Posted on 2010-09-09
8
2,106 Views
Last Modified: 2013-12-21
I had setup a SIP trunk between an Avaya IP Office 6.0.14 and an Asterisk 1.6.0.26-FONCORE-r78. Everything works fine. The Avaya phones and the Asterisk phones works in the two ways for all extensions.

PSTN access is behind the IP Office. The problem is when I want to make PSTN outbound calls from any Asterisk extension via the IP Office.

I see the request arriving in the IP Office but the IP Office search a user and do not try to use the ARS. Because the user does not exist the IP Office rejects the call and I get a congestion signal on the Asterisk.

All extensions must dial 9 for PSTN calls.

Asterisk has a Outbound route with Dial Pattern 9. sending the call to the Avaya trunk.

IP Office has the SIP line with Incoming Group 180.

IP Office has an Incoming Call Group with Line Group ID 180 and Destination "."

IP Office has a Short Code 9N, feature Dial Speech, Telephone Number N, Line Group ID 50. That is the same short code that all Avaya extensions use to place outbound PSTN calls and works perfectly.

IP Office has an ARS route ID 50 that places the call.

This is the Asterisk log for the call:

[Sep 9 10:03:16] VERBOSE[3200] logger.c: == Using SIP RTP TOS bits 184
[Sep 9 10:03:16] VERBOSE[3200] logger.c: == Using SIP RTP CoS mark 5
[Sep 9 10:03:16] VERBOSE[3200] logger.c: == Using SIP VRTP TOS bits 136
[Sep 9 10:03:16] VERBOSE[3200] logger.c: == Using SIP VRTP CoS mark 6
[Sep 9 10:03:16] DEBUG[3153] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [94162312309@from-internal:1] Set("SIP/801-00000016", "EMERGENCYROUTE=YES") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [94162312309@from-internal:2] Macro("SIP/801-00000016", "user-callerid,SKIPTTL,") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/801-00000016", "AMPUSER=801") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/801-00000016", "0?report") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/801-00000016", "1?Set(REALCALLERIDNUM=801)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/801-00000016", "AMPUSER=801") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/801-00000016", "AMPUSERCIDNAME=Francisco Braz") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/801-00000016", "0?report") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/801-00000016", "AMPUSERCID=801") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/801-00000016", "CALLERID(all)="Francisco Braz" <801>") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/801-00000016", "0?Set(CHANNEL(language)=)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/801-00000016", "1?continue") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Goto (macro-user-callerid,s,19)
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/801-00000016", "Using CallerID "Francisco Braz" <801>") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [94162312309@from-internal:3] Set("SIP/801-00000016", "_NODEST=") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [94162312309@from-internal:4] Macro("SIP/801-00000016", "record-enable,801,OUT,") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/801-00000016", "1?check") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Goto (macro-record-enable,s,4)
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/801-00000016", "recordingcheck,20100909-100316,1284040996.24") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Sep 9 10:03:16] VERBOSE[11683] logger.c: recordingcheck,20100909-100316,1284040996.24: Outbound recording not enabled
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- <SIP/801-00000016>AGI Script recordingcheck completed, returning 0
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/801-00000016", "") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [94162312309@from-internal:5] Macro("SIP/801-00000016", "dialout-trunk,1,94162312309,,") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/801-00000016", "DIAL_TRUNK=1") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/801-00000016", "0?sub-pincheck,s,1") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/801-00000016", "0?disabletrunk,1") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/801-00000016", "DIAL_NUMBER=94162312309") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/801-00000016", "DIAL_TRUNK_OPTIONS=tr") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/801-00000016", "OUTBOUND_GROUP=OUT_1") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/801-00000016", "0?nomax") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/801-00000016", "0?chanfull") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/801-00000016", "0?skipoutcid") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/801-00000016", "DIAL_TRUNK_OPTIONS=") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/801-00000016", "outbound-callerid,1") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/801-00000016", "0?Set(CALLERPRES()=)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/801-00000016", "0?Set(REALCALLERIDNUM=801)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/801-00000016", "1?normcid") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Goto (macro-outbound-callerid,s,6)
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/801-00000016", "USEROUTCID=") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/801-00000016", "EMERGENCYCID=") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/801-00000016", "TRUNKOUTCID=") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/801-00000016", "1?trunkcid") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Goto (macro-outbound-callerid,s,12)
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/801-00000016", "0?Set(CALLERID(all)=)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/801-00000016", "0?Set(CALLERID(all)=)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/801-00000016", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/801-00000016", "0?AGI(fixlocalprefix)") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/801-00000016", "OUTNUM=94162312309") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/801-00000016", "custom=SIP/avaya") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/801-00000016", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/801-00000016", "dialout-trunk-predial-hook,") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/801-00000016", "") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/801-00000016", "0?bypass,1") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/801-00000016", "0?customtrunk") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/801-00000016", "SIP/avaya/94162312309,300,") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: == Using SIP RTP TOS bits 184
[Sep 9 10:03:16] VERBOSE[11683] logger.c: == Using SIP RTP CoS mark 5
[Sep 9 10:03:16] VERBOSE[11683] logger.c: == Using SIP VRTP TOS bits 136
[Sep 9 10:03:16] VERBOSE[11683] logger.c: == Using SIP VRTP CoS mark 6
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Called avaya/94162312309
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- SIP/avaya-00000017 is circuit-busy
[Sep 9 10:03:16] VERBOSE[11683] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-dialout-trunk:20] Goto("SIP/801-00000016", "s-CONGESTION,1") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Goto (macro-dialout-trunk,s-CONGESTION,1)
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/801-00000016", "1?noreport") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Goto (macro-dialout-trunk,s-CONGESTION,3)
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/801-00000016", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [94162312309@from-internal:6] Macro("SIP/801-00000016", "outisbusy,") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- Executing [s@macro-outisbusy:1] Playback("SIP/801-00000016", "all-circuits-busy-now,noanswer") in new stack
[Sep 9 10:03:16] VERBOSE[11683] logger.c: -- <SIP/801-00000016> Playing 'all-circuits-busy-now.ulaw' (language 'en')
[Sep 9 10:03:16] WARNING[11683] rtp.c: RTP Read too short
[Sep 9 10:03:18] VERBOSE[11683] logger.c: -- Executing [s@macro-outisbusy:2] Playback("SIP/801-00000016", "pls-try-call-later,noanswer") in new stack
[Sep 9 10:03:18] VERBOSE[11683] logger.c: -- <SIP/801-00000016> Playing 'pls-try-call-later.ulaw' (language 'en')
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-outisbusy:3] Macro("SIP/801-00000016", "hangupcall") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/801-00000016", "1?skiprg") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Goto (macro-hangupcall,s,4)
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/801-00000016", "1?skipblkvm") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Goto (macro-hangupcall,s,7)
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/801-00000016", "1?theend") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Goto (macro-hangupcall,s,9)
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/801-00000016", "") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/801-00000016' in macro 'hangupcall'
[Sep 9 10:03:20] VERBOSE[11683] logger.c: == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/801-00000016' in macro 'outisbusy'
[Sep 9 10:03:20] VERBOSE[11683] logger.c: == Spawn extension (from-internal, 94162312309, 6) exited non-zero on 'SIP/801-00000016'
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [h@from-internal:1] Macro("SIP/801-00000016", "hangupcall") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/801-00000016", "1?skiprg") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Goto (macro-hangupcall,s,4)
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/801-00000016", "1?skipblkvm") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Goto (macro-hangupcall,s,7)
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/801-00000016", "1?theend") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Goto (macro-hangupcall,s,9)
[Sep 9 10:03:20] VERBOSE[11683] logger.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/801-00000016", "") in new stack
[Sep 9 10:03:20] VERBOSE[11683] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/801-00000016' in macro 'hangupcall'
[Sep 9 10:03:20] VERBOSE[11683] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-00000016'
[Sep 9 10:03:20] DEBUG[3153] pbx.c: FONALITY: This thread has already held the conlock, skip locking

This is the Avaya monitor log for the call:

  96951512mS SIP Rx: UDP 192.168.14.20:5060 -> 192.168.14.10:5060
                    INVITE sip:94162312309@192.168.14.10:5060 SIP/2.0
                    Via: SIP/2.0/UDP 192.168.14.20:5060;branch=z9hG4bK751c00d1
                    Max-Forwards: 70
                    From: "Francisco Braz" <sip:801@192.168.14.20>;tag=as0e9c944a
                    To: <sip:94162312309@192.168.14.10:5060>
                    Contact: <sip:801@192.168.14.20>
                    Call-ID: 159589f5207126ef11b8f3f1198c9636@192.168.14.20
                    CSeq: 102 INVITE
                    User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
                    Date: Thu, 09 Sep 2010 13:15:10 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                    Supported: replaces, timer
                    Content-Type: application/sdp
                    Content-Length: 278
                   
                    v=0
                    o=root 1972484822 1972484822 IN IP4 192.168.14.20
                    s=Asterisk PBX 1.6.0.26-FONCORE-r78
                    c=IN IP4 192.168.14.20
                    t=0 0
                    m=audio 13598 RTP/AVP 0 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv
  96951515mS CMCallEvt:    0.14234.0 -1 BaseEP: NEW CMEndpoint f5b2f1dc TOTAL NOW=6 CALL_LIST=1
  96951518mS SIP Tx: UDP 192.168.14.10:5060 -> 192.168.14.20:5060
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 192.168.14.20:5060;branch=z9hG4bK751c00d1
                    From: "Francisco Braz" <sip:801@192.168.14.20>;tag=as0e9c944a
                    To: <sip:94162312309@192.168.14.10:5060>;tag=69ffa9237dd7054b
                    Call-ID: 159589f5207126ef11b8f3f1198c9636@192.168.14.20
                    CSeq: 102 INVITE
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
                    Supported: timer
                    Content-Length: 0
                   
  96951519mS SIP Tx: UDP 192.168.14.10:5060 -> 192.168.14.20:5060
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 192.168.14.20:5060;branch=z9hG4bK751c00d1
                    From: "Francisco Braz" <sip:801@192.168.14.20>;tag=as0e9c944a
                    To: <sip:94162312309@192.168.14.10:5060>;tag=69ffa9237dd7054b
                    Call-ID: 159589f5207126ef11b8f3f1198c9636@192.168.14.20
                    CSeq: 102 INVITE
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
                    Supported: timer
                    Content-Length: 0
                   
  96951521mS SIP Rx: UDP 192.168.14.20:5060 -> 192.168.14.10:5060
                    ACK sip:94162312309@192.168.14.10:5060 SIP/2.0
                    Via: SIP/2.0/UDP 192.168.14.20:5060;branch=z9hG4bK751c00d1
                    Max-Forwards: 70
                    From: "Francisco Braz" <sip:801@192.168.14.20>;tag=as0e9c944a
                    To: <sip:94162312309@192.168.14.10:5060>;tag=69ffa9237dd7054b
                    Contact: <sip:801@192.168.14.20>
                    Call-ID: 159589f5207126ef11b8f3f1198c9636@192.168.14.20
                    CSeq: 102 ACK
                    User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
                    Content-Length: 0

Any help would be much appreciated.

Thank you.
0
Comment
Question by:fbraz
  • 4
  • 4
8 Comments
 
LVL 9

Expert Comment

by:Alex Bahar
ID: 33639205
According to the debugs, Avaya receives a call with destination "94162312309", however it cannot find the route for this call.
Check you number plan for calls coming from the SIP trunk. Is there a matching pattern for 94162312309 ? Does it strip the digit "9" before it dials "4162312309" to the PSTN?
0
 

Author Comment

by:fbraz
ID: 33640249
I agree that is the problem: IP Office does not know what to do with 94162312309.

As far as I know I cannot define number plans for a specific trunk. What I did is:

- defined a Incoming Call Route with ID matching the Incoming Group ID set at the SIP URI tab for the SIP line;

- this Incoming Call Route has the Destination set to ".". I understand the "." will pass whatever number is received. This works fine for any extension;

- IP Office will decide what to do with the number based on the Short Codes and ARS.

- There is one Short Code for "9N", striping the "9" ans sending "N" to ARS.

- ARS dials the number using its rules.

Please correct me if the process is not right.

Seems that something is missing and the number is not getting to the Short Codes or ARS evaluation.
0
 
LVL 9

Expert Comment

by:Alex Bahar
ID: 33642532
It looks good to me. I assume you have defined the * entry to match all incoming numbers from the SIP line group. Is this correct? Check the incoming call routing example>
http://www.carrollcommunications.com/manager1/incallrouteover.html
0
 

Author Comment

by:fbraz
ID: 33642778
Thanks for the hint abahar.

Actually I had the Incoming Number field blank on the ICR for the SIP trunk because Avaya help says: "A blank entry matches all calls that do not match other entries".

I tried to put the * there but no luck. IP Office still tries to find a user to match the 94162312309 number and does not find it.
0
How your wiki can always stay up-to-date

Quip doubles as a “living” wiki and a project management tool that evolves with your organization. As you finish projects in Quip, the work remains, easily accessible to all team members, new and old.
- Increase transparency
- Onboard new hires faster
- Access from mobile/offline

 
LVL 9

Assisted Solution

by:Alex Bahar
Alex Bahar earned 125 total points
ID: 33642938
As I understand IP trunks are treated a bit differently. Does the following paragraph spark anything ?
IP Trunks
Line short codes are used if Small Community Networking (SCN) is not being used or no SCN user extension match occurs on the digits received. If no line short code match occurs then normal incoming call routing is applied.

E1 and BRI ETSI Trunks
These types of trunks use line short codes immediately. If no short code match occurs then normal incoming call routing is applied.
 
0
 

Author Comment

by:fbraz
ID: 33651111
So I should have a line short code for the sip line. The question is how to create it for PSTN outgoing calls in tandem with the SIP line?

I have a short code that matches the Asterisk extensions number plan (8XX/N/sip line). It works fine. If a user dials 801, for example, Avaya uses the sip line to place the call to the Asterisk.

When Asterisk sends the call to Avaya, if it is an existing extension Avays finds it and places the call. If it is a 9N number it only tries to find a user and does not use the existing short code for 9N that sends the call to the ARS.

I tried to create a short code "sip:"9N"@192.168.14.10" sending to the ARS but I get the same error.

The piece that is missing in this puzzle is how to tell Avaya to use the short code 9N (or any other that matches) when is receives the call from the Asterisk via the SIP line.
0
 

Accepted Solution

by:
fbraz earned 0 total points
ID: 33677179
I found the setting that was missing reading the post at http://www.avayausers.com/showthread.php?t=27459 and found the explanation at the Avaya documentation:

"For IP Office Release 5, if the wildcard * is used in the SIP trunk's Local URI, Contact and Display fields, that SIP trunk will accept any incoming SIP call. The incoming call routing is still performed by the IP Office incoming call routes based on matching the values received with the call or the URI's incoming group setting."

On Line / SIP URI I had Local URI, Contact and Display Name fields set to "Use Internal Data". I changed the 3 fields to "*" and everything is working perfect now.

I did not touch the Incoming Call Routes, Short Codes or ARS.
0
 
LVL 9

Expert Comment

by:Alex Bahar
ID: 33678513
I am glad using "*" got it working.
0

Featured Post

What Should I Do With This Threat Intelligence?

Are you wondering if you actually need threat intelligence? The answer is yes. We explain the basics for creating useful threat intelligence.

Join & Write a Comment

Article by: user_n
How Sip Phone (User Agent) works and communicates with sip servers 1.  There is a sip server and a sip registrar.  The sip server and sip registrar can be one server or two different servers. The sip registrar is the server on which it is record…
Implementing Avaya's One-X portal is pretty painless, until you want to deploy this to the Android and iPhone clients when these clients are outside of your network. The clients will also work within your local network. Here is our experience and so…
When you create an app prototype with Adobe XD, you can insert system screens -- sharing or Control Center, for example -- with just a few clicks. This video shows you how. You can take the full course on Experts Exchange at http://bit.ly/XDcourse.
This video demonstrates how to create an example email signature rule for a department in a company using CodeTwo Exchange Rules. The signature will be inserted beneath users' latest emails in conversations and will be displayed in users' Sent Items…

707 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question

Need Help in Real-Time?

Connect with top rated Experts

14 Experts available now in Live!

Get 1:1 Help Now