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Patton 4554 dail-out issue

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Spyklee

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Hello,

I'm a newbe to 3CX and currently have two issues with C3X v3 and a Patton 4554.

Although the Patton has 2 BRI connectors currenlty only BRI-1 is connected. BRI-0 is not connected.

I I try to dail-out with my LinkSys SPA942 it shows the error message
[CM103005] Call(6) is rejected: Service Unavailable
[CM104008] Call(6): Call from Ext.102 to 06XXXXXXXX terminated; cause: 503 Service Unavailable; from IP:10.1.9.1
[CM103002] Call(6): Incoming call from 102 (Ext.102) to sip:[email protected]

Active Calls in the Patton:
Call Call-Leg State Address Display Charge
-----------------------------------------------------------------------------------------------

00821060
sipif0-00822f08/acti CONNECTED 102 n/a n/a
router-00821560/inco TRYING 06XXXXXXXX n/a n/a
0081fab8
router-00821560/outg CONNECTED 102 n/a n/a
isdnif0-00819b90/act TRYING 06XXXXXXXX n/a n/a

If the Patton is trying to use the BRI-0 port to dail out (isdnif0) it seems logical I'm not able to dail out, since only BRI-1 is connected to the provider. However in 3CX I configured that virtual number 1001 is the dail-out line. Is this not connected to the BRI-1?

Is it possible to connect virtual number 10001 to the second BRI?

Another issue is the caller-id, I used V4 prior to downgrade to V3. Within V4 I had no caller-id at all. Wihtin v3 it's truncating the first digit.

I noticed it's within the Patton already truncated:
00818948
isdnif1-00824098/act CONNECTED 6XXXXXXXX n/a n/a
sipif1-00822c98/acti ALERTING 7XXXXXXXX n/a n/a

I've also an Asterisk PABX running the caller-id within Asterisk is working fine. Most likely a Patton issue as well.

Any ideas?

Cheers! Eric
 
Post the Patton gateway's config file here for us to investigate your issue.
 
Ricky,

Thank you for your reply, I've created with V4 the following config file:
Code: 
webserver port 80 language en

system

  ic voice 0
    low-bitrate-codec g729

system
  clock-source 0 0

profile ppp default

profile call-progress-tone NL_Dialtone
  flush-play-list
  play 1 1000 425 0

profile call-progress-tone NL_Alertingtone
  flush-play-list
  play 1 1000 425 -7
  pause 2 4000

profile call-progress-tone NL_Busytone
  flush-play-list
  play 1 500 425 -7
  pause 2 500

profile call-progress-tone NL_Releasetone
  flush-play-list
  play 1 250 425 -7
  pause 2 250

profile call-progress-tone NL_Congestiontone
  flush-play-list
  play 1 250 425 -7
  pause 2 250

profile tone-set NL_default
  map call-progress-tone dial-tone NL_Dialtone
  map call-progress-tone ringback-tone NL_Alertingtone
  map call-progress-tone busy-tone NL_Busytone
  map call-progress-tone release-tone NL_Releasetone
  map call-progress-tone congestion-tone NL_Congestiontone

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile voip fax_enabled
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  fax transmission 1 relay t38-udp

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface eth0
    ipaddress 10.1.9.1 255.0.0.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context cs switch

  interface isdn isdnif0
    route call dest-interface sipif0
    use profile tone-set NL_default

  interface isdn isdnif1
    route call dest-interface sipif1
    use profile tone-set NL_default

  interface sip sipif0
    bind gateway sipgw0
    service default
    route call dest-interface isdnif0
    remote-party-id called-party
    address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none
    address-translation incoming-call called-e164 request-uri

  interface sip sipif1
    bind gateway sipgw1
    service default
    route call dest-interface isdnif1
    remote-party-id called-party
    address-translation outgoing-call request-uri user-part fix 10001 host-part to-header target-param none
    address-translation incoming-call called-e164 request-uri

context cs switch
  no shutdown

gateway sip sipgw0
  bind interface eth0 router

  service default
    domain 10.1.8.6
    authentication 10000 password 10000 default
    default-server 10.1.8.6 loose-router
    registrar 10.1.8.6
    user 10000

gateway sip sipgw0
  no shutdown

gateway sip sipgw1
  bind interface eth0 router

  service default
    domain 10.1.8.6
    authentication 10001 password 10001 default
    default-server 10.1.8.6 loose-router
    registrar 10.1.8.6
    user 10001

gateway sip sipgw1
  no shutdown

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface eth0 router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    protocol pmp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      encapsulation cc-isdn
      bind interface isdnif0 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pmp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      encapsulation cc-isdn
      bind interface isdnif1 switch

port bri 0 1
  no shutdown

Thanks in advance!

Eric
 
Although the Patton has 2 BRI connectors currenlty only BRI-1 is connected. BRI-0 is not connected.
Click to expand...
Please try to shutdown BRI 0 0

port bri 0 0
shutdown

If the problem still not fixed. Then enable gateway's debug. Post the log as well.

#debug ccisdn error
#debug ccisdn signaling
#debug isdn error
 
Thanks,

I'll try this ...
Any idea about the truncate of the first digit of the caller-id?

Cheers!

Eric
 
Enable gateway's debug mode. Issue one call into to your Patton gateway. Capture the log then post here again. Since the config generated by 3CX. It should be a pink config file. There is not found any special digits mapping. Therefore, the 1st digit may be truncated by your provider. But I not sure, since different provider have different behavior. For me, my provider needs sent back the 8 digits full e164 calling party number. However, 3CX box unable to provide this function. It's default just can only use extension number as e164 calling party number. Therefore, I need to add the prefix 12345+ext# to make it as our calling party number. As I know that most gateways without this function. This function should handled by PBX. However, Patton's gateway did.
 
Thanks for the fast reply Ricky(!)

Any idea where to find the debug option on the Gateway?

Eric
 
Please find the steps as below:

1. Telnet to your Patton gateway
2. key in userid and password
3. gatewayname>enable then enter
4. gatewayname#debug ccisdn error then enter
5. gatewayname#debug ccisdn signaling then enter
6. gatewayname#debug isdn error then enter
 
cool stuff the debugging:

Code: 
SN4554/2BIS/EUI#16:39:33  ICC   > [isdnif1] << Message: primitive=64
16:39:33  ICC   > [isdnif1] Added endpoint isdnif1-00820050
16:39:33  ICC   > [isdnif1] NEW CALL. Allocated Endpoint isdnif1-00820050
16:39:33  ICC   > [EP isdnif1-00820050] << [080005]
SETUP (DSS1 User)
  [04038090A3]
  Bearer capability : speech - CCITT
    circuit mode - 64kBit/s - G.711 A-law
  [180189]
  Channel id : 1 - exclusive
    basic rate interface - is not d-channel - CCITT - b-channel units
  [6C0B2181363534363738393437]
  Calling party number : 6XXXXXXXX
    national number - E.164 numbering plan
    presentation allowed - user provided verified and passed
  [700AA1373537353031313330]
  Called party number : 7XXXXXXXX
    national number - E.164 numbering plan
  [7D029181]
  High layer compatibility : telephony
    CCITT
  [A1]
  Sending complete

16:39:33  ICC   > [EP isdnif1-00820050] State: NULL, Event: TERMINAL SETUP IND
16:39:33  ICC   > [EP isdnif1-00820050] Set state to CALL PRESENT
16:39:33  ICC   > [EP isdnif1-00820050] Set state to INCOMING PROCEEDING
16:39:33  ICC   > [EP isdnif1-00820050] >> [080002]
CALL PROCEEDING (DSS1 User)
  [1E028582]
  Progress indicator : destination address is non-ISDN
    private network serving remote user - CCITT

16:39:33  ICC   > [EP isdnif1-00820050] State: INCOMING PROCEEDING, Event: PEER
TRYING
16:39:33  ICC   > [EP isdnif1-00820050] Hold State: IDLE, Hold Event: PEER TRYIN
G
16:39:33  ICC   > [EP isdnif1-00820050] State: INCOMING PROCEEDING, Event: PEER
PROCEEDING
16:39:33  ICC   > [EP isdnif1-00820050] Hold State: IDLE, Hold Event: PEER PROCE
EDING
16:39:33  ICC   > [EP isdnif1-00820050] State: INCOMING PROCEEDING, Event: PEER
ALERTING
16:39:33  ICC   > [EP isdnif1-00820050] Set state to CALL RECEIVED
16:39:33  ICC   > [EP isdnif1-00820050] >> [080001]
ALERTING (DSS1 User)

16:39:33  ICC   > [EP isdnif1-00820050] Hold State: IDLE, Hold Event: PEER ALERT
ING
16:39:40  ICC   > [isdnif1] << Message: primitive=50
16:39:40  ICC   > [EP isdnif1-00820050] << [08004D]
RELEASE (DSS1 User)
  [08028290]
  Cause : normal call clearing
    public network serving local user - CCITT - Q.931

16:39:40  ICC   > [EP isdnif1-00820050] State: CALL RECEIVED, Event: TERMINAL RE
LEASE IND
16:39:40  ICC   > [EP isdnif1-00820050] Set state to NULL
16:39:40  ICC   > [isdnif1] CLEARING CALL isdnif1-00820050
16:39:40  ICC   > [isdnif1] Removed endpoint isdnif1-00820050
16:39:40  ICC   > [isdnif1] Destroying finished calls.
16:39:40  ICC   > [isdnif1] Destroyed endpoint isdnif1-00820050

Seems that the first digit is not there either. How did you add the prefix so the caller-ID is shown correctly on the extensions?

Thank you for all your support(!)

Cheers, Eric
 
Referred to your debug log. The issue should be your provider pass the incomplete e164 calling party number to your BRI line. You should contact your Telco to fixed this issue. Since almost all the gateways won't truncated any digit(s) at calling party number. All the digits are from Telco. If the missed digit is fixed, the digit mapping at gateway will help to solved your issue. Otherwise, you must in need to work with your Telco to fixed the caller id issue.
 
Well, I've also an Asterisk server running, but this system has no issue with the caller-id number.

Regards, Eric
 
Ricky,

Called-ID is solved.

Within the web-gui;

Advanced-GUI>Telephony>Call-Router>Configuration>E.164 Number Prefixes

You're able to adjust here the national prefex and the international prefix.

Cheers!
 
Hi Eric,

Good to hear, your issue had been fixed. In your case, I also learn more at Patton's gateway. Since our region ISDN BRI for Video Conference only. Only ISDN PRI for telephony.

Thanks
 
Ricky,

I've tested the shutdown config for the BRI 0 but it seems the gateway is not redirecting the call to the second line. Must be a config somewhere in the gateway.

to be continued ... :wink:

Eric
 
Please try following options.

AT GUI Admin mode > BRI 0 0 > Configuration
Port State: disable

(ISDN Layer 2):
Permanent Activity: disable

(ISDN Layer 3):
Untick => Bind to BRI Interface
 
Ricky,

If tested your settings, but it seems not te work.
I've redirected sipif0 to isdnif1 instead of isdnif0 and that seems to work, but if both ISDN cables are plugged to the Patton, it will always use isdnif0 / BRI-0 to dail out, that is not the way it should work (I think).

sipif0 should dail out on BRI-0
sipif1 should dail out on BRI-1

Look like there is a setting to always redirect to the sipif0/BRI-0.

Eric
 
Eric,

Any impact for if will always use isdnif0 / BRI-0 to dail out? How about if two extensions in-use at BRI 0. The 3rd one will use BRI-1 or get the busy tone? How about your calling party number? Is it working fine or not?
 
Ricky

Good point!

solution for me is to set the secondairy ISDN line to the BRI/0
and the primary to the BRI/1 in that case the primary number will me at least available most of the times, even if 2 people are dailing out. Since they are using the secondairy ISDN on the BRI/0


Thanks for all your help!
 
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