@gemtag wrote:

We run an outbound call center, but do have a number of inbound calls come in. We want to use queues to handle the inbound calls, but have some issues to resolve.

Our agents can be logged into only one of a number of inbound queues at a time. When the agent logs in, the queue is determined, and the agent is then logged into the queue.

Since our call center is mainly outbound, I want to only have the agent unpaused between outbound calls, once after call work is performed. So basically I want to toggle pause for the agent. Note that I do not want to use the Wrap Up Time setting. This cannot be an arbitrary value. The queue must be programmatically toggled to paused once they end the previous call, and toggled to unpaused once the agent has finished dispositioning the previous call. This could be 20 seconds or 120 seconds.

So my issue is, how can I suppress the toggle pause voice message, so it does not become extremely annoying to the agents? we are currently running 5.211.65-21.

I tried setting Feature Code Beep Only, but that has no affect. I'd really like to 'hide' completely the pause toggle of the queue, so it is invisible to the agent.

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@munozj wrote:

Our campus requires for us to have a method to be able to alert everyone of emergency situations on our campus. I setup an IVR that has
1)" Record your announcement"
Which directs callers to the Feature Code Admin / Edit Recording "emergency-annoucment"

2) "Send your recording as a page"
Sends callers to Paging and Intercom : Emergency Page with the "emergency-annoucment" recording set to play back before the page.

3) "Send your recording to external phones"
Sends callers to the Broadcast : Start Emergency Announcement Campaign. Which uses the "emergency-anncoument" recording for both the answered and unanswered recording. Any phone number registered with us gets' this call, useful for those who are not typically in a building, (maintenance workers, drivers, etc)

The current issue i'm having is that the person using the IVR must hang up, redial and proceed to the next steps manually. Is there a way to have these steps proceed some-what automatically? Also is there a way to reset the broadcast campaign through the IVR?

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@Bw_directron wrote:

We were using SIPStation for testing our server until our regular provider got our new fiber connection up and running. Inbound and outbound calls worked normally with Sipstation. No changes to the server, other than disabling SIPStation and setup of new SIP trunk.

We can make inbound and outbound calls. The other end can hear us, but we hear no audio from them. If we check the recording, the server recorded their audio. So, audio is getting to the server but not the phone. Log from Asterisk -vvvr doesn't show anything that jumps out at mean as a cause for the problem.

Same phone, same extensions, everything is the same except for the trunk. SIP Trunk connection is to a session border control device that does not require username and password, just host IP address. Trunk config is below.

Anyone have any ideas how to resolve this?

Outgoing Settings
Trunk name - providers name
peer details:
host - valid local ip address of eSBC
type-peer

Incoming Settings
user Context - company name
type - user
context - from-trunk

Register String:
192.168.223.2 (valid local ip address of eSBC, no NAT between this device and freepbx)

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@avayax wrote:

Where do I change the attended transfer timeout?
I.e. after a few rings and no answer the transferring party gets reconnected to the caller. I would like to turn this off.

How would I do that?

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@BluePukeko wrote:

I'm building a system where calls are routed to external numbers rather than extensions but still require to drop down to voice mail if unanswered.
I don't want to create a bunch of unused extensions but rather have one catchall voicemail box that all calls get routed to if unanswered BUT I want to record the ring group that the call was routed from in the email.
My intention is to use this ring group number to later rereoute the email to the end users email (this to be done on another machine)
Any suggestions, pointers to how I can achieve this.

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@mjcp wrote:

I have a new install of FreePBX from the x64 iso.

It has 2 x NICs - One for the internal LAN (internal LAN also has access to the external world), One for the External SIP trunk only.

External calls are working as expected.

Internal calls are working as expected.

However, when the users transfer a call (blind or attended) to another internal user, the call drops about 30 seconds after the destination takes the call.

Internal -> Internal calls do not suffer with this issue.

The simple logs are shown below.

Rport settings (in Chan_SIP section) are (No) 30, 300, 0 I have also tested with YES and the same numerical settings.

Please help

[2015-09-03 10:31:42] VERBOSE[23104][C-0000024b] pbx.c: -- Executing [s@macro-dial-one:44] Dial("SIP/spitfire-000005b4", "SIP/221,,Ttr") in new stack
[2015-09-03 10:31:42] VERBOSE[23104][C-0000024b] netsock2.c: == Using SIP RTP TOS bits 184
[2015-09-03 10:31:42] VERBOSE[23104][C-0000024b] netsock2.c: == Using SIP RTP CoS mark 5
[2015-09-03 10:31:42] VERBOSE[23104][C-0000024b] app_dial.c: -- Called SIP/221
[2015-09-03 10:31:43] VERBOSE[23104][C-0000024b] app_dial.c: -- SIP/221-000005b5 is ringing
[2015-09-03 10:31:52] VERBOSE[23104][C-0000024b] app_dial.c: -- SIP/221-000005b5 answered SIP/spitfire-000005b4
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] pbx.c: -- Executing [h@macro-dial-one:1] Macro("SIP/spitfire-000005b4", "hangupcall,") in new stack
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] pbx.c: -- Executing [s@macro-hangupcall:1] ExecIf("SIP/spitfire-000005b4", "0?Set(CDR(recordingfile)=.wav)") in new stack
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] pbx.c: -- Executing [s@macro-hangupcall:2] GotoIf("SIP/spitfire-000005b4", "1?theend") in new stack
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] pbx.c: -- Goto (macro-hangupcall,s,4)
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/spitfire-000005b4", "") in new stack
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/spitfire-000005b4' in macro 'hangupcall'
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] pbx.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/spitfire-000005b4'
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] app_macro.c: == Spawn extension (macro-dial-one, s, 44) exited non-zero on 'SIP/spitfire-000005b4' in macro 'dial-one'
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] app_macro.c: == Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'SIP/spitfire-000005b4' in macro 'exten-vm'
[2015-09-03 10:32:12] VERBOSE[23104][C-0000024b] pbx.c: == Spawn extension (ext-local, 221, 2) exited non-zero on 'SIP/spitfire-000005b4'

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@mvogel4949 wrote:

As far as I can tell QXact reports do not keep track of transferred calls. A call that is answer by Jim and then transferred to his boss Mark and then transferred to Emily all shows up under a single call to Jim. Is there a way to have the call broken up in the QXact reports? Thanks

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@neoian1277 wrote:

Hello everyone,

I have a client that has got a FreePBX installation setup and running, however we seem so be having problems. First abit of info about the network, there are currently 6 branches connected to the HQ via MPLS links the HQ has a subnet of 192.168.0.X (the FreePBX server is 192.168.0.248/24) with the other branches being 192.168.X.X etc. There is a Endian Firewall at the HQ as well as a Juniper SRX for the internet and MPLS breakout, all other branches have asus routers with tomato firmware installed. From the branches we have connectivity to the FreePBX server.

The problems start when calling the branches, we can hear them but they cannot hear us at all, i will include a debug from HQ to a branch office, any help would really be great

<--- SIP read from UDP:192.168.9.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK105219e6;rport=5060
From: "Nedson Phiri" ;tag=as65b438b4
To: ;tag=1848821140
Call-ID: 28835d107021001f6abefe5e624235e4@192.168.0.248:5060
CSeq: 102 INVITE
Contact:
Supported: replaces, path, timer
User-Agent: Grandstream GXP1400 1.0.7.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 589

v=0
o=9001 8000 8000 IN IP4 192.168.9.201
s=SIP Call
c=IN IP4 192.168.9.201
t=0 0
m=audio 5004 RTP/AVP 9 0 18 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 99 98 34
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<------------->
--- (12 headers 22 lines) ---
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
[2015-09-03 14:55:49] WARNING[2688][C-000005e4]: chan_sip.c:10105 process_sdp: Ignoring video stream offer because port number is zero
Capabilities: us - (ulaw|g729|g722|h263|h263p|h264), peer - audio=(ulaw|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|g729|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.9.201:5004
Peer doesn't provide video
list_route: hop:
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.9.201:5060
Transmitting (NAT) to 192.168.9.201:5060:
ACK sip:9001@192.168.9.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK6f5f7206;rport
Max-Forwards: 70
From: "Nedson Phiri" ;tag=as65b438b4
To: ;tag=1848821140
Contact:
Call-ID: 28835d107021001f6abefe5e624235e4@192.168.0.248:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

<--- SIP read from UDP:192.168.9.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK105219e6;rport=5060
From: "Nedson Phiri" ;tag=as65b438b4
To: ;tag=1848821140
Call-ID: 28835d107021001f6abefe5e624235e4@192.168.0.248:5060
CSeq: 102 INVITE
Contact:
Supported: replaces, path, timer
User-Agent: Grandstream GXP1400 1.0.7.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 589

v=0
o=9001 8000 8000 IN IP4 192.168.9.201
s=SIP Call
c=IN IP4 192.168.9.201
t=0 0
m=audio 5004 RTP/AVP 9 0 18 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 99 98 34
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<------------->
--- (12 headers 22 lines) ---
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.9.201:5060
Transmitting (NAT) to 192.168.9.201:5060:
ACK sip:9001@192.168.9.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK6897e0d0;rport
Max-Forwards: 70
From: "Nedson Phiri" ;tag=as65b438b4
To: ;tag=1848821140
Contact:
Call-ID: 28835d107021001f6abefe5e624235e4@192.168.0.248:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Reliably Transmitting (NAT) to 192.168.9.201:5060:
OPTIONS sip:9001@192.168.9.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK1eaf7854;rport
Max-Forwards: 70
From: "Unknown" ;tag=as5469fc93
To:
Contact:
Call-ID: 30a36e376149041b0b5c45ae49832b2d@192.168.0.248:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Thu, 03 Sep 2015 12:55:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<--- SIP read from UDP:192.168.9.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK1eaf7854;rport=5060
From: "Unknown" ;tag=as5469fc93
To: ;tag=1961335533
Call-ID: 30a36e376149041b0b5c45ae49832b2d@192.168.0.248:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1400 1.0.7.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->

<--- SIP read from UDP:192.168.9.201:5060 --->
BYE sip:1011@192.168.0.248:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.201:5060;branch=z9hG4bK2055290134;rport
From: ;tag=1848821140
To: "Nedson Phiri" ;tag=as65b438b4
Call-ID: 28835d107021001f6abefe5e624235e4@192.168.0.248:5060
CSeq: 103 BYE
Contact:
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1400 1.0.7.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.9.201:5060 (NAT)
Scheduling destruction of SIP dialog '28835d107021001f6abefe5e624235e4@192.168.0.248:5060' in 19136 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.9.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.9.201:5060;branch=z9hG4bK2055290134;received=192.168.9.201;rport=5060
From: ;tag=1848821140
To: "Nedson Phiri" ;tag=as65b438b4
Call-ID: 28835d107021001f6abefe5e624235e4@192.168.0.248:5060
CSeq: 103 BYE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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@Hawkeye wrote:

Hi,

cli is showing failures to register coming from 62.210.250.141 but fail2ban is banning the freepbx servers address

No matching peer for '501' from '62.210.250.141:5110'
[2015-09-03 13:03:26] NOTICE[7280][C-00012a8a]: chan_sip.c:25526 handle_request_invite: Failed to authenticate device 501sip:501@107.6.xxx.xxx;tag=f40d1547

/var/log/asterisk/full shows:

[2015-09-03 10:28:31] NOTICE[7280][C-000128d0] chan_sip.c: Failed to authenticate device 501sip:501@107.6.xxx.xxx;tag=19758342

the 107.6.xxx.xxx = our freepbx server.

Thanks.

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@Ajbryant wrote:

I have a system running freePBX version 12.0.76.1 on Asterisk Version 11.10.0. We are using Grandstream GXP2000 phones (I know they're a little old but other than a few bugs they seem to work fine) I'm having a problem where when calls ring in to the queue one of the extensions in the queue will try and pick up and the phone won't pick up the call for anyone in the queue. I can not for the life of me figure out what the problem is. My only guess is that the phones and the version of asterisk we are running are having some kind of compatibility issue. Unfortunately I am far from a SIP expert and so I'm not really sure where to go from here.

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@techs wrote:

How would I change the voicemail email attachment from WAV to MP3?

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@mvogel4949 wrote:

Anyone know of a good way to setup up TLS encryption on a remote phone extension? Thank you

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@muhajreen wrote:

Hello experts,

In a FreePBX queue:

1. I would allow call centre agents and supervisors to be able to track which unanswered calls have been responded to later. In other words, agents may be away when somebody calls the sales queue, after which they have to call back the same number. We would manage this process in order to allow maximum sales performance.

2. We also would get basic call centre reports about call lengths, call waiting times...etc

Would you please feedback which modules could cover the above requirements?

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@deanot26508 wrote:

I have joined a conversation before about this, never had an answer to it.

Dynamic DNS via DynDns, DynDns notices an IP change, yet Sipstation module does not update the new IP and just sits there until I do a reboot.

Is there a way to make Sipstation more aware of the change in IP? or maybe it has nothing to do with Sipstation as the settings are within Freepbx/Asterisk.

PBX Firmware 6.12.65-29

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@mogan wrote:

Hello, my configuration is:
Asterisk (Ver. 11.18.0)
FreePBX 12.0.76

I'm using Grandstream HT503 (HT-503 V2.0A Software Version: Program -- 1.0.14.1; Bootloader -- 1.0.0.18; Core -- 1.0.14.1; Base -- 1.0.14.1; Extra -- 1.0.14.1; CPE -- 1.0.1.48) to connect to a PSTN line and it's working, i can place call using an outbound route and incoming calls ring on ring group correctly.
The problem is that it doesn't hang up correctly, but only after (exactly) 60 seconds that I terminate the call on the pbx's phones.

Here are some config files i use, If you need more details just ask.

TRUNKS

Trunk name 503
Outbound CallerID 20002000
Maximum Channels: 	1
*Outgoing Settings*
Trunk Name: landline
PEER Details:
type=peer
authname=20002000
secret=mysecret
host=192.168.100.128
port=5062
disallow=all
allow=ulaw&alaw&gsm
canreinvite=no
dtmfmode=rfc2833
context=internal-phones
qualify=yes
hanguponpolarityswitch=yes
*Incoming Settings*
USER Context: empty
USER Details: empty
*Registration* 
Register String: empty

EXTENSION

Display Name = 20002000
Secret = mysecret
DTMF Signaling = RFC2833	
Can Reinvite = no
Context = from-internal
Host = dynamic
Trust RPID = yes
Media Encryption = none
Send RPID = send P-Asserted-Identity Header
Connection Type = friend 	
NAT Mode = no
Port = 5060
Qualify = yes
Qualify Frequency = 60 	
Transport  = All-UDP primary
Enable AVPF = no
Force AVP = no
Enable ICE Support = no
Enable Encryption = no

Here you can find grandstream config settings (add http cause I can't put images in topic).
://www.marcogiannini.it/temp/basic.png
://www.marcogiannini.it/temp/advanced.png
://www.marcogiannini.it/temp/fxs.png
://www.marcogiannini.it/temp/fxo.png

Thanks for helping.
Marco

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@dks wrote:

I upgraded our system to FreePBX 12 and m having trouble configuring with my follow me settings. Prior to this version I was able to suffix internal extensions with a # and follow me would then continue to honor the follow me settings of the forwarded extension(s). FreePBX 12 continues to support this setting when migrated from the previous version, however if I try to create or edit any follow me settings the # is dropped from any internal extension and, if the forwarded extension is not available, instead of continuing with the follow me policy of the forwarded extension, calls are directed to the Destination if no answer for the original extension. This is true regardless of where I try to modify this setting.

Is there any way to restore the previous behavior? If it isn't possible through the GUI, is there a config file somewhere that I can edit. Again, I know the configuration is supported because it is still working on any extension previously configured this way that hasn't been modified.

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@duncanidaho wrote:

I recently attempted to use my gui update to 12. However I am getting a lot of FOP, PHP and other disabled modules errors. I also am getting a security threat error that I can not seem to find answers to resolve.

I am running firmware 5.211.65-9 service pack 1.0.0.0

Following is the GUI screen errors

There is 1 module vulnerable to security threats

is 1 module vulnerable to security threat

fw_ari (Cur v. 2.11.1.5) should be upgraded to v. 12.0.5 to fix security issues: SEC-2014-002
This is a critical issue and should be resolved urgently

There is 1 module vulnerable to security threats



There is 1 module available for online upgrade

webrtc 12.0.2 (current: 2.11.0.0beta7) (I can't upgrade this module

config.php?display=modules

15 minutes, 42 seconds, ago



Could not reload FOP server

I constantly can not get FOP to reload



You have 3 disabled modules

You have 3 disabled modules

The following modules are disabled because they need to be upgraded:
endpointman, webrtc, xmpp

You should go to the module admin page to fix these. Again I can not upgrade these modules whenever I try I get errors. This also will not install
PHP-XML isn't installed.

PHP-XML is a requirement of PHP SysInfo. System Statistics are disabled until that package is installed.

This can be resolved by installing the 'php-xml' package on most distributions, and then restarting Apache

PHP-XML isn't installed.

PHP-XML is a requirement of PHP SysInfo. System Statistics are disabled until that package is installed.

This can be resolved by installing the 'php-xml' package on most distributions, and then restarting Apache

Show New

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FreePBX Statistics

PHP-XML isn't installed.

PHP-XML is a requirement of PHP SysInfo. System Statistics are disabled until that package is installed.

This can be resolved by installing the 'php-xml' package on most distributions, and then restarting Apache

Any help to get my system properly updated would be appreciate. I also want to remove security issues.

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@Marbled wrote:

Hi!

I would like my PBX to support additional languages besides English so I have configured more than one language.

*65, "Speak Your Extension Number", mostly works but doesn't say the words in the right order and tries to use a sound that doesn't exist in the other language so the English word is spoken.

If I look at the definition of that feature code it looks like currently only English and Japanese are supported. Is this a correct assessment and can we open tickets to have new added (it, of course, the no additional sounds are required).

Please let me know...

Thank you!

Nick

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@ss4sgoku wrote:

hi im having a lil trouble, i have a gsm gateway and i want to call my service provider for example in a cell phone i press 001 but i cant do that in the gsm trunk my calls have a prefix 99 so when i try to call *001 and press 99 and the phone hang up what can i do how can i call my service provider
how can i call to a *XXX
please help

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@celson wrote:

Hi all,

We have this setup below, all calls are working except the caller-id when making outgoing calls. Their existing system is not capable of setting outgoing caller-id and can only use the extension number as the caller id when passing the call to FreePBX. How I can map in FreePBX such that, when the extensions from another system make outgoing call to the ISDN line which is connected to the FreePBX it will use a specific caller-id (based on the block of numbers)- currently as it will use the mainline number for all outgoing calls. Is there any workaround for this scenario and if possible we want the modification/customization as our last option.

Setup Connectivity

Inbound Settings of FreePBX when receiving calls from the existing system.

The CallerID number is the caller-id of the extension from the other system, so if FreePBX receives a caller-id such as 200 simply route the call directly to the ISDN trunk.

Thank you!

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