1. Brekeke Product Name and Version:
Brekeke PBX, Version 3.16.5.0/576.2, Pro
2. Java version:
OpenJDK 64-Bit Server VM, Version: 11.0.23
3. OS type and the version:
CentOS Linux release 7.9.2009 (Core)
4. UA (phone), gateway or other hardware/software involved:
None
5. Your problem:
We want to route incoming call to a pre-recorded audio file but getting User busy (17).
Dial Plan (for testing we are routing all incoming calls to extension 1234)
Matching Patterns:
$request = ^INVITE
Deploy Patterns:
To = 1234@192.168.x.x:5056 (Brekeke IP Address)
Would you kindly advise how to deal with this?
Thank you.
Regards,
Mansour
Routing a call to an ivr extension (Dial Plan)
Moderator: Brekeke Support Team
If it is the bundled SIP Server in Brekeke PBX, you don't have to add or modify any DialPlan rules. The default DialPlan rules make all calls go to the PBX.
Instead of tuning DialPlan, let you use IVR in Brekeke PBX to meet the requirement.
FYI:
https://docs.brekeke.com/pbx/auto-attendant
Instead of tuning DialPlan, let you use IVR in Brekeke PBX to meet the requirement.
FYI:
https://docs.brekeke.com/pbx/auto-attendant
Hi James,
Thanks for sharing below details.
I have disabled the custom dial plan and kept default rules enabled only. I have also created the ivr extension as described in your link below. The calls are passing & matching "To PBX" rule:
Matching Patterns:
$request = ^INVITE|^SUBSCRIBE
Deploy Patterns:
$pbx.in
$auth = false
But we are getting User busy (17) and ivr message not playing. I am not sure if it is related to configuration issue or something else. We want all incoming calls to play the uploaded ivr greeting message.
Dial Plan History Details:
No. 2
Session ID 111
Rule To PBX
Time (received) 07/23/24 07:34:25.249+0000
Time (Dial Plan IN) 07/23/24 07:34:25.250+0000 (1ms)
Time (Dial Plan OUT) 07/23/24 07:34:25.251+0000 (1ms)
Source IP 192.168.14.53:5060 (UDP)
Destination IP 127.0.0.1:5052 (UDP)
Action com.brekeke.net.sip.sv.session.plugins.InviteSession
Incoming Packet:
INVITE sip:3002@192.168.14.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bK0cB7891e3d10f840648
From: <sip:96176878xxx@192.168.14.53>;tag=gK0c791e48
To: <sip:3002@192.168.14.204>
Call-ID: 403473803_14772759@192.168.14.53
CSeq: 869783 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:96176878xxx@192.168.14.53:5060>
P-Preferred-Identity: <sip:96176878xxx@192.168.14.53:5060>
Supported: timer,replaces
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session; handling=required
Content-Type: application/sdp
Content-Length: 311
v=0
o=Sonus_UAC 859557 219655 IN IP4 192.168.14.53
s=SIP Media Capabilities
c=IN IP4 192.168.14.47
t=0 0
m=audio 34466 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
Outgoing Packet
INVITE sip:3002@192.168.14.204:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK689e34dc185e8-30-19658f
From: <sip:96176878xxx@127.0.0.1>;tag=gK0c791e48
To: <sip:3002@127.0.0.1>
Call-ID: 403473803_14772759@192.168.14.53
CSeq: 869783 INVITE
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:96176878xxx@127.0.0.1:5060>
P-Preferred-Identity: <sip:96176878xxx@192.168.14.53:5060>
Supported: timer,replaces
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session; handling=required
X-Remote: 192.168.14.53:5060
X-Session-Info: 111
Content-Type: application/sdp
Content-Length: 307
v=0
o=Sonus_UAC 859557 219655 IN IP4 127.0.0.1
s=SIP Media Capabilities
c=IN IP4 192.168.14.47
t=0 0
m=audio 34466 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
Looking forward for your update.
Thank you again.
Regards,
Mansour
Thanks for sharing below details.
I have disabled the custom dial plan and kept default rules enabled only. I have also created the ivr extension as described in your link below. The calls are passing & matching "To PBX" rule:
Matching Patterns:
$request = ^INVITE|^SUBSCRIBE
Deploy Patterns:
$pbx.in
$auth = false
But we are getting User busy (17) and ivr message not playing. I am not sure if it is related to configuration issue or something else. We want all incoming calls to play the uploaded ivr greeting message.
Dial Plan History Details:
No. 2
Session ID 111
Rule To PBX
Time (received) 07/23/24 07:34:25.249+0000
Time (Dial Plan IN) 07/23/24 07:34:25.250+0000 (1ms)
Time (Dial Plan OUT) 07/23/24 07:34:25.251+0000 (1ms)
Source IP 192.168.14.53:5060 (UDP)
Destination IP 127.0.0.1:5052 (UDP)
Action com.brekeke.net.sip.sv.session.plugins.InviteSession
Incoming Packet:
INVITE sip:3002@192.168.14.204:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bK0cB7891e3d10f840648
From: <sip:96176878xxx@192.168.14.53>;tag=gK0c791e48
To: <sip:3002@192.168.14.204>
Call-ID: 403473803_14772759@192.168.14.53
CSeq: 869783 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:96176878xxx@192.168.14.53:5060>
P-Preferred-Identity: <sip:96176878xxx@192.168.14.53:5060>
Supported: timer,replaces
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session; handling=required
Content-Type: application/sdp
Content-Length: 311
v=0
o=Sonus_UAC 859557 219655 IN IP4 192.168.14.53
s=SIP Media Capabilities
c=IN IP4 192.168.14.47
t=0 0
m=audio 34466 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
Outgoing Packet
INVITE sip:3002@192.168.14.204:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK689e34dc185e8-30-19658f
From: <sip:96176878xxx@127.0.0.1>;tag=gK0c791e48
To: <sip:3002@127.0.0.1>
Call-ID: 403473803_14772759@192.168.14.53
CSeq: 869783 INVITE
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:96176878xxx@127.0.0.1:5060>
P-Preferred-Identity: <sip:96176878xxx@192.168.14.53:5060>
Supported: timer,replaces
Session-Expires: 1800
Min-SE: 90
Content-Disposition: session; handling=required
X-Remote: 192.168.14.53:5060
X-Session-Info: 111
Content-Type: application/sdp
Content-Length: 307
v=0
o=Sonus_UAC 859557 219655 IN IP4 127.0.0.1
s=SIP Media Capabilities
c=IN IP4 192.168.14.47
t=0 0
m=audio 34466 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
Looking forward for your update.
Thank you again.
Regards,
Mansour
Have you made users in the PBX?
https://docs.brekeke.com/pbx/brekeke-pbx-quick-start
https://docs.brekeke.com/pbx/brekeke-pbx-quick-start
We are not aiming to register users on brekeke for initiating calls.
We have a voice switch which will forward certain dialed numbers to our Brekeke PBX to play/run pre-uploaded ivr voice message.
For this we've created an ivr extension with uploaded audio file. When passing the calls to brekeke we are getting User busy (17) and ivr message not playing. Please note that we have disabled authentication for register & invite towards brekeke from our voice switch.
Regards,
Mansour
We have a voice switch which will forward certain dialed numbers to our Brekeke PBX to play/run pre-uploaded ivr voice message.
For this we've created an ivr extension with uploaded audio file. When passing the calls to brekeke we are getting User busy (17) and ivr message not playing. Please note that we have disabled authentication for register & invite towards brekeke from our voice switch.
Regards,
Mansour