Posted in AudioCodes, Skype for Business

GWAPP REQUESTED CIRCUIT NOT AVAILABLE

Scenario:

You have configured the AudioCodes Gateway (M800, M1K, M2k) with the best of your knowledge. Numbers manipulations are all correct in your view. You are sending the call to PSTN however call is simply dropping and you encountered below errors on logs.

nocircuit_1

Other errors on the logs:

  • pstn recv <– CALL_RELEASED Trunk:0 Conn:0 RetCause:104 NetCause:44
  • Abnormal Disconnect cause:44#GWAPP_REQUESTED_CIRCUIT_NOT_AVAILABLE Trunk:0 Conn:0
  • LOCAL_CALL_RELEASED_EV(Trunk:0 Conn:0 Bchannel:28 TpEv=68)
  • LOCAL_CALL_RELEASED_EV State:WAIT_FOR_BOARD_ANSWER Substate:sub_None
  • RELEASE_EV (send) GWAPP_REQUESTED_CIRCUIT_NOT_AVAILABLE

So what’s wrong??!!

First, ask your customer if how many E1 or T1 channels they purchased. Usually as Voice engineers, we are used to use 1-24 for T1 and 1-31 for E1 in Trunk Configuration however if for example your customer only bought 10 channels, you must specify it in Trunks configuration correctly like 1-10 range only. Otherwise, you will meet above errors.

Hope it helps you!!

 

 

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Posted in AudioCodes

Configuring AudioCodes Debug Recording – SIP & Media

Firmware version: 7.0

The procedure below describes how to configure the address of the debug recording (capturing) server to where the device sends the captured traffic. Once you configure an address, the device generates DR packets for all calls.

Configuration tab > System menu > Logging > Logging Settings

debugrecording

In the “Debug Recording Destination IP”, put the IP address of your PC where you install and run wireshark.

Configuration tab > System menu > Logging > Logging Filters Table

debugrecording_2

Set as below:

  • Filter Type: Any
  • Log Type: Signalling & Media
  • Mode: Enable

Activate a wireshark which will gather all the SIP and RTP traffic coming to and from the GW or SBC.

Reproduce the issue, then stop the wireshark. Send the logs to AudioCodes Support.

Note: Disable the Debug Recording on the GW or SBC.

 

Posted in AudioCodes

Configuring AudioCodes Syslog Debug Level

You can configure the amount of information (debug level) to include in Syslog messages.

To configure the Syslog debug level:

Configuration tab > System > Syslog Settings

syslog_1

  • Enable Syslog: Enable
  • Syslog Server IP Address: IP address where Syslog Viewer or Wireshark is installed
  • Syslog Server Port: 514
  • Syslog CPU Protection: Enabled
  • Debug Level: Detailed

Syslog Viewer can be downloaded here http://www.audiocodes.com/downloads

 

 

Posted in AudioCodes

AudioCodes SBC Voice Transcoding Example Configuration

Diagram

transcoding_diagram

Scenario

  • Enterprise LAN users in LAN #1 served by SIP entity server #1:
    Voice coder: G.711
    SIP transport protocol: UDP
  • Enterprise LAN users in LAN #2 served by SIP entity server #2:
    Voice coder: G.729
    SIP transport protocol: TCP

As the two SIP entity servers use different voice codecs, you need to configure the SBC to perform transcoding between the servers. In the example, the codec support is as follows:
 SIP Entity Server #1 uses G.711 A-law or G.711 μ-law, and does not allow any other coder in the SDP offer-exchange coder list
 SIP Entity Server #2 uses G.729

The configuration for the example uses the following terms related to coders:
 Extension Coders: Voice codecs supported by the SIP entity. The SBC adds these coders to the SDP offer sent to the SIP entity. Extension coders are required for transcoding when the two communicating SIP entities support different coders (i.e., supported coders do not appear in the SDP offer).
 Allowed Coders: Coders that are permitted to be listed in the SDP offer that the device sends to the SIP entity. This is required for SIP entities that accept only SDPs that include specific coders (for whatever reason). The Allowed coders would include the Extension coder as well as other coders.

Step 1: Add Extension Coder Groups for SIP Entities

A Coder Group (or Extension Coder Group) defines the codecs supported by the SIP entity. Even if the original SDP offer does not include the coder supported by the SIP entity, the SBC adds it to the SDP before sending it to the SIP entity.
In the example, you need to configure a Coder Group per SIP entity server with the supported coder:
 SIP Entity Server #1 – G.711 A-law and G.711 μ-law
 SIP Entity Server #2 – G.729

Configuration tab > VoIP menu > Coders and Profiles > Coders Group Settings

Coder Group for SIP Entity Server #1

transcoding_g711

Coder Group for SIP Entity Server #2

transcoding_g729

Step 2: Add Allowed Coders Group for SIP Entity Server #1

In the example, SIP Entity Server #1 allows only the G.711 A-law and G.711 μ-law coders to be listed in the SDP offer sent to it by the SBC. If other coders are listed in the SDP, the SBC removes them before sending them to the SIP entity. Therefore, you need to configure an Allowed Audio Coders Group with these coders.

Configuration tab > VoIP menu > SBC > Allowed Audio Coders Group

Allowed Audio Coders Group for SIP Entity Server #1

transcoding_g711_allowed

Step 3: Add IP Profiles for SIP Entities and Assign their Coder Groups

An IP Profile defines a set of configuration settings that can be assigned to specific calls. In the example, you need to configure an IP Profile for each SIP entity server and assign it the supported codec (i.e., Coder Group) that you configured in the previous steps:
 SIP Entity Server #1: Supports only G.711 (A-law and μ-law) and does not allow other additional coders to be listed in the SDP. Therefore, the IP Profile must be assigned the following:
• Extension Coders Group (Index 1): G.711 (A-law and μ-law)
• Allowed Audio Coders Group (Index 0): G.711 (A-law and μ-law)
 SIP Entity Server #2: Supports only G.729, but accepts SDPs listing other additional coders. Therefore, the following configuration is required:
• Extension Coders Group (Index 2): G.729

Configuration tab > VoIP menu > Coders and Profiles > IP Profile Settings

IP Profile for SIP Entity Server #1

transcoding_row1

IP Profile for SIP Entity Server #2

transcoding_row2

Step 4: Assign IP Profiles to SIP Entity IP Groups

To associate the voice coders with the SIP entity servers, you need to assign the previously configured IP Profiles to the IP Groups of the SIP entities.

Note: IP Profiles should already been created for SIP entities.

Configuration tab > VoIP menu > VoIP Network > IP Group Table

Assigning IP Profile to IP Group of SIP Entity Server #1

transcoding_sipentity1

Assigning IP Profile to IP Group of SIP Entity Server #2

transcoding_sipentity2

That’s it!

 

 

 

Posted in AudioCodes, Skype for Business

Upgrading AudioCodes SBA to Skype for Business -Requirements

AudioCodes has created an SBA Upgrade Kit meant for Skype for Business. Essentially, upgrade depends on customer current M1K model or SBA Platform.

The following table lists the different Mediant 1000B SBA platforms and their corresponding SBA upgrade kits.

sba_kit

Upgrading HDMX Module from HDD to SSD

  1. Maintain power to the Mediant 1000B SBA chassis.
  2. Shutdown the SBA server
  3. Remove the current HDMX module from the chassis:
    a. Gently pull the HDMX module’s handle out to the first notch; the handle is now partially pulled out and the module undergoes a shutdown sequence, indicated by the slow-flashing LED on the module.
    b. When the LED stops flashing and is constantly lit, indicating that the shutdown sequence is complete, grip and gently pull the HDMX module’s handle to slide the module out of the slot.
  4. Insert the new HDMX module:
    a. Take the HDMX module out of the box in which it was shipped and remove it from the protective plastic packet.
    b. Hold the new HDMX module in the correct orientation, as shown in the figure below, and gently insert the module into the slot, sliding it along the slot’s guide rails until it makes contact with the card-edge connector located on the backplanesba_hdmxmodule
  5.  Push-in the HDMX module’s handle until it clicks firmly into the slot.

Installing the SBA Skype for Business Image

The SBA Skype for Business application is provided on the USB dongle that is shipped in your ordered SBA upgrade kit. The image also includes Microsoft Windows Server 2012 R2 installation.

Warning:
• Before installing the SBA Skype for Business image, make sure that your OSN server (SBA server) is running a compatible BIOS version (5603E216 or later). If not, you need to upgrade the BIOS.
• When you install the SBA Skype for Business image, all previous SBA settings are restored to default settings.

To install the SBA Skype for Business image:

  1. Establish a connection with the OSN server:
    • Direct connection through HDMI
  2. Connect the SBA USB dongle:
    • Direct connection through HDMI: Plug the SBA dongle into one of the available USB ports on the USB hub:sba_hdmi
  3. Reset the OSN server using the OSN module’s handle
    • Even though the handle can be pulled out two notches, gently pull it out only to the first notch; the LED light starts flashing blue:sba_firstnotch
    • Wait until the LED light changes to solid blue, indicating that the SBA server has shut down.
    • Gently push the handle all the way back into the module; the LED switches off, indicating that the SBA server is undergoing a reset:sba_firstnotch2
  4. At the prompt, type the following:
    X:\windows\system32>gorecover.bat
    The following appears:sba_recover
  5. At the confirmation prompt, type the following:
    X:\windows\system32>Y
    The Skype for Business and Windows installation begins. When complete, the following appears:sba_complete
  6. Unplug the SBA dongle from the USB port.
  7. At the prompt, type the following:
    X:\windows\system32>exit
    • For HDMI Connection: The SBA server restarts to complete installation.
  8. (For OSN3 Module Only) To activate the Windows Server 2012 R2, access the BIOS of the SBA server and enter the Microsoft license (supplied).

Done!

Proceed first to do the usual pre-requisites for SBA installation like:

  • Adding SBA to Active Directory
  • Defining Branch Site Topology

And continue the SBA configuration through Web-based (HTTP) SBA Management interface. This procedure I will tackle on another blog.

Posted in AudioCodes, Skype for Business

Deploying AudioCodes Skype for Business Cloud Connector Edition (CCE)? What you need to know!

Plan your network design

  • Below design is based on AudioCodes M800 CCE
  • The two rear Ethernet ports will be used (GE1:Corpnet & Ge2:Internet)
  • Front Ethernet port will be used for GW/SBC

cce_network_design_v3-0

Prepare the required IP addresses

  • 6 Internal IP addresses
  • 1 External IP NAT’ed to 1 Public IP address
  • BaseVMIP is optional
  • You may put the gateway or SBC same range as Internal IP of the CCE or place it on your internal corporate network range (e.g. 192.168.1.10)
  • Below IP addresses are mere examples only

cce_ip

Public DNS Records

  • Other Office 365 DNS records should already be created at this point
  • SRV records are created as well

cce_publicdns

Office 365 with CloudPBX license

  • E5 license
  • E3 + Cloud PBX

Prepare the following Administrator during installation

VM ADMIN loremarc@contoso.com
Domain Admin loremarc@contoso.com
Safemode domain Admin loremarc@contoso.com
Office 365 Global Administrator loremarc@contoso.com

Prepare the Cloud Connector ini file

  • I prefer it to be pre-populated before starting the installation
  • Values will depend on above information
  • Each parameters are explained well here: https://aka.ms/deploycloudconnector

Public Certificate

  • Common Name: accesspool1.contoso.com
  • Subject Alternative Names: accesspool1.contoso.com, sip.contoso.com

Open the Required Firewall Rules

  • Here’s the drill, once you have finalize the network design make sure to let company’s Security team approve it as this part usually gets some exciting discussion.

Internal firewall

Source IP Destination IP Source Port Destination Port
Cloud Connector Mediation component SBC/PSTN Gateway Any TCP 5060**
SBC/PSTN Gateway Cloud Connector Mediation component Any TCP 5068/ TLS 5067
Cloud Connector Mediation component SBC/PSTN Gateway UDP 49 152 – 57 500 Any***
SBC/PSTN Gateway Cloud Connector Mediation component Any*** UDP 49 152 – 57 500
Cloud Connector Mediation component Internal clients TCP 49 152 – 57 500* TCP 50,000-50,019

(Optional)

Cloud Connector Mediation component Internal clients UDP 49 152 – 57 500* UDP 50,000-50,019
Internal clients Cloud Connector Mediation component TCP 50,000-50,019 TCP 49 152 – 57 500*
Internal clients Cloud Connector Mediation component UDP 50,000-50,019 UDP 49 152 -57 500*

External firewall – recommended configuration

Source IP Destination IP Source Port Destination Port
Any Cloud Connector Edge External Interface Any TCP 5061
Cloud Connector Edge External Interface Any Any TCP 5061
Cloud Connector Edge External Interface Any Any TCP 80
Cloud Connector Edge External Interface Any Any UDP 53
Cloud Connector Edge External Interface Any Any TCP 53
Cloud Connector Edge External Interface Any TCP 50,000-59,999 Any
Cloud Connector Edge External Interface Any UDP 3478; UDP 50,000-59,999 Any
Any Cloud Connector Edge External Interface Any TCP 443; TCP 50,000-59,999
Any Cloud Connector Edge External Interface Any UDP 3478; UDP 50,000 – 59,999

Once you have done above, you should be ready to start your installation! Hopefully, I could still have time to share my step-by-step installation on next blog 🙂

Stay tune! Cheers!

Posted in AudioCodes, ShoreTel, Skype for Business

AudioCodes SBC Configuration

Diagram

sip-trunk-ippbx-3

Scenario

Configuration entities configured with the name “LAN” are used for interfacing with the LAN (e.g., IP PBX) and those configured with the name “WAN” are used for the interfacing with the WAN (e.g., SIP Trunk). IP addresses used above are only set as examples.

Device/Software

  • AudioCodes Mediant Software
  • Firmware Version: 7.00A.049.003

Step 1 – Assign Ethernet Ports to Ethernet Groups 

mediant-sw-ports

 LAN: Ethernet Group 1 with ports GE_1
 WAN: Ethernet Group 2 with ports GE_2

Configuration tab > VoIP menu > Network > Ethernet Groups Table

mediant-sw-ethernet-group

Step 2 – Assign VLAN IDs to Ethernet Groups

 LAN: VLAN ID 1 assigned to Ethernet Group 1
 WAN: VLAN ID 2 assigned to Ethernet Group 2

Configuration tab > VoIP menu > Network > Ethernet Device Table

mediant-sw-vlan

Step 3: Add Logical IP Network Interfaces for LAN and WAN

 LAN: IP address 172.16.0.234
 WAN: IP address 212.199.200.100

 VLAN 1 (Ethernet Group 1) is assigned to the LAN interface
 VLAN 2 (Ethernet Group 2) is assigned to the WAN interface

Configuration tab > VoIP menu > Network > IP Interfaces Table

mediant-sw-logical-ip

Step 4: Add Media Realms for LAN and WAN

Configuration tab > VoIP menu > VoIP Network > Media Realm Table

Media Realm for LAN Interface

mediant-sw-realm-lan

Media Realm for WAN Interface

mediant-sw-realm-wan

Step 5: Add SIP Interfaces for LAN and WAN

Configuration tab > VoIP menu > VoIP Network > SIP Interface Table

SIP Interface for LAN

mediant-sw-sip-int-lan

SIP Interface for WAN

mediant-sw-sip-int-wan

Step 6: Add Proxy Sets for IP PBX and SIP Trunk

The Proxy Set defines the actual address of SIP server entities in your network. Therefore, you need to add a Proxy Set for the following entities:
 LAN: IP PBX with address 172.16.0.231
 WAN: SIP Trunk with address 212.199.200.10

Configuration tab > VoIP menu > VoIP Network > Proxy Sets Table

Proxy Set for IP PBX (e.g. Skype for Business)

mediant-sw-proxy-lan

Select the table row of the Proxy Set that you added, and then click the Proxy Address Table link located below the table

mediant-sw-proxy-ip

Proxy Set for ITSP (e.g. Telstra)

mediant-sw-proxy-wan

Select the table row of the Proxy Set that you added, and then click the Proxy Address Table link located below the table.

mediant-sw-proxy-ip-wan

Step 7: Add IP Groups for IP PBX and SIP Trunk

The IP Group represents the SIP entity. In the example, you need to add an IP Group for the following entities:
 WAN SIP Trunk (server-type IP Group)
 LAN IP PBX (server-type IP Group)

Configuration tab > VoIP menu > VoIP Network > IP Group Table

IP Group for LAN IP PBX

mediant-sw-lan-ip-group

IP Group for WAN SIP Trunk

mediant-sw-wan-ip-group

Step 8: Add IP-to-IP Call Routing Rules

For call routing between the SIP entities, you need to add IP-to-IP routing rules for the following call directions:
 Calls from the WAN SIP Trunk to the LAN IP PBX.
 Calls from the LAN IP PBX to the WAN SIP Trunk.

Configuration tab > VoIP menu > SBC > Routing SBC > IP-to-IP Routing Table

Call Routing Rule from WAN SIP Trunk to LAN IP PBX

mediant-sw-wan-to-lan

Call Routing Rule from LAN IP PBX to WAN SIP Trunk

mediant-lan-to-wan

It will look like this

mediant-sw-ip-to-ip

Note: A destination SIP Interface is not specified for the routing rules. The ‘Destination SIP Interface’ parameter is applicable only if the ‘Destination Type’ parameter is configured to any value other than IP Group.

Step 9: Add IP-to-IP Manipulations

Essentially, manipulation will depend on your scenario and requirements. Like for example, if your IP-PBX is a Skype for Business you need to configure your incoming destination number to be in E.164 format like +65912345678. On the other hand, the destination number for outgoing calls must strip + sign or + and country code. And usually add 00 for international calls.

I will discuss more of this on another blog so just tune in.

Hope it’ll help you to initially setup your AudioCodes Mediant SW SBC!!

Thanks!

Posted in AudioCodes, Lync Server 2013, Skype for Business

AudioCodes AA – ACD call cannot transfer to any internal SfB User

Diagram

ee-and-sba

Scenario

  • AudioCodes Auto-Attendant is installed as virtual machine inside the SBA.
  • SBA is associated to a SfB Enterprise Edition in Central Pool.
  • AudioCodes AA configured with Automatic Call Distribution (ACD) in Call Work Flow
  • When a PSTN caller calls to AA IVR and chose ACD option (e.g. transfer to Receptionist), Receptionist answer the call but when it tries to transfer to any internal SfB user within the branch site, the transfer failed.

Errors on SfB logs captured on SBA

  • The call cannot be transferred
  • SIP/2.0 504 Server time-out
  • ms-diagnostics: 1038;reason=”Failed to connect to a peer server”;ip-address=”192.168.10.10″;peer-type=”InternalServer”;winsock-code=”10061″;winsock-info=”The peer actively refused the connection attempt”;source=”SfBPoolinCentralSiteFQDN”
  • SIP/2.0 400 Bad Request

 

Solution

Open TCP port 15001 from SfB servers in Central office to AudioCodes Auto-attendant.

SfB requires that all servers, that receive REFER and start a new call as result of the REFER, have access to the referring server, for authenticating the referred-by header.

In the case above all servers should have access to AudioCodesAA.local:15001

 

Thanks to AudioCodes R&D, they found the solution!

But this port is not on Firewall requirement on AC document so we have requested it to be added.

 

 

Cheers!

Posted in AudioCodes, ShoreTel

Incoming call disconnects after 32 seconds!!!

Scenario:

  • Inbound: Optus -> SBC -> ShoreTel
  • Incoming call from Optus to Shoretel drops after 32 seconds

Diagram:

  • Optus -> SBC -> ShoreTel

32-seconds-hangup

As you can see on the SIP ladder diagram above, there’s unusual multiple 200 OK response from ShoreTel.

Solution: Unfortunately, the solution wasn’t shared by ShoreTel engineers, however the fixed was done on their end. So the next time you see above trace within SBC, immediately asks assistance from ShoreTel engineer to address the issue on their end.

Cheers!

Posted in AudioCodes, ShoreTel

488 Not Acceptable Here

Happy New Year!

After spending a vacation for Christmas, I had my first AudioCodes virtual SBC implementation for this year and the error above took me a while to resolve. A while means few hours, thank God not days 🙂

So here’s the scenario.

ShoreTel -> SBC -> Optus

Essentially, the error above only happens to few outbound numbers. Almost all other numbers are working fine. Diagram below shows where the 488 Not Acceptable Here error comes from.

488-not-acceptable-here

Based on the error, when Optus is sending a valid 180 Ringing to SBC, for some reason the SBC automatically sending “488 Not Acceptable Here” error to ShoreTel thus call is not able to connect successfully.

Solution: On the SBC’s IP Profile for ShoreTel leg, you have to disable “Early Media”. Search for parameter IpProfile_EnableEarlyMedia.

Note: When EarlyMedia is enabled, the device sends a SIP 18x response with SDP, allowing the media stream to be established before the call is answered. However, on above case the SBC doesn’t have Media channels and DSP resources on Feature Key license that’s why it can’t construct the SIP 18x response with SDP. So the only option is to disable it.

Thanks!