Axtan - Packet Telephony solutions

Frequently Asked Questions

Answers

1 Audio issues

1.1 There is noise on my calls.

There are many posible reasons for apparent noise on calls going through IPCallBox units. First you will need to determine if your call is going across ISDN only or if it is going across SIP or H.323.
If the call is going across the ISDN Interfaces only , then the cause of the noise is likely to be external to the unit on the ISDN line or the attached PBX. It may also be due to lack of syncronisation of the internal ISDN bit clock on the PBX with the ISDN bit clock in the unit. If the calls are going across SIP or H.323 then the problem could be associated with:-

syncronisation between the PBX ISDN bit clock and the unit's ISDN bit Clock

packet loss on the IP network

mismatch on RTP packet size being sent / received

The internal gain for audio within the unit has been set too high

Check the gain settings in the configuration file applied to the unit (Configuration Editor->Interfaces->SIP or H323 dialogs). Try setting the value to zero. If the problem still persists make a note of the PABX channels being used and double check that there is no noise on those channels. Also check the VoIP service by making a packet capture and extracting the audio.

1.2 One way audio

If the IPCallBox is setup behind a router that is performing NAT and you can't hear audio from the service provider then please see question 3.4. Another possible reaon for one way audio may be mismatched codecs, with one or other end using a codec that is not supported by the other end. You can see the audio codec being used for a call using the out 170 command.

1.3 Talk over and/or clipping

This is caused by the packet sizes being too large. Try selecting a lower packet size. Recommended values are 20ms for G.711 & G.729 and 30ms for G.723. You will need version 9 of the Configuration Editor to alter the packet sizes.

2 ISDN

2.1 What debugging options are there for ISDN?

User debugging is performed through a telnet connection. Connect a telnet client to the box (in Windows Start->Run, type...telnet <ipaddress of box>). Then use one of the following commands.

out 70 - this will provide a small amount of debugging information
out 170 - this will provide a greater amount of debugging information

2.2 The line LED is alight and the route in the configuration file is correct yet calls don't go out over line!

This is usually because the IP address of the DNS server is not set. Depending on the Ethernet port you use, either set it in the Management Station for the LAN port (Selected Unit->IP Config) or in the Configuration Editor for the WAN port (Interfaces->WAN Ethernet).

2.3 Calls are not being deflected as intended.

This is caused by one or more of the following:-

Configuration file on the unit doesn't include a call deflect rule or number (Unit Settings->Advanced)
The line is not subscribed to call deflection or call rerouting. These are the two types that we support.

3 IP

3.1 When behind NAT do I need to setup port forwarding?

Yes, you will need to setup the following ports to be forwarded:-

23 TCP (telnet)...if the external port 23 is in use then use port 2323 externally
80 TCP (HTTP)...if the external port 80 is in use then use port 8080 externally
10190 UDP (management station)
10191 UDP (management station)
5060 UDP (SIP signalling)
6001 TCP (management station)
6002 TCP (management station)
6003 TCP (debug)

3.2 Which ports must I allow access for in a firewall?

In addition to the ports in the previous question, you should also allow access for the following:-

If rtpportbase is not configured: [8000,8199] UDP (RTP media)
If rtpportbase is configured: [alpha, (alpha + 999)] where alpha is the value of the RTP port base
[16000,16100] UDP (SIP signalling and T.38 fax)

RTP port base is configured in the configuration file. This can be changed by Configuration Editor version 9 upwards (in previous versions you must add the string "central rtpportbase 9000" as an Edit->Extra command).

3.3 How can I analyse packets going out over IP?

Although we don't provide any network diagnostic tools there are plenty of good quality freely available tools to choose from. One good tool that we can recommend is WireShark

3.4 I only see audio packets sent out from behind NAT (Network Address Translation), not the other way.

This is the classic problem when using NAT. To overcome it you must usually either supply a STUN server that exists in the same network as the gateway/proxy or else turn on a NAT helper on the gateway/proxy. We do not support the use of outbound proxies. This can also be caused by a firewall so check that as well.

3.5 I'm using DHCP (Dynamic Host Configuration Protocol) on an interface but it doesn't appear to be picking up the default gateway.

This is usually caused by having DHCP default gateway configured on both Ethernet interfaces with both Ethernet layers up. In this situation whichever interface was the last to receive the information will be the default gateway. This is undesirable behaviour so care should be taken not to let this happen.

4 VoIP

4.1 What debugging options are there for SIP?

All SIP debuging on the IPCallBox is performed through the telnet interface. SIP messages will be printed out with the "out 70" and "out 170" commands. The latter providing a greater levels of information. To turn this off use the "out 0" command. To see the actual SIP messages being both received and sent use the "log sip" command. To turn this off use "log sip 0".

4.2 What debugging options are there for H323?

4.3 Why are partial numbers being dialed?

This is due to one of two reasons. Either a dial string rule is being hit before the intended rule or else you need to use the "incomplete" tick box to signal that there are more digits to follow and for it to use the interdigit timeout.

4.4 I have other VoIP issues, can you help?

If your question isn't answered here then either try calling us (0870 7066073) or for more generic problems we can recommend a good website...VoIP Trouble Shooter

5 Configuration Editor

5.1 Why is SIP/H323 greyed out?

There are many different configuration options between SIP and H323. Because of this when a new configuration file is created you must choose between SIP or H323. The other VoIP protocol will be greyed out and you will not be able to alter that configuration.

5.2 What does "incomplete" mean?

Incomplete in the dial string filter dialog indicates to the IPCallbox that it must wait for more digits until the interdigit timeout expires. This is useful when sending number of varying lengths to SIP carriers as they rarely support partial number dialing (overlap).

5.3 How can I divert fax calls through my line rather than VoIP?

This is useful when you are using the IPCallbox to add VoIP to an otherwise ISDN only circuit. If you know the number of the fax machine then you can set rules that match on Calling Party Number. This is found by right-clicking over the link arrow between a source and destination and selecting "Calling Party Numbers".

5.4 I've put in a routing rule but it isn't being hit.

The order of the rules is significant. A rule that is higher in the routing list will be matched first.

6 Management Station

6.1 I can't see a unit I've just added

If you have just added a unit and can't see it appear in the Management Station then this may be because it already exists at another IP address. To rectify this you must remove the other instance, close down the application and then restart. Only now will you be able to add the unit.

6.2 How do I upload a configuration or new firmware?

New firmware is uploaded to an IPCallBox using the following steps in the Management Station

Select the unit from the left hand side
Either right click over the unit or go to 'Selected Unit' in the menu and choose 'Code and Configuration'
Select the configuration/code(firmware) tab
Click the Select button to choose the firmware to upload
Choose when to reboot the unit using the radio buttons
Now click 'Send to Unit'. A message saying 'Operation Complete' will be displayed when done. The new configuration/firmware will be used as soon as the unit reboots.

7 PPP

7.1 Another IPCallbox is unable to dial in and bring up a PPP link.

The static route is probably missing in the PPP interface setup, or the password been set incorrectly. The IPCallboxBox determines the type of the cincoming call by looking at the type of Berarer Capabvility being used. The advanced units setting window determines which bearer capability is matched in order to terminate the call within the IPCallBox PPP stack. This is in fact the case although there is no need to enter this route into the routing table. Then imagine a filter, whilch matches which has been set into the unit settings advanced window. All calls matching the particular bearer capability will then terminate in the PPP stack of the IPCallBox. The PPP stack will then try to match the calling party's password, name, subaddress and any other fields that may be present to its PPP settings. However, so that the IPCallBox can reply back to the calling party a 'static' route' must be set on the PPP interface back to the caller.

7.2 If I dial into another unit, what is the IP address of the remote unit that I enter into my voice routing configuration?

Unlike a normal IP router that expects to route incoming data on the PPP link on to another router or host on its LAN, the IPCallBox epects to terminate to VoIP packets itself. THerefore the IP address of the remote IPCallBox that you would use in your local IPCallBox voice call routing configuration would be the IP address of the distant end of the PPP link. Since it is normally the unit that answers a PPP call that assigns the IP addresses for the PPP link, the IP address of the distant end of the IP link is the "server", or remote IP address in the PPP link.

8 Miscellaneous

8.1 Under what conditions does the red alarm LED light?