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End point FAQs and troubleshooting
Near Optimal solution - Gigaset N720IP
Gigaset offer this solution for mid-to-large scale (up to 100 handset) roaming solutions. It seems to be a very capable self-installed solution. It would still be advisable to have a professional site survey to determine the best locations for the installation of radios.
It will be automatically provision if you are using recent PABX firmware, and the flexibility of the provisioning will improve over time.
- Upside - Almost "Perfect" solution. Mostly auto-provisioned.
- Downside - Need 3rd party for radio survey/installs.
Simple Alternative 2 - Gigaset N300IP/N510IP with no repeaters
These Gigaset DECT basestations support 6 handsets and 3 simultaneous calls. You need to be sure to buy the correct handsets to support all of the call and roaming features. You may also need more base stations due to their lower capacity, but handsets can be set to switch to best signal as the phone moves. They still do not roam in-call.
- Upside - Slightly better roaming solution. Mostly auto-provisioned.
- Downside - Lots of manual configuration, more base stations.
Alternative 3 - Gigaset N300IP/N510IP with DECT repeaters
In the lab we have managed to configure a DECT repeater onto a Gigaset base station, so some of the issues from alternative 2 can be solved by having fewer base-stations, and linking up repeaters (a repeater is dedicated to a basestation). This technology was tested a long time ago, and we are not up-to-date on DECT repeater technology, so you'd need to take some advice on them, or test it out for yourself.
The device we used allowed you to repeat 1-hop in any direction from the base. You can have up to 6 repeaters on a base.
- Upside - Full roaming solution within repeater range. Base stations mostly auto-provisioned.
- Downside - Significant manual configuration of repeaters. One-hop limited range.
1) It is possible that the phone is requesting its configuration from the wrong server (some phones come with this field pre-set). Try doing a factory reset of the phone.
2) The DHCP server that the phone gets its IP address from must have the "066: TFTP Server Name" parameter set. The PABX administration guide has details on configuring this setting on Unix or Windows DHCP servers. If your DHCP server does not support custom parameters, auto-configuration will not function.
3) The DHCP server will also be handing out DNS server addresses to the phone. It is required that ALL of these DNS servers can resolve the full hostname of the PABX to the correct IP address. A common error is to use a made-up domain-name for the PABX, but still use and Internet DNS server that cannot resolve this name on the phone.
4) Auto-configuration of more recent models of phone may not be supported by your version of the PABX software. An upgrade may be necessary.
The username and password is sent to the phone based on the current system configuration. If the configuration has been changed but the phone has not recently reloaded its configuration, it may be out-of-sync. A reboot should fix this.
Some phones have a "user" login. This can be seen by looking on the PABX web interface "System -> Phone Hardware" screen, and clicking the "Show Info" icon.
Other phones have only an administration login. The password for this login is set in "Global Settings" in the PABX web interface.
1) Is the call being routed over the Internet, or other WAN link? If so, then you may simply not have enough bandwidth on the link. It may help to configure the link to use low-bandwidth codes.
2) If the problem is occuring on calls made internally, or via an ISDN link, then it is possibly a LAN traffic issue. Try connecting the phone to a switch port that is closer to the PABX and see if that improves the quality. Ensure that the network is switched, and does not use hubs. Use a network monitor to check that there is not a large amount of broadcast traffic on the LAN.
This is generally caused by one of:
1) The DHCP server has stopped functioning.
2) The DHCP server has handed out a duplicate IP address (often caused if an ASDL router is serving DHCP, rather than a server)
3) The DNS server is no longer serving the name of the PABX correctly.
4) There is a LAN issue, perhaps a faulty switch or router.
This is caused by a facility called "RTCP", which is some extra control signalling in the voice data. Some devices support it, and some devices do not. If a call switches from a device which supports it (eg. snom) to a device which does not (eg. an ISDN call) then some softphones can timeout and hangup the call.
This can be resolved by disabling the timeout. On X-Lite, Eyebeam and Bria, this setting is found under Network/Advanced in the preferences menu.
Most phones allow the microphone volume to be configured for the handset and headset. This is configured by using the phone's own menus or web interface. The PABX will not attempt to set these values.
Phones are DC powered, un-earthed, and susceptible to mains AC noise. This can normally be solved by providing a better earth to the phone by using a SHIELDED twisted pair cable, and ensuring that the socket the cable is connected to has a path to earth.
(Note: if the phone has 2 network ports, and one is connected to a PC, using an STP cable between the phone and PC will be the easiest solution)
Please contact your telephony service provider and request that this facility is enabled. The PABX will always present a supplied caller-ID. If it is missing, it is probably because it is not being sent.
By default the PABX sends your extension number as the caller ID. The ISDN network, or other far-end service provider will then try to resolve this to a "valid" number. This resolution may vary between providers, so the results will vary.
To send a different number or to bar the sending of caller ID, edit the caller id override field in the edit-extension screen of the PABX web interface.
Extensive per-provider caller-ID configuration options are available. Click the '123' icon next to the provider that the call is being sent to to check these settings. Different providers have different requirements, and behave differently if invalid numbers are sent to them.
Below are basic and generic configuration instructions. Please contact our support department for information on the latest compatibility with SIP software for Android and IoS.
First create a phone entry on the PABX:
- On the PABX web UI, manually add a generic phone to the PABX from the System -> Phone Hardware menu.
- Use a made up, unique MAC Address and set the phone type to Unknown/Other.
- Unless you use static network settings (definitely not recommended) don't enter any network settings
- Set Monitor/Keepalive Override to Enabled.
- Having created the phone, click on Show Phone info (magnifying glass symbol) next to the phone and note the Line 1 SIP Login/Pass username and password which you will need when setting up the mobile.
Using the PABX hostname, and the SIP username and password from above, follow the configuration instructions for the handset software. It will probably be necessary to disable all audio codecs except "alaw/g711a" and perhaps "gsm".
1) See the section above entitled "The manual says our phone can be auto provisioned, but it is not working." Most often this issue is caused by an issue with DHCP or DNS.
2) The firmware on the handset may be very old. In this case it is sometimes necessary to update the phone to a minimum required version before it can proceed.
3) Some manufacturers do not allow us to distribute handset firmware. Verify that the handset firmware version is recent.
4) Check that the PABX has the correct firmware on it. Some firmware files are large and are not included by default. Go to the "System->Upgrades" screen. Click the refresh link, and then use the handset-mode screen to update or fetch the required firmware.
We don't support entering contacts directly to the phone. Contacts should only ever be added via the PABX using one of the following methods:
Logged in as admin...
- PABX -> Phonebook Mgmt. -> CSV Upload Entries
- PABX -> Phonebook Mgmt. -> Additional Entries
- PABX -> Users -> Manage Phonebook
Logged in as a user...
- User -> My Details
- User -> My Details -> CSV Addressbook
This has the advantage of being able to replace a phone in the event of a failure and still retain existing contacts.
Address book order
Handsets normally display the addressbook alphabetically. The address book is filled from the PABX in the following order:
- Auto-generated (Modified via PABX -> Phonebook Mgmt. -> Ex-directory Status)
Please note all phones have a limited number of phonebook entries, once this has been exceeded additional contacts will be ignored.
PDQ and Modem devices will struggle to work on an ATA device except for brief 9,600 baud connections. SIP is not a reliably clocked protocol, so frame-slippage occurs in the translation from analogue to SIP to ISDN and means it is generally very unreliable except for short transmissions.
If the communication is over the Internet, there is virtually no chance that it will work as the jitter introduced as the data crosses the Internet is variable.
For extended PDQ and Modem usage, we suggest connecting them to a real analogue line. The vast majority of services where PDQ/Modem equipment is used are replacing these devices with online equivalents, which will eventually make the problem disappear.
Information updated as of June 2014
Some Yealink products are listed in their datasheets as Hearing Aid Compatible, and others not, but having discussed this with their UK technical staff, we understand that the T41PN, T42GN, T46GN and T48GN models are fitted with a 'T' mode inductive loop in the handset.
Our supplier has informed us that none of the current snom handsets have support for hearing aids. The snom m3 DECT did but it is now a discontinued product.
From the Polycom datasheets on the IP and VVX range handsets:
"All Polycom handsets are Hearing Aid Compatible (HAC) and have telecoils that magnetically couple to most forms of wearable hearing aids per FCC section 508 (compliant to ADA Section 508 Recommendations: Subpart B 1194.23)."
The soft-fax facility is 99% reliable. Sadly, it is still only a piece of software pretending to be a fax and is ulikely to ever be 100% reliable.
We are constantly monitoring progress as this software improves, so if you have issues, let us know and we can keep you informed if things improve.
It is also possible that the Fax receive is working, but the email sending is not. If this is the case, you can find a copy of the received fax on the Users->Messages screen if logged in as administrator. Email configuration is available on the System->Global->Email screen.
There are 2 main reasons for difficulty with sending faxes from an ATA. Firstly, network quality is very important - Jitter or packet loss will prevent faxes from working as a fax machine cannot tolerate an imperfect audio stream; This can be monitored with a network trace, and using a PC product such as "Wireshark" to analyse the results.
Secondly, and more usually, the ATA units have a large number of analogue configuration options, and some fax machines are intolerant of incorrect line settings, some suggestions follow, which apply to Sipura ATA devices:
(Set on the "Regional" settings tab of the ATA's administration interface)
- FXS Port Impedence: The correct UK setting is "370+620||310nF" - For an American device it is "600"
- Caller ID Method: ETSI FSK With PR(UK)
- Caller ID FSK Standard: v.23
- Fax Passthru Method: ReINVITE (Asterisk does not currently support NSE, but this setting should be unimportant)
- Disable any Echo Cancelling and silence supression options, and/or set to disable EC on FAX tone detection
Once the above is set, it may then be necessary to modify the input and output gain settings. Using -6 in both fields is usually the best, but values of 0, -3 and -6 are usually worth trying in various combinations. Set and save a value, then try sending several (perhaps 10) faxes to see what the success rate is, then change one setting and re-test until the results are acceptable. It is usually possible to get a FAX machine working using this method.
T.38 is a relatively new protocol which can be used in the following ways on a digital system:
1) T.38 defines a method for a fax end-point (a fax machine or gateway) to transmit a fax over a network using the UDP protocol natively. This does not fit into Asterisk's means of transmitting data as it describes a method of exchanging data that is unique to the T.38 protocol. Using this method would basically take the PABX out of the loop entirely (which may be acceptable)
It may be possible to use a 3rd party T.38 service directly with a T.38 enabled ATA to achieve bi-directional Fax over the Internet, but this is beyond the remit of the PABX configuration.
2) Asterisk is able to terminate a T.38 call, and the IPCortex PABX will automatically use T.38 if offered on a SIP trunk call that terminates on a soft-fax number. At present there is no facility to scan and send documents outbound from the PABX, only to receive and email them.
3) T.38 can also be represented in a manner compatible with existing SIP devices. At present, Asterisk is only capable of acting as a pass-through agent for T.38 streams, so cannot convert it to other formats - Pass-through allows two endpoints which understand T.38 over RTP to exchange faxes. This might include two ATA devices within the same company, but excludes fax machines connected to an ordinary telephone line, so is generally not particularly useful to our customers.
The PABX does not currently support this mode, but we are constantly monitoring progress as this software improves. It is expected that the "Next generation" PABX release will include a T.38 gateway facility to convert from T.38 to regular fax and vice-versa. This should be particularly helpful to enable sending and receiving faxes using a fax machine over ISDN or over a T.38 enabled SIP trunk.
Sorry, no. A combination of the number of different email document formats, and the processing power needed to convert them to fax (TIFF) format means that this facility is not supported. Alternatives are:
1) Connect a fax to an analogue phone socket - Most people have a spare socket on their ADSL line.
2) Connect an ATA to the network. This is a fairly reliable solution to faxing.
3) Use an external agency to provide a fax to email gateway.