FAQ

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== Can I use more than one account? == == Can I use more than one account? ==
-The system is not designed to handle more than one account for the same customer. In particular you should avoid registering one terminal to several PBXes accounts at the same time.+The system is not designed to handle more than one account for the same customer. In particular you should avoid registering one SIP User Agent to several PBXes accounts at the same time. If you wish to ring a series of extensions, create a Ring Group and set it as the default destination for your DIDs.
== Are ENUM lookups performed automatically? == == Are ENUM lookups performed automatically? ==

Revision as of 21:48, 19 September 2008

Contents

How are classical extensions set up?

Within PBXes an extension on a trunk is called a "classical extension". All you have to do is create some of these for the PSTN numbers you want to forward to. The number format depends on your trunks. If you are not sure about this, find it out by using your VoIP provider from a softphone first.

How can voicemail be configured?

These voice prompts may only be configured from a phone by dialing *97 or *98.

Should music on hold be uploaded at any specific sampling rate?

Music on hold may be uploaded as MP3 or WAV files. If MP3 does not work try WAV, which need to have 8000 Samples per second, Mono, PCM, 16-Bit format. MP3s may be at any sampling rate. Sometimes it became neccessary to reset the PBX (by clicking "Submit & Start" in "Personal Data") to get music on hold running initially.

We recommend Goldwave to edit sound files and convert them.

Announcements for queues and digital receptionists may be uploaded in WAV format only.

How can I reach SIP addresses?

For reaching SIP addresses on external servers not registered to I have added an option. It's available under the Device Options of a SIP extension and called "dial". Try putting your SIP address there, e.g. sip/21488@fwd.pulver.com. Since this is an advanced option it only appears when editing an existing extension.

How do callback and callthru work?

Start with callthru by adding an inbound route with the number of your phone entered as Caller ID. Leave Trunk Name empty. If it doesn't work check your call monitor for the right format of your Caller ID. When dialing the callthru destination you may always press the * key to redial and # to end digits. After the called party hangs up you will hear another dialtone for your next call. To hangup by yourself transfer the call to an invalid destination by dialing *2.

What handsets are recommended for PBXes?

The Nokia E-series phones can be used as a GSM mobile when you are on the way, and as a PBXes extension when you are within WLAN coverage. Be sure to update them to the firmware of August 31 (or newer) to avoid loosing registration when moving across areas of poor WLAN coverage. PBXes supports all features (including putting on hold and call transfer) as well as call forwarding to your mobile number if you are not registered to WLAN. Numbers can be automatically converted from +xxx international format for IP calls by writing 011++|. into the trunk's rules (substitutes + by 011). Only available to paid accounts are ConnectionCheck for immediate detection of leaving WLAN and EarlyConnect for better quality on outgoing calls.

On the other side of the spectrum, any SIP based desktop IP phone should work properly with PBXes. We have gathered reports that Linksys and Grandstream models integrate well and are easy to configure. In particular the Linksys SPA9xx line along with the Grandstream GXP and BT lines of IP phones.

Last but not least, in order to keep using your existing fixed-line handsets (cordless, custom-made, etc.) ATAa (analog telephone adaptors) are a very economical alternative to PC based soft-phones. Again, all Linksys (ex Sipura) and Grandstream units integrate very well with PBXes. Another category which should be mentioned here, are the ATAs built-in to broadband routers and/or ADSL modem/routers. We have discovered a high degree of compatibility with the Draytek Vigor routers which have built-in FXS ports.

Can I use more than one account?

The system is not designed to handle more than one account for the same customer. In particular you should avoid registering one SIP User Agent to several PBXes accounts at the same time. If you wish to ring a series of extensions, create a Ring Group and set it as the default destination for your DIDs.

Are ENUM lookups performed automatically?

No. You can use an ENUM trunk for peering to other VoIP subscribers without a provider. When adding the trunk write the following rules into it to bring the numbers to ENUM format (e.g. 011|., 1aaa+NXXXXXX, 1+NXXNXXXXXX) and put it on topmost position into the trunk sequence of your outbound routes (Creating a sequence of trunks is limited to Premium and PRO Accounts).

What can be done if the PBX does not work any more?

If one of your providers fails for more than 30 minutes it gets deactivated. So it remains grey in the status display. To activate it again re-apply one of your settings (no matter which) and press the red bar. Paid accounts automatically recover from such failures.

In rare cases a PBX had to be reset to get functional again. Symptoms were extensions that could not register and music on hold that was not played any more. A reset can be done by clicking Submit & Start in Personal Data.

In case of a server outage paid accounts are automatically moved to another server. For this to work be sure to always enter "pbxes.org" as your SIP server. If there is a partial server outage (very rare!!) moving manually to another server can help. Choose a different timezone and click Submit & Start to move your account. You can find an overview of all servers at pbxes.org/status.

Calls ring but drop immediately when answered.

If you have such problems with incoming calls it is likely that a trunk was registered twice. Check if you have still registered to your SIP provider directly from a phone. Delete those! A reason can also be that a trunk has recently been renamed. In these cases it can be neccessary to wait up to a day to clear the registrations.

Another reason can be found in the advanced options "audio bypass" and "G.729 passthru". That's why they are disabled by default. Disable to find out whether one of your phones or providers don't support them.

I am using provider A for incoming DID and provider B for outgoing calls. How can the Caller-ID be shown?

That depends entirely on how provider B (trunk) handles Caller-ID. Some trunk providers suppress the display of Caller-ID, some of them display a Caller-ID of their own choosing, and some of them allow you to set your own.

If your provider falls in the 3rd category, then you can set your Caller-ID in two places. In every extension at the "Outbound CID" field, on in the trunk at the "Outbound Caller ID" field.

How can I pass the Caller ID when forwarding calls to my mobile?

You need to find an outgoing provider that is able to pass Caller ID to the countries that you are mostly calling to. See forum for further info.

Why does Paypal ask me for a credit card?

Paypal takes subscription payments from local funds when available. Entering a credit card is required by Paypal just in case you run out of local funds. According to their customer support this is a general policy.

You must not remove your credit card from your paypal account after subscribing because this would cancel your subscription immediately. It has to be present as long as you want to keep your subscription running.

That's why we recommend the following procedure if you want to change credit cards within the Paypal system: enter a second credit and verify it, then change the funding source of the subscription to your new card, and after that you may delete the first card. Watch the messages on the Paypal pages carefully while deleting a card.

Outgoing calls work but incoming calls are always busy or answered by voicemail.

In rare cases a single or all extensions need to be saved again (without changes).