RE: No service 23.03.2021 18:34

Forum: Bugs

Trying to figure out my options. Are you expecting to restore service for non-static IP accounts, or is our only option to get back up to upgrade to pro?

Thanks,

RE: Sip registration issues 13.03.2015 15:20

Forum: Bugs

I am having a similar problem. When I come in for the day none of my phones are registering. The pbx logs show no activity. But if I reboot pbxes from the personal data page everything starts working again.

Any ideas?

Trunk Dial rules 27.03.2013 20:02

Forum: Miscellaneous

I'm confused about the Dial Rule Field on the Trunk Screen.

I see the pop up explanation for Dial Rules but the two default rules already present:

b - Pref = "100", Pattern = "9(\*)([0-9]*).*", Replacement = "\1\2"
a - Pref = "200", Pattern = "9*", Replacement = ""


dont seem to follow the structure in the pop up. Do I just ignore these?

The first ones I was trying to create are: 1) a rule that adds a starting 1 if it isnt already there, and 2) a rule to delete any parentheses or dashes in the number since my DID provider rejects numbers containing them.

Any help would be appreciated.

Thanks

RE: Unable to access account 09.01.2012 18:46

Forum: Bugs

Well I am on www4 so that explains what happened.

Interestingly, If I am reading it right, the Pbxes status screen shows w4 as having had 100% uptime for the past 7days..????

Should maintenance have resulted in the whole server going down? why couldnt you have done a failover to another server like you usually do?

If it was planned and a service interruption was unavoidable shouldnt this have been scheduled for late at night and some notice sent out ahead of time?

As I said in my last email, you guys usually do such a great job. Still recovering from New Years Eve?

RE: Unable to access account 09.01.2012 04:44

Forum: Bugs

my Pro account was also down for much of the day. My phones could not register and I couldnt log on to pbxes.com either.

It seems to be up now but there was no failover to another server and no acknowledgment when the site went down or when it came back up.

What happened? If this had happened on any day but a Sunday it would have been very disruptive to my business.

This is not typical of the very good service I have had from Pbxes.com. and I guess i would like some kind of explanation and reassurance.

Thanks,

MB

wrong caller id when forwarding calls to gvoice 14.11.2011 16:21

Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

I have all my voicemails forwarded to my email.

I thought I would try substituting Google Voice for PBXes internal voice mail since Google voice puts their transcription of the message in the body of the email forwarding each voicemail.

I have tried putting my gvoice number into the field for the number to forward to. It works but the incoming caller id on the message is some strange phone number in the 626 area code.

I then tried creating a gvoice trunk in pbxes and assigning it to an outbound route to be used when a call is made to my gvoice number. No difference. Same (626) caller id shows up.

Any ideas?

RE: Voicemail in MP3 format 02.02.2011 20:57

Forum: Feature Requests

I would vote for this too.

But, sticking with wav is fine if Pbxe's wav files could be made to be compatible with android platform.

Anyone know what it is about the way pbxes encodes the wav file that android doesn't like?

Mark B

RE: Outbound Caller Id 18.09.2010 19:52

Forum: Feature Requests

after adding to the thread on this issue in the Providers Forum, I found the thread here.

When will a switch to toggle the sendrpid variable be available?

RE: Change CID 16.09.2010 20:59

Forum: Providers

I am also having trouble having the CID I have entered on either the extension or trunk screens passed through to my DID provider, Callcentric. The correct outbound CID is showing up in the PBXes.com call monitor but is not getting to Callcentric.

They say that sendrpid must be set to yes and trutrpid must be set to no in the Asterisk sip.conf file.

Looking at the PBXes.org source view for sip.conf., neither variable appears even when I have placed an phone number entry for outgoing caller id in the trunk or extension screen.

Could this be the problem? If so, could you had a setting to the UI to allow these variables to be optionally set?

Thanks.

RE: call forward from one extension to another 01.09.2010 03:56

Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

Thanks so much for your reply.

The problem was that I was entering userid-nnn in the call forward field instead of just nnn.

It would be great if the extensions screen had a note clarifying this next to the data entry field.

thanks also for you explanation of the dial field function. i am going to look at that now.

Best regards.

RE: call forward from one extension to another 31.08.2010 18:26

Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

>>There's a "call forwarding" section. Why not use the "all calls" option?

I tried but didnt work what how should the entry be formatted assuming I wanted calls forwarded to userid-22


>>I use the "dial" section to have an extension set as a callthru.

Can you explain?

call forward from one extension to another 31.08.2010 00:30

Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

What is the function of the dial field on the extensions page?

According to an old message I read it was created to allow for forwarding to unregistered external pbxes. But when a new sip extension is created it seems to be populatedby default with: SIP/<userid>-<nnn> as if it was set to call itself.

Beyond the general question of the field's function, my real question how to get one extension act as an alias to another, ie, autoforward to another extension on the same system. I have tried putting SIP/<user-id>-<nnn> in the forward field without success. Am I using the wrong syntax? Should I be using that Dial field?

Any help would be appreciated.

TAPI for screen pops 23.08.2010 04:04

Forum: Terminal Equipment

I am new to asterisk and Pbxes.org. I have gotten my pbxes acct working with a LA941 hard phone and also a softphone.

I would like to have the TAPI functionality I had on my old analog PBX which gave me the incoming call info which I ould then send to my case management system so it could pull up the caller's info in a screen pop..

Is this a feature that needs to be activated from Pbxes or is it a matter of having some kind of Tapi software on my workstation, or both.

If someone can point me to a product or website where I can learn more, I'd appreciate it.

MB