similar to: Advice

Dovecot has been working fine for months. We are using 1.0. Come in to work today and dovecot quit working POP3 and IMAP. I restarted the service and everything started working except IMAP? We use Dovecot as an authentication method for outgoing mail and it works. I can't telnet to myhostname.com 143, Squirrelmail will NOT work either. Where is a good place to start looking for signs of

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

Has anyone had experience using a SIP trunk provided by Paetec over MPLS? With or without FreePBX Regards, Jared Baxley -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120926/436ee928/attachment.htm>

Are there plans to expand the logging capabilities in OpenSSH, so that the details of what files were moved using sftp is included? If not, does anyone know of a good way to capture this information? Thanks in advance, - Sue Susan K. Diller UNIX Systems Administration PAETEC Communications, Inc. 600 WillowBrook Office Park Fairport, New York 14450 *(585) 413-2320 * susan.diller at

Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

I need 100 DID's in a specific rate center (916-854-xxxx). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas

Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

How do you get the called number on incoming SIP calls? I've never had multiple DID's via SIP from one provider before and somehow I never realized that with IAX it just works, and SIP is different. If I don't set an extension in the register command the incoming invite has <sip:s@me.com> in the To field. Now if I have multiple DID's that I want routed to different

I thought maybe one of the providers here could answer this question. When using IAX to make calls, it passes the called number. Being designed to work with a PBX this makes sense. SIP works differently though and I'm curious why providers don't have a way to pass the called number on their DID's, or choose not to do so. Providers that themselves use upstream SIP proxies are

I know I can get free DID's with SIP, is anyone giving out free DID's with IAX? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

OK.. I'm up to my eyes in LD BS! I can't for the life of me understand how any carrier, either VoIP or traditional service provider can make heads or tails of how to hand off an * based call to an LD provider. Every provider I talk to, says I have to have a traditional T1 put in to their respective networks. I don't want to do this. I want a LD provider that can take a IP, SIP, IAX

I have a pretty strange problem. I have about 100 DID's that come down a PRI from SBC in the United States. On Friday afternoon, one of my DID's flipped out. When you call it, the SBC operator comes on and says that the line has been disconnected. I contacted them and they ran test and they are telling me the problem has to be on my end. My problem is that the CLI never shows the

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from

I wrote a week or two ago about a problem I was having with a TDM400P with 2 FXS. The problem was reliable fax transfers. There were often slow or had communication failures during transmission. Richard replied to my post and told me this is an ongoing problem that might have a solution in the weeks to come. However, I was wondering if anyone had any alternate solutions for faxing with an