search for: hastingsutil

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may

1. RE: Start recording automatically when xferring to anextension? (Don Pobanz) Message: 1 Date: Fri, 25 May 2007 11:54:33 -0500 From: "Don Pobanz" <dpobanz@hastingsutilities.com> Subject: RE: [asterisk-users] Start recording automatically when xferring to anextension? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <67F14E0CCFEA374CADFD6A8AB17A450D1943EC@es2006.HUDOMAIN.local>...

We have had a few different times when a user has forwarded their phone to himself. This has overloaded the communications to our operator panel (FOP). One user should not be able to effect the whole phone system! Is there a way that the number of times that a call can be forwarded could be limited like to 10 or even 100? Then even if a user does something stupid like forwarding their calls to

Thanks anyone, I found the problem in rhconfig.h. After the fix I successfully compiled zaptel. V. --- Ron Wellsted <ron@wellsted.org.uk> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Have you built your kernel on that machine? > > The errors suggest that while the kernel sources are > installed, the > kernel has not been built. > > Check on

When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507)

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

Hi list, Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo

I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes.

Since Colin Anderson -- in a previous thread -- asked the question about whether ADSI was dead, I thought it was worth discussing. We've been using Nortel Vista 350s in our office up until now. The phones are from Telus, I don't know if there's any way to unlock them. It would appear Telus hasn't done much in the way of updating software for these phones; or if they have, they

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

There has been discussions about the voicemail menus and some of us would like to see an overall plan for the voicemail menus. There are 3 primary ways of arranging the menus. First is a tree structure, second is a random access structure and the third would be a hybrid of the two. (Comedian mail is currently a hybrid.) As was pointed out by Brad Bergman, the ideal would be to have it

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

I managed to get Asterisk working with my PBX using T1, now I am moving on to trying to make PRI work. I have my zaptel.conf and zapata.conf configured as follows: Zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf: [channels] transfer=yes immediate=yes callprogress=yes language=en context=default switchtype=national signalling=pri_net group=1

Another issue to consider is T1 framing. If your application is putting bits onto the T1 at the rate of 1.544 Mbit/s then the T1 would need to be unframed. I don't believe this is an option in zaptel! If however, it is putting bits on at a rate of 1.536 Mbit/s and adding 8000 bit/s for framing then you may be able use the suggestion below. Don Pobanz On Thursday, April 03, 2003 3:28 PM,

<snip> > Paul Crick wrote: > > Brad Bergman wrote: > > my thinking is that Comedian Mail is its own thing with > > its own interface and users who have become accustomed to > > it, and it needs refinement before it needs an Octel emulator > > I guess it's each to their own. Maybe * could come with a default > Comedian Mail configuration file then have