search for: pobanz

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 --------------

...cemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz

...ensions.conf I have added the 'd'ata option. exten => 333,1, Dial(Zap/19,,d) In zapata.conf echo cancel has been turned off and I have experimented with different size jitterbuffers, none of which seem to help. I couldn't find anything else in the archives. Other suggestions? Don Pobanz

...on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also welcome. Don Pobanz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061119/0fab481e/attachment.htm

...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

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

...rid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the warning "file chan_zap.c line 4155(ss_thread):callerID returned with error on channel'Zap/6-1' advice and suggestions will be appreciated........ On Thu, 23 Oct 2003, Don Pobanz wrote: > On Thursday, October 23, 2003 8:52 AM, austino@skannet.com > [SMTP:austino@skannet.com] wrote: > > > > > > hello * users, > > i am using a zplex 10b(8fxosand 16fxs) connected to a T100 digium > > card, > > RH8.0. > > > > soon after run...

...ude/asterisk/term.h:47: previous declaration of `term_color' term.c:98: conflicting types for `term_prompt' include/asterisk/term.h:49: previous declaration of `term_prompt' make: *** [term.o] Error 1 [root@localhost asterisk]# Can anyone give me a hint of what the problem may be? Don Pobanz

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

Hi all, I am having a problem on to different boxes and to different channel banks. I can't get dial tone out of either one. I can still send and receive calls but no DT. This is the error that I get: Jun 8 15:23:54 WARNING[5021]: chan_zap.c:6283 handle_init_event: Unable to play dialtone on channel 95 asterisk-1.2.9.1 zaptel-1.2.6 zapata.conf: signalling=fxo_ls context=gene

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...

...ser 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 himself, it wouldn't cause problems for others. Don Pobanz

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

Is there something special that needs to be done to allow a T100P/T400 to respond to a remote loop request? I am having some issues with a Point to point T1 line using zaptel ppp. This line gave us small problems when we had a pair of Cisco 2600's on either end but now with the zaptel ppp it is going down every couple of minutes for 15 intervals. The phone company says the line is good to

...or to a Grandstream sip phone. So, I do not know whether this is a Sipura 2000 problem or an * problem. Does anyone have any light to shine on the subject. Asterisk CVS-HEAD-06/14/04-09:03:15 built by root@localhost.localdomain on a i686 running Linux Sipura 2000 software version 1.0.33 Don Pobanz

...ribute to a slip but contribute to a LOF (loss of frame). > When set for B8ZS the "error" is somewhat > expected and ignored. Some errors are expected and the expected errors do not count as a bipolar violation. There can also be bipolar violations on a B8ZS line coded T1. Don Pobanz

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

Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 -> voicemail -> Secretary 1 555-1235 -> voicemail -> Secretary 2 Any help would be greatly