similar to: SIP Phones Features

Every time I send an e-mail to the * list, I receive this "AntiSpam UOL" E-mail. is anybody else experiencing the same? How can I get rid of it? Uriel -----Original Message----- From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] Sent: Wednesday, September 24, 2003 11:51 PM To: uriel@adelphia.net Subject: RE:RE: [Asterisk-Users] SIP / GrandStream Configuration Ol?,

I have seen a couple of messages on the Xten and the work done by William Walsh (Kudos). It is not clear in my mind the advantages of SIP phones versus using GnoPhones (once we complete the work for the Windows version). Since I lack the experience with IP SIP phones, can someone, high level, tell me when it makes sense to use them. Is it complicated to set up on the Asterisk side? Thank you.

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Where can I learn the syntax for DBput and DBget? is it working with MySQL? do I need to set up tables? URiel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030508/a2598dc8/attachment.htm

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

How is a callgroup used in the dial plan?? I can't see an example in the extensions.conf.. > See configs/sip.conf.samples search for pickupgroup > > RTFSC (Sample Configs) > > > Jeremy McNamara > > > > > WipeOut . wrote: > > >Is there a way to pickup a call whn using a SIP phone? > > > > > > > >> From

Callgroups/pickupgroups are allocated per channel, not in the dialplan. sip.conf and zapata.conf are the two files you're interested in. -wade ---- Original Message ---- From: wipeout@linuxmail.org To: asterisk-users@lists.digium.com, Subject: RE: [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension) Date: Sun, 20 Apr 2003 16:39:15 +0000

I have one of my asterisks off an xDSL modem with a dynamic IP address. The IP address keeps on changing so I am curious as to what other people are doing to work around this issue. I am contemplating to register for a dynamic DNS service but then my communications become dependent on the availability of this service. The "host=dynamic" for SIP seems interesting. Any suggestions?

vmail.cgi rocks (if I can borrow the expression for Mark Street). As Mark pointed out, the /vm/INBOX messages are created with 0700 security and vmail.cgi is not happy. Apache/cgi/vmail.cgi cannot play them unless I fool around with the Apache wrapper or chmod 755 *.* thefiles myself. (Tedious, that is why I like computers). Obviously this is not acceptable. I took a trip to the

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

On Wed, 30 Apr 2003 00:24:19 -0400, Uriel Carrasquilla wrote: > >Gary: >I just copied the content from chan->exten to chan->dnis. I am calling from How are you doing this coying ? >one extension to another. >Have you got DynExtenDB to work? nope, haven't got over the first problem yet. Gary .

"classwt" in the current version of the randomForest package doesn't work too well. (It's what was in version 3.x of the original Fortran code by Breiman and Cutler, not the one in the new Fortran code.) I'd advise against using it. "sampsize" and "strata" can be use in conjunction. If "strata" is not specified, the class labels will be used.

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. PRI/T1----- | | | ---------------- | | | Channel Bank | | |

Andy; Well, that about does it.... I'm copying this one back to the list for the benefit of those who may hit this thread while searching the archives. Your changes to the code run just fine on the my Windows machine, but gives the vector length error on the G4 whether I'm using the OS X build of R (as in Raqua) or the X11 build (for Darwin). It is worth noting that I have nearly twice

Sorry for the repost, but I've really been looking, and can't find any syntax direction on this issue... Just browsing the documentation, and searching the list came up short... I have some unbalanced data and was wondering if, in a "0" v "1" classification forest, some combo of these options might yield better predictions when the proportion of one class is low (less

Hi all, I need a suggestion or just to know if it's even possible to achieve the following. There is a "central" vpn server which is my main network. I have a few other gateways (customers) which should connect to this central server (there's a firewall on this machine too) which have behind the customer network. Then I have a few single servers which still connect to my

Full_Name: David L. Van Brunt Version: 1.8-1.9 beta OS: OS X 10.3 Submission from: (NULL) (68.74.58.109) As posted on R-Help (after which another user replicated the problem): --------------- This is the conclusion from a prior thread ([R] " cannot allocate vector of length 1072693248") which ended with no other answer but that there must be a problem in the OS X version of R, or in

Hi all, Can any one please help me in intergrating PHP/Mysql with my running asterisk server to configure IAX or SIP users? I will highly appreciate any help in this regard. Regards Nawaz. --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, >