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2008/10/5 robert.boardman at gmail.com <robert.boardman at gmail.com> > Kevin P. Fleming wrote: > > Olivier wrote: > > > > > >> 2. R Hook-flash key is now available to transfer calls. > >> In s450IP web management server, its defaults settings are : > >> Application-type: dtmf-relay > >> Application-signal: 16 > >> >

Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123

Hi, My Gigaset S450ip allows 2 simulatneous calls when each incoming call are targeted to different phones. When both calls target the same extension, the second one is forwarded to voicemail. I couldn't check yet SIP messages but has anyone met this limitation (one simultaneous call per phone) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, How to set a Gigaset S450IP up to support R-key transfers ? Mine is enabled with firmware 02223. In Settings/Telephony/Advanced Settings, I set: DTMF Send Settings: SIP info Call transfer: Use the R-Key to initiate call transfer: yes Transfer call by on-hook: yes Derive target address: from SIP URL Find target address: automatically Hold on transfer target: attended and unattended Then,

Hello, My goal is to provision C450IP or S450IP models. Has anyone a hint to provision them from configuration files ? Usually, we use dedicated menu embedded inside Gigaset handset. An http server also exists but I couldn't find any dhcp-tftp combination to configure them. Any clue ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Could someone please help me with this? I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS, where an incoming line is plugged and also analog phone plugged to the FXS port. Am faced with the problems below. - For conversations between analog phone and sip phone, Analog phone can't here the SIP user

Hello, When sending this AMI request ... 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123 192.168.64.5 -> Context: local 192.168.64.5 -> Priority: 1 ... I've got this INVITE from Asterisk INVITE sip:9121 at 192.168.100.198:5060;user=phone SIP/2.0

MWI-related SUBSCRIBE message to send NOTIFY messages changing phones MWI status. This is fine for me but I'm wondering what if I were using SIP hardphones refusing any such NOTIFY without prior SUBSCRIBE (does such phones exist ?) ? 1. In this case, which URI shall use a hardphone to build its SUBSCRIBE message ? Here is a hand written example. Which value should I substitute to foo (in this

Hi, Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges). If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here ... interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table

Hi, Is it possible to create Rose Diagrams of wind data (speed & direction) with R?? Best regards, Lars Peters ----- Lars Peters University of Konstanz Limnological Institute D-78457 Konstanz Germany phone: +49 (0)7531 88-2930 fax: +49 (0)7531 88-3533 e-mail: Lars.Peters@Uni-Konstanz.de web: Lars Peters <http://www.uni-konstanz.de/sfb454/tp_eng/A1/doc/peters/peters.html>

Hi, I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces. I'm using web interface at the moment. Setup is: ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone> I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in when a call comes in. Previously,

I am doing a little application to originate a call through Asterisk via AMI (Perl Asterisk::Manager). It logs in successfully, does an originate command with Exten: 0020 (which is set up to answer and wait for 60 then hang up) Channel: SIP/5101234567 at test-host (which comes to my desktop machine also running Asterisk). At the target machine I see only a CANCEL to which it immediately responds

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

Hi, Has anyone met something like this ? dialor*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 7541/7541 (Unspecified) D 0 UNKNOWN 7540/7540 (Unspecified) D 0 UNKNOWN 7534/7534 (Unspecified) D 0 UNKNOWN 7533/7533 (Unspecified)

Hi, I can't find a way to let users defined in a specific section to use a given timezone. Am missing something obvious ? voicemail.conf: [general] tz=paris24 envelope=on [zonemessages] paris24=Europe/Paris|'vm-received' q 'digits/at' R [mysection] 7530 => 1234,foo at example.com,,attach=yes|tz=paris24 7531 => 1234,foo at example.com,,attach=yes When 7530 is

Hello, I've got a big problem. I'm using R for geostatistical analyses, especially the field-package. I try to generate plots after the kriging process with help of image.plot(..., col=terrain.colors, ...). Everything works fine, but I want to reverse the color-palettes (heat.colors, topo.colors or gray()) to get darkest colors at highest data-values instead the other way round. Could

Hi, I usually use lme from the nlme library. Now I have read an article about lmer in Rnews and lmer seemed to me more comfortable to use. Unfortunately, I didn't find out how to use covariance structures (e. g. corSymm(), corAR1()). Is there a way to use them similarly as in lme ? Is it implemented ? If somebody knows, please let me know. Thank you very much in advance, Stephan

Dear all, I've got a big problem. I try to analyse my data using R with a mixed model ANOVA without useful results and success. My data are as follows: 3 factors (Treatment, Site, Subsite) with 'Subsite' as random factor and nested into 'Site'. I want to analyse the effects of the three main effects (factorial design to a specified degrees (2)) with the interactions between

hi rsync-list, i have an authentication problem with my rsync-server. @ERROR auth failed on module XXX i`ve searched for an solution that fixes this problem in the mailinglist archives of rsync. but no one solves my problem. i`m running rsync version 2.5.6cvs protocol version 26 on debian 3.0 with kernel 2.4.22 my rsyncd.config looks like this: #######################################

Dear R community members, I think I am asking a very simple question, but I really looked up in the faqs and manuals and found nothing helpful. I am trying to plot a data frame with the following structure (this is just a small extract): glo conc odor line series X1 X2 X3 X4 X5 X6 X7 X8 X9 X10 X11 X12 X13 1 0 AIR LN1 UP -0.488