search for: tels

Hi All I have a symmetric matrix of genes ( 100x100 matrix). I also have a matrix (100x2) of two columns where column 1 has the gene names and column 2 has the cluster it belongs to (they are sorted and grouped based on the cluster no). I would like to order the rows and columns of the 100x 100 matrix such that the first n genes correspond to cluster 1 and next n genes correspond to cluster 2

Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone system via a SIP trunk using the IPRC card? -- Marvin Horst -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/5dbe435a/attachment.htm

Hello, I was relying on withTimeout (from R.utils) to help me stop nlme when it ?hangs?. However, recently this stopped working. I am pasting a reproducible example below: withTimeout should stop nlme after 10 seconds but in this example it does not. I tried this both on a linux (64 bit, CentOS 7, R 3.4.1, nlme 3.1-131 R.util 2.6) and mac (Sierra 10.13.1, R 3.4.2, same versions or nlme and

Hello All, Just looking some comments from gurus about this proprietary systems and phones: Inter-Tel Eclipse2 Model name: IP PhonePlus I did not find anything useful or reasonable about their products on their website or even in Internet.... except sales. -- Thanks and regards, Vasyl Rublyov

Dear rsync users, I'm trying to copy my Time Machine Backup from my old USB hard drive to a new one. I got error messages, because rsync don't create group and other permissions. I use rsync 3.1.3. Did I use the correct parameters? Who has an idea, what mistake? Thanks! Best, André -- The bash script: #!/bin/bash source=/Volumes/LaCie\ d2\ Quadra\ v3/Backups.backupdb

Hi list, Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a system that uses exclusively tel: uri on inbound and outbound calls. I could not find documentation or sample config about tel:uri. Is this doable? If not possible with PJSIP, is chan_sip a better option? Any pointer would be greatly appreciated. Thanks, -- Jean-Denis Girard SysNux Systèmes

Ich werde au?er Haus sein von 18-06-99 Bis 28-06-99. Ich werde Ihre Nachrichten nach meiner R?ckkehr beantworten. Bitte wenden Sie sich in dringenden F?llen an: Frank R?seler, Tel: MON 3855 Uwe Spiess, Tel: MON 3743 Guido Zimmermann, Tel: MON 3962 Ferdinand Gro?, Tel: MON 2368 lieber Gru? Axel Hellmig

Has anyone had success patching chan_sip.c so that Asterisk will recognize the tel: header for RDNIS information? exten = get_in_brackets(tmp); if (!strncasecmp(exten, "sip:", 4)) { exten += 4; } else if (!strncasecmp(exten, "sips:", 5)) { exten += 5; } else { ast_log(LOG_WARNING, "Huh? Not an

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23] WARNING[6227]:

Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the

Hello, Is there any way I can use the gregexpr functions (or a different function) in a manner that will also return overlapping (i.e. non disjoint) regular expressions? For instance, when running gregexpr("AAA","AAAAAA"), I get two matches, one at position 1 and one at position 4. I'd like to receive 4 matches at positions 1, 2, 3 and 4. Thanks, Schraga

I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... Dustin Moore

Hello! I am using package "VAR". I've fitted my model: mymodel<-VAR(mydata,myp,type="const") I can extract the Log Liklihood for THE WHOLE MODEL: logLik(mymodel) How could I calculate (other than manually) the corresponding Akaike Information Criterion (AIC)? I tried AIC - but it does not take mymodel: AIC(mymodel) # numeric(0) Thank you! -- Dimitri Liakhovitski

Hello, I want to make an 2D plot of .xyz data. So plot on x-axis and y-axis and use a color scale for the z. For every x-location along a cross section of the soil at several depths a resistivity must be displayed. This must result in a picture/graph which shows the resistivity for that cross section. The depths differ for each x-location. How can I do that? In fact it is an cross

i''m doing a site for someone that has a form set up to collect contact information from the user. i installed the vpim gem and was hoping to give my client a way to download all of the contact''s information as a vcard. using the examples, i was able to get it working in ruby, but i''m not quite sure how to generate and download a vcard on the fly. has anyone done

Would you mind repeating the test with canreinvite=no set for all you phones and mobile phones? What is your upload bitrate? Is it guaranteed? I would try also to test the PMTU: Try: ping -M do -s 2000 ${ip address of the sip server} You should receive icmp asking for lowering the packet size. The LTE phones could have lower MTU and thus overcome PMTU problem. Marek 2020-06-22 21:48

Hello! I would like to know if there is a simple way to reorder levels of a given factor.Let's say that the vector testf<-factor(c("red","red","red","blue","blue","white")) levels(testf) : blue red white should have reordered levels such as levels(testf) : red blue white (this is for presentation purposes) I guess

On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23]

Hi All, Is someone implemented Tel uri support in the latest asterisk ? If yes, can you guys share some info on it Regards, Ramananda AS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091224/02ea76f9/attachment.htm