search for: tel

Displaying 20 results from an estimated 33461 matches for "tel".

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2011 Jun 26
2
Ordering a matrix based on cluster no
...o (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 and so on. The order of genes is taken from the sorted matrix(100x2). Can someone tell me how to do this in R. I tried the grep() but I get a message saying that the length of pattern >1 so only first element will be compared. But i want to check for each gene in the 100x100 matrix for its cluster number and then group it. I also tried the order() but it did not help either. T...
2010 Oct 06
3
integrate Intertel Axxess with Asterisk
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
2017 Nov 26
0
withTimeout does not timeout nlme anymore
...7355399,2.99573227355399,2.99573227355399,2.99573227355399,2.99573227355399,2.99573227355399,2.99573227355399,2.99573227355399,2.99573227355399,2.30258509299405,2.30258509299405,2.30258509299405,2.30258509299405,2.30258509299405,2.30258509299405,2.30258509299405,2.30258509299405,2.30258509299405<tel:(850)%20929-9405>,2.30258509299405,2.30258509299405,2.30258509299405,1.6094379124341,1.6094379124<tel:(609)%20437-9124>341,1.6094379124341,1.6094379124341,1.6094379124341,1.6094379124341,1.6094379124341,1.6094379124341,1.6094379124341,1.6094379124341,1.6094379124341,1.6094379124341,0.91629...
2004 Jul 01
5
Inter-Tel Eclipse2 (IP PhonePlus)
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
2018 Mar 19
4
rsync time machine backup permissions
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
2019 Jan 31
2
tel URI
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 Linux en Pol...
1999 Jun 20
2
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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
2014 Aug 28
1
RDNIS with tel: vs. sip: header
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, &quot...
2008 Nov 07
1
Help with asterisk and avaya SIP trunking
...erisk 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]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564)? [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' rejected because extension not found." A SIP Debug of the packet when this happens on asterisk CLI is "<--- SIP read from 10.10.8.2:5060 ---...
2013 Mar 21
9
Asterisk disconnecting SIP Calls after 15 Minutes
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 session-timers to refuse but th...
2008 May 05
2
Finding non disjoint regular expressions
...ssions? 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 Schwartz Department of Human Molecular Genetics and Biochemistry, Tel Aviv University Medical School, Tel Aviv 69978, Israel. Tel: +972-3-640 6894 email: schragas@post.tau.ac.il [[alternative HTML version deleted]]
2005 Mar 03
4
MGCP to Inter Tel system
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
2013 May 21
1
Calculating AIC for the whole model in VAR
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
2010 Sep 21
2
plot xyz data in 2D
...e example) X Depth Resistivity -3 0 1 -2 -0.25 2 -2 0 1 -1 -0.5 3 -1 -0.25 2 -1 0 1 0 -0.75 2 0 -0.5 3 0 -0.25 2 0 0 1 1 -0.5 3 1 -0.25 2 1 0 1 2 -0.25 2 2 0 2 3 0 2 Acacia Water Jan van Beaumontstraat 1 2805 RN Gouda The Netherlands Tel. office: +31(0) 182686424 Tel. mobile: +31(0) 6 12143599 Fax: +31(0)182686239 Email: jacob.oosterwijk@acaciawater.com <mailto:jacob@acaciawater.com> Website: www.acaciawater.com <http://www.acaciawater.com/> Acacia Water Jan van Beaumontstraat 1 2805 RN Gouda The Nether...
2006 Aug 10
4
exporting vcards from rails
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
2020 Jun 22
2
Voice broken during calls (again...)
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
2004 Dec 22
4
ordering levels
...els(testf) : blue red white should have reordered levels such as levels(testf) : red blue white (this is for presentation purposes) I guess I'm looking for a generalized version of "relevel"... Thanks Anne ---------------------------------------------------- Anne Piotet Tel: +41 79 359 83 32 (mobile) Email: anne.piotet@m-td.com --------------------------------------------------- M-TD Modelling and Technology Development PSE-C CH-1015 Lausanne Switzerland Tel: +41 21 693 83 98 Fax: +41 21 646 41 33 -------------------------------------------------- [[alternative HTM...
2005 Aug 20
0
Re: Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
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 1 > T...
2008 Nov 07
0
asterisk - avaya ip office SIP trunking
...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]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564 <tel:+32564>)? [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' rejected because extension not found." A SIP Debug of the packet when this happens on asterisk CLI is "<--- SIP read...
2009 Dec 24
1
Tel uri Support
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