similar to: OT: ad-hoc polycom network

Displaying 20 results from an estimated 1000 matches similar to: "OT: ad-hoc polycom network"

2006 Jun 20
0
Anyone using VoIP WiFi phones?
The only advantage is when you travel. Last year I took my wifi sip phone to Astricon in Madrid and everything worked as expected. I am just packing it and heading for Paris... Wojtek -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Tuesday, June 20, 2006 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:
2006 Mar 02
0
* dials out zap line first 6 digits, pause, then last digit
Hello, This seems to be a weird one. I'm at work now and will get some more-verbose logs later when I get home if nobody has any ideas about what's happening here. I've got a tdm card with 1 FXO and 1 FXS. Asterisk is in the 1.2.x line, so is zaptel. astlinux to be specific. I can get the versions at home later if it might help. It's running on a silent epia 5000 board
2005 Oct 05
4
dropped calls when g729 is used on sip leg
Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all
2006 Mar 13
2
Simple php script to monitor asterisk calls
Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It
2005 Jul 15
2
seems-to-be-inexpensive source of polycom 301 and501
i have ordered 500s from tritechcoa.com several times over the past 4 months. great service and delivery, and the prices are lowish, only problem is, they add a $20 handling fee per phone, on top of phone price, and shipping, making the lower price not as good -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Friday, July 15, 2005 12:01
2007 Jul 21
0
asterisk-users Digest, Vol 36, Issue 61
Please, unsuscriber, this group. regars Nestor Castillo ----- Mensaje original ---- De: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> Para: asterisk-users at lists.digium.com Enviado: viernes, 20 de julio, 2007 11:00:04 Asunto: asterisk-users Digest, Vol 36, Issue 61 Send asterisk-users mailing list submissions to
2006 Mar 31
0
Re: Asterisk-Users Digest, Vol 20, Issue 226
That was how I reset the black Iaxy I have used; I've never used a blue one. What I found was the initial provisioning would work fine, but if I tried to change the settings after having already provisioned the device, the provisioning program would hang, so I Googled for instructions on resetting the Iaxy to the factory settings. This was the procedure described at
2008 Mar 26
5
Asterisk parking hold and transferdigittimeo ut
> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >
2005 Sep 08
5
data manipulation
Dear All, I would be grateful if you can help me. My problem is the following: I have a data set like: ID time X1 X2 1 1 x111 x211 1 2 x112 x212 2 1 x121 x221 2 2 x122 x222 2 3 x123 x223 where X1 and X2 are 2 covariates and "time" is the time of observation and ID indicates the
2006 Mar 27
0
Question about Polycom 601 and expansion module.
Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"
2012 May 16
1
Merging multiple data sets
Hello R user, I have four data sets in dir "D:/Bharat Warule/Rdata_file" which are output_data_prod_1.rda, output_data_prod_2.rda, output_data_prod_3.rda, output_data_prod_4.rda. Each data set is huge size like number of rows 343297 and columns are near to 50. For example: x1 <- data.frame(x11=c(1,2,3,4,5),x112=c(10,10,10,10,10)) x2 <-
2007 Jul 08
3
Sparc
By chance, has there been any progress on a 5.0 version for an Ultra Sparc? -- Best regards, Chris Registerd Linux user number 448639
2006 Nov 22
1
G729 issues on 1.4 beta 3
Hello Everyone, I just upgraded to the latest beta version and I am running into one problem. We purchased g729a licenses from digium and they aren't loading anymore. If I roll back asterisk to 1.2.10 the codecs work fine. I've downloaded the new 1.4 version of the codec from their website and re-registerd everything with no luck. Here is the error message: error loading module
2007 Oct 20
1
Asterisk and Cisco
Hi I have asterisk ip-pbx on my network, with some grandstream ip phone and i have cisco gateway that is connetced to VOIP service providers . Cisco is 3700 series and is using H323 . i have compiled H323 on asterisk . now i want to make a call from ip phone that is registerd to asterisk , and route call to VOIP provider so this call should goes to teh cisco from asterisk and then
2009 Feb 07
1
put the hostname of asterisk in the callerid uri to avoid NAT problems
hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The
2009 Dec 09
1
SkypeForAsterisk
Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks in
2012 Feb 01
2
Double Copies Double Copies
Hey Y'all, why am I getting double copies of every email on this list today when it wasn't happening yesterday? Isn't happening on any of my other email. -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004
2012 Apr 22
1
Missing dependency
Hey all, Anyone know where I can find source packages for X86-64 CentOS 6.2: dev86 is needed by my project iasl is needed by my project spice-server-devel >= 0.8.2-4.el6 is needed by my project I can't seem to find them in the source repositories. Thanks. -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 www.counter.li.org ****
2012 Jun 17
3
Rhythmbox Replacement
Hey Y'all, What application replaces the functionality of Rhythmbox for Cent OS 6.2? In particular I am interested in handling podcast feeds. -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 www.counter.li.org ****
2012 Oct 29
4
RPM file download
I see a package "x246-0.120-5.20120303.el6 (i686)" available in my Add/Remove Software tool. I can't find said package in any of the repos I've searched through. Can anyone point me to a link where I can download that package to my local repo? -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 https://linuxcounter.net/ ****