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Hi All: I have a Cisco3620 with a proper T1/PRI card installed with asterisk running on the same LAN. Since I have lit up the line, I can dial out and make calls to regular lands lines. However when a call comes back in it rings the destination phone once and disconnects. Here is an error from my router 15:40:45: ISDN Se1/0:23 SERROR: L3_GetUser_NLCB: EVENT 0X45 No NLCB 2 15:40:45: ISDN

Hi, We are experiencing a very slow network when loading files from the Samba server (Samba V1.9.18p4 on Digital Unix V4.0B). A file transfer of a 5 MB file from the samba server to a NT Workstation with FTP takes about 10 seconds. The transfer of the same file with using the NT-Explorer takes also about 10 seconds. The load of the same file (it is a tiff file) with a application called

The Asterisk Development Team has announced the release of Asterisk 11.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.16.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.16.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

I have a small pile of them from customers who were too lazy to send them back after switching to our local voice service... Is there any hope of ever using these things with Asterisk? Vonage does not want them back and they won't unlock them either. A terrible shame! Should I just toss them? Steve

Hello All, I was recently fighting with an optimum online connection in NY. I finally got in touch with someone that confirmed they are throttling my upload connection. I just wanted to make everyone aware of it, so if you have problems if your ping times jump erratically, this could be the cause. Their suggestions were, although you can upload a lot, do not do it constantly. They do not want

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

A small example before I begin my query: > educ <- read.table(efile, header=TRUE) > educ Education Age_Group Count 1 IncompleteHS 25-34 5416 2 IncompleteHS 35-44 5030 3 IncompleteHS 45-54 5777 4 IncompleteHS 55-64 7606 5 IncompleteHS >64 13746 6 CompletedHS 25-34 16431 7 CompletedHS 35-44 1855 8 CompletedHS 45-54 9435 9

> -----Original Message----- > From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Matt > Garman > Sent: Thursday, February 2, 2017 8:52 PM > To: CentOS mailing list > Subject: Re: [CentOS] Spotty internet connection > > On Thu, Feb 2, 2017 at 7:13 PM, TE Dukes <tdukes at palmettoshopper.com> > wrote: > > Lately I have been getting slow and

Hi, I wrote a mail a few days ago concerning my setup with a front/back firewall, shorewall being front and ISA server 2004 acting as back firewall. I said that ISA server is logging some "intrusion attempts" namely requests coming from external interface to the internal network. As this shouldn''t happen (all intrusion attempts should be stopped by shorewall) I begun to

Hi All, We are using Asterisk 1.2.18 in a CentOS box. Implemented a queue (maqueue) structure for handling customer calls. There are 4 queue members (85744,85766,85511,84888). These 4 members are logged in using AgentCallbackLogin application. But at some point, one of the agent's SIP phone does not ring for an incoming call to this queue. I checked the agent status and it is not in paused

Hi, I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to register with TLS to an Asterisk 11.10.0 box. The registration fails and I see this in the Asterisk console: == Problem setting up ssl connection: error:00000000: lib(0):func(0):reason(0) [Jun 8 15:33:39] WARNING[8555]: tcptls.c:274 handle_tcptls_connection: FILE * open failed! Anyone know what that error means? The

I have a collection of .csv files in a directory, and want to read them into R data.frames whose names are the same as the file names, without the .csv extension e.g., from > (files <- list.files(pattern="*.csv")) [1] "Allstar.csv" "AllstarFull.csv" [3] "Appearances.csv" "AwardsManagers.csv" [5]

Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU

I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and