similar to: one way voice

Hi, Has anyone else encounter the following; While testing the new CentOS 6.5 kernel-2.6.32-431.1.2.el6.x86_64 on a Dell PowerEdge 2950 in the lab it kernel panics on boot up, complaining about the scsi_wait_scan Module signed with unknown public key. The Dell Poweredge 2950 is running the latest 2.7.0 BIOS and BMC Firmware v2.50 with PERC5/i integrated (running latest firmware 5.2.2-0072,

http://grandstream.com/BETATEST/HT488_496_386/

Hi all, I have Centos 7 which is doing great. I tried to install Cisco Anyconnect VPN client but could not launch it. It appear on my Application tab but not launching. Kindly assist me the a working rpm on Centos 7. Regards.

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user

Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r

CentOS Errata and Bugfix Advisory 2014:0356 Upstream details at : https://rhn.redhat.com/errata/RHBA-2014-0356.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: 7fb765a31da9410aedb81de9973ea1e8806237908c4cf3bd0855c98c2bf63179 rrdtool-1.3.8-7.el6.i686.rpm fc0368f9cd4f1c287c6ea6b3ebba86b38338c1c0aa332f7684a5334afc93e091

CentOS Errata and Bugfix Advisory 2014:0356 Upstream details at : https://rhn.redhat.com/errata/RHBA-2014-0356.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: 7fb765a31da9410aedb81de9973ea1e8806237908c4cf3bd0855c98c2bf63179 rrdtool-1.3.8-7.el6.i686.rpm fc0368f9cd4f1c287c6ea6b3ebba86b38338c1c0aa332f7684a5334afc93e091

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

I am getting a TON of these errors on the console. I Googled and wikied and greped found the error in the source but cannot understand why it is happening. The system works fine, no dropped calls, no echo, it will even run for weeks with this error. But it just scrolls and scrolls on the console. Temporary fix was to turn off the console monitor! :-) Any ideas. Apr 16 10:40:12

I see that Asterisk@home and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the inbound numbers.. Soon I will be adding one or more IAXy devices.. Would either Asterisk@home's or

Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the

Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just

Having trouble running make on res_perl: [root@charlie res_perl]# make perl -MExtUtils::Embed -e xsinit gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \"-DASTVARLIBDIR=\"/var/lib/asterisk\"

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

Hi, I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, Do you know if it's possible to push the status of an extension (a phone) to a phone like blinking a light on the phone ? And do you know wich brand of phone can do this ? I'd like to make the same as the secretary phones that can see the status of lines before putting a call on it or transfering someone to. As i know that the Flash Operator Panel get the global

I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the