similar to: Cisco 7960 SIP Firmware

Hey, Can anyone comment on the difference between the 7905 and it's upgrade, the 7905-G ? Has anyone used these phones in a configuration? Is SIP well implemented in the 7905 ? Thanks in advance, Mat

Hi all, I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. Any comments on your experiences would be very much appriciated. Best regards Matt

Has anyone tried the new ariavoice $89 VOIP desk phone with Asterisk? ` http://www.voip-info.org/wiki-AriaVoice -- Jim James H. Thompson jht@lj.net

Maybe not the place for this but thought I'd post the info for others. I purchased a cisco 7960 off ebay and needed to convert to SIP for *. I know * supports SCCP but I wont go into that here. I'd read on voip-info.org that a contract could be purchased for approx $8 to allow me to download the firmware. I though, being in the UK, i'd get one through a reseller in the UK.

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

Does anyone know if version 3.1 is Call Manager or SIP? Thanks.

Is the 7905G Firmware the same as the 7960 firmware? -- MBM

I have some Cisco 7960's and want to use them with SCCP - I have gotten it working with a few different firmware versions but all seem a little flakey. I know that SCCP is not as solid as SIP but am wondering, which firmware version is advised for use with chan_sccp from http://chan-sccp.sourceforge.net/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk Monitor seems to be working fine. Though the problem I am having is the two files (in & out) muxing. I added ,m to the string, yet the call records two files still, and I get the resulting error, at the bottom. monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4 8:23-in.gsm

I am using the Monitor() application (with soxmix for combining the audios) and the voice connected to the phone network is recorded at a lower volume then the voice connected directory to the Zap analog phone card. How can I get both the audios to be at the same volume on recording? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been trying for sometime to upgrade this phone to either sip, or the latest sccp (at this point I don't care) with no success. It has an sccp image (don't know the version off the top of my head) that is incompatible with chan_sccp. When I boot up the phone, it asks the tfpt server for the .tlv file, then the SEP<mac> file which it receives successfully. It then asks for

Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this: exten => 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)

Hi, is there anybody who knows how to force Asterisk 1.4 to use soxmix instead of sox? Thank you. Giorgio

Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix

On Thu, 29 Jun 2017 at 22:51, Victor Nomura <victor at mezine.com> wrote: > Thanks for the reply. What would be the best course of action? The data > on the volume isn?t important right now but I?m worried when our setup goes > to production we don?t have the same situation and really need to recover > our Gluster setup. > > > > I?m assuming that to redo is to

By any chance are you having any redundant peer entries in /var/lib/glusterd/peers directory? Can you please share the content of this folder from all the nodes? On Tue, Jul 4, 2017 at 11:55 PM, Victor Nomura <victor at mezine.com> wrote: > Specifically, I must stop glusterfs-server service on the other nodes in > order to perform any gluster commands on any node. > > > >

Hello All, I am using monitor with soxmix, however the quality seems somewhat low after sox converts to mp3. Does anyone know a way to get a higher quality file? Some of my lines are coming in on isdn. Regards, Greg

Thanks for the reply. What would be the best course of action? The data on the volume isn?t important right now but I?m worried when our setup goes to production we don?t have the same situation and really need to recover our Gluster setup. I?m assuming that to redo is to delete everything in the /var/lib/glusterd directory on each of the nodes and recreate the volume again. Essentially

I've just recieved a copy of the new SIP firmware for the Cisco 7970, those of you with Cisco accounts may wish to try it (shock horror I'm sticking with SCCP). This coincides with the release of v8 firmware for all Cisco phones (and for those of you running Sergio's chan_sccp v8 works fine) The firmware is now also (and for the 7970 SIP, only) distributed in ".cop" files,

Does soxmix works with asterisk ver. 0.9? I have ver. sox-12.17.5 on Gentoo but the option "m" does not combine two WAV files (In and Out) into one file. I have two separate files in /monitor folder. exten => 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten => 711,2,Monitor(wav,${CALLFILENAME},m) exten => 711,3,Dial(${sales_support},20,r) exten =>