similar to: Cisco 7905 vs Cisco 7905-G

This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. An even if we were to

Hey list (again - annoying bastard I am) I've played with Firefly/* for a while and I have yet to find a way to have * send voicemail notification to Firefly. It appears possible using SIP (no clue whether Firefly supports it) in the sip.conf file, but there's no mention of anything voicemail-related in the IAX.conf file. I'm using IAX with Firefly, so that might just be the

Hey list! Some people are auctioning Cisco CP-7905 phones on Ebay, and I was wondering if those supported SIP. I think I saw something on voip-info.org about this, but I would like someone who has the same phones to confirm this. Anyone can? Good quality? OR If anyone knows of a place that sells those phones for a good price, I'd be interested. Buying a 250$US Cisco 7960 from Canada is

Can someone point me in a direction where I can acquire the Cisco 7960 SIP firmware? I just purchased two phones of ebay, but they are both in Call Manager mode. Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030515/6265b777/attachment.htm

Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this

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

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

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

I'm deploying a fairly large number of Cisco 7905 with SIP connected to an asterisk PBX. The problem: the 7905 has this nice "feature": You can set a "voice mail number" in configuration, so that you can listen your voice mail just by hitting the "messages" key on the phone. It just autodials "8500" or whatever else. The customer wants this, so

How to configure cisco 7905 with asterisk ,if you please can send me step by step configuration steps . Thanks Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 ********************************************* No employee or agent is authorized to conclude

My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes

I've got asterisk working great with cisco 7905 sip phones. I've just got one issue that I can't figure out. I have a 5300 connected to the PSTN via PRI. When I send a call from the 5300 to asterisk it will ring the 7905 phone for 4 seconds then drop. This is because the only message asterisk sends to the cisco is the 100 trying message. The 5300 receives that and sends a call

Hi Everyone.... Got a newbie type question regarding MOH & Cisco phones. I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems.

Hi, I'm using a Cisco 7905 (with SIP image CP7905010300SIP050414A) on Asterisk 1.07. Whenever the user gets voicemail, the MWI goes on, and when the user listens to the voicemail, the MWI goes off. However, the phone is configured (as in the default config by Cisco) to re-read its configuration from tftp every hour. This 'warm reboot' also turns off the MWI, and it won't

How to disable silence suppression (or Voice activity detection - VAD) on Cisco 7905 phone? On Cisco 7940 I use "enable_vad: 0", but I can't find anything similar for 7905. -- Tomislav Parcina tparcina#lama.hr

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

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 feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for "7960 sip firmware" and find that fast). Thanks for the help, Jeb Campbell jebc@c4solutions.net