similar to: Is this a $50 wifi or wireless USB VOIP phone ?

In that example you could make outgoing calls only correct? (since incoming likely needs port forwards) I guess the questions becomes "how often are you going to do that to justify the extra $100 or so you going to pay for a wifi sip phone?" Paul Fielding (paul.fielding@shaw.ca) wrote: > > I think some people are missing the point. You can't throw your cordless >

Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because

I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want

Had a good question for the list, it seems whenever I work in an Asterisk console or on the machine normally I get jitters on any audio going through it. Especially if you did file copies or a 'ps ax' for example. I was wondering if there was a proper way to 'nice' the asterisk proc's? Cisco does this for example to it's "EXEC" and icmp processes, I tried

I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

I'm using a Grandstream IP phone to call someone through our asterisk pbx. The PBX is running "Asterisk 1.0.3-BRIstuffed-0.2.0-RC3" and uses 2x ZAP-HFC cards. When I call someone, if the call isn't answered and then I hang up, I get "487" coming up on the grandstream phone. If I pick up the receiver again and then hang up, the PBX starts calling me back and when I

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

This senario works great! I have a Snom phone coupled with a Linksys Wireless game adapter. When on a event trip I was able to have the snom ring where ever there was a signal. So the proof of concept passed. Unfortunatly all the wifi phones I have read about seem to suck! -----Original Message----- From: asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com>

Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've come to the same conclusion as the wiki: it's probably better to avoid this

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

My problem is: No matter what machine I install and configure Asterisk on it I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work. CAN PLEASE ANYONE HELP. -------------- next part -------------- A non-text

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

Hi, all I am trying to use the label_wrap_gen function in this website. https://github.com/hadley/ggplot2/wiki/labeller I tried to make a long name like this Light and heavy good vehicles (diesel) -\nGVX f2 = facet_grid(vehicle ~ ., labeller=label_wrap_gen(width=15)) eventually, I got something like this in my label... *Light and heavy good vehicles (diesel) - GVX* I suppose the

> Hi! > >> We are in the process of an Asterisk call center deployment using IAX2 >> G711 ulaw softphones. Outbound sound quality is terrible. RESPONSE: This is Bill McCready from PCPhoneline.com . To address your sound quality issues, you may want to check out our VTA1000 Skype+SIP+IAX2 Tri-Mode Gateway which plugs into the USB port of a Windows computer. You can use it

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

Hi, I have recently upgraded to dovecot v2.3 on Ubuntu 16.04.3 from repo.dovecot.org and I switched to IMAPSieve for rspamd with the guide from the wiki. When a mail is moved from Junk to another Inbox or vice-versa imap segfaults. I have posted debug logs and versions below, any help will be greatly appreciated. Thanks, Rhys Williams lux at bifrost:~$ dovecot --version 2.3.0 (c8b89eb) lux at

I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and

pigeonhole 0.5.0 has been already released... ---Aki TuomiDovecot oy -------- Original message --------From: Daniel Kenzelmann <dovecot.org at k8n.de> Date: 03/01/2018 21:31 (GMT+02:00) To: Rhys Williams <lux+mailinglists at melted.me>, dovecot at dovecot.org Subject: Re: dovecot v2.3.0: imap segfault when sieve_extprograms_plugin.so ? called Hi, see the following thread:

Does anyone have a way of producing solid shapes other than spheres in rgl? I am using rgl to produce a simple visualisation of a forest model results using "lollipops". Its just a bit of fun, but as many of the trees are pines I would like to depict their crowns as cones. If there is a solution I need it to work under windows. Here is the example. library(rgl) library(misc3d)