search for: ring2

...is files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream GXP2000 1.0.0.3" [Wed May 11 16:47:17 2005] [error] [client 201.133.125.152] File does not exist: /usr/local/apache/htdocs/voip/firmware/ring2.bin 201.133.125.152 - - [11/May/2005:16:47:17 -0500] "GET /firmware/ring2.bin HTTP/1.0" 404 289 "-" "Grandstream GXP2000 1.0.0.3" [Wed May 11 16:47:18 2005] [error] [client 201.133.125.152] File does not exist: /usr/local/apache/htdocs/voip/firmware/ring3.bin 201.133.1...

...answered at least once without changing the CallerID. I am not worried about attended transfers since then the internal ring tone is what should be used and that is what happens now. I just need to deal with blind transfers. What I have now is: 1. Outside call => ring1 2. Internal call => ring2 3. Transferred call => ring1 || ring2 (depending on 1 or 2) What I want is: 1. Outside call => ring1 2. Internal call => ring2 3. Transferred call => ring3 (regardless of 1 or 2) If everything went though ## then that would be simple enough. The trick is that most (all) users emp...

If you wan't to create a ringtone with makering.pl for firmware 1.0.50, be sure to create it as ring.bin and then rename it to ring1.bin / ring2.bin or ring3.bin. This seems to be the only change between the format from 1.0.4.68. Regards, Maron

On Thu, March 5, 2015 05:30, Ruben R?gels wrote: > > > Am 05.03.2015 um 01:09 schrieb James B. Byrne: >> I am trying to determine how the transfer button on the Snom-870 >> works >> with Asterisk. Is the ## special code employed as when it is >> entered >> through the handset or is the blind transfer through the phone >> function accomplished in a

...ot;Use NAT IP" field. Can you help me to start using BT-100 in both directions when I am behind the Firewall? This is the configuration of the phone: Software Version: Program--1.0.4.71 Bootloader--1.0.0.17 HTML--1.0.0.32 VOC--1.0.0.6 Custom Ring Tone: ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) detected NAT type is open Internet IP Address: 10.1.1.110 SIP Server: 194.12.230.167 Outbound Proxy: SIP User ID: xxxx Authenticate ID: Authenticate Password: NAT Traversal: Yes, S...

...acts like it was never called ? Or is it just something I missed ? If it matters for something I have call waiting enabled on the phone. Product Model: BT100 Software Version: Program--1.0.5.11 Bootloader--1.0.0.18 HTML--1.0.0.37 VOC--1.0.0.6 Custom Ring Tone: ring1--1.0.0.0 ring2--1.0.0.0 ring3--0.0.0.0 Regards Kiss Karoly

Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I can't seem to make the "Hold" button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press "Hold" button 3) Hang up phone What I expect is for the call to go on hold until I pick up the receiver again. (Like my SPA-2000's, except it's a flash, but I can hang up the phone and the call waits there for me to pick it up

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

Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for