similar to: grandstream ringtones - makering.pl usage for 1.0.50

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

Anyone used this ? I am having a bit of trouble got the right perms on makering.pl . Should that file be somewhere in particular ? use the reccommended command sox inputfile -r 8000 -c 1 -t ul - rate | makering.pl ring1.bin but i get bash: makering.pl: command not found Can ya help ????? Best Regards Simon Garvey

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

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

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 have just successfully got the TFTP file remapping to work such that I can have unique ringtone files for each and every extension. I added the following to my server_args line in the xinetd configuration for TFTP: -m /home/asterisk/grandstream/ringmap.cfg Now the entire line reads: server_args = -v -s /home/asterisk/grandstream -u asterisk -m /home/asterisk/grandstream/ringmap.cfg (There

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

Hello List. I am setting up asterisk as a central dialplan, voicemail and conference solution, connected to 12 Cisco 1760 Routers running Call Manager Express IOS distributed around the world. This is all done over VPN. These routers all have PSTN access in their respective country. So far all is good, and Asterisks interopability with the Cisco CME using SIP is very good, although

http://www.kerneltraffic.org/kernel-traffic/kt20031201_243.html --- Relevant section --- Nuno Silva mentioned: The good people at Cambridge made a (very nice) VMM that exploits ring0/1/3 to let one machine run various kernels independently (the kernels need to be ported to the xen arch). Xen itself executes in ring0 and the "guest" operating systems execute in ring1.

Hi, I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. I've tried making the sounds, but for some bizarre reason they're not working. I used to make quite a few ringtones for the 79xx's, but I seem to have forgotten how to do it! And to top things off, I can't even find the documentation on Cisco's site

The about-to-be-released Grandstream firmware now supports multiple ringtones, but (so far) I haven't been able to unearth any documentation as to how one uses them. Anyone out there know anything about this? I've googled, read the firmware READMEs and combed the GS site without any luck. Thx. B.

Hello, I want to know if there is any way to avoid source() stopping when there is an error. Here is the content of my Main.R script: source("~/R/source/Constructor1.R") # Object1 should be constructed ifelse(exists("Object1"), # It's an S4 object print("Object1 exists"), # I can't avoid using 'validity'

Hello, We have a DID partner sending traffic to Asterisk via SIP, but we are not hearing ringtones. When we call the same extension via SIP, we can hear that's it"s ringing (virtually).. Is is something related with call-progress not recognized by DID provider ? Thanks, ________________________ a b d o u l aba at gcomnetworks.com SIP: (131) 229-1002 at sip.freeipcall.com

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel).

Hi all, I've been modifying the ringtone on a group of Snom phones like this, depending on certain dial-plan conditions: Exten => s,1,SIPAddHeader("Alert-Info: <http://192.168.0.200/tel_ring01.wav>") exten => s,n,Dial(SIP/mjc_office&SIP/mjc_home&SIP/mjc_lab&SIP/mjc_server,20,trj) Now, I'm migrating slowly to Digium D70 phones, which have a different

Hi, It seems the patter matching on CallerID rule is an exact matching with this firmware. ie: if you configured "30" for 2nd ringtone then callerID "30" will match and callerid "301" will NOT match. This doesn't correspond to the wiki description ( http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone ). Can anybody confirm? Is there a workaround

Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the

Hi, I'm wondering how I could notify to a dumb analog phone that a voicemail message is waiting. My goal would be to change the tone that is heard just before user starts to dial. Any idea on that ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090706/6a2c88d5/attachment.htm

Hello Everyone. I usually find my own solutions for problems but this time, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the