similar to: Grandstream Ringtones on a per phone basis

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

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

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, 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).

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

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

I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 & 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories & buddy lists. Unfortunately some of the

We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone.

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

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

The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort

I have an Intel e1000e NIC which I put into passthrough for an HVM domain under Xen 4.2. All the corresponding hardware protections are enabled on my system (DMA + Interrupt remapping), however, once in a while I get a SERR NMI in dom0 (NMI - PCI sys error (SERR) in xl dmesg). I am wondering about its exact reason. I am thinking in the following way: [+] Under Intel VT-x, interrupts are

I have Asterisk configured to record calls. Both in and out record ok but SoxMix fails to join the two files. The error from the CLI is as follows: Execute of ( nice -n 19 soxmix /var/spool/asterisk/monitor/Support-in.wav /var/spool/asterisk/monitor/Support-out.wav /var/spool/asterisk/monitor/Support.wav && rm -f /var/spool/asterisk/monitor/Support-* ) & failed. If I run exactly the