similar to: New CVS and Sipuras

Can someone tell me where I can get just app.c from. Mine somehow got corrupted, and no updates or anything else will fix it. I just need the one file from the latest cvs. 8-1-04. Please help

Has anyone been able to change the way that asterisk performs transfers? Instead of using the # key, I would like to due something else, such as flash. # is just causing too many problems with transfers and menus when calling out.

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy

This question isn't entirely Asterisk related, but I'm hoping that someone here may have the knowledge to respond to me anyway. I'm using Asterisk with several Sipura SPA-2000 SIP devices as FXS adapters. I would like to have my SPA's automatically provisioned through http or tftp, but I can't find any information on how to do so. Sipura's tech-support has not been very

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

I am now at a total loss. Using Sipura spa-2000s connected to *, I get DTMF working just fine for internal extensions, voicemail, etc. If making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I get no dial tone. I am working unsuccessfully with Cisco right now on this, but they cant find anything wrong. I have tried all suggestions I can find from the list and elsewhere.

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's). We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys

Any help with this will be greatly appreciated. When re-compiling * to include voicemail access from a MySQL database, I recieve the follwing error. Anybody know how I can fix this? Am I missing packages somewhere? app_voicemail.c:44:25: mysql/mysql.h: No such file or directory In file included from app_voicemail.c:247: mysql-vm-routines.h:7: parse error before '*' token

Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just

Hi everyone, I'm having an odd problem with one of my sipura boxes. The box registers the first time with asterisk properly after being plugged in. After which, some of the subsequent registration tries fail and the box becomes unregistered. However, after a few hours, it finally successfully re-registers and the cycle continues. I have not been able to figure out the problem but

Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the

To All, I recently purchased a Sipura 3000 for home use. Generally it seems to work fine, but it also majorly screws up on a frequent, almost daily, basis. Terrible echo problems that come and go, and sometimes it just seems to "lose its way" and can't re-register. For example, I'll be talking to an incoming caller and an echo starts quietly in the background. Within a minute

Hi list, Well I really didn't want to see things get to this point, but Sherman at Sipura along with their President Jan F. leave me no other choice. SIPURA has been provided a letter from our attorney for Breach of Contract and damages. They have yet to respond. A quick background. 1. Sherman (SIPURA's Director of Marketing), stated that we would do a join press release for the Oct

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

I'm experiencing a very unusual problem with a few of my sipuras. They keep over heating after minimal usage. I unplug them and let them sit for 15 minutes and it starts working fine. The other way around is that I put a fan on it or something to keep it cooling. Has anyone else experienced this or are they just defective units? Regards, Mohammed Salim EZZI Telecom, Inc.

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main number and then dialing an extension for the modem they want to talk to. What are my options for supporting these modems with Asterisk? Here are