similar to: sipuras 841 bad sound

I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem?

Guys. Ive been testing how to receive faxes using TDM400P cards and so far, after playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno luck, faxes come in as garbage or broken or with blank lines. Anybody has successfully done this? Any tips.. Also I have some ideas: 1. Is it really possible to get fxes on a fax machine using ATAs like the sipura 2002? Even using ulaw

Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :)

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

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

Is anyone else having problems with Sipuras not being able to re-register to Asterisk after applying the cvs update last night? Just curious if I need to roll back or take all of my Sipuras out back and beat them.

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to

Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call [Jun 28 09:40:02] VERBOSE[3172]: [Jun 28 09:40:02] -- Unicall/4 released [Jun 28 09:40:02] VERBOSE[3084]: [Jun 28 09:40:02] Asterisk cleanly

Guys. I have a te100p with unicall and an E1 and Im having problem with DTMF tones but the weird thing is, I only have problems sending the tones to certain phone numbers, anybody seen this behavior? Asterisk shows on the console the dtmf tone been pressed but seems the other side is not getting them, and this just happens with certain phone numbers, not all.. Any clues(tips?

Hello! I'm tryuing to bring up an R2 connection but eventhough I've followed the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems to be missing. When an incomming call is generated I get: Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 <- 0001 [1/ 1/Idle /Idle ] Nov 24 06:01:17 WARNING[-197416016]:

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

Hi All, I have installed chan_unicall and MFC/R2 successfully, and is runnign fine. But I noticed that once unicall is installed, asterisk CPU usage as reported by 'top', jumps to 99% every few seconds. I have no incoming calls, and I have even removed the E1 lines from card and I tried almost everything possible but I was not successful in determining the cause of this high cpu

I am so fed up with Digium cards. My company first owned a TE410P, I installed it in a Dell server and "enjoyed" its instability (we bought it months before Digium warned about the incompatibility issues). Then we switched to a TE411P for the hardware echo cancellation. Now we want to receive fax (< 20/day) on it and guess what ? Since April 2006 (again a few months after we bought

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

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

Guys, is Steve's ftp site down? DNS says ftp.soft-switch.org doesn't exist. Anybody else seen this?

Hello, How good is :SPA 841 form SIPURA. Thanks Varun

Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten => s,7,ChanIsAvail(IAX2/anton:intrudercom@armando-gw) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any