similar to: VOIP carriers and asterisk

I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak

Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone

Sorry, I am newbie. Sample codes are from https://github.com/oneman/libflac/tree/master/examples/cpp. I used FLAC__stream_decoder_process_single function but it still gives exception. Maybe I could not control read callback, you're right. I will check it and write result in this thread. Thanks for help. 2013/7/1 Martijn van Beurden <mvanb1 at gmail.com> > I'll top-post this

Martijn, I don't use any metadata when encoding and decoding. When I call *FLAC__StreamDecoderStateString[FLAC__stream_decoder_get_state(m_decoder)] * * * it returns FLAC__STREAM_DECODER_SEARCH_FOR_METADATA enum value. Is it an error ? 2013/7/2 Burak Or?un ?zkablan <borcunozkablan at gmail.com> > Hi again, > > I can not solve problem. I want to mention my source code, so

I am trying to obtain adjusted means and standard errors for a three way ANOVA I have three effects, two continuous; fire frequency and annual precipitation, and one categorical; soil type in an unbalanced design. I am testing the effect of annual precipition (AP), soil type (ST), and fire frequency (FF) on stem count (SCt) My data table looks as such: ST FF AP SCt 3 Coy

Hello, I was wondering if there is a way to ignore the error message you get when some of the data means you compare are constant in some lines of your data frame. I'd like to go ahead with t.test and get the calculated p-values anyway in such a case. Thanks -burak

Anybody using Cisco Call Manager and connecting to any SIP termination service like voipjet, voxee, etc? Please msg me offlist. AK

Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Hi, I am developing an audio network system and using boost, OpenAL and FLAC library in C/C++. I can stream raw audio data over network but I want to encode audio before streaming in current PC and decode after streaming in other PC because of bandwidth limit. I run your sample codes, encode.c and decode.c, about file encode / decode. Then, I run streaming encode / decode with two different

Hello, I added this to the configuration: namespace { type = public separator = . prefix = public. location = maildir:/var/mail/public subscriptions = no list = yes } and I can list maildirs inside these folders correctly. however, I can't see any subfolders. The issue seems to be that /var/mail/public dir itself is interpreted as a maildir directory. However, what I'd

Yes, I have tried plain wav data with file and stream functions of decoder and encoder. They work succesfully. You're right about FLAC can't find the first block in stream, because I didn't add any metadata in stream. The documentation of FLAC says that metadata callback function is optional for stream function of decoder, so I didn't need adding metadata. Actually, I am sure

Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from

> -----Original Message----- > From: dovecot-bounces at dovecot.org [mailto:dovecot-bounces at dovecot.org] > On Behalf Of Burak G?RER > Sent: Thursday, May 30, 2013 10:34 AM > To: Nikolaos Milas > Cc: Dovecot Mailing List > Subject: Re: [Dovecot] Dovecot 2.2 build rpm on Centos6 > > On 27-05-2013 16:56, Nikolaos Milas wrote: > > On 27/5/2013 1:07 ??, Birta

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone.... Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com

Hi again, I can not solve problem. I want to mention my source code, so you may answer easily. This is decoder init stream function. *FLAC__stream_decoder_init_stream(m_decoder, decoderReadCallback, NULL, NULL, NULL, NULL, decoderWriteCallback, NULL, decoderErrorCallback, input_pile_array);* * * Then, callbacks * * /// \brief read callback function of decoder FLAC__StreamDecoderReadStatus

i have install 2.1.1 and configured in dovecot log this error is coming every 20 seconds: dovecot: imap-login: Disconnected (no auth attempts in 0 secs): rip=**********, lip=**********, TLS handshaking: SSL_accept() syscall failed: Connection reset by peer and this error every 2 seconds: dovecot: imap-login: Disconnected (no auth attempts in 0 secs): rip=**********, lip=********** what can

All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently last Thurs evening thru Sat morning. On Sat morning, some of our employees were attempting to log-in

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?