similar to: Anybody success using Asterisk 1.2.6 and Spa nDSP 0.0.3 pre 6?

HI all, My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Best regards, Marco Mouta

hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. 3x welemon

Hi all, I an using MeetMe and I would like to use the -i function to annouce the join/leave of the user. However, this require that users record their names. Is there anyway to remove this? I just want MeetMe to annouce somethig like "A new user has joined the conference" and that need not to record user's name. Is there a way to do this?? Pim

Hi All, I'm trying to add this function to my find-me application: when all available numbers are dialed in parallel , if any number is busy, take it at busy and go to voice mail. I read the Dial() Application but there's nothing written about this. My question is, is it possible to do this with Asterisk? Thank you, Pim

Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim

Hi, I was using Asterisk with Hylafax via IAX Modem. It works fine until I upgraded to Asterisk 1.2.7.1 I didn't change any configuration but it seems that Asterisk does not get the call from IAXModem anymore. I'm doing something like this Asterisk <--> IAXModem <--> Hylafax Usually when I use sendfax -n -d 260XXX somefile I'll see Asterisk receiving the call in

Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten =>

On the /proc/zaptel it was lost interrupts, you haven't got any so that's good! On opencall.org there's a known issues link and that mentions some fax machines that have issues, might be worth a quick check there. I can receive from our fax machine this end, if you'd like I'll send you a test fax if you send me your fax number, if you can receive from us and tpc then it's

Hi all, I want to be able to see the status of my Agents on a web interface. I have no idea how to do so. I have found a few sample script to communicate with queues manager to view queues.But I couldn't find any on viewing the agent status. Could anybody give me a clue? Regards, Pim

Hi, I am able to use SpanDSP 0.0.2 (all pre version) with Asterisk 1.2.6 to recieve fax successfully. Today I tried to change to SpanDSP 0.0.3 pre 6 but I just couldn't complie the app_rxfax and txfax application. The SpanDSP 0.0.3 was successfully complied though. I'm running it on Debian. I've searched the space for the answer, there're so many people asked this but I just

Sorry to bore you more with the clock issue, but have you check /proc/zaptel/<span> to make sure it's not missing interrupts? There's also an option to record the audio for the fax, you could listen to that vs a recorded file that will receive correctly on a fax machine and see whether there is an obvious difference? (Good luck, that'll be really scraping the barrel!!) Does it

I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk and

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

Hi Samba world, Have been struggling with this for the past 10 days, we are running Samba 3.0.22 on VCS zone, we have end users saving files onto Samba mapped drive, and complained that it 5mins to save 300 files, now it takes 30 mins. There is recently a change in the topology. Before Enduser ----------- samba mapped -------------- server (local attached storage)

Type "swift" at the command line so you can see the -options. Then modify the line to use the correct switch and specify the name of the voice you want to use. Thanks, Steve -----Original Message----- From: Pimjai Wesnarat [mailto:pw@nummerndirekt.de] Sent: Fri 4/21/2006 6:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] How

hi, it'd be very useful to include the quota warning patch in official release. without it the quota support is not really useful since mail simple dropped when quota is over. and most enduser never know what happend, they just recognize mails are not coming:-( thanks. -- Levente "Si vis pacem para bellum!"

Hi, I just noticed that the path to some config files in the manpages get's replaced during the build of the debian package with the pathes which are used during the build. This is not good, since the path will not be the same for the enduser, who installs the deb. Is it intented that the pathes in the manpage get generated during compilation and should I know generate a script for debian to

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Jean-Marc Valin wrote: > That's quite strange. The only thing I can say is that the bug is most > likely *not* around line 414. It's probably some sort of memory > corruption somewhere else (quite possibly outside of Speex). Do you have > any more information? What CPU? What's the value of best_ntarget[j]? Is > SSE enabled? What's the allocation method

Hi to everybody, I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port). This is my problem: the incoming call doesn't arrive to asterisk. In the spa web page i configured this dialplane: (<:line01@192.168.1.220:5060>) where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip and 5060 is the asterisk sip port.