similar to: No rtp activity

Hello List; I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the following debug, can someone advise me about the solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE sip:22021782 at Asterisk_IP_Address:5060 SIP/2.0?Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1?From: "1828444" <sip:1828444 at

Hello, I just received what seems to be a nice SIP<->DECT gateway but can't make it work with asterisk. The manual is very unclear (written in "chinese" english) and the web configurator is ambiguous as well. Has anyone succeeded in making one of these babies work with * ? info: http://www.planet.com.tw/product/product_dm.php?product_id=367&menu_id=3

Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -> Dial Plans -> Dial Plan 8 (<S0:66610>) Should I put extension [66610] in sip.conf with a context in extensions.conf that

--- august <august@alien.mur.at> wrote: > > > sorry to bother again, > > but I can narrow down my problem to certain parst of my flac file. > > for example, when I seek to 19*44100 samples or to 81*44100 samples, > the > seek callback runs off in an endless loop. > > there aren't any while loops in my code, so I don't think I am > causing it.

Me and Dennis Kasprzyk changed CompRect to be more intuitive and easily replace XRectangle use. This patch changes CompRect and whole Core. Thanks for attention. -- Eduardo Gurgel Pinho (GELSoL-UFC) (Gentoo) Linux User #415930 http://edgurgel.wordpress.com http://alu.dc.ufc.br/~eduardo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've just installed DAHDI at two PBXs as follows: *PBX-1 PBX-2* FXO ------------- FXS When I try to send calls from PBX-1 to PBX-2 I just receive the message: "Starting simple switch on 'DAHDI/1-1" It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard at

Hey guys, I just joined the list. I've been programming an external for PureData <http://pure-data.sourceforge.net/> & <http://pure-data.org>, as well as for Max/MSP....that reads multiple sound formats. But, I've been having massive problems with getting reliable seeking with FLAC. sometimes when I call FLAC__seekable_stream_decoder_seek_absolute, I get a false

Hello, I encounter a problem using the UPS MGE Evolution 850 VA USB with Debian Etch and the nut package available for this version. I got this error: ==syslog== Can't connect to UPS [MGE850VA] (usbhid-ups-auto): No such file or directory upsd[2767]: Startup successful upsmon[2769]: Startup successful upsd[2767]: Connection from 127.0.0.1 upsd[2767]: Client MGE at 127.0.0.1 logged

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, maybe my problem was discussed before, but I'm just too busy to browse the whole mailinglist at the moment. I'm running Samba 3.0.14a(sernet) on a Debian Sarge with LDAP-Backend. After some days, the users aren't able to load their profiles from the server. In fact, no datas can be transfered from/to the sambaserver. The cpu

On Feb 26, 2009, at 2:02 PM, John Mosby wrote: > Hello LLVMdev, > > I have been working with LLVM for just over a year now, mainly in > the area of compilation for HDLs like SystemVerilog and SystemC. > Most of this work dealt with translation to LLVM IR, representing > concurrent languages with LLVM and using LLVM analyses and transforms > for compiling onto proprietary

Hello LLVMdev, I have been working with LLVM for just over a year now, mainly in the area of compilation for HDLs like SystemVerilog and SystemC. Most of this work dealt with translation to LLVM IR, representing concurrent languages with LLVM and using LLVM analyses and transforms for compiling onto proprietary simulation acceleration hardware. All of this work used the C back end exclusively,

Dear All, I have created an inbound context in sip.conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes under General context...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

Hi, I''m new to rails and finding it hard to figure out what this error is referring to. Can anyone help please?!! NoMethodError in Registeredphone#list Showing app/views/registeredphone/list.rhtml where line #24 raised: You have a nil object when you didn''t expect it! The error occured while evaluating nil.txtForename Extracted source (around line #24): 21:

First, thanks very much for your comments! On Sat, Feb 28, 2009 at 8:05 PM, Evan Cheng <evan.cheng at apple.com> wrote: > > On Feb 26, 2009, at 2:02 PM, John Mosby wrote: > > It is limited to X86 presently since that is the only target I have > > access to at the moment. > > What part of this is target dependent? Is this due to emitPrologue / > emitEpilogue being