similar to: Talking SIP to Vocal

Anyone have a working sip.conf for Intervivo? (with bidirectional audio, dtmf and authentication!) Thanks David -- |> /+\ \| | |>

I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my "plan" with them. Now all my calls get intercepted immediately, "We're sorry, but your account is temporarily unavailable." Incoming calls work just fine. I contacted their so-called

Before I submit a bug report, perhaps the following has been seen and dealt with before: I found that it is possible for dir-merge filter rules to inconsistently leak outside the directory where they are defined. This happens when using --delete-after and -R and when asking for transfer of a child of the directory where the filter rules are defined, with an other transfer directory later on the

Hi, I'm going to migrate from SAMBA 3.2 to 3.5 (Debian Lenny -> Squeeze). The server is an AD member and uses ACLs. Are there any preliminary steps to make the upgrade as smooth as possible? What kind of problems could I expect? Best Regards, Christian -- __________________________________________ Christian Reischl Fraunhofer Institut f?r Verfahrenstechnik und Verpackung Giggenhauser

Hello Everybody, we have authentication problems with the mentioned configuration. Current situation: We have two Windows 2008 R2 domains (currently on 2003 level) in separate forests. Recently we created a two-way trust with selective authentication between them. A Debian Squeeze LTS machine running SAMBA 3.6.6 (latest Backports version) is member of our primary domain and provides file

I am trying to get my connection to IConnecthere.com working. I didn't have a register command in sip.conf at first, so I believe that is why it was not working. However, I can't seem to get the register command correct, it just keeps timing out. Below is what I have: register=<username>:<password>@natrelay.deltathree.com I know that there is supposed to be

Hi All, I am using PWLIB-1.6.6-1 and Openh323 1.13.5-1 and running a RH7.3 machine and I am unable to compile asterisk due to these errors. say.c: In function `powiedz': say.c:1633: parse error before `int' say.c:1636: `i1000E6' undeclared (first use in this function) say.c:1636: (Each undeclared identifier is reported only once say.c:1636: for each function it appears in.)

Hi All, Are multi-channel (more than 2) formats fully supported in the OggVorbis specification ? I couldn't find any information about multi-channel support on xiph.org. I've used 'oggdropXPd' to encode a 5.1 wavefile and the Xiph OggVorbis libraries (vorbisfile.dll) to decode the file successfully, however the order of the channel interleaving is different to the original wave

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I

I'm a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w I'm planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate (mstupak@comcast.net). Here's what I'm planning: === Parts List === 1 Digium Wildcard TDM400P w/

how does one free segment(offset=77984887808 size=66560) on a pool that won''t import? looks like I found http://bugs.opensolaris.org/view_bug.do?bug_id=6580715 http://mail.opensolaris.org/pipermail/zfs-discuss/2007-September/042541.html when I luupgrade a ufs partition with a dvd-b62 that was bfu to b68 with a dvd of b74 it booted fine and I was doing the same thing that I had done on

Hello, I need help here about plockstat on X86 platform (Sun X4600 AMD) # plockstat -A -p 20034 plockstat: processing aborted: Abort due to systemic unresponsiveness # plockstat -e 5 -s 10 -A -x bufsize=100k -x aggsize=20m -p 20034 plockstat: processing aborted: Abort due to systemic unresponsiveness # ps -ef | grep 20034 algodev 20034 1 2 07:00:54 ? 86:17

Dear list, I am trying to learn how to use Zap-things in Asterisk. While loading Asterisk verbosely I get this error: [chan_zap.so]Warning, flexibel rate not heavily tested! => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Mar 12 17:19:01 WARNING[5563]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 12

Hello, I use Brendan''s sysperfstat script to see the overall system performance and found the the disk utilization is over 100: 15:51:38 14.52 15.01 200.00 24.42 0.00 0.00 83.53 0.00 15:51:42 11.37 15.01 200.00 25.48 0.00 0.00 88.43 0.00 ------ Utilisation ------ ------ Saturation ------ Time %CPU %Mem %Disk %Net CPU Mem

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a

I'm sure there must be a setting I'm missing somewhere, so I thought I might was well ask here. Conversations are punctuated by sudden replacement of a given syllable or so of conversation with a DTMF tone. I would hope perhaps there's some kind of setting that has to do with the way it detects inband DTMF? I'm pretty sure it's an artifact of this particular ATA; my

I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways..... my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.........am expecting 1000 -> 5000 users.. your thoughts would be appreciated. _________________________________________________________________ Don't

> I'm about to try to integrate SPEEX into the VOCAL project. Good. Just a detail, but the correct spelling in "Speex". > 1) To encode, it appears I need an array of floats. If > I am playing wav files, what is the best way to turn these > into something speex can encode? Speex version 1.0.x (stable) expects float's in the -32768 to +32767 range, so it's just

Jean-Marc Valin wrote: >>I'm about to try to integrate SPEEX into the VOCAL project. > > > Good. Just a detail, but the correct spelling in "Speex". > > >>1) To encode, it appears I need an array of floats. If >>I am playing wav files, what is the best way to turn these >>into something speex can encode? > > > Speex version

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to