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

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

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

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

> 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

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

Hi all, we face a problem with SMS reception sended to _landlines_, at least in France. Normally operators -tested with France Telecom and SFR- are sending voice SMSs from a particular CID number, so no problem. But today we discover that -at least SFR- send from time to time voice SMSs with original callerID which means that the call is terminated like a normal call and not recognized as

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

Hello! I'm about to try to integrate SPEEX into the VOCAL project. If anyone has any pointers as to the best way to do this, please let me know. After reading the speex api man page, I have a few questions: 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? 2) Are there any commercially

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/

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group The scenario that I tested was as follows. A call was originated from extn. 1001 on Vocal and the call was made to

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