similar to: * Compatible VSP Service in Ukraine?

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

Hello, everybody: My friends in Ukraine are starting a research lab at a national university and they asked what programs to use. I said "R" of course, and they then asked me 'what support does it have for Cyrillic'? i've done some snooping in the R website and all the references i find to foreign languages concern c and fortran, not Ukrainian or Russian. Since i'm an

Hi All, My friend is looking to for a fulltimer to work on various VoIP gateways (mostly Cisco) and few Asterisk servers. Some development skills and knowledge of Asterisk's API would be an asset. At least familiarity with Asterisk's AGI and Perl/C/PHP would help. He is located in Kiev, so obviously Kiev residents are preffered. Remote work to be discussed. Again, this is not a contract,

Dear All, could anybody be so kind and explain me such situation: I set up OpenSSH client on Windows 7. Then I generated keys pair by 'ssh-keygen -t dsa' and with empty (no) passphrase. Then I inserted appropriate info into authorized_keys file on my Unix host. But when I try to connect to my Unix host 'ssh' asks for passphrase for 'id_dsa' and than wants to authenticate

Hi All, I seem to be having a problem with all my VSPs. When I am registering with them I don't seem to be passing my port number. This problem causes other users the inability to call my VoIP number with the VSP. My VSP showed me what they are seeing. I have changed my useragent to be: Linksys/SPA941-4.1.15 Linksys/SPA941-4.1.15 Contact sip:1234321234@aa.bb.cc.dd with no

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support

Our project is a service for C2C and B2C segments of the global online market. It''s orientation is a wide segment of customers – users of online markets, classifieds, auctions, etc. Project is designed for the worldwide usage, starting with the biggest markets: USA, China and Western Europe. Our goal is to obtain the senior position in the worldwide B2C market, through taking a

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

Hi All, I think I have missed something as I am resisted with 4 VSPs and I can not dial in using any one of them using the corresponding VoIP numbers assigned with the VSP. I can make outbound calls to another VoIP number to the same provider. The weird thing is that I have a DID with a VSP and I have that working fine but try using the associate VoIP number and nothing happens. When

Greg, Chris, and Jay, Thanks! You've given me plenty of info to digest. I really appreciate the responses. Apologies if my list manners aren't up to snuff! Thanks again, Jason

I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by root@gf-002-pbx-001 on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing.... I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right

hi all. it was happened that i have no upgrade my system long time. Before that i used for upgrading "classic" way. make buildworld make buildkernel make installkernel mergemaster make installworld But in current moment as i can see the upgrading is totally changed. As i understand commands sequence is next: #freebsd-update fetch #freebsd-update install #freebsd-update upgrade -r

I have a phone (a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more

I'm not really looking for working configurations as much as I am looking for people who can say "This is a solid product and I trust my business to a solution running Asterisk." As far as pre-sales work... Well, tell that to my consultant. I'm quite excited about *. I've got my company sold on it, they just want some reassurance that it's ready for prime-time

https://bugzilla.samba.org/show_bug.cgi?id=11423 Bug ID: 11423 Summary: rsync 3.1.x is creating empty backup directories Product: rsync Version: 3.1.1 Hardware: All OS: All Status: NEW Severity: major Priority: P5 Component: core Assignee: wayned at samba.org

Hi i have the running i both sides, client and server sides the puppet 0.25.4 Get this error on server side: puppetmasterd[5363]: Report puppet_dashboard failed: wrong Content-Length format And receive this error on my client side: warning: Value of ''preferred_serialization_format'' (pson) is invalid for report, user default (b64_zlib_yaml) I am getting any reports on my

On Thu, Feb 13, 2014 at 5:52 PM, Renato Golin <renato.golin at linaro.org> wrote: > On 13 February 2014 13:47, Evgeniy Stepanov <eugenis at google.com> wrote: >> Hm, I see that -funwind-tables on arm-linux-androideabi target >> replaces this "cantunwind" with a proper unwind table. >> Hence http://llvm-reviews.chandlerc.com/D2762. > > If Android is