similar to: how to route outgoing calls on IP-level

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on

Dear Sirs, let me repost my question again, probably the last one was lost in a huge amount of messages during weekend. I'm actually looking for web-based tool which can do two level of administration: 1) high level, Administrators, can create "domains" 2) lower level, Users, can manage extensions within certain domain. much like asterisk2billing. so, I want users to manage

Dear Sirs, I'm looking for a tool which can do the following: 1) higher level of administration, only one person, it can create "domain"s and per-domain administration accounts 2) lower level of administration, many persons, each can add new extensions and change passwords with their domains. somewhat similar to asterisk2billing, but with privilege separated into

hello there... our telecom sold us VoIP-numbering, managed by Nortel MCS I successfully registered Ekiga to it ( http://sol.chel.skbkontur.ru/ekiga.png) What exactly do I have to write in sip.conf to make Asterisk register on this SIP ? Cheers, Kate -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I was looking for a Java FastAGI interface for Asterisk I came across asterisk-java first and didn't realize there was more than one out there. It seems to work fine and I've got my first project working with it, but I was wondering which Java FastAGI implementation is the most popular and how they compare against each other. So I'm aware of: asterisk-java JastAGI

Can You please advice me free softphone which supports SIP registrations ? Cheers, Kate -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/539fb7a2/attachment.htm

Some light relief .... SNOM say "Please note that you will not be able to reach us by phone." http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

> get.hist.quote(instrument="INR/USD", provider="oanda", start="2005-10-20") trying URL 'http://www.oanda.com/convert/fxhistory?lang=en&date1=10%2F20%2F2005&date=11%2F01%2F2005&date_fmt=us&exch=INR&exch2=&expr=USD&expr2=&margin_fixed=0&&SUBMIT=Get+Table&format=ASCII&redirected=1' Content type

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

Hi, Our operator frequently gets requests to transfer a call directly to voicemail in order for the caller to leave a message without disturbing the callee. Basicly, I'm looking for a blindxfer to vm. My first thought was to prepend a digit (eg 7) to the extension but this does not fit well with our dialplan. According to an article on voip-info.org Asterisk@Home appears to implement

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like

Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

I was on vacation the last week and wrote some code to run a 500-day correlation between the Nasdaq tracking stock (QQQ) and 191 currency pairs for 500 days. The initial run took 9 hours(!) and I'd like to make it faster. So, I'm including my code below, in hopes that somebody will be able to figure out how to make it faster, either through parallelisation, or by making changes. I've

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Marc, On 25 May 2017 10:25: > You can build Samba yourself. See Wiki. Thank you, that looks like the solution. Although I'm tempted to wait for 4.7 in case I break something.. > A migration documentation (package to self-compiled, self-compiled to > packages, or package to other packages) is currently work in progress. > Maybe I have it finished next week. Am I correct in

Hello; I am trying following but getting a warning message : Warning in rbind.zoo(...) : column names differ, no matter whatever I do. Also I do not want to specify column names manually, since I am just writing a wrapper function around getSymbols to get chunks of data from various sources - oanda, dividends etc. I tried giving col.names = T/F, header = T/F and skip = 1 but no help. I think

Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was

I have installed the kate editor on Centos 6.7 but it seems to be a very old version, 3.3.4, installed as part of kdesdk. On Centos 7 I can simply run 'yum install kate' but, alas, not on Centos 6. What is the recommended way of updating kate on Centos 6? Thank you.

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but