similar to: Asterisk vertical service activation codes

Source: http://www.nanpa.com/number_resource_info/vsc_assignments.html See also: http://bugs.digium.com/bug_view_page.php?bug_id=0000071 Some (which?) of the codes below are hardcoded into Zap channels only. Is there a European equivalent for this (or ITU / IETF)? Greetings, Philipp VERTICAL SERVICE CODES (US Standard) ASSIGNMENTS *00 - Inward Voice Activated Services (English) *01 -

Greetings- I'm in the process of turning up an Asterisk box for a customer and was wondering if anyone had any good code they could share for implementing vertical service codes within Asterisk. I'd really rather not have to spend hours making new wheels if someone has one or more that will fit. One of my issues is that I've had a very hard time finding out exactly which

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

I can't seem to get *80 to do its thing on a Zap channel. Looks like *8 is being seen by asterisk first, and *80 is basically inaccessible. What *80 is intended to do, by the documentation on the wiki and by inspection of the source code, is add the last callerid to the blacklist. Looking at the source, I see the same behavior coded in chan_zap, chan_mgcp, and chan_skinny. While *8

Hi there, after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

Hi all, Really excited to be using Asterisk and learning about VOIP and PBX's. I'm a complete beginner at telephony but have built and installed Asterisk 1.4.5 and read several of the Asterisk books online and have successfully connected to FWD with IAX2 and to GIZMO using SIP. Just purchased a Motorola Wildcard X100P and installed it into a clone PC running Fedora Core 6. Analog

Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a

Greetings Asterisk Users, I'm looking for some friendly advice on setting up a asterisk PBX for our small business. I've played with Asterisk and setup a soft-phone open323, though even on my ethernet network this showed very poor performance. Got a phone call through to digium, but had a difficult time either hearing (low volume) or understanding (line breaking up). Hoping a hardware

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

Thank you for the suggestions Andrew. I have not come across some of them before and will give them a shot. Based on my reading, changing the motherboard should have minimal impact unless that motherboard and the TDM400P don't get along (aka. IRQ sharing). I have disabled everything that is not needed and I do not believe I have any IRQ problems and I am NEVER wrong ;). Calls are crisp and

Hello All, Just a question, I have an adit600 and I am looking for a way to pull the incoming cid into asterisk. Does anyone know if this is just not possible via t1? Or is it only available on PRI? Thanks, Greg

We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14 debs from http://pkg-voip.buildserver.net When misdn stuff (misdn-init start) is not started, everything is fine, our 8 FXO (Channel 1-8) 4 FXS (21-24) are working well. If we start the misdn stuff (one card, port 1,2,3,4 in misdn-init.conf and no TELCO cable plugged in the card), the dialtone disappear on TDM lines :-( Does

Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

Hello, I am plotting many histograms together using facet_grid in ggplot2. However, I want to then add a vertical line to each histogram, or facet, each of which vertical lines are at different x-values. The following example adds all vertical lines to each facet: ggplot(data,aes(values)) + geom_histogram() + facet_grid(.~variable) + geom_vline(xintercept=c(5,10,15)) How can I add a vertical

Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __________________________________ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com

Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like " I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him "wait a sec" and push "Flash" and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to"...