similar to: Semi-OT: useful things to do with XML browsers in phones

Requests 1. Directories linked to their databases 2 weather broadcasts 3 local traffic info. 4 local news headlines 5 sms send / receive 6 alarm on the phone of calendar events - not a call back, simply a beep and notice without pulling out a pda or opening outlook Couple of other "one of's" too, some very esoteric like evening's TV lineup, Dave Bour Desktop Solution

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Greetings list, I've noticed over the last couple of weeks that, unsurprisingly, nearly every new PC seems to be coming with Vista these days. I expect it'll only be a matter of time for all of us before clients start needing Vista-compatible softphones (if it's not already happened). So, what's the story with Vista compatibility amongst the softphones currently out there?

Greetings list, Does anyone have experience with SIP conference phones? I need to source a couple for a client, but I'm not really familiar with the market - i.e. what's available, what's decent quality, etc.. A cursory googling has led me to the Polycom Soundpoint IP4000 at around the ?450 mark - any thoughts on this? If anyone knows a good Polycom wholesaler in the UK, I'd be

Greetings list, I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.). Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

Greetings list, Wondering if anyone's come across this before. I've configured a couple of our servers with a "privatedundi" context to allow calls to still flow between extensions even if they're registered to different servers . The DUNDi lookups seem to work fine, evidenced by the following on the originating server: -- Called

Greetings list, I posted this to the -biz list a few days ago. In hindsight, I think it would have been more appropriate posted here, so apologies to those on both lists who've now seen this twice. I have had a request to provide 2x PRIs to a site in Lausanne, Switzerland, but my knowledge of the Swiss Telco market is non-existent. Are there any folks on the list who've experience in

Greetings list, Not exclusively asterisk-related, but I've noticed the CentPBX site has been offline the last few days. Anyone know the reasoning behind that, and more importantly, is anyone mirroring it? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons

Greetings list, Wondering if some kind soul can help me with an issue with chan_sip segfaulting as soon as it loads... Basically, if sip.conf contains any peers with "host=dynamic" in them, asterisk won't start. Doing -vvvdddc yields the following: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Segmentation fault As

Greetings list, Thanks to all who replied to my thread a few days ago "SIP devices with packet loss tolerance". One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which

Don't know if anyone's got experiences on this they'd be able to share... I'm trying to obtain numbers for Kenya/Nigeria, but I'm struggling to find a company selling them. There's one for Nigeria listed on didx.com, but $22/month seems a little steep. Has anyone had any luck getting DIDs for countries in Africa, and are there any companies out there selling them? Thanks

Greetings all, Not specifically an asterisk query, but a couple of transfer queries that I'm sure are obvious to folks who use these phones all the time: 1) how does one do a blind transfer? When a call is answered and one hits the transfer button, followed by an extension, one has to wait for the other party to answer, then hit transfer again, before the call is released. I'm sure there

Greetings list, A quick question regarding extensions.conf #include behaviour if I may. I'm sure someone will know the answer off the top of their head... How does asterisk handle "overloading" of contexts. For example, say an extension exists in extensions.conf as follows: [incoming] <some stuff> Then one includes a, b and c.conf, each of which also contains: [incoming]

Greetings list, I'm trying to establish if there's an issue whereby certain telcos in certain countries have not updated the London, UK numbering plan to include some parts of the 020 3 range, despite it being in operation for some two years now. To help with this, I'd be most grateful anyone outside the UK could make a test call to +44 203 3393 7389. This is a simple test number

I've been working my way through the softphones listed on voip-info over the last few weeks and I've not really found anything to fit the bill. Has anyone had more luck? The environment is a small call centre of 5 users. Operators often need to be able to transfer calls to other operators with different specialties, so the softphone needs to be easy to use and quick to transfer calls.

Hello all, One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN BRI delivered to asterisk via zaphfc, then dropped into a context as follows: exten =>

Greetings list, Hoping someone might have experience with poorly-performing net connections and which devices work best over them. One of our clients has a number of employees that work from home, and are given a SIP phone to take with them and hook up to their broadband. For the most part, this works fine, but there are an increasing number where sound quality is poor ("chops" in and

Hi, Is it possible to dial voicemail from a particular phone line and automatically enter the extension that is being dialed from, thereby only prompting for the password? I've been searching around to find if this is possible, but I haven't been able to find an example of this. I have a feeling it's more of a endpoint function, but I thought I'd ask if anyone has accomplished