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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, 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, 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

We manage a *lot* of asterisk boxes at client locations at the end of = DSL connections. We have a schedule to make sure each box is updated = once a month (e.g. "these 10 boxes are updated in week 1 by Marcus, then = in week 5 by Tom, etc."). If we can login and run a couple of simple = commands to bring everything up to date, that saves us many hours every = month. For better or

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, 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, 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

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 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

Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of

Hi all, I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. An IAX gateway is used for outbound calls. At the moment, when an incoming call comes in, asterisk dials every SIP phone like so: Dial (SIP/1&SIP/2&etc.) This has worked fine for some months, but I noticed a few days ago

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've just been upgrading one of our servers from 1.2.17 to 1.2.21.1-r1, and noticed that it's not picking up any of my macros written in AEL. Upon further examination, it looks like pbx_ael is missing. Is this a deliberate change, or is this something I need to address in the pre-compile configuration? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For

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

Greetings list, It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do

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.

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, I've been using DUNDi for some time now to prevent calls between users going out via PSTN if there's no need, set up as follows: [macro-dundi-e164] exten => s,1,Goto(${ARG1},1) include => dundi-e164 [dundi-e164] include => in-e164 switch => DUNDi/e164 My outbound call macro tries [macro-dundi-e164] first, if that fails then goes onto PSTN connectivity.