similar to: Project independent VoIP mailing lists?

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

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

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

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, 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

Folks, First of all, this email is sent to -users and -biz, but please follow- up to the -biz list only. I have set the reply-to, but I fear mailman will strip it off ... Please don't flame me for posting to -users, I'm just not sure who lives on -biz as the (signal/noise | net.kook | troll) factor has been pretty bad on that list lately (poor Rehan!!) ;-) Telephony Depot

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'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, 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 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.

Greetings list, There have been quite a few posts on the list over the last few months about using DUNDi to ensure users are always reachable even when logged into different asterisk boxes (as part of a load balancing cluster). For example, yesterday, this was in a post: (Olle Johansson) " In combination with Dundi and the regexten= system, it's even more dynamic." Are there any

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, I've been having a go at preparing some music on hold from CDs clients have supplied, but quality seems really rather poor over compressed channels (tried g729, GSM and Speex). I've been doing the following: sox -v 0.15 <filename.wav> -t raw -r 8000 -s -w -c 1 <filename.sln> resample -ql As I understand it, I'm reducing volume to 15% (which sounds about

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

have an urgent requirement, I'd go with the S450 for now. Regards, Chris --=20 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, 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, I've been asked by someone to help them set up a SIP link between an asterisk system and an Alcatel OmniPCX (v6 software). The asterisk bit's fine, but I know nothing about the Alcatel except that it does apparently allow the setup of SIP trunks. Does anyone have experience with the Alcatel unit? Any obvious pitfalls to watch out for? Any suggestions gratefully

Greetings list, One of our asterisk boxes has been spitting out the following error this morning: Oct 15 12:31:50 WARNING[22300]: acl.c:306 ast_ouraddrfor: Cannot create socket Looking at the list archives, it seems this is usually caused by insufficient file handles on very heavily loaded systems. Is there anything else that can cause it? For now I've killed and restarted the asterisk