similar to: R: queue behaviour

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

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply

Hi folks, need some help on queue behaviour. What I'm trying to do is accepting a call from pstn, put it into a queue, while callee is waiting contact some numbers till one responds, then bridge the two calls. What I can't manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I also tried AGI and Asterisk Manager, with the same

Hi everybody. Just noticed that when calling a mobile phone, Asterisk doesn't bridge the voice message by telco if mobile is unreachable, but keeps on ringing till it receives a hangup signal. I think this is due to the fact that the message is played without the call has been answered, but I'm wondering if there's some way to let Asterisk realize it. All I see in the CLI is the line

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

I apologize, but I can not find a complete list of directives in dovecot.conf possible. where can I find? thanks excuse my for my orrible english -- Caselle da 1GB, trasmetti allegati fino a 3GB e in piu' IMAP, POP3 e SMTP autenticato? GRATIS solo con Email.it http://www.email.it/f Sponsor: Apri Conto Corrente Arancio entro il 28 febbraio 2011 e ricevi 100 euro da spendere su Media

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, 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 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, 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, 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 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'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

Hi Charles, attached there is a patch for a new version of riello drivers: Changes: - small changes in riello_usb.c and riello_ser that solved problem with UPS that not support realtime autonomy calculation (and also internal temperature); - the second problem I think is related to the first because the shutdown command that I sent to UPS by NUT monitor works without problems; Cheers, Elio

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

Thanks, chdir dir /data error seems go away, but at 12:00 o'clock the following error appear: Feb 20 11:45:03 brulx01 vsftpd: Mon Feb 20 11:45:03 2006 [pid 12273] [job260] OK DOWNLOAD: Client "10.90.1.1", "/data/mde/DATA/.././ACTUAL/mdeact260.msg", 12481 bytes, 174.76Kbyte/sec Feb 20 11:56:10 brulx01 smbd[12410]: [2006/02/20 11:56:10, 0]