similar to: Number and name for SIP extension at the same time ?

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN

Hi, I'd like to test my links with remote locations. I wonder if there are any echo asterisk server that could be called for quality estimation .... Regards, Robert.

Hi, I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? I'd like to dial some default number if user doesn't dial anything or give him some message - but I don't know what gets executed after DISA if nothing is dialed .... I'm reading this on wiki, but

Can anyone provide me with a step by step on how to set up Windows Messenger on a Windows XP Pro box as a SIP client with asterisk? I'm interested in doing various tests of my asterisk server from the Windows perspective of the world. In the alternative if someone could provide information on another Windows based fully functional easy to configure iax or SIP client that would suffice as

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Hello. I'm interested in running guests as read-only to turn them into a sort of virtualized "live=cd". The goal is to leave no forensic evidence on the host disk or virtual one which would lead to traces on the host still- similar to how TAILS works but with the added convenince and flexibility of running in a VM. If I set the qcow image to read-only as per the manual, will any

Hi, thank you for last info. we've tried to use te110p but failed. We're quite surprised that cable wasn't included with the card as any documentation, at least on HW setup and installation, yet cable pinout for connection to PRI interfaces.... 1. We have followed instructions on your site and from Beronet guide, but card just keeps blinking and nothing happens (also no useful

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

Hi, I'd like to use Asterisk with ISDN interface and normal analog interface to door phone (or any other low cost connection type to door phone). What would be your recomendations for needed HW in Europe? Is it possible to have this in one PCI card? Are there any lower cost voip door phones? Thanks in advance, Robert.

Hi, I'm testind Diax. I have flashing note about 1 new voice message. Can I hear it somehow from Diax gui, or must I call pbx to get message ? Thanks, Robert.

Hi, I had music on hold working but now don't know what happened. I get : WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on hold (class '') on channel SIP... Any ideas what is wrong ? Regards, Robert.

Hi, I'd kindly ask for any guidance how to setup Netmeeting to work with Asterisk. I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call local extensions (no calls into PBX functions) but get no sound. Any hint, advice ? Anyone using Netmeeting (maybe also windows messenger) with Asterisk sucessfully ? Thanks in advance, regards, Robert.

Hi, I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall (all ports we're set according to instructions) on DSL line. When pbx connects to Digium demo server( I'm located in Slovenia, Europe), it gets first few words, then silence and then comes back when enumerating dial possibilities ("4 for accounting ...). Same happens from SIP or IAX local extension.

Hi, I'd like to implement scenario to send user to operator's queue by default (if doesn't dial any extension) but only if there is operator agent logged, so user could get response. If not I'd like to send it to voicemail... Any quick advice ? Thanks in advance, Robert.

Hi, I have P4 2.8 HT machine with Asterisk. I spot 3 processes of mpg321 taking 20% of CPU each even when no call is made to Asterisk or any other PBX activity is on ? Is this behaviour normal ? Regards, Robert.

Hi, I've found approximate same pricing for both. Sipura 2100 seems to have more features... What are differences between those two ? What about their reliability (specially regarding fact, that they deal with analog phones) ? Thanks in advance, regards, Rob.

Hi, I'm curious if anyone is using firefly as SIP client and if qualify=yes works for it. In my case Asterisk just keeps retransmitting of OPTION SIP message and Firefly doesn't seem to respond - but all this could be just wrong settings ? Anyone has working SIP configuration with qualify ? Thanks , Rob.

Hi, what would be general choice between those two options? They are related to two different things: security, performance. But what would I loose if I run as non-root without realtime priority ? Thanks in advance, regards, Rob.