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Hi! I've went through the READMEs and could not answer this question: During installation, the Setup program asks: Would you like update/upgrade wanpipe drivers? (y/n) For a pure Asterisk TDM installation - is it required to patch the kernel or is this only when using the sangoma cards as WAN router? regards klaus

Hi! What are the typical ways to work around the 64 groups limit? thanks klaus

Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -

Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? Regards Bilal

Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE

Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango

Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,

Hi! I wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) thanks klaus

Hi! All the AEL examples have a semicolon after the closing curly bracket, e.g: context test { 1 => Hangup(); }; but without ; it works fine too, e.g: context test { 1 => Hangup(); } So - what is the reason for the ; after the closing curly bracket? thanks klaus

2009/3/4 Klaus Darilion <klaus.mailinglists at pernau.at> > core show function SIPPEER Thanks : that's exactly what I was looking for !! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/7d05f172/attachment.htm

Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. thanks klaus

Hello, Is there a list of software phones which will work with Asterisk? For Linux and Windows? I don't have any hardware yet, and before I buy anything I would like to know how Asterisk really works (with software "phones" for example). Tomek

Hi! I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed

Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called "asterisk-1.4-current.tar.gz" This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. Thanks Klaus

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-5555: the TAPI driver would call

Hi there, Regarding the calling card system, I would like to know what software tools are suitable to write the CDR system? 1) Perl (but what kind of database should be used?) 2) PHP + MySQL 3) or others suggestion? Also, which one is easier to access AGI? Any source and information would be appreciated. Best regards, Max Chow -------------- next part --------------

...> > Content-Type: text/plain; charset=us-ascii > > Hello, > > --- Tomasz Chmielewski <mangoo@mch.one.pl> wrote: > > > Is there a list of software phones which will work with Asterisk? > > See the 'SIP Phones (SIP User Agents)' section here: > http://pernau.at/kd/voip/bookmarks-sip-rtp-ua.html > > Regards, Girish > > > > __________________________________ > Do you Yahoo!? > The all-new My Yahoo! - What will yours do? > http://my.yahoo.com > > > ------------------------------ > > Message: 3 > Date: Wed, 01 D...

Hi! I'm using a PRI card. When a call arrives, I want to answer the call and dump the binary data received on the B-channel into a file or stdout or to the console (for debugging the B-Channels). Is this possible? regards, klaus

Hi! I've bought the EICON DIVA V-2BRI card an begin installation soon. Are there any issues with certain kernels? E.g. the standard debian kernel 2.4.27-2-386 and kernel-image-2.6.8-3-686. Further current kernel 2.6.16.14 would be an option. Thanks Klaus