similar to: include and hint in extensions.conf with new realtime feature - how?

hi, i just ported the patch of David Hinkle http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html to the current cvs version of asterisk. the theory is that the leds of supervisioned extensions are blinking until a call is established whereafter the leds should be constantly lit. however it's not working. the asterisk server is sending the following xml notify to the

Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added "exten => 403,hint,SIP/403" in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can this be done? Please Help Altus

hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips.

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax

hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set

I recently changed to all IAX providers for my DIDs, and none of them offer incoming caller-names. Back in the days when I did have incoming caller names, I found the names provided by the various phone-companies fairly useless -- "WIRELESS CALLER", "YOURCITYHERE" etc. Last Friday I finally set off to roll my own, in order to meet the following requirements: - uniformely

hi, after stumbling over the compile time flag in zaptel and after reading the new features of the 2nd generation firmware of the TE405P/TE410P, i was wondering if the cards are capable of upgrading the firmware in field? regards frank

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Hi, I am trying to use the support for monitoring extension states of a snom 220. In the extensions.conf file I added: exten => 770,hint,SIP/770 It means that when the snom phone boots, it will register it-self to asterisk as a monitoring phone for 770: Asterisk knows that the SIP/770 is monitored. I am using the 3.56q-beta firmware. It is in the DESTINATION option as it said on the tiki

hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a "Caller ID no screen" capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. the customer is currently using a paging system (small receivers which display a callback number and a base station (transmitter) with several antennas at the site) the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6

hi, since my latest libpri update i get these messages: !! Unable to handle ROSE operation 36 !! Unable to handle ROSE operation 30 i searched through ITU X.219 and X.229 but can't find any values for the Remote Operations Service Elements. are these AOC-E messages? regards frank

One of the 3 ups's that I am trying to configure is a Tripp Lite SU5000RT4UHV. The ups is recognized by the IETF 1.4 MIB, but I am getting Unhandled ASN 0x80 and 0x81 when starting upsdrvctl with the -D option. When I do a upsc {name of ups} the message returned is Error: Connection failure: Connection refused. Thank you for your support V/R Bill -------------- next part

Hello, we are a firm who wants to develop some VOIP solutions. The first infrastucture we choose for development is: - an Asterisk machine connected to a traditional PBX (s0). In this way people is not (yet) obligated to migrate its extisting PBX (and analog phones) to VoIP. - The PBX will be then configured to redirect specific outgoing calls (i.e. a remote branch office) to Asterisk, that

I'm looking to use Asterisk to replace my current PBX system. I'm in Australia so I need to use Austel approved equipment. My plan is roughly as follows: - Get a box with a suitable card and install Asterisk - Connect our existing PSTN lines to the Asterisk box - Get suitable softphones/ip phones to connect to the Asterisk system - Route all intra-office calls over our VPN - Route all

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

Hi! First I have to say, that I'm not very familiar with CVS and patching. I tried to patch & compile CVS-HEAD. First I checked out zaptel, libpri and asterisk with this command: "cvs co zaptel libpri asterisk" But the latest patch sipsubscribe-20050812.rev806v2.txt from http://bugs.digium.com/view.php?id=3644 didn't worked, so I tried to check out an older CVS-Versions