similar to: SV: Set(CALLERID(all) not working with 'unknown'call?

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Hi, This is really strange (but probably simple solution). The CALLERID(all) setting doesn't seem to work when the incomming callerid is 'unknown'. Dialplan looks like this: exten => _3072,1,Answer exten => _3072,n,Set(CALLERID(all)=DIRECT <0850553072>) exten => _3072,n,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004&SIP/2201&SIP/2202&SIP/2

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Can someone explain what the parameters pridialplan and prilocaldialplan are? What do they do and do I need them? I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx. The pbx technican complains about the format of the nr asterisk sends. Asterisk sends all numbers in on piece the pbx expects the numbers devided into

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm using the biult in feature attended transfer. If someone calls me, I hit the #, dial another extension and connect these two extensions. When hitting # and dialing the nr, asterisk only diales the new nr for 15 seconds. Is it possible to increase this time? I've only found the timeout for the digits, not for the call time. Anyone

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I' try to implement an automatic callback mechanism, just for local SIP calls.. Callback on busy and on no answer. If the other party doen't answer, it should be possible to press 5 to place an callback. Here is my dial: exten => _X.,1,Set(EXITCONTEXT=callback) exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT) And here the script for

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

Hi, I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops near "Apache2 starting...". I started my system with "recovery" kernel, and tun off misd, then my system works fine. I think it's problem with memory. Has anybody debian and misdn working fine? Maybe you can

Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I've used CISCO 79XX phones, and they're great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? ---------------------------------------- Rod Bacon

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from

Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier

Hey all, I''m not sure if this is a ZFS bug or a hardware issue I''m having - any pointers would be great! Following contents include: - high-level info about my system - my first thought to debugging this - stack trace - format output - zpool status output - dmesg output High-Level Info About My System --------------------------------------------- - fresh

Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way:

Hi All, I have read many posts about problems with Asterisk on some systems. I also set up Asterisk on many different boxes. But I have never seen the following... There is an IBM X346 (3.4GHz Xeon) with one Sangoma A104. This system is currently idle, that means there is nothing running except Asterisk (1.2.7.1). We are handling no calls now, but if I do a vmstat, I get peaks in system load up

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I wonder if someone else discovered that behavior and hopefully fixed it. I've two asterisk boxes, both have a user 102. If 102 from Box A calls 105 on Box B, Box B want's 102 from Box A to authenticate. But it's an incoming call, there shouldn't be a authentication. Box A doesn't send an authentication so the call

Hi, I've connected Asterisk with 4 PRI to a Siemens HiPath 4000. For CALLERID(name) feature I wanna use Q.SIG as switchtype. Cause Siemens PBX orders Channels logical I need the parameter qsigchannelmapping=logical. Here is my chan_dahdi.conf trunkgroups] [channels] language=de context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes