search for: andresin

...enzaptelcong doesn't generate any config for the Astribank. Can anyone send a copy of their working Astribank settings? I am concerned about the span used by the Astribank. I just cannot find anything that gives me a clue... Thanks in advance. Regards, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

...cilityenable = yes musiconhold = default ;overlapdial = yes overlapdial = no immediate = no txgain = -4.0 rxgain = -4.0 signalling = pri_cpe channel => 1-15 ;channel => 17-32 channel => 17-24 ;toneduration=100 toneduration=300 ;relaxdtmf=yes Thanks, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/.) for using round robin, and

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

Dear all, I have small question in sip.conf I added [service] type=friend ;username= ;secret= qualify=900 host=X.X.X.X dtmfmode = rfc2833 disallow=all ;allow=g729 allow=gsm allow=alaw allow=ulaw and I can proccess incoming call from soft phone only I calling on number that is used in extensions.conf(in example below it is 1) exten => 1,1,Answer; exten => 1,2,Playback(hello-world,skip);

I am aware of how to match a particular caller-id or a caller-id pattern and do something with the call like this: exten => 15554441212/_888NXXXXXX,n,Playback(GoAway) What I am curious about, is the best way to block unknown, private and 000-000-0000 calls. I know I can do this for 000-000-0000 calls: exten => 15554441212/0000000000,n,Playback(GoAway) Is there a better way to catch

...be broken when the asterisk packages are updated. Would anyone consider just install everything from source (branch 1.4) as the best option? I would like to keep an easy upgradeable system like Debian packages, but could use source code if necessary. Cheers, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all I am working in a new set up with Grandstream GXP-2000 handsets. I like those phone, but they lack a feature I need: the phone cannot be locked by the user. What I actually want is a user to be able to avoid someone else making calls from his phone without giving him access to SIP configuration access to the phone. i.e. let say I want user

...: (not actual coonfig) exten=> 111111,1,Dial(Zap/41) exten=> 222222,1,Dial(Zap/42) Do I need any other piece of software? I know 1.6-beta is capable of managing faxes properly, but I won't upgrade my * if any other option is available. Thank you. -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's