similar to: Linux Voice Mail Application??

I have wanted to setup an Asterisk system for several months but have been unable to do so because I lacked the funds for a T1 card to connect my channel bank to the server. Recently, I acquired several Brooktrout T1/E1 interface (PRI-PCI48VC/PRI-PCI64V-C) cards for the PCI bus. According to the Brooktrout web site http://www.brooktrout.com/products/netaccess_pri_pci/specs.html Linux

I have the option to purchase an Brooktrout PRI-ISA48 dual-span T1 card, which, upon checking with brooktrout, is supported for linux 2.x, but before I do this, I want to check and see what the opinions of your, the list members, and Mark, of course, as far as asterisk being able to use this card. ANY information would be helpful, as this offer will expire to me very soon. Thank you. :)

Don't want the opening line of my post to be misconstrued. "Anyone familiar with the BrookTrout TR1034 Fax Board, supports T.38 and PSTN fax in an Asterisk environment?" That's a question, not a statement. Curious if anyone has played with one of these as an extension of an Asterisk server for T.38 faxing, did not mean to imply it could, should or would play well with

HI, I'm trying to config a span from port 2 of a DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been working fine for some time now. No problem with dialplan PSTN calls. Now I'd like to "route" specific DID numbers in thru the TELCO T1-PRI and out thru the second port (span) on

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com

I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323. Trying to find a way I could use my laptop to send a fax over H323 to the BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface on a Cisco router and setup a h323 dial peer to the BrookTrout, but I didn't the router with me! ----------------------------------------- Disclaimer: This

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

Hello all I would like change the all announcemennt(Voicemailmain,Voicemail etc.). But I don't know how to change the these each prompt. Do we have any guide book for this? Please teach me about changing the voicemail or other prompt. Thanks

In attempting to make my asterisk server the latest and the greatest tonight I attempted to upgrade my CVS. In the asterisk directory I ran the make clean, executed the CVS update -d, and after all the files completed ran make upgrade. My problem is that when I pull up the CLI the cvs version that is showing is the same date as my initial install. Does this mean that the upgrade did not go

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

Hello everyone. This is a message I've sent before on Sunday, no one replied so I'm reposting it (guess not everyone's at work 7/7) I've got this really annoying and beyond-my-knowledge-to-debug problem. The line connected to my FXO port gets marked "out of order" by my telco operator. I don't know how to explain this further. If I dial my own number from a

Hi, I have a strange problem trying to update Asterik on one of my boxes. I have done the following: - delete all the old source files - download the new file using: cvs checkout zaptel zapata libpri asterisk - compiling the new source file using: make clean ; make install Everything is compiled fine, except.... when I stop and start Asterisk server, it is still the old CVS version:

Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the *

Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can