similar to: 1.6.x SIP allow incoming calls based on from ip address?

Hi All, I'm poking around with 1.6, tried to compile the addon package, but it doesn't see mysql_config installed. I have mysql-client, mysql-common and mysql-server installed. I'm running debian etch. Any suggestions? Thanks. JR -- JR Richardson Engineering for the Masses

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init

Hi, I have a data set look like this: ================================= Fruit Quty apple 20 banana 10 orange 17 apple 30 apple 15 orange 26 banana 15 .........and so on .......... ================================= The level of fruit is 30, that is, there are 30 different fruits. I'd like to compute some simple statistics for each different fruit and get output like this:

Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed "pedantic=yes" the other day. After years of

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

Hi All, I have a Cisco 2600 PRI gateway being hosted on an Asterisk server. The PRI on the cisco is pointing to a customer legacy PBX, the SIP VoIP side of the cisco is pointing to an Asterisk server (1.2.X). In Asterisk, the SIP peer is setup with callerid="some name"<5551212> In a SIP call from the cisco to asterisk, there is no CID name info in SIP debug, so Asterisk

Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

Hi All, Strange issue, recently I started getting a lot of .lock files in the voicemail /INBOX folder preventing proper access to voicemail. I can delete the .lock files and everything is normal. After searching around, I found some SIP lock file stuff but nothing specific to voicemail. Can someone point me in the right direction to resolve this? I'm runnning 1.2.9 on Debian Sarge.

Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses

Hi All, I'm running an AGI, calling a perl script the does number lookups to a remote server. I would like to put a timeout in the script. The problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. Here is my clean script:

Hi All, In the Asterisk extra sound file, there is listed in the text file an 'On' and an 'Off' sound prompt but I only have the 'On' file. I have searched through various versions of 1.6, 1.4 and 1.2, and can not find the file. Does anyone have this prompt and can send it to me, gsm or ulaw? Thanks. JR -- JR Richardson Engineering for the Masses -------------- next

Hi All, After reading the sparse info and attempting to get this running, I'm unsuccessful and could use some guidance. I already have a MRTG server up and running serving hundreds of router interface graphs. I would like to add SIP/IAX channel graphs for all our asterisk servers. I'm running asterisk 1.2 and MRTG 2.4.17. I tried the script from http://karlsbakk.net/asterisk/ but get

Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the

> I have 2 PBX's, one in each office (say one in New York, one in Boston). I > have mobile users that can show up at either office and connect their soft > phones. > > > > Is there a very simple DUNDI config available which describes how to set > this up? > > Also, can I have the same outbound trunks setup in each office, so that > calls don't have to

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure

Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified field. ""-any custom value that you wish to store."" My question is how do you setup more custom fields in the cdr and be