similar to: Failover Design

I have2 Trixbox Servers. Each has an IAX trunks to the other. One works the other fails: -- Executing [s at macro-dialout-trunk:19] Dial("SIP/526-09eec7c8", "IAX2/InterOffice/210,300,tr") in new stack -- Called InterOffice/210 -- Hungup 'IAX2/InterOffice-7578' == Everyone is busy/congested at this time (1:0/0/1) The only difference I am aware of is

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

Using a spa3000 with asterisk cvs head, and the spa3k is config'ed with a dialplan that essentially routes any call starting with an "8" to asterisk. All other US 7 and 10 digit calls, 911, etc, route via the spa3k's fxo port. Is there a way in extensions.conf to: - inspect the dialed exten number, - if first digit is "8", drop the 8, - continue through each

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 I have an installation where I'll have a site to site data DS1 for use between two corporate offices. We'll have one asterisk server at each office. I'd like to be able to route calls over the 24 channels on that DS1 between the offices, instead of over the voiceT at each location to maximize savings on interoffice calls. An

The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies

After about one weeks time I've gone from no VoIP to a completely configured system for two of our offices to be able to page/communicate interoffice as well as handle existing PSTN communications (okay, waiting onf hardware for the PSTN side and I've likely jinxed myself now). I was sweating getting the two boxes talking to each other and I knocked that out in no time without even

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

I'm rolling out an installation with snom 360s in the near future. Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I have the 360's set up to subscribe and notify for the line use lights, which works like a charm for interoffice calling (between the 360's, anyway. The IAXy, 200 and,

The registry's are stored in DB. Just export your database with 'database show' Schedule it with cron to run every 5 minutes or so. You can do that with -rx command line switch for asterisk. Send the file across to other node and pipe it through awk/perl/cut or whatever you like and import it when you bring the other node up. You will have to stop and start asterisk I think. I

Hi, Guus I encounter the below log messages that after tinc run for a while, it seems box1 begin to buffer traffic to abc, and box1 is behind a firewall to connect abc from firewall’s inside to outside, and abc is on the public internet. Normally the major traffic is actually from abc to box(download direction), but not sure how this happens, and could you suggest something we can avoid this to

Here is a patch which adds mailboxcmp() and mailboxcasecmp() functions, similar to mailbox_equals(). Names were chosen to match strcmp() and strcasecmp(). I needed this for Johaness Berg's dspam plugin. It watches a folder "SPAM" and forcing this to be uppercase is unacceptable for me. (I also had to modify the plugin to use the new routine.) It's against dovecot-1.0.beta8

--delete option is not working for me. The following is a minimal example. $ mkdir box1 box2 $ touch box1/letter box2/extra $ rsync -a -vv box1/ box2/ sending incremental file list delta-transmission disabled for local transfer or --whole-file letter total: matches=0 hash_hits=0 false_alarms=0 data=0 sent 92 bytes received 34 bytes 252.00 bytes/sec total size is 0 speedup is 0.00 $ tree

Hi. My goal is to use rsync to syncronize box1 and box2. On box2, I have a tapedrive which I write the data I syncronize from box1 every night. On box1, my /etc/rsyncd.conf looks like this: root@box1# cat /etc/rsyncd.conf max connections = 1 syslog facility = local6 [tmp] path = /tmp read only = yes comment = export of /tmp hosts allow = box2 auth users

> Message: 13 > Date: Fri, 9 Jul 2004 11:42:01 +1200 (NZST) > From: =?iso-8859-1?q?Eugen=20Cristea?= <tecristea@yahoo.co.nz> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] asterisk to asterisk config > Reply-To: asterisk-users@lists.digium.com > > Hi, > > I would like to set two separate asterisks to talk to > each other. > Any

Hello everybody and Timo. I have the next problem. With LDAP tree like ou=mail | - dc=example1.com,ou=Mail | - mail=box1 at example1.com,dc=example1.com,ou=Mail | - mail=box2 at example1.com,dc=example1.com,ou=Mail ? | - dc=example2.com,ou=Mail | - mail=box1 at example2.com,dc=example2.com,ou=Mail | - mail=box2 at example2.com,dc=example2.com,ou=Mail ? and settings in dovecot

I have two Asterisk boxes, Box1 has TDM22B and Box2 has a TDM11B. From Box1 I can pick up Zap/1 and dial 300 causing Zap/1 on Box2 to ring. From Box2 I can pick up Zap/1 and dial 100 causing Zap/1 on Box1 to ring. PSTN calls come into Box1. They ring thru to Box2 on Zap/1 via IAX, then if no answer they ring Zap/1 on Box1. When such a call comes in and is answered on Box1, I can transfer to

Hi All, I''ve got an interoffice IPSEC VPN in place that I''m trying to give priority to terminal service (tcp 3389) traffic. I''ve created rules at each end, but have hit a bit of a dillemma. As the data is encrypted I must also give highest priority to protocol 50 otherwise the priority is lost as the packet gets encrypted. When I do this however, I can''t

Hi All, I've been experimenting during this weekend with asterisk as a softswitch, talk about me being completely lifeless, but let not talk about that. I've been conducting some really funny tests, trying to get an optimal SoftSwitch functionality. Here is my current setup: Source: Windows XP Pro + SJphone Box 1: Asterisk running in PassThorugh mode Box 2: Asterisk running in

I am thinking about a change to our company's phone "layout" and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to