similar to: Asterisk failover solution

Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve -- Steve Foy steve@narnian.org

Questions: Does anyone have a really STABLE asterisk system running about one year without need to restart the service or the SERVER ? Does anyone have a production Call Centre saled that don't lockup and is stable for 6 months ? I'm asking this questions because we have choose Asterisk for our call centre solution but, since the bugtracker only grows and people still want to stuck more

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

I love TrixBox, with the custom config files you can tweak pretty much with TrixBox too, I have at least done some. Plan to do a plain Asterisk install later, but for now I learn a lot about the config files just with TrixBox. Some things might be a bit harder with TrixBox due to some of the premade dial plans, but can get it to work :-) _____ Fra: asterisk-users-bounces@lists.digium.com

Dear All, I was trying to enable call forwarding, following the steps of the link on voip.org regarding this issue it doesn't work and the phone I am trying to implement on is still ringing. below is my conf in extensions.conf and the CLI output during the process. My configuration is : exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2}) exten => _*5X.,2,Hangup exten =>

Hi, I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this

Hi All, Is it possible to go beyond 250 concurrent ZAP channels with some tweaking or workaround ? Meetme uses zap channels, so we could have a max of 250 conference ports. Is it possible to higher this ? "An Asterisk system can only handle a *max. of 250 concurrent ZAP channels*. This is due to the design limit (255) within the ZAP channel driver." Thanks, ~Vamsi -------------- next

Hi, Realtime SIP and Extensions are working fine but facing some problems with Voicemail. Added an entry to extconfig.conf voicemail => mysql,asterisk,voicemail_users Created the corresponding table and an entry for mailbox 201. This is also reflected in the CLI as shown below. CLI> realtime load voicemail mailbox 201 Column Name Column Value

Hi, I was able to work out SIP trunk between Asterisk and CCM 4.x without any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk. Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not replying. For the same reason Asterisk is marking it as UNREACHABLE. Anybody got Asterisk and CCM 5.x intergation working. How can I fix the problem which I'm facing with CCM 5.x?

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

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

Dear All, I have installed Asterisk everything is OK until I tried to configure meeting room, configuration was simple enough when I try I get a message that it's not a valid meeting room, Now I don't have a Zaptel device on my machine, so I found that you will have to use ztdummy to make a dummy zaptel device on your machine and this is because of timing issues. My question is ztdummy

Dear All, I am totally new in this arena and I am still waiting for my installation process on freebsd to finish, but I wanted to make sure of the following: - Call routing between IP telephones can be done regardless of who made the phones? - Asterisk does support SIP V2? - it does act as SIP Proxy and Register? -- Thx MAG -------------- next part -------------- An HTML attachment was

Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but

Hi, I am trying to understand the building of Huffman codes from the code lengths. In the Tremor code first I see that the codewords are being generated by the function _make_words() and then sorted. After this I see some magic code and something related to unused entries. Does the code generate code words for unused entries too? Are these unused entry code words used during the decode

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar – get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

How to encrypt and decrypt the url using blowfish in ruby? ex: url= http://localhost:3000?username=vam&paswd=1234&street=hyd&contact=999999999&company=raymarine&city=hyd&state=UP&country=ZP&zip_code=543211 please help its very urgent. Thanks in advance - Vam -- Posted via http://www.ruby-forum.com/. -- You received this message because you are subscribed to

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??