similar to: UK SIP Provider

If your phones are setup to connect to the asterisk box by name, then a smart DNS server can just point phones to the backup box after failure. However, since asterisk running on the backup box doesn't know about the phones, this is only half the solution ________________________________ From: Mohamed A. Gombolaty [mailto:mgombolaty@noorgroup.net] Sent: Thursday, June 30, 2005 8:30 AM To:

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

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 =>

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

Hi there... I've to setup an Asterisk system for a small office, I haven't done one of these in at least a year and was wondering if someone could just let me know what sort of phones are doing well these days. It just needs 9 phones in the office, for general use, no fancy things required for that, just accept calls, transfer calls etc. 1 Master phone for a receptionist. Is there an

Hi there, I'm wondering if there is a way to assign a different Caller ID to each Zap interface. I have 3 Digium X100P cards, and I'm sure there must be some way of configuring zapata.conf to allow each line to identify itself with a different Caller ID string. Many thanks, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone

Hi there, I'm having some trouble with getting Asterisk to make a call, I think it should be quite easy, but anyway... Using the following file contents: ## Channel: Zap/3/<TEL NUMBER HERE> MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: phones Extension: 502 Priority: 1 ## Extension 502 is simply one that plays a sound back. When I dump this file into

This strikes me as something that should be really very simple to do, but I can't figure it out. Is there a way of logging all SIP debuging info to a file somewhere? It would help me greatly! Cheers, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

Hi there, Just wondering if there is a way to speak to the person you are transferring a call to before actually connecting them to the incoming call. E.g. "Hi, Colleague, I've got Bill from Microsoft on the line here... putting you through now" Then actually transfer the call. Does that make sense!? -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077

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, 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

Hello there fellow R-users, I have received some data which comes in the following format: example1<-"200301" The first 4 digits correspond to the year and the remaining 2 digits correspond to the week of the year. I have tried to convert this to a date by using strptime as follows: strptime(example1,format="%Y%U") where U (looking up strptime) is the week of the

Hi, I've had an inquiry for a small UK call centre, mostly outbound calls. I get the impression they are mainly calling 3G mobile phones, monthly phone bill, with calls is approx ?5,000 for several feature lines. How feasible is something like this with asterisk? I guess one big question is which type of circuit to use, ADSL in the UK is only 256kbs upstream, some providers do bonding but

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

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, 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?

Hi: I want to connect a wi-fi phone to my Asterisk box through a wi-fi AP so I can make voip calls. please send me your recomendation about what wi-fi phone I should be looking for. Anybody tried the HOP1502 Wi-Fi IP phone. Its listed price $39. Regards; Chawki ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

Good morning :) I want to use some c code to read statistics as well as the configured settings ... .. Just give me some links for starting reading :) Have a nice day ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com