similar to: Re: asterisk-users Digest, Vol 32, Issue 106

Hello all, is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Thanks. Lito -------------- next part

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. dbc.

I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxxxxxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it "forget" what it has programmed and

It looks like this meetup group is becoming the venue for contact as most "Toronto" respondents have signed up here. Can I request that the remainder who responded to the "Toronto" call signup? Shidan is the organizer and has proposed a date. I'd hate to confirm it without the remainder having input - especially those with travel times like Andrew K. out in Listowel.

This might be a newbie question but I'm just wondering how would it be possible to have 30 analog lines using asterisk for PBX by just using TDM40B and X100P (or are there any device>), if an ordinary PC support just 4 PCI slots? the maximum scale i guess would just be 2 x 8. Adding a new PC just for this purpose would be costly. I would appreciate your comments. Thanks.

I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Loadid: SW: P0S3-06-3-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 Here is the section dealing with time in my SIPDefault.cnf file. Does anybody see anything wrong with it or have any other ideas? # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)

Asterisk as a phone system makes perfect sense in a condo. You can get all the DID's you want and eliminate costs for the owners. You can offer standard FXO for people who don't care and IP sets for people who want to "upgrade" to feature sets. Your door openner is a piece of cake. 1. Create an option in your dialplan only in the "from-access-door" context that reads

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

If you are having trouble reading this email, read the online version<http://now.eloqua.com/es.asp?s=491&e=78675&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> . <http://app.en25.com/e/er.aspx?s=491&lid=215&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

Hello, First, thanks for releasing Mocha. As someone who''s been practicing TDD much longer than I''ve been using Rails, one of my biggest complaints has always been the fact that the "Unit" tests are more like intefration tests because of how tightly coupled the domain model is when using ActiveRecord. Stubba looks like it could really help aleviate this problem. The

Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour

Did the console "Dial" command go away in recent releases? I just went to use is on my server (now at 1.0) and its gone. Help from the CLI doen't show it. -- David Cook

I'm looking to give the SPA-3000 a whirl as I'm having too much difficulty with the irq sharing thing inside the box. I'm reading the book but without having one in-hand to play with it appears a little obtuse at this time. Before I drop down my money can someone with some hands-on with one of these confirm if the SPA-3000 can: a) detect inbound distinctive ring (this looks to me like

2017-04-21 1:08 GMT-03:00 Epsilon Minus <theepsilonminus at gmail.com>: > 2017-04-21 0:38 GMT-03:00 Epsilon Minus <theepsilonminus at gmail.com>: >> >> >> I'm desperate. My English is terrible, but I will try to be as clear as possible. >> >> I have Samba 4.4.4 in a Centos 7. >> >> I have many connections for each ip. >> >>

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

Dear shorewall users: I am going to implement VPN IPSEC pass through. Got shorewall as the main firewall interfacing the internet. I found Tom''s documentation on how to do it. The question is: there''s an argument that VPN pass through (pass through NAT) will undermine the importance of AH (source: freeswan site). Is it true? ------------------------ Lito Kusnadi

Hi, I have a rule: DNAT net dmz:a.b.c.d tcp 25 - k.l.m.n The problem: I want to DNAT port 25, 143, 110 k.l.m.n is alternate public ip (using vrrp, just like alias) Can I abridge the above line using macro, instead of writing 3 separate lines? I can do: MailPorts/DNAT net dmz:a.b.c.d But this refer to the physical public ip I have more elaborate requirement to forward around 20