similar to: which ATA SIP is better with asterisk

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Whey you 'destroy' a Sip extension you are only removing the entrys that allow you to make and receive the auth needed to do so. Your voicemail files are not tied to an extension but are independent and are only 'married' when you specify it in your sip.conf or other channel configs. Removing a confi from a channel will not touch voicemail. You need to go into the voicemail.conf

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

Hi all, I've a some users on my network, reporting this: Sjphone is registered , and some times just looses registry in Asterisk, I don't know if it is expiration ( instead of loosing registry). Then to get registered again they need to restart their own PC. Why could this beeing happening? Best regards, Marco Mouta

Is it possible to set up the following? public IP Asterisk Server to ata 186 behind a XP server firewall. I think my biggest problem is that I don't know how to make XP forward the RTP port to the private ata address. I would put up some configs, but was hoping that someone one who has this working can share the working configurations incase mine are all messed up (which is likely.)

Hello, Currenly I've ASterik@Home 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: "sip

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any

HI all, My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Best regards, Marco Mouta

Hi, I must say that i'm not very used with customization of FOP. I've a box runing Flash Op.Panel, and i notice that the screen is full of buttons from my sip users, as well as Zapata channels. The problem is that i have more Zapata channels as well as SIP users, is there any way to get a scroll on this to display everything? do i need to resize the buttons? For sure someone now how to

Hi all, I'm using Asterisk@home 2.5, and i've done: 1-Create a SIP extension. 2-Leave there a Voicemail message 3-Remove SIP extension Then I've create another SIP extension but with the same number of the above one. I found imediately a voicemail message in my voicemail box. Is this a bug? Am I doing something wrong? Best regards, Marco Mouta

Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta

Hi all, I'm planning to connect 2 office from one company. I'm the developer, so i hope i can get all the features working well. Ast@Home(Portugal)---------IAX2/VPN--------Ast@Home(Brazil) 1- First i'm integrating Asterisk in Portugal's company office, one Asterisk@Home with TE110P connecting to an old PBX. (the same is done in Brazil, but only VoIP no TE110P) For *@Home PCs:

Dear all, I've a live system that needs to be upgraded but, before I proceed to the upgrade I want to assure the rollback process. That's why I'm requesting your feedback, in fact this asterisk in live system isn't going so bad but.... the upgrade is essential NOTICE that the upgrade will keep the same version 1.2 not from 1.2 to 1.4 Requirements: -backup /usr/sbin/asterisk

Hi all, I would like to hear from you, SjPhone has the option to Do not Send silence (audio options, advanced), should i use this or remove this option? Everything ran well until now, but there was few people on my server, i'm increasing sip extensions and i want to avoid complains from the users:) Best regards, Marco Mouta

Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on

Hi guys, I'm trying to reuse Mitel 5215 from proprietary system now into Asterisk :) ! I've them already with SIP and handling calls sucessfully! I've followed instructions from: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Mitel+5220 Additional Servers * Outbound Server: Off * Outbound Server URL: blank * Outbound Server Port: blank * Voice Mail

Boa tarde, Ap?s alguma experi?ncia com o Asterisk, e com muito ainda para aprender, gostaria de saber se h? algu?m nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed...

I must be doing something incorrectly, or something is wrong with ATA-186 reINVITEs in SIP. Perhaps someone more enlightened than me can lend me a hand. I have been attempting to get two SIP phones to reINVITE to each other, and I've been unable to think of or uncover the correct method. The calls always go through the Asterisk server, no matter what I try. I've simplified things