similar to: how to get caller ID

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

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All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Thanks, Chris

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any

Is there a DS# (T3) card that will work with asterisk? OR a card that supports more that 4 T1's per card? Kyle

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Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does Asterisk support

Hi, Has anyone got incoming IAX to work on the above router. I can call out, but incoming calls are not reaching the * box. Has anyone got this working? Could they give me some configuration hints. Thanks John

Hi, I have one Q: why doesn't work h-extension in macro? I want to setup fax macro, to send tif-file after fax channel hang-up, but h-extension doesn't work in macro, so I must put them to global context. But in this setup, exten => h,1,system(/var/lib/asterisk/bin/mailfax ${FAXFILE} ${EMAILADDR} "${CALLERIDNUM} ${CALLERIDNAME}") mailfax binary will be executed after any

Hi All, Is anyone use the sipcall.co.uk 'professional' account with a UK geographic number? What do you think of the service? Alternatively, who else are you using to terminate a UK geographic number on asterisk? Thanks, Nathan. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.529 / Virus Database: 324 - Release Date:

Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious.

Hi Everyone, I'm new to * and I'm trying to setup a small configuration of SIP clients. Eventually when I get this working I plan on expanding with a Digium developers kit to add analog phones and PSTN access. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

Obviously, there are no DS3 TDM cards that are currently compatible with Zap channels. (or are there?) Does anyone know of an inexpensive DS3 card that could perhaps be used with Asterisk if one were to try to port the Zap drivers to such a card? PCI, of course, would be the bus of choice. I think there are quite a few discouraging comments to be made on that question. Firstly, most

Thanks a lot for help in advance. I am switching from matlab to R and I guess I need some time to get rolling. I was wondering why this code : > fit.0 <- lm( Response ~ 1, data = ds3) > step(fit.0,scope=list(upper=~.,lower=~1),data=ds3) Start: AIC= -32.66 Response ~ 1 Call: lm(formula = Response ~ 1, data = ds3) Coefficients: (Intercept) 1.301 is not working

Hi, Can U tell me the Vonage ATA 186 settings? I would like to try to have a web interface on my adapter :-)) Best regards, Chris Hariga