similar to: Question on bridged calls

I apoligize in advance for this newbie question on what I perceive as a mostly advanced level list... I did some searching, but would like some of your expert opinions. I'm building an asterisk server to be used in the home, both to learn, and as proof of concept of applying this solution in a home. To keep costs down, I'm considering one X100P ($25 ebay clone) card to connect the

It seems to me that ITSP's like to use a US dialing code eg 1-xxx Wouldn't it be cool to have an Internet dialing code?? I don't know what the structures are or how the allocations work but it would be so cool to know that 1-xxx was USA , 44-xxx was UK and yy-xxx was an internet phone.. That way the whole internet phone space could be consolidated into a single dialing structure

----- "Chandrakant Solanki" <solanki.chandrakant at gmail.com> escreveu: > Hello > > I have successfully compiled OSLEC for echo cancellation for DAHDI > channel. > > Is there any way to do echo cancellation for SIP Channel. > > Is any, please suggest me.?? > > Thanks in advance.. > > -- > Regards, > > Chandrakant Solanki Short

Hello I'm looking at my options to build a compact, silent, headless Asterisk server to handle one or two FXO ports. Out of curiosity, I got one of those babies on eBay for 20E: http://silicon-verl.de/home/flo/software/netstation-8364/ Before I spend time on this, can someone tell me if... 1. it's easy to install Linux on those things (am willing to add a CF-to-IDE adapter to use a

I have a client that has 10 POTS lines incoming. There is no other option for lines here. I have 3 options I can see: 1. 3 TDM400 cards 2. An external SIP/FXO gateway 3. A T1 card plus a channel bank. Does anyone have any thoughts on these 3 or suggestions on keeping the cost down? -Darren

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

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

I have been able to setup three different providers successfully, but only one at a time. I would like to have all active in a fail over configuration so that one failing would not be noticed by the users. I know it's probably easy to configure but I have not been able to find out how. Can anyone give me an example? Chris Mason

Problem 1 : Incoming conversations from the SIP-provider come into the [default]-context and to the 's'-extension. I am unable to change this, even if I have : sip.conf [general] ;context=default ; Default context for incoming calls register => 092779077:XXXX at 85.119.188.3 ; incoming [092779077] type=user host=85.119.188.3 context=from3starsnet So I define no

Hi All, I am looking for a way to troubleshoot issues with TDM (E1) trunks with a provider. Currently with SIP trunks I am using tcpdump to perform packet captures between our gateways and the SIP providers IPs, capturing traffic on all ports, to include both the SIP messages and the RTP stream. How can I achieve a similar result on TDM links connected to TDM cards in Asterisk servers, where by

I wrote a week or two ago about a problem I was having with a TDM400P with 2 FXS. The problem was reliable fax transfers. There were often slow or had communication failures during transmission. Richard replied to my post and told me this is an ongoing problem that might have a solution in the weeks to come. However, I was wondering if anyone had any alternate solutions for faxing with an

All, I use various IAX providers to terminate my outbound calls. I set the caller-ID to one of several DIDs, based on the called number. There doesn't seem to be any rhyme or reason as to what the called user sees, however. Calls to most cell-phones show *exactly* the number I submit. Calls to land-lines sometimes show, other times not. Calls to other voip-providers most often show, but

I received an e-mail from a vendor who says: "We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP." I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send

With the recent service outage at Broadvoice, there has been a lot of discussion here, on broadband reports, Voxilla, etc., regarding whether VOIP is mature, or "ready for the masses", etc. One particular point I've seen repeated, and with which I agree: "we're willing to deal with less than five 9s, even one or 2 9s, as long as we have good communication regarding the

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

FYI to the list.... I gave up (for now) trying to make spandsp work with the digium TDM card. Instead, I signed up with www.trustfax.com at a cost of $9.95 per year plus $.10/page. Since we only deal with an estimated 120 pages per year, the total cost of about $22/year seemed like a very reasonable alternative. (At least until we can find out why the TDM card does not function properly with

We are developing an IVR application and when I am testing locally on my machine using a softphone (iaxcomm) the digits I press for GET DATA work every time. I am testing with a local extension that goes right into my routine. However when I try to call in to the system using an analog or cell phone GET DATA drops some digits that are pressed. There doesn't seem to be a pattern to which

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on