I’ve already installed Asterisk, Zaptel, and XLite and Ekiga softphones on a Fedora Core 6 box. I don’t have any phone adapter cards yet, but I do have an X100P coming from Ebay. I have a number of questions I was hoping you guys could help me with. I’m trying to see if I can get by with just a softphone for what I want to do, hence I went ahead and installed Asterisk and Zaptel without a card (I know…Zaptel is for the card). All I want to do for now is just do some call broadcasting to play pre-recorded messages to recipients on a phone list. I plan on experimenting with other features, but this is my goal for now. Call broadcasting will be a learning curve in and of itself, but I know I read somewhere where you can do this with Asterisk. First I want (and need) to get Asterisk working with the softphones. I’m working through the pdf manual as I go. OK…the questions:
If and when I get the X100P, will I have to uninstall Asterisk and Zaptel and re-install them; Zaptel first to setup the card and then Asterisk; or can I install the card with both packages already on my computer?
2)I’m able to dial 500 on the softphones and hear the demo recording, but when it tries to connect to the misery.digium.com demo site, the recording says it can’t connect to the internet at this time. /Var/Log/Asterisk/message log shows “pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually.” Where do I set this manually and is that the fix for reaching the demo site? My Fedora box connects to my LAN via a wireless adapter. I am behind a gateway server and firewall that’s got Squid’s proxy server running in transparent mode, but I have port 5062 assigned in sip and open in my firewall rules. I chose 5062 because I have a Vonage phone adapter using ports 5060:5061 and Ekiga says 5060 is blocked when I try to use it for the softphones, but it doesn’t seem to have a problem with 5062.
3)According to the pdf manual, I don’t need a phone adapter card to use Asterisk with softphones. Is this the case and will I be able to dial out on a softphone extension and do call broadcasting? When I get the X100P I plan on plugging Vonage’s Motorola phone adapter into the FXO port and my telephone into the FXS port and seeing how it performs. Is this going to be a better way to do call broadcasting, or is it feasible to think that I can use a softphone?
4)I currently have a Windows box and the Fedora with Asterisk box on my home network. I have the XLite softphone on both boxes and the Ekiga softphone on the Fedora box. As I said, I can dial 500 and hear the recording, so I think they’re working and I setup an extension labeled “internal” and used this in a context in the softphone sip configurations, such that dialing 100 from the Windows box rang the Fedora box. Now I also have XLite on a Windows box at work, which is behind a proxy server requiring authentication. Anyone know how to configure XLite in Windows behind a proxy with authentication required to reach my Fedora box at home?
5)I have the Astguiclient application installed also, but I can’t reach my Asterisk setup through my browser when I enter 192.168.x.x:8088. Can someone point me in the right direction for this? Thanks.
No, the firewall is disabled on the Asterisk box, but I do have a gateway server that I’ve opened port 5062 on, which is what I’m using in my sip configuration. The gateway server does have its proxy server running, but it’s in transparent mode. This means that all http traffic is routed through not only the content filter, but the proxy server. I wouldn’t think this would have an effect on sip since we’re not talking http.
By broadcasting I mean calling one phone number after the next and playing an audio file message to each.
I’m pretty certain it’s got two ports; one to plug your PTSN into (FXO) and one to plug your phone into. If the phone port isn’t an FXS, what is it? Just a standard modem phone port?
No, the firewall is disabled on the Asterisk box, but I do have a gateway server that I’ve opened port 5062 on, which is what I’m using in my sip configuration.
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That’s exactly the problem. The 500 demo uses IAX to connect to Digium. (See my update on my own thread shortly after; I fell for this one myself ) Moreover, you’ll also need to open RTP ports. Check iax.conf and rtp.conf. Remember, all are UDP.
I see. This is usually called prerecorded message and not broadcast. Sematics aside, there shouldn’t be any reason why you have to use a Zap channel to play an audio file. (What’s a SIP phone for?)
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I’m pretty certain it’s got two ports; one to plug your PTSN into (FXO) and one to plug your phone into. If the phone port isn’t an FXS, what is it? Just a standard modem phone port?[/quote]
My understanding is you have to have a route in * to send a prerecorded message to a PSTN phone number, hence I was thinking SIP. With SIP you have to have an SIP account with someone like Vonage to do the interfacing for VOIP to PSTN on the recipient end. I’m assuming you can do this with IAX, but that’s just as greek (if not more so) to me as SIP. I was just using the softphone to check the SIP connection out. I’ve got a long way to go getting it all setup because there’s so much to take in. Can you say for sure if I would be able to send prerecorded messages via SIP or IAX, such that I wouldn’t need the X100P card? Although, the advantage to using the X100P is that I could plug my phone line in and use the PSTN and have it cost me no more, whereas through VOIP I would have to pay someone for the link. Thanks for your insight.
So I understand that your final destinations are on PSTN, not your local users. And your messages are low in volume. In this case, signing up with a SIP/IAX provider is a requirement if you don’t use a telephony card. (BTW, Vonage is not a SIP provider so forget them.)
There are many combinations in which you can do this. X100P + POTS line has the advantage of a cheap card. But as you have already learned, you don’t have an FXS. (And the card does weird things beyond possible quality issues.) As you move on, you’ll also learn that POTS line also have severe limitations when it comes to remote messaging. (Search answer supervision, disconnect supervision, etc. and you’ll hit plenty.)
Signing up with a SIP/IAX provider would not require a telephony card, but if you want to use your regular phone, you’ll need to obtain an analogue telephone adapter (ATA). This costs around US $90. As I said, Vonage does not allow you to use their SIP account, so they are as useful as (in fact, less useful than) a POTS line. You’ll need to find a real VoIP provider.
In this approach, the shortcomings of POTS line is eliminated because the provider uses digital lines to the PSTN side.
In addition to the cost of ATA, you also have to consider whether you want to keep your existing POTS line. If yes, that’s another cost.
Another option is to give up your POTS line for a basic rate ISDN (BRI), and buy an BRI card. (Also in the range of $100 I believe.) You pay more than a single POTS line, but it’s digital, and much more reliable than VoIP.
I’m working on installing an X100P in a Fedora Core 6 box. I already have a VOIP account with Vonage and am using their Motorola phone adapter, plugged into my LAN. The call quality is really nice and it winds up being cheaper than my local phone company with all the benny’s. Right now I have the Vonage phone adapter plugged into the FXO port (i.e. line) of the X100P card and a phone in the other port and I still am getting good call quality coming in and going out. Can’t I use this setup to call out prerecorded messages using the X100P card, rather than paying more money for an SIP account?
I see what you mean about Zaptel going off hook and the message playing prematurely. Is there a way you can delay the audio file playing in the following extension:
hoping it would make the call and then play the file, but it doesn’t work. When the call is answered it stays that way until the person hangs up and then never plays back the audio. I’m sure this is the wrong use for Playback. I’m too new at this to know any better. As it stands if I leave the extension as shown in the first line, the message is half way done before anyone answers the call.
You need to use call file to do broadcasting. Generally Dial is designed to establish a live call, and not to interact with the remote. So it’s unsuitable for broadcast. Take a look at voip-info.org/tiki-index.php … o-dial+out would help.
As to dealing with premature playbacks, people use various tricks to cope with them, but none is ideal. If you don’t have to leave message in people’s answering machine/voicemail, things could be much simpler. You could use the “c” channel identifier and tell your (live) users to remember to press “#” after picking up. (c.f. voip-info.org/wiki/index.php … p+channels). Well, asking users to do that may or may not be feasible depending on your environment.
Thanks, Valley. I’m going to mess around with the autodial feature and see if I can get it to work with a Zap channel, although the quality of the connection is deplorable on my X100P, but then I knew that was going to be a distinct possibility. I’ve recently gone through Ekega and Diamond to get some SIP minutes to dial telephone numbers through a soft phone. This way I can DIAL via SIP rather than ZAP. I’ll give autodial a whirl via SIP as well. Is using IAX a better method in the way of VOIP and if so, does it configure much the same way as the SIP and can I buy minutes for that also, meaning do I need a softphone account or something of the likes?
Op3r: I would love to try out VICIDIAL, but I can’t access it in my browser because it won’t let me in. Please reference this post, http://forums.digium.com/viewtopic.php?t=15394&highlight= , to see what I’m talking about. Personally I think it’s because the php files located in my Apache htdocs directory don’t know exactly where to find the MySQL database tables (the ‘asterisk’ table in specific since this is what I created when following the readme files that came with the astguiclient tarball). This is the table that houses the VICIDIAL_user and password values, which I’ve verified are there and the login still won’t work. I have * installed on a fedora 6 box and the only option for installing astguiclient was the install.sh file that came with the tarball I downloaded. There was no configure or make files; just the install.sh. It worked…to an extent. I think the dbconnect.php files are good, and the http.conf file in /etc/asterisk, but still no luck. I’ve since installed asterisk gui (as opposed to astguiclient) and got that working, but it can’t do call broadcasting. Basically it’s just asterisk configuration files in web page form.
Perhaps one of you knows what direction I can take for astguiclient? In the mean time I’ll mess with the autodial feature associated with /var/spool/asterisk/outgoing. Thanks guys.
Yes, you can find IAX providers. If you are talking about voice quality, the two wouldn’t make a difference. The main advantage of IAX is to work behind NAT because it’s a simpler protocol. The main disadvantage is that it’s not widely adopted (partially because it’s less versatile).
) Moreover, you’ll also need to open RTP ports. Check iax.conf and rtp.conf. Remember, all are UDP.