Re: [Asterisk-Users] Vonage Linksys Router - Life after Vonage
On Wed, 23 Mar 2005, GP wrote: I've read that someone was able to do it by contacting vonage and getting instructions for clearing the router of the vonage information. Does anyone have the instructions for completing this or is this something that only the vonage people can provide. I've spent the last 2 hours and 30 minutes in the vonage phone system trying to get to tech support to find out how to do this without any luck. Their phone system and indeed their customer support in general is really lacking. on the Cisco ATA-186s which they used to use (before Cisco EOL'ed them) Vonage had them configured to use a (64-bit?) RC4 key to encrypt the configuration and programmed a random password to each device. Every time an account change prompted an update to the ATA's configuration, it would be encrypted with a new RC4 key and get a new password. So don't expect to find a universal back-door password, I doubt you'll find one. I too have heard of people persuading a vonage tech to provide the password to log into and factory reset their device, but I get the impression that it is an uncommon occurrence.. you'd be lucky, basically. (On top of which, they charged me a $40 termination fee to terminate my account - just a parting shot I guess). yup.. That right there was the primary reason why I skipped over them in the beginning. It's buried in their terms of service somewhere, along with all their other over-restrictive junk.. If anyone has any suggestions, or the instructions for clearing the MTA on this linksys router, I'd be very greatful. you could try faking out the box by setting up some DNS and network spoofing to make the box think it is talking to vonage when in fact it is talking to your * box. If you use autocreatepeer in sip.conf then * will accept connections from the PAP2 even though you haven't added the credentials in sip.conf. This is quite a kludge, however.. When it comes down to it, I think Vonage thinks it is better to make people throw equipment away rather than allow it to be reused. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How NuFone.Net's customer service works.
On Mon, 14 Mar 2005, nik martin wrote: > Matt Riddell wrote: > >> Hmmm...I've had 2 problem with my NuFone service in the year or more > >> I've used them. Each time I've treated them professionally when > >> reporting the issue and received the same treatment in return. The > >> issues were also resolved promptly. > > > > > > :) > > > > I've had no problems and hence no need for support. > > > I've had problems, and general usage issues, and have gotten very poor > support, on several occaisions. I'm always surprised by how many people claim to use NuFone.. I've tried, on more than one occasion, to contact them both by phone and email. After waiting on hold for a while, their phone system offered to let me leave a message and somebody would contact me as soon as possible. I did so, but never heard from them. Not in an hour, not the next day, not even within a week. Never. Peculiar sales strategy, to say the least. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
On Sun, 13 Mar 2005, Jess Coburn wrote: > So you basically want an SMS or IM callback app right? > > One way to do this would be send an email to an address like > ([EMAIL PROTECTED]) and have a cronjob query/pop this email > address for your specific message and then when it finds it have it > create a .call file to call you and connect you to whatever context > you setup, etc. Shouldn't be very hard at all. This would allow for > emails and I'm pretty sure every cell can send an email. As an extension to that idea, if your asterisk host is also your email host, the you could send mail to whatever user Asterisk runs as. Set up a .forward file and use procmail etc to filter the message and "authenticate" or validate the source (remembering that email is easily spoofed, so don't let this gateway have access to anything terribly sensitive), then create a .call file from there. This approach is nice because it's asynchronous: as soon as the email arrives it'll be processed and .call will be generated. The cron approach means that you'd have to wait until the next job runs before any action will be taken, although if your email doesn't get delivered in such a way that you can do scripting in .forward, then it may be the only choice. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Multiple "lines" {Scanned}
On Fri, 11 Mar 2005, David Shaw wrote: > I also have multiple line with Broadvoice. I would like to have each > incoming line ring a different extension and configure an internal user > to use his or her own broadvoice line.. For dialing out, Dial(SIP/[EMAIL PROTECTED]) should send it through account X. For incoming, try setting each account to use the same context (sip.conf). Then append /exten-no to the register strings ( /111, /112, whatever). When a call comes in, it should start in extensions.conf in the context you specified at the extension after the /, rather than at the default s extension. That should make it easy enough to craft a dialplan which routes calls based on which account/number they arrived through. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage a provider?
On Fri, 11 Mar 2005, Frank Abernathy wrote: > I am new to the mailing list, but I am very interested in running my small > home business office phone system using Asterisk. However, Broadvoice, a > VoIP provider of choice based on my research, is not available in my area. > > I currently use Vonage VoIP. Their website mentions nothing about being > able to link to Asterisk. I was wondering if any US subscribers have been > able to configure Vonage with Asterisk. Or if anyone has found Vonage to be > a non-compatible provider. The only known (to me) way to connect Asterisk and vonage is to buy their normal service using their provided terminal adapter, and then connect that to a card in the Asterisk box, or to add on a "softphone" account and feed those credentials to Asterisk so that it can connect with their servers directly. If you go to google and search for asterisk vonage site:lists.digium.com you'll find references to several sample configurations. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT Far End Traversal
On Wed, 9 Mar 2005, Michael Graves wrote: > On Wed, 09 Mar 2005 12:54:34 +0400, Jean-Michel Hiver wrote: > > >Leo Ann Boon wrote: > > > >> > >>> > >>> Another question... Are you aware of a SIP ATA or phone that has some > >>> kind of VPN (i.e. PPTP) client embedded in? This would make the NAT > >>> problem go away nicely and provide added security... > >> > >> > >> The Zulty's phones support VPN. Then again, many firewalls don't pass > >> through VPN traffic nicely. Would be cool if we can have a phone that > >> supports SSL VPNs like OpenVPN. > > > >Agreed. In my experience, OpenVPN is a breeze to work with. > > Forgive my lack of depth in this area, but aren't SSL based VPNs > fundamentally IP centric? Whereas RTP & IAX2 streams are UDP? I had > read some time ago about a company that was planning to revolutionize > voip through SSL based VPNs, they met with much scorn from those who I > thought were knowledgable people. TCP. IP carries both tcp and udp protocols. RTP, IAX, and many other media-type services use udp. This is because udp doesn't use acknowledgements for every packet, which means some may get lost and the sender may never know. With media streams, especially live streams, by the time you notice a packet got lost it's already too late to try to recover it. Life goes on and the codec deals with the loss gracefully (or not, depending). TCP is used by SSL and other data-oriented services where the bits received absolutely must match those sent, and must also be complete. TCP provides for acknowledging that packets have been received and re-transmitting those that got lost. So, except for the minor slip of using the wrong acronym, you've got the right idea. Putting media streams through tcp is probably not a good idea unless you can somehow guarantee that the link will have _extremely_ low lost or out-of-order packet rates. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk box and verizon calling it
On Fri, 4 Mar 2005, Randy Johnson wrote: > I set up an asterisk box with a broadvoice sip connection for incoming > connections > > it works great when I use a cell phone, vonage line, calling card to > call the asterisk box, but when I try to call it from our verizon land > line it is busy and asterisk logs do not show incoming call. > > Any ideas on what the issue is? I'm putting my bet on a misconfigured switch. I had a similar problem when I tried to use VoiceGlo over a year ago: They assigned me a number, and I could reach that number via 3 of the local cell carriers, through a calling card, and from several COs in the area. But calls from at least one Qwest CO couldn't complete and only gave a busy tone. I talked to some Qwest techs who determined that it must be a problem in the configuration from when that number was ported to VoiceGlo's provider. Unfortunately, I never could get them to fix it (or even recognize that a problem existed) and had to drop their service. Your issue sounds similar to the one I had. Try originating calls from a few other "places" like other cell carriers in your area and from landlines which are serviced by different COs than the one you've already tested. Then call BV's support and see what they think of the results of your research. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DISA and a long delay; ideas?
On Sun, 27 Feb 2005, C. Tomlinson wrote:
> I have just setup a DISA setup whereby people can dial in, authenticate, are
> given a dialtone and can then call out.
>
> Everything works however there is a 10 second delay after the user enters
> the number and presses #, until the system does anything.
>
> Here is the relevant section from my extensions.conf:
>
> [dialtone]
> exten => s,1,Authenticate(1234)
> exten => s,2,DISA(no-password|dialtone_outgoing)
>
> [dialtone_outgoing]
> exten => _01.,1,Dial(${OUTGOING}/44${EXTEN:1},30,L(6:3:1))
> exten => _07.,1,Playback(pbx-invalid)
> HOWEVER there is a 10 second delay between the dialing (followed by #) and
> the system doing anything.
My first guess would be digit timeouts. Your patterns are _01. and _07..
These don't give asterisk any hints about how many digits to expect, so
its only choice is to wait for the maximum digit timeout period to be sure
that it doesn't make a decision early before you've entered all your
digits.
The "best" thing (in my view) would be to completely specify the digit
patterns you want users to be able to use. This gives you the opportunity
to control which numbers may be called and which may not, and it also
gives asterisk hints about what kinds of digit patterns it should expect.
These hints allow it to make faster decisions about whether a digit
pattern is complete and/or valid. An alternative would be to use the
DigitTimeout application to set a lower timeout period.
Greg
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Re: [Asterisk-Users] Asterisk With Broadvoice
On Fri, 25 Feb 2005, James Taylor wrote: > I have two Broadvoice "lines" and there's three people in the office. > Any way to: > > 1) "Pool" the connections for "trunking", where any one can get a "free" > line? > 2) Prevent more than 1 simultaneous call per "line"? (So I will not get > hit for 3.9 cents a minute. have a look at ChanIsAvail, SetGroup, and CheckGroup. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXY DNS possibilities??
On Thu, 24 Feb 2005, John Bohman wrote: > Is there any way to hard code an ip in the IAXY then re-direct that ip > to a dynamic ip via dns.. Some sort of ip forewarding...?? I'm don't think I understand exactly what you're trying to accomplish. Maybe a good place to start would be the DNS howto (ibiblio.org) to get a good understanding of how DNS operates and how the mapping between IPs and names works. That'll probably help clear up what you're trying to accomplish, and will probably also help you to figure out the answer to your question. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura to dial extension automatically
On Thu, 17 Feb 2005, Oswaldo Arratia wrote: > Has anyone figured out how to make a Sipura to dial an extension > automatically as soon as you pick the the handset? > > I need to make all my users go thorugh a menu to place a call. Users should > not be able to dial directly, only through the menu. You can get the manual for a Sipura from their web site. If you read it, specifically the section on the dial plan, you'll find that you can use pattern substitution with a zero delay to effect a "hotline" function. You could even search the pdf for that word (hotline) and that should get you to the right page quickly. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help With Broadvoice
On Tue, 15 Feb 2005, Max Clark wrote: > I have experimented with several configs based on different pages and > threads but nothing is working. How do I properly configure my > broadvoice account? > > [general] > register => [EMAIL PROTECTED]::[EMAIL PROTECTED] the register I'm using looks like this: register => 310584:@sip.broadvoice.com > [broadvoice] > type=peer > host=sip.broadvoice.com > secret= > fromuser=310584 > fromdomain=sip.broadvoice.com > context=incoming > dtmfmode=inband > canreinvite=no > nat=yes > qualify=yes try: [broadvoice] type=peer username=310584 secret= host=sip.broadvoice.com port=5060 context=incoming fromuser=310584 fromdomain=sip.broadvoice.com dtmfmode=inband canreinvite=no insecure=very permit=147.135.8.128/32 qualify=yes and adjust your permit= line to match the IP of the BV proxy you've set in your /etc/hosts (proxy.[chi|lax|dca|one other I forgot].broadvoice.com). Try a blend of this stuff with whatever the most recent recommendation on their support page says. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice international dialling question
On Sun, 13 Feb 2005, Malcolm Taylor wrote:
> I'd be grateful if someone could point me in the right direction.
>
> I have a Broadvoice trunk attached to Asterisk which I use for frequent
> calls to the UK using the following in extensions.conf
>
>
> exten => _0[1-68].,1,Ringing
> exten => _0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1})
> exten => _0[1-68].,3,Hangup
>
> The caller hears immediate ringing, though it seems that Broadvoice takes a
> long time to make the international connection and sometimes fails
> altogether
This is because you've told Asterisk to play a ringing sound before it has
even attempted to place the call with BV. Take out your Ringing line and
that behavior should stop.
Greg
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Re: [Asterisk-Users] list administrator.....???
On Tue, 1 Feb 2005, Randall Shimizu wrote: > I keep recieving multiple digests per day. Need to find out if there is > a way to limit the number digests that are being sent to me. Tried > contacting the list administrator, but I have not recieved a response. > Does anyone have a alternate email for him...??? heheh.. I'm not sure we have an administrator. The list was created in a big bang and goes on by itself now.. :) My guess at what's happening (entirely supposition) is that there may be a limit to the digest size. Once the daily traffic exceeds that limit, a digest is kicked out immediately and the counter restarts on a second (or third, etc) for the day. ..so can anybody confirm the guess? If the first n-1 digests of the day are roughly the same size, that might support the theory. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BroadVoice Troubles
I had a little billing problem once.. my credit card on file had expired, and it took a bit of work to finally find somebody there who could get their system to acknowledge my updated card number and quit sending me "pay up or be disconnected!!" warnings. My recommendation: call them on the phone. Keep calling until you get a live body. Tell the person (be kind the first time, at least) that you have an unresolved billing problem and request to speak with a manager or supervisor about it. You'll probably be able to tell when you're speaking to a rep whether they're the type who is empowered enough to fix the problem. Oh, as for calling: if you don't want to incur any long distance charges by dialing their regular number (978 area code?) dial 611 on your broadvoice line instead. When all else fails, call the credit card company and ask them to send a chargeback letter. In my dealings with another VOIP company I was amazed by how quickly this strategy got their attention -- and THEY called ME. :) Greg On Tue, 11 Jan 2005 [EMAIL PROTECTED] wrote: > > >My question is simply, has anyone received a deposit from these people once > >you return the equipment in good order? I've been unable to contact them now > >for almost 2 whole months. > > > >Thanks, > >Bill Church > >[EMAIL PROTECTED] > > > > > Bill, > Although this is quite OT, I'll reply. > I signed up for their service under the BYOD (bring your own device) > plan, which clearly > states that there is NO disconnect fee when you cancel. After a great > deal of grief, I canceled > per their rules (e-mail only) and waited almost two months before they > finally acknowledged the > cancellation. Of course, they were happy to charge my credit card > during that period of time, and > then they charged me a disconnect fee. It took them two weeks to reply > to my complaint, telling me > they would get back with me. > That was a week ago, and still no reply or refund. > > Niles > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BroadVoice
Did you try BroadVoice's site yet? www.broadvoice.com, click Support, click Installation, click Asterisk, follow instructions there. Greg On Tue, 11 Jan 2005, Vitalie Apostu wrote: > Following links says: HTTP 404 - File not found . Is it a right link > http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel > Jafferali > Sent: Tuesday, January 11, 2005 11:09 AM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: RE: [Asterisk-Users] BroadVoice > > > Can you give me example of sip.conf and extention.conf which work with > > broadvoice? I want users who registered with Messenger through sip to > > be able to make a call thought broadvoice. > > I posted this just a few days ago: > http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm > l > > -- > Nabeel Jafferali > Tel: +1 (416) 628-9342 Toronto > +1 (646) 225-7426 New York > FWD: 46990 > Email/MSN: nabeeljafferali.net > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Glophone/Voiceglo and Asterisk
On Thu, 6 Jan 2005, John Voss wrote: > Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this > posting. > > http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html > > I've tried copying the config in this listing with no success. > > One thing that I have noticed is that all the listings that I have found > mention the use > of 10 digit numbers. They now give you 14 digit numbers which shouldn't > matter. However, > it does make me wonder if anything else has changed. > > Any help anyone can supply will be greatly appreciated. 14 digit numbers..? I could imagine 13, with 011 prepended to the numbers.. hmm. The config in the post you reference looks similar to the one that I used, which is (approximately): [voiceglo] type=peer username=801203 secret=NEERHFD ;nat=yes host=myphone.voiceglo.com disallow=all ;disallow=g729 allow=ulaw ;allow=alaw ;allow=gsm ;allow=g729 canreinvite=no ;qualify=400 restrictid=no fromdomain=myphone.voiceglo.com dtmfmode=inband I did have it working at one point, however, I didn't (still don't) have the g729 codec for my asterisk. I could only place calls through Voiceglo by using their bundled SJ Labs software (which did include g729) and setting it to register through my *. This way * never needed to listen to the RTP stream anyway and could just pass it through. At the time, g729 was the only codec you could use. And they also used inband DTMF -- a very bad combination. I cancelled my service after they failed to correct (or even recognize) a significant problem: the DID they assigned me was provisioned incorrectly (routing config problem, evidently) and could not be reached from at least one local (to me) ILEC exchange. In fact, they didn't even recognize my (repeated) requests to cancel the account. Funny thing was, when I asked my credit card company to chargeback Voiceglo, I got a call within just a few days from a Voiceglo rep, who acted surprised to have received a chargeback and wanted to know why I hadn't contacted them first to see if we couldn't resolve any problems. I nearly hung up on her. Maybe they've done some hiring and firing since then and run a better shop now. This stuff is all I know about them, and it's nearly a year out of date. Anyway, if there is anybody else who can provide service in the area you need, I think I might recommend going that route instead. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BroadVoice WiSIP with Asterisk
On Fri, 31 Dec 2004, Adi Linden wrote: > BroadVoice sells a wireless SIP phone for $149. Does this phone as sold > by BroadVoice work with Asterisk or is it a locked down device like the > Vonages ATA186? You'd probably have to ask them that. Just so you know, you can buy that phone elsewhere. It is made by Pulver Innovations (www.pulverinnovations.com). The fact that it lists at $199 on Pulver's site suggests that it would probably be tethered to the BroadVoice service, or at the very least you can count on paying the difference in the disconnect fee whenever you close your account at BroadVoice (they don't chage a disconnect fee if you brought your own device instead of buying a discounted model from them). Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do I need to build up DID services?
On Sat, 25 Dec 2004, Ronald Wiplinger wrote: > To complete my project, I would like to setup DIDs in several areas. > > What do I need to do that? Another Asterisk box or can I use gateways > instead? Which hardware can I use? Who has experience? You either set up your own points of presence, or buy service from somebody who already has them set up. So you could install an * with a zap card, a stand-alone spa-3000, or some other device to establish the POP where you want it (and you'll need broadband there too), or you could just buy the service from a VOIP carrier. The latter, IMHO, is better because you won't have to deploy hardware all over. Deployed hardware requires a place to live, somebody to feed it and provide TLC whenever necessary. Usually it's better to keep hardware in as few places as possible to simplify that task.. If you don't like the VOIP delivery, telcos for years have offered "market expansion" lines which give you a number in a remote rate center which automatically forwards to your real number. (funny thing is, when I talked to Qwest about that a month ago, this forwarding service cost more than regular residential phone service!) Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting up asterisk for one user in private ip NAT.
On Sat, 18 Dec 2004, Anders F Eriksson wrote: > I've never tried softphones on Linux, but my guess is that since you run > kphone and asterisk on the same server you get a port conflict. If the > client uses port 5060 (default sip port) it would defenitely have > problem connecting to an asterisk on the same port. that isn't quite how ports work.. True, asterisk listens for udp connections on 5060. But the softphone won't make its outgoing connection on 5060. The OS will automatically choose an unused port number for the outgoing connection. So (for example) you might have the softphone talking on port 23107 to asterisk on 5060, and asterisk on 5060 talking back to the softphone on 23107. No port conflict. Now one place where you could have conflict is if asterisk is trying to use your soundcard for its console. Then the softphone client may have trouble getting the soundcard port opened. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio levels via sip
On Sat, 18 Dec 2004, Doug Langley wrote: > I see from reading the mailing list theres a way to set audio levels on > the zap channels but I'm wondering if there's a way to set audio levels > on either sip or iax channels. I'm using some BT-100's and people are > saying the audio levels are a little low and I would like to bring them > up a bit. audio settings are most likely to be done at the device which does the A/D conversion.. which is why Asterisk allows you to configure it for a zap channel (x100p etc anyway.. don't know about the TDM cards). But for a SIP/IAX/H323/etc channel, Asterisk is just something of a gateway or proxy. The A/D happens in your handset or softphone, so that's where you'd have to adjust microphone gain. This would be especially true in a reinvite situation, where the rtp (audio) stream goes directly between the endpoints -- Asterisk doesn't even get to see the audio packets! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NPA NXX data
On Fri, 17 Dec 2004, Jon Bebeau wrote: > HI all - I know, slightly off list, but.. I'm looking for a NPA NXX > database with City and State. Actually it's for an Asterisk routing app > I'm working on. I see several vendors that want a few bucks to those > that want an arm and leg. I expect this is published somewhere by some > government agency, but Google hasn't got me to it yet. I dug up this message from the list archives back in September. Hopefully it'll help you find the info you're after: Date: Thu, 23 Sep 2004 18:25:01 -0400 From: Brian Rozmierski <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Billing Fun - anybody know where to get aNPA/NXX db? You might also want to try http://www.telcodata.us. Don't hit his servers hard, tho. If you do need bulk access to the data he has been quite helpful in the past. -- Brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Suffill Sent: Thursday, September 23, 2004 5:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Billing Fun - anybody know where to get aNPA/NXX db? There used to be an NPA NXX sql on 1 of the asterisk site's. http://www.fnords.org/~eric/asterisk/ I doubt you will find a nice complete 1 for free unless you parse the npana data yourself which you could do. I did it recently not exactly fun. Still might not be 100% though. -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Optimizing Sipura/Asterisk for DTMF?
On Fri, 17 Dec 2004, Brent Goran wrote: > We have an application which is primarily DTMF driven (automated on both > sides), which we are trying to deploy over VOIP and Asterisk (using some > Sipuras and some IAXY's). > > We are finding that in around half the cases, the Asterisk server can't > decode the DTMF digits from the field office (or at least some of them). > Though, when we place voice calls for testing, we can hear eachother > quite well. > > I was wondering if there are any settings in Asterisk and/or in SIP > clients such as the Sipuras, which will optimize the connections for > DTMF rather than voice? Depends on whether you're using dtmfmode= inband, rfc2833, or info. Inband is sent, of course, "in-band" (as audio). g711 is the first step toward success for inband DTMF. Evidently the other codecs work okay for voice, but really make a mess of DTMF to the point where it's not likely to even work. I believe the other two dtmfmode's actually send DTMF as SIP notify messages, so the codec would be irrelevant and the DTMF should always arrive fine.. so long as the endpoints do a good job of detecting and regenerating the tones. One thing which some have found helps with picky IVRs is to set the endpoints to use dtmfmode=inband and set Asterisk to use dtmfmode=rfc2833 (or use the sipsetdtmfmode app, or whatever its name is, in the exten.conf to change the setting on a per-call basis). This way Asterisk doesn't notice the DTMF tones being passed inband and doesn't try to intercept and regenerate them. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Total newbie here looking to do a VoIPconfer ence call?
On Fri, 17 Dec 2004, Patrick Campbell wrote: > Come to think of it since the DTA310 uses DNS to find the SIP server, > you could setup a DNS cache and override the DNS entry for what packet8 > uses (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP > of your own SIP server? Kind of a hack but it should work as long as > it's running on port 15062. I am very new to this so I don't know if > there's a port standard for SIP like there is for HTTP, SSH, FTP, etc.? once upon a time I was tinkering with a vonage-locked ATA186. The first thing I did was connect it to a linux box and run ethereal. When I powered up the unit, I saw that it made DHCP requests. So I set up a DHCP server on the linux box. Next it made DNS requests for a few vonage hosts. So I set up a DNS server, and configured it so to reply that I am the vonage host the 186 wanted to speak with. It's an interesting excercise to configure a spoofed network to make some locked device do what you really want it to do. Unfortunately, the thing melted down (literally) before I finished my testing with it. I was going to see if I could make calls to/from that box into Asterisk by using the autocreatepeer option in sip.conf.. oh well. Maybe this serves as a little food for thought for you. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Total newbie here looking to do a VoIP conference call?
On Fri, 17 Dec 2004, Patrick Campbell wrote: > I don't have a great grasp as to what Asterick is capable of, but my > thoughts were that perhaps with VoIP telephone lines (either hooked up > to the company's network or just using a 3rd party VoIP provider such as > Packet8, which is whatI have for personal use) and an Asterick server, > that we could setup a VoIP conference bridge. it's spelled asteriSk. :) > Can someone enlighten an unknowledged as to whether or not this is > possible, and if so, how might it be done? Would the Asterick server > need X number of VoIP lines? I.e. If there's 10 participants, it'd need > 10 VoIP lines? I've only played with the meetme conferences feature in Asterisk a tiny amount, but it wasn't terribly difficult to set up. You can start playing with it with a few softphone clients. The only cost would be your time to tinker with it. (you'll need a timing source on the asterisk machine. This can come from any (?) Digium card or it can be derived in software from some onboard USB chipsets, and maybe other sources. I haven't kept up on it for a while. Search the wiki for timing sources and read more.) If all the participants have sufficient bandwidth to run a voip call, and you can locate the asterisk server someplace which has enough bandwidth to handle everybody together (it's not going to multicast; every participant will receive a separate audio stream), and if you're pleased wth the function of the asterisk conference room feature, then you should be able to get it up fairly easily. You won't even need to buy any voip service from anybody, because those are typically to bridge voip to the PSTN. If everybody who needs to be in the conference already has IP bandwidth, then just keep the whole conversation as voice over IP and leave the PSTN out of the loop. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: DSL without voice
On Fri, 17 Dec 2004, Michael Graves wrote: > On Fri, 17 Dec 2004 09:42:28 -0600 (CST), Joe Greco wrote: > > >> www.Covad.com > >> > >> I have their TeleSoho dedicated loop DSL. It costs the same as the > >> bundled loop. > > > >ADSL or SDSL? (I haven't looked at Covad's pricey offerings for a while) > > > > ADSL > > 3.0 Mb down / 768k up. $99/mo. The dedicated loop service requires a > "professional" installation that costs $175 (I think) incidentally, Speakeasy.net resells Covad's service. They're charging about this price for the 6 MB/768k level with 8 IPs etc. I found some attractive offers through an ad on slashdot, visit www.speakeasy.net/promos/osdn for more info. > I was having trouble with the bundled DSL dropping when my home POTS > line rang. SBC and Covad were hopeless at diagnosing this, and the > unbundled service was available so I simply switched. SBC droppped a > clean, new pair to the house. Covad's tech did his install in less than > 10 minutes. it's kind of annoying that they call this a "professional install" when it typically consists of little more than plugging in the DSL adapter and making sure that it trains properly. And especially annoying to have to pay $175 for the job! > The also told me that I had to buy their DSL mode/wireless router > combo. I did, but the cost was rebated. Then I put my trusty Siemens > Speedstream/m0n0wall combination back in the line ;-) nice work..! Sometimes you just have to work "around" the system. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: DSL without voice
On Fri, 17 Dec 2004, Joe Greco wrote: > > Joe Greco schrieb: > > > > > Don't forget, you ought to have a conventional phone line for E911 > > > purposes, including "what happens when a hurricane goes through and my ISP > > > becomes toast". VoIP is a neat technology but it lacks the resiliency of > > > the traditional phone system. > > > > For this you can take your mobile. When my local company (T-Com) decides > > to allow ADSL without a phone line I will take it. I've got my mobile > > for cases of emergency. > > > > And since in germany there is really no danger of a hurricane the > > stability of the mobile nets should be sufficient. ;-) > > > I do think the thing that worries me about this trend is the unexpected > scenario. Right now, we have a fairly high quality E911 system (dunno > about where you are) and people expect that they can dial "911" and the > right things happen. > > So what if you've got some friends visiting your house and you have a heart > attack and no 911 on your POTS-via-VoIP? Are they expected to know your > cell phone's unlock code? Are they required to bring their own cells as a > prerequisite for visiting? Or is it acceptable for them to have to go > finding a neighbor who has a usable POTS phone? This random thought just popped into my head: Seems like I've read that any cell handset will place a 911 call, regardless of whether it is associated with a valid and paid-up account. Is that true? If so, then maybe we could just attach GSM interfaces to our asterisk box to provide communications in the unlikely emergency (so long as the LAN and * box have power to operate, that is). Whaddaya think? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with broadvoice outbound plz... ;)
On Fri, 26 Nov 2004 [EMAIL PROTECTED] wrote:
> *sigh*
>
> Ok, I have fought and fought with this. I have read all the FAQ's, I have
> scanned the list archives. I can receive calls on * from my Broadvoice
> acct, but I cannot place calls...
>
> I have the 'World Unlimited' plan, and like 5 area codes that are local to
> me in Dallas.
>
> Can anyone help me?
>
> here are my config files...
A sip debug would be helpful, too.. ("sip debug" / "sip no debug") With
this turned on, you'll be able to watch the dialog between * and BV when
you try to place a call. This may help you identify where things are
breaking down.. (or, if you can't find it, then it could help somebody
here identify the problem)
Greg
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Re: [Asterisk-Users] Calling an outside number along side otherinternal extensions?
On Fri, 12 Nov 2004, Paul Fielding wrote: > Hmmm... Interesting that you mention it's not a problem with VOIP > companies as they use PRI. The analog trunk I'm connecting to is > actually a Vonage line. Thing is, it seems to me that by connecting via > the Zap channel to the Vonage ATA I'm effectively cancelling any > advantage that Vonage's PRI might have... (?). I don't believe I have > any other alternatives for connecting to Vonage's service, but perhaps > I'm wrong about that. yeah... well, Vonage probably does use PRI.. in their office (yay, that really helps at your end!). It's rather unfortunate that they insist people convert back to the analog domain to use their service. One option available to you is to buy the "softphone" option on your account. In the archives for this list, within the last two months or so, you'll find config examples of how to get Asterisk to connect to vonage with a SIP channel on the softphone account. That would make your current goal easier, probably, but more expensive too. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF and Access Codes
On Wed, 10 Nov 2004, Nathan Bowyer wrote: > I have a problem which I've found quite strange, to say the least. I > have a client who uses long distance access codes from their LD > provider. The codes are 4-digits, nothing extraordinary there. The > problem is, if you dial the digits quickly, without pauses inbetween > them, the LD company does not recognize those digits. If you dial the > code slowly, everything works. > > Phones I'm using are Cisco 7960G phones, * is connected by PRI to the > PSTN. 7960Gs are on SIP v6.3 > > If I set the phones to use inband DTMF, and Asterisk to use rfc2833, the > LD codes work no matter how fast or slow I key them in. It Just Works. > If I set both Asterisk and the Phone to either inband or rfc2833, fast > digit dialing breaks the LD codes. > > Anyone ever see anything like this before, or know of any way to fix it? My guess at the reason why "It Just Works (tm)" with the phone set to inband and Asterisk set to rfc2833 is that Asterisk ignores your digits. That is, by setting * to rfc2833 mode, you disable its built-in tone detection routines. The tones go through just like your voice, Asterisk is none the wiser, and the far end is happy to receive your data. The reason for the breakage when Asterisk and the phone are using the same DTMF method probably has something to do with Asterisk not relaying the digits quickly enough, or maybe the duration of the tone generated by Asterisk isn't long enough for the far end to catch it. Try calling another phone, instead of the LD carrier, and listen to how the tones sound, this may help narrow down the guesses about what's happening. I had a similar problem accessing my Broadvoice voicemail. I was able to use any IVR with my dtmfmode=inband for the broadvoice context in sip.conf, with the exception of their voicemail system. It Just Didn't Work. Someone suggested that when I dial any of their * codes, I set the dtmfmode to rfc2833 for the duration of the call: exten => _*XX,1,SIPDtmfMode(rfc2833) exten => _*XX,2,Dial(SIP/[EMAIL PROTECTED]) I've been able to listen to voicemail without any trouble, and other IVRs still work too. As I'm writing this I'm starting to doubt whether this will work for you, though.. I suppose your users dial the long-distance number they wish to call, and then an LD carrier IVR prompts them to enter the access code. Upon success, the call gets connected. Right? I suppose you could still try the above two-step process for all LD calls which go through that carrier and see what happens.. (ie, whether the carrier picks up the access code digits, and whether you can interact with an IVR in that call) Maybe another option would be to track down how Asterisk decides how long the DTMF tones should be generated, and increase that time. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, X-Lite, and * and # keys
On Wed, 10 Nov 2004, Stanley Cline wrote: > Has anyone else had issues with Asterisk rejecting calls from X-Lite > softphones when the dialed number contains the * or # keys (e.g., dial #86 on > X-Lite "keypad" and then press "send", and Asterisk rejects the call with a > 404 error)? > > It turns out that X-Lite isn't sending the actual * and # characters, but is > converting them to hex, and Asterisk doesn't like it: > > (X-Lite build 1103m, doesn't work) > From: Cline W S Jr ;tag=528200022 > To: > ^^^ > (Sipura SPA-2000 w/ firmware 2.0.10(e), works) > From: Cline W S Jr ;tag=a7b8c513a0ee55aco0 > To: > ^ > (I'm running the most current stable build of Asterisk, obtained via CVS a few > days ago.) > > Any suggestions on what to do here, short of not using the * and # keys in my > dial plan? ;) Asterisk "should" parse that input to look for embedded sequences like this. The SIP URIs are intended to work similarly to HTTP URIs, and the encoding of characters by the %xx method is part of that. Reminds me of the problem somebody had a few months ago where their username assigned by the SIP provider contained an '@' character. The person needed to register using something like [EMAIL PROTECTED]@proxy.sipprovider.tld. I don't remember how that one was resolved.. So.. I guess you could sift through the chan_sip.c code to see whether Asterisk really is checking for these encodings. And then maybe you can fix it. :) Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question: forwarding call from PSTN to VoIP
On Thu, 4 Nov 2004, Luciano Macedo Rodrigues wrote:
> That's problaby a easy question to solve but I couldn't figure out how to do
> what I need.
>
> My PSTN line is connected to a phone and a FXO card. What I need is when
> someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to
> my office, where I'll pickup that call. Or I want to configure that, for
> example, if I know that nobody will be at home, I want to set that the phone
> immediately is forwarded to my office.
>
> Configuration examples? Where I configure this? Please?
It would be configured in extensions.conf. Maybe a timeout on your Dial()
will help accomplish what you're after. For example:
exten => 100,1,Dial(ZAPexten,10)
exten => 100,2,Dial(SIP/youroffice,10)
would ring the zap extension for 10 seconds, then try the SIP extension
for 10 seconds, and then would drop off to somewhere (you might want to
route to voicemail, or play an automated greeting, or simply hang up the
call).
Use the help facility in the CLI ('show application dial') to find out
more.
Greg
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Re: [Asterisk-Users] OT COs/Providers Cannot Reach Others
On Wed, 27 Oct 2004, Steve Totaro wrote: > I have had several experiences where certain providers or COs could not > call other providers. When dialed I would get a fastbusy or similar > message "were sorry, this number" > > I have just realized that this is the case with Voicepulse. Many > different COs and providers cannot reach my VP DID. I know its not one > particular CO or switch or provider so I tend to think its something > that Voicpulse needs to address. > > Not sure how it works in the telephony world but maybe they just have to > broadcast their route or something? > > Since I am not new to these types of situations, I request some feedback > so when everyone blames everyone else, I have some recourse and > knowledge under my belt to say, "No, you need to ." I had the same situation with a number provided to me by VoiceGlo. It was reachable by any of the cell carriers in the market, as well as the more distant "local" COs, but not reachable by the CO where I lived. I couldn't get any useful info from VoiceGlo's support, nor from Qwest, so I visited the CO in person. The techs there were very nice and looked up some stuff on their computers, trying to track down the problem. They finally concluded that the number hadn't been ported to the CLEC properly and that something in the routing had gotten messed up. Ultimately, I think it was up to VoiceGlo to talk to their CLEC providing my DID to get things moving. They were totally unresponsive and I cancelled the account (what good is an unreachable DID? I certainly won't pay money for one!). Anyway.. my feeling was that it takes some pressure on the provider of the unreachable DID to get it fixed. Good luck. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice
On Sat, 23 Oct 2004, Tim Jackson wrote: > We just got setup with Broadvoice yesterday for LD. This isn't something > I REALLY need (No local numbers avail so we just got a Houston number), > but I'm just curious. I can make outbound calls to Broadvoice and they > work great, but I can't do inbound. I have bv's voicemail turned off so > all I get is a busy signal when I call our bv number. I've tried this > with both type=peer and type=friend and I get the same results, any > ideas? in the * CLI, use 'sip show registry' to find out whether you're really registered with the BV servers. Also use 'sip debug' and then place a call. See whether your screen gets filled with a transcript of the conversation between your * and BV. If it does, then read through every line to decide whether what it says seems reasonable (or not). You should at least see attempts by your * to register with BV. If these don't get any reply, then you're probably fighting NAT or some other network issue. Once the registration is successful, then BV should know where to find you so that they can route your inbound calls. These may also be getting dumped by a router/nat somewhere along the network. Hopefully these tips will aid you in diagnosing the problem! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
On Sat, 23 Oct 2004, Terry Evans wrote: > I just signed up for the BroadVoice service a few hours ago, but for > the life of me I can't get any incoming voice. The incoming > connection is fine as it rings my extension from outside, but I can't > hear anyone talking. Outgoing voice is working fine though. (snip) > I have the following ports forwarded to my linux server (it's behind a > NAT router): > > 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of > those have both TCP and UDP forwarded for now. It really sounds like a NAT problem to me.. If your NAT supports the notion of a "DMZ host" then give that a try. Or if the NAT has some sort of logging feature to let you know when the nat receives unexpected packets and discards them, then look through the log. It may be that BV isn't sending RTP in the 2-21000 port range, and that these packets are being dropped by the NAT. Outgoing RTP (voice) would work fine, of course, because the NAT is designed to work that direction. FYI, I just placed a call to my BV number and ran 'netstat -nupa'. UDP connections showed up on ports 14704, 14705, 19838, 19839. These disappeared when I hung up the call. While it might be a config issue, I'm inclined to believe that NAT is making life unpleasant for you. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk dropping last digit of phone number
On Mon, 18 Oct 2004, Demian wrote: > I've recently installed and configured Asterisk. I'm having some > problems with phone numbers which look like 1 021 123 4567 > > (1 for an outside line) and then the phone number. Asterisk will always > drop off the last digit and dial 1021123456 instead. I thought this was > a problem with my contexts however I've recently added a SIP phone and > it's initial context is the same as the analogue phones that display > this problem the SIP phone works fine. Any ideas where I should be > looking? I'd start in extensions.conf.. double-count your X's (or N's) in the exten=> lines to make sure they match the number you're trying to dial. You didn't mention much detail about how the analogue calls get into your *, nor how calls get out. I guess it shouldn't matter much; they'll all get routed through extensions.conf regardless. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Don't go with vendor lock-in or other traps
Hmm, something must've gone wrong with the quoting below I didn't write that which was attributed to me here! Greg On Sun, 10 Oct 2004, Jerry Glomph Black wrote: > Not seen unlimited flatrate? You are not looking very hard. > > Umm, you are ignoring Broadvoice, which has no crappy bonding to their hardware, > or bogus soft-phone ripoff. Broadvoice has unlimited plans that work with > Asterisk. I have it, it works great. > > > Digium rules, and DOES deserve everybody's support. > > On Sun, 10 Oct 2004, Wolf Paul wrote: > > > Greg Hill <[EMAIL PROTECTED]> wrote: > > > >> You probably mentioned this before.. but why Vonage? Since you're using > >> the PSTN for receiving calls and VOIP for placing calls, why not go with > >> another, more enlightened, vendor? Everybody offers US domestic long > >> distance cheap as dirt, and most other vendors' plans are less expensive > >> than Vonage's anyway. The main advantage here is that you get to work with > >> a vendor who will provide you with SIP (or IAX, or something else) > >> credentials to make your * box talk directly to them. > >> > > I have not seen another UNLIMITED FLATRATE offer. That alone, if I expect to > > make many long calls, > > would make me go with Vonage, even if I have to go out through the ATA. Of > > course that leave one with the problem that you cannot make multiple > > simultaneous calls that way (but that is probably the intention, and why > > their softphone account is not available with unlimited calls. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] POTS failover relays (was Vonage, PSTN, 911, and hardware question)
On Sun, 10 Oct 2004, Rajeev Sharma wrote: > Yeah, thanks, I was thinking of doing something similar to that. > Actually, I was gonna spice a cable in my computer's power supply and > use that. Why? Because if it's on a UPS, then the switch will throw at > the same time as the computer looses off. I dunno, I might not even use > a UPS, just a surge protector, but I'll see. Thanks for the idea. I was going to suggest this, but didn't because it's (slightly) more involved. Due to the physics of a relay (that it's constructed with a coil of wire), there is some inherent inductance. When you try to interrupt the current to an inductor, the voltage across it spikes. This is considered a Bad Thing in sensitive electronics like a computer. This isn't really an issue with the wall-wart power adapter, because there isn't likely to be anything terribly sensitive in there (and they're cheap to replace in the event of failure). In a sensitive computer environment, you should include a reverse-biased diode in parallel with the relay's coil. This diode, because it's reverse-biased in normal circuit operation, won't conduct any current. But when power is lost, the voltaged induced by the inductor will forward bias the diode, and the voltage spike will be clamped by the diode rather than going out into other components via the power bus. Most any common diode will work fine (1N4148 small-signal diode, 1N4001/2/3/4 rectifier diode, etc) for this purpose. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to get caller ID
On Sat, 18 Sep 2004, vrushank wrote: > thanx andrew > first of all > your messages are in Plain Text format! plain text format is the preferred format for this (and most?) mailing lists. Replying to a digest and not trimming the unrelated portion before posting isn't a good way to earn points here.. (but you did change the subject line. Thank you for that.) Both of these issues (plain text and trimming replies) are (at the very least) a courtesy to whoever buys the bandwidth this list consumes: posts in HTML are typically double the size of a plain-text post. This means double the workload for the mail server and double the bandwidth bill. Now consider what happens when messages with only 2kB of content and 24kB of un-trimmed junk: the message ends up requiring 13 times more bandwidth. These also forces every subscriber to use more of their own bandwidth and mailbox storage for your message. As for your question... well, you didn't really ask anything. It's hard (for me, at least) to tell what you're looking for. Greg > i hv monitored Asterisk both managerAPI console and Asterisk main console to > see wht is actually going on .when a new incoming connection comes. > > when the phone is ringing.it gives > > starting simple swithc on 'ZAP/1-1' > > and on manager API i get newExten event with exten 's' > from channel 'zap/1-1' > > i hvnt picked up my fone. > > i think its its ringing 7-8 times and asterisk doesnt do anything . > > > > > - Original Message - > From: <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, September 16, 2004 9:22 AM > Subject: Asterisk-Users Digest, Vol 2, Issue 152 > > > > Send Asterisk-Users mailing list submissions to > > [EMAIL PROTECTED] > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > > or, via email, send a message with subject or body 'help' to > > [EMAIL PROTECTED] > > > > You can reach the person managing the list at > > [EMAIL PROTECTED] > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of Asterisk-Users digest..." > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Question:CLI dial cmd
On Wed, 15 Sep 2004, Marconi Rivello wrote: > On 15 Sep 2004 06:26:29 -, Murali <[EMAIL PROTECTED]> wrote: > > Hi friends, > > > > I tried to dial 111 from CLI without any hard/soft phones. > > Well, the ellegant solution is to disable OSS/ALSA and use a softphone :) > I suggest SJphone if you want a SIP client, or iaxcomm if you rather use IAX2. The question is a good one, though. You can answer a call on the console, why not dial from it? I've used something like dial(sip/[EMAIL PROTECTED]) to place calls from the console in the past (I think that's how I did it!). One thing I've never figured out, though, is how to send DTMF digits from the console. For example, suppose I encounter an IVR, or maybe I'm testing my own IVR by calling it from the console -- how do I send the digits? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Use ISP's SIP account for IP-PSTN gateway
On Tue, 14 Sep 2004, Kuniyoshi Murata wrote: > I'm thinking of introducing Asterisk on Linux for IP PBX. > > Now I'm using ISP that has VoIP service and I have VoIP terminal box for > that ISP and a SIP account for SIP server of the ISP. > > Now, what I would like to do is the following. > > A. Setup IP PBX on Linux by using Asterisk. > B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and > connect to my ISP's IP telephony service. > > Is that possible? Hey, this is Linux. It can do anything. If you can code it. :-) You'd just have to set up a context for that service in your sip.conf. Probably need to register=> with the service as well. You'll need your sip username and password in order to do this. Look at the examples in the sample sip.conf, or look through the mailing list archive for example sip.conf's. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrong ID going out...
On Tue, 14 Sep 2004, Evert Meulie wrote: > Hi all! > > I'm trying to have asterisk route all outgoing calls out via my VOIP > provider. > exten => _NXXX,1,Dial,SIP/[EMAIL PROTECTED] seems to have them to in > the direct direction. However, debug shows that my asterisk doesn't > identify itself correctly: > > > Sip read: > SIP/2.0 100 Trying > From: "Evert";tag=as0aca53fa > To: > Call-ID: [EMAIL PROTECTED] IP] > CSeq: 102 INVITE > Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868 > Content-Length:0 > > > 7 headers, 0 lines > > > Sip read: > SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) > From: "Evert";tag=as0aca53fa > To: ;tag=87f2a0d5-13c4-4146d5ea-1a4b30eb-3af4 > Call-ID: [EMAIL PROTECTED] IP] > CSeq: 102 INVITE > Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868 > Content-Length:0 > > ( [my IP] is my external IP, [voip IP] is the IP of the SIP server of my > VoIP provider. [dialled number] is the number I dialled) > > > I don't see any sign here of the username/password being passed to my > provider. is that ok? > IMHO I think it should identify me as [username]/[password], instead of > 'asterisk' to my VoIP provider. > > > What am I doing wrong...? you'll need to use fromuser= and fromdomain= in the provider's context in sip.conf. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk newbie questions
On Sat, 11 Sep 2004, John Stegenga wrote: > [sarcasm on] > Thank you ALL for your warm welcome to this list. I posted this message > yesterday, and since I'm only getting Digest I figured I'd see a response in > a day... > [sarcasm off] > > C'mon. This is the Asterisk Users mail list, isn't it? This is where > the Voip WIKI tells me to go for information on how people are using *. > Even if you only point me in the direction of some other information, it > would be great if I could hear SOMETHING from you guys and gals out > thereI humbly seek YOUR wisdom. One thing you'll learn quickly is that on this list, the subject line in the headers is extremely important. People decide to read or delete entire threads based on those few words, so make 'em count. Apparently the subject line you picked didn't exactly entice us to read or reply.. :-) > Reposted message: > > Hi everyone. > I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. > I'm also brand new to *. > I've been reading the Voip.org wiki, and perusing the list archives for a > while since I've been asked to investigate using IP telephone / soft phones > for a call-center type scenario. People (marketing folks) have pointed me > at Cisco, but I really don't wanna. I'd rather be the hero and pull this > off with a much smaller budget. > > Here is a scenario - 40 person call center, all with PC's (windows) and > soft-phone. > -any recommendations on hardware to run *? soft phones? 90% of calls would > be IP / IAX coming to the center. I can't make any hardware recommendations (I use * at home on a P200) for your application. As for soft phones, try a few and see how you like them. See how your people respond to them. You're going to need to build at least a small development system to evaluate and test before going live with a full install. > I read in the list archives about an ACD application / extension to * that > would probably to what I need in that regard. > - thoughts? it's in there, but I haven't played with it. > In remote locations I would also run *, and hook it up to an extension > on an existing PBX. Excuse the complete newbie question, but how many > 'wires' do I need to bring between the PBX and the * box to support > multiple simultaneous calls? These calls would come from any extension > on the TDM pbx to asterisk to the call center. In a typical scenario > there would NOT be a lot of simultaneous calls unless the system we're > supporting went down hard. quantity of wires depends on A) how many simultaneous calls do you want to be able to carry and B) what's going to run on the wires. If you're going to use an analog interconnect, then it's a 2 wires per call relationship. If you're going to use a T1 link, then you could do 24 (or is it 23?) on the 4-wire cable. Find out what your PBX lets you do. > How would / could? one configure * at the remote location to communicate > with * at the call center? IAX trunking? > How would / could? one configure * at the remote location to use the > existing TDM PBX as failover to call the support center via 1-800 if the > IP circuit died? Hmm. If I understood correctly, you have a PBX with its phones distributed to each of your people's desks. They pick up their phone and dial a number, and you want the PBX to use * as a gateway to transport the call over IP to your support center. Correct? If the IP link is down, you want calls to go directly from the PBX out onto PSTN, skipping *. You may be able to set up * to reject the call if it can't route it out (IP down) and configure the PBX dialplan to re-try the call on a PSTN route. You could also let the PBX be ignorant of the situation, and write your dialplan on * so that when a call can't be routed through the IP link it is transferred back to the PBX to dial out the PSTN link. This latter setup could require 2 channels between * and PBX for every actual call in progress when the IP link is down. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What would be required for this?
On Fri, 10 Sep 2004, Jon Miron wrote:
> I have a question that I'm curious about. I want to set up a 4 phone
> system in my home with 2 actual lines coming into the house. Both or
> just regular lines (not sure of this matters?), one being VoIP and the
> other just a regular analog line. For now though I just want the VoIP
> line coming in, but would like the ability to expand to 2 lines in the
> future. What type of hardware is required for this, and how much would
> it cost?
Digium's X100P card takes a PCI slot and plugs into your phone line just
as an analog handset or modem would. I assume that your VoIP service is
delivered through some type of analog terminal adapter; an X100P could be
used to connect that as well. (You may be able to configure Asterisk to
connect directly to your VoIP provider. This ability depends on whether
you can persuade the provider to give you the account credentials.) Three
or four X100Ps seems to be the maximum people have been able to use in a
single box (some people may have hit a lower limit?). When your quantity
of lines gets into that range or above, you should consider the 4-port
relative of the X100P or a T1 card and an external channel bank.
> For now though, this is what I want to do and for as cheap as possible..
> I have a VoIP line that has free long distance on it and I want to be
> able to dial into Astrisk from my cell to be able to reach any number I
> want (eg extention that dials an outside line). Any ideas on how to go
> about this? Thanks in advance!
This is relatively straightforward to implement in a dialplan
(extensions.conf) either by implementing extensions direction or by using
the DISA application. Keep in mind that a system which allows an incoming
call to make an outgoing call has some inherent security issues
('wardialing' hasn't gone away).
Greg
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Re: [Asterisk-Users] New to *
On Fri, 3 Sep 2004, Bill Andersen wrote: > I just ran across the * site. Looks great. I do not need a PBX at this > time, but DO need to replace an old voice mail system. I'll do my > homework and figure out the specifics, but before I dive into it all and > spend a bunch of time only to find out "I didn't understand", is it > reasonable to think I could configure * to simply act as a voicemail > system off an existing PBX? It looks possible to me. > > Who knows, I might learn enough about how it all works to actually end > up replacing my PBX. But for now, with proper configuration, could it > "act" as a voice mail system? yes, that's entirely reasonable. Probably the trickiest bit will be the actual connection between your PBX and your Asterisk box. This connection could be made via an x100p card connected to an analog station port on your PBX (or multiple connections of the same style). From there you'd have to work out how to transfer a call in the PBX out to Asterisk via the analog port extension, and how to signal to Asterisk what mailbox is wanted (or simply make the transfer and use an IVR in Asterisk so that the caller can choose a mailbox him/herself). In any case, it's a relatively inexpensive experiment. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extens and number converting so that i can dial following one standaard.
On Sun, 29 Aug 2004, Johannes van Hulst wrote:
> For asterisk I am using more than one sip providers.
> The provider in Holland would like to have the international calls like
> 00 31 20 1234567 but the provider in the US likes it like 0011 31 20 1234567
>
> Can I make a rule in asterisk so that I can dail 00 31 20 1234567 and
> asterisk dails 0011 31 20 1234567 to the US provider?
>
sure. You'd use a Dial() command like this for the provider in Holland:
Dial(SIP/[EMAIL PROTECTED])
and something like this for the provider in the US:
Dial(SIP/0011${EXTEN:[EMAIL PROTECTED])
so to route any extension starting with 0031 through the US provider:
exten => _0031.,1,Dial(SIP/0011${EXTEN:[EMAIL PROTECTED])
for example. You didn't mention how you want asterisk to know/decide which
of the two providers a particular extension should be routed to. You'll
likely need to write a different exten => line than the sample I gave.
Greg
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RE: [Asterisk-Users] Asterisk to Vonage
On Tue, 24 Aug 2004, Paterson, Mark wrote: > Honestly, do you think I would ask for help on the list if I hadn't come > up with any successful results on my own?? > > Just asking if anyone has made this work. If so what rev of * were they > running and what do their configs look like. Jay does have it working. The search suggestion he gave is likely to bring up the message he posted to this list some weeks ago. That message does contain an example for sip.conf which shows how he has his configured. Greg > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk > Sent: Tuesday, August 24, 2004 6:06 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Asterisk to Vonage > > Yes, search google for > > asterisk vonage working site:lists.digium.com > > > -Original Message- > > From: Paterson, Mark [mailto:[EMAIL PROTECTED] > > Sent: Tuesday, August 24, 2004 11:19 AM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Asterisk to Vonage > > > > > > I'm trying to connect my Asterisk server via sip using my vonage soft > > phone account. Has any anyone successfully got to work? I get error > > from > > asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing > > '/etc/asterisk/sip.conf': Found > > Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: > > Unable to lookup '216.115.25.199:5061' when trying to register with > > the vonage sip proxy. Any examples would be greatly appreciated. > > > > > > Rgs, > > mark > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/aster> isk-users > > To > > UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newb question regarding DTMF
On Tue, 24 Aug 2004, Erik Anderson wrote:
> On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
> <[EMAIL PROTECTED]> wrote:
> > x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to
> > transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
> > for your x-lite user.
>
> That's what I've read, and I have added dtmfmode=rfc2833 in my
> sip.conf...see this snippet:
>
> [xlite1]
> ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
> ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
> type=friend
> username=xlite
> callerid="Jane Smith" <5678>
> host=dynamic
> nat=yes ; X-Lite is behind a NAT router
> canreinvite=no; Typically set to NO if behind NAT
> disallow=all
> allow=gsm ; GSM consumes far less bandwidth than ulaw
> allow=ulaw
> allow=alaw
> dtmfmode=rfc2833
>
> I've applied that change and restarted asterisk, but no dice...
Dial the extension, then on the * CLI use 'sip show channels' to get the
name of the active channel. Next use 'sip show channel ___' to get info on
that particular channel (you can type the first few characters and use tab
completion; no need to type the whole string!). Scan through the output to
see whether asterisk is really using rfc2833 for that channel. If it is,
then the problem is likely in the x-lite config. If not, try moving
dtmfmode to the general section of sip.conf
Greg
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[Asterisk-Users] Re: [Asterisk-Dev] Asterisks
On Tue, 24 Aug 2004, Huddleston, Robert wrote: > Take mercy on me - I'm a newbie w/ Asterisks... Here's what I'm trying to do > - and please someone let me know if this can be done... > We have a commercial VoIP network... The gatekeeper supports MGCP/H.323 and > allows for calls to be made to the PSTN cloud. > I would like to build Asterisks with H323 (or MGCP if need be) and have it > attach to our gatekeeper to access the PSTN. > Instead of installing a T1/E1 or ISDN or POTS card we would like to use the > existing VoIP network. > Anyone ran into this before - can provide some direction? so you want your Asterisk machine to connect to your gatekeeper "via existing VOIP" .. you mean over an ethernet-type interface, right? Sure you can do that. A better place to ask for this type of help is the asterisk-users list, though. The asterisk-dev list is for topics related to development of asterisk itself, not development and implementation of solutions involving asterisk. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ex-girlfriend logic not working in latest CVS?
On Tue, 24 Aug 2004, James Sizemore wrote: > Ex-girlfriend logic not working in latest CVS? > Incoming sip calls don't work. Anyone else seen this > problem? > > Extension logic looks good: > > exten => 6153248305/_931NXXX,1,Queue(queue1); > exten => 6153248305/_615NXXX,1,Queue(queue2); > ;exten => 6153248305,1,Queue(queue3); > > > show dialplan looks good: > > -- Added extension '6153248305' priority 1 (CID match > '_931NXXX')to vantage > -- Added extension '6153248305' priority 1 (CID match > '_615NXXX')to vantage Your lines have 7 X's in addition to the N. Did you mean to type six of them instead? (10 digits total) Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newb question regarding DTMF
On Mon, 23 Aug 2004, Erik Anderson wrote: > Hello all - I'm just starting to play around w/ asterisk, and I've run > into a seemingly simple problem that has really manged to frustrate > me... > > I'm running the latest cvs version of *, and am trying to dial in to > the default extention 1000 demo using x-lite. I can dial and hear the > greeting no problem, but when I try and send any DTMF tones, I don't > get any response. Is there something specific I need to set in my > sip.conf to allow DTMF? x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to transmit inband). You need dtmfmode=rfc2833 in [general] or in the section for your x-lite user. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Internal extensions giving weird static-like dialtone
On Tue, 17 Aug 2004, Lucas Wrenn wrote: > Hello all I have a RH8 machine running the latest CVS of asterisk > against digium hardware including a TDM400P (with two fxs modules) and a > X100P. I am having some really weird problems with m internal > extensions. (I am trying to get them working before connecting to my > POTS line) The dialtone I get is loud and has lots of static. I was > thinking that it might simply be me overdriving phone but that seems > unlikely. Anyone have any ideas? does regular audio, after a call is connected (to voicemail, Playback(), another extension, whatever) sound normal or is it distorted as well? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] couple basic questions
On Tue, 17 Aug 2004, John Williams wrote: > My desire to run Asterisk is finally giving me the reason to install a > Linux box at home. > Is RH9 the only distro that Asterisk will run on, and can anyone > recommend a good source for a cheap Linux (RH9) box? > For example, walmart.com has microtel boxes with no OS. Will RH9 and > Asterisk run on these boxes? oh, no no.. Asterisk should run on nearly any distro (provided basic things like C libraries and stuff). I've found that the property surplus store at the local university is a good source for cheap, though dated, PCs. I've seen PII 300+ MHz boxes in the US$60 range. This is MUCH more power than you'll need for a home asterisk install (I have two; both run on P-200 class machines with slackware 9.0). And as Chris mentioned.. it'd be wise to get comfortable with Linux generally before diving too quickly into Asterisk. It would be easy to get overwhelmed and lose enthusiasm if you bite off too much all at once. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?
On Mon, 16 Aug 2004, Olle E. Johansson wrote: > Greg Hill wrote: > > > On Mon, 16 Aug 2004, Olle E. Johansson wrote: > > > > > >>James Freire wrote: > >> > >> > >>>Hi All, > >>>I am trying to setup another sip trunk in addition to what I am already > >>>using. The sip provider we are using right now gives you your username > >>>as your email address. So IE. If my email is [EMAIL PROTECTED] that is > >>>my username . Now... When I put this in the sip.conf file I have found > >>>that Asterisk is not able to parse it correctly and instantly goes to > >>>the email server to authenticate the sip user upon registration > >>> > >>>Here is the line below in my sip.conf file > >>> > >>>register => [EMAIL PROTECTED]:[EMAIL PROTECTED] > >>> > >>>THe error is below > >>> > >>>Aug 16 11:30:05 NOTICE[114695]: chan_sip.c:3922 sip_reg_timeout: > >>>Registration for '[EMAIL PROTECTED]@sip.voipamericas.com' timed > >>>out, trying again > >>>Aug 16 11:30:06 NOTICE[114695]: chan_sip.c:6575 handle_response: Failed > >>>to authenticate on REGISTER to > >>>';tag=as1c528b93' > >>> > >> > >>That's obviously an error. Please add it to the bug tracker and we'll solve it. > > > > > > > > I disagree.. although the sip rfc (I'm looking at 3261, June 2002) doesn't > > specifically state that the user field cannot have the @ character in it, > > the language there suggests that '@' is supposed to be the separator > > between the user string and the host string. In addition, it is stated > > that the sip URI scheme follows RFC 2396, which states that all of > > [;/?:@&=+$,] are reserved characters. See section 2.2: > >The "reserved" syntax class above refers to those characters that are > >allowed within a URI, but which may not be allowed within a > >particular component of the generic URI syntax; they are used as > >delimiters of the components described in Section 3. > > > > I think Asterisk's behavior is correct and the syntax > > '[EMAIL PROTECTED]@sip.voipamericas.com' is debateable at best. It's > > possible that replacing the @ in the intended user portion with %40 may > > allow it to slip through Asterisk and get un-escaped by the server on the > > far end.. Anyway, the issue may warrant some more dialogue before > > declaring it a defect. > > We are not talking about the user field, it's the digest auth username > that is the important field for authentication. I belive that field is > defined as "quoted-string". I've seen use of @-constructs in http, > logging in to a web site with my e-mail name as the username, so I guess > it's valid. > > http digest auth, used in SIP, is defined in RFC 2617. > Don't mix this with the username part of the SIP URI. > > I have to check if my proposed realm authentication scheme I use > in chan_sip2 can handle this. oh, I see.. So is this similar to the fromuser and fromdomain options that are used elsewhere in sip.conf? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID Questions
On Mon, 16 Aug 2004, Mike Roberts wrote: > sip debug made no changes > > I can makes calls with my asterisk using X-lite softphone, I can even > call the 877 number and it works perfectly! But when I call the 877 > number over the PSTN, it does nothing. A busy signal. Try calling from a different provider, a calling card, a cell phone.. anything that uses a different part of the PSTN. Maybe you could try the toll-free gateway on iaxtel, fwd, or another. I had a similar problem with Voiceglo: they set up a DID and it worked fine for calls from any of the local cell providers, as well as my calling card. But it simply wouldn't work to call that number from the local Qwest exchange. Their techs told me (I went to the source.. the CO) that it appeared that the CLEC Voiceglo was using didn't have the number ported correctly and that they (the CLEC) would have to fix it. Voiceglo never did anything about it, so I ended up dumping them. Anyway, if it turns out that you get different results when calling from different parts of the PSTN, then you're right -- it's definitely a provider problem. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID Questions
On Mon, 16 Aug 2004, Mike Roberts wrote: > Is there anyway to test if this call is touching my servers? Everyone > is telling me the DID is fine. But I can't confirm that for sure. And > I don't want to go ahead and start making changes to my config when > the DID isn't even working in the first place. as Eric suggested, iax2/sip debug is your friend. You could also run a packet sniffer if you wanted to. It is possible that the DID could be fine and your config broken: for example, if your box isn't registering with the far end, then it may not know what to do when somebody calls. The busy tones would be their switch saying "sorry, I can't connect you." Likewise, a broken config could keep calls from originating at your * and going out to them. A failing register command would get announced on the CLI, as should a failure to establish a call via sip/iax2. Actually, I don't think you mentioned what transport you're using. We just assumed that it's sip/iax2. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?
On Mon, 16 Aug 2004, Sean Cheesman wrote: > It might make sense for * to parse the register line from right to left. > Then it wouldn't be an issue. Or am I missing another issue that would > arise? That's an interesting idea. It might be tricky, though, because you have to read the scheme (sip:) off the front of the URI to identify what it is and what fields it might have. Some fields are optional, too, like the password and port. It would take some careful handling, but I suppose it could be done. As an aside, many MTAs will simply drop a message when they find that a to/from header has an address with multiple @'s. They consider it to be an attempted attack (by the sender) trying to break some security. For example, if I were to send mail to [EMAIL PROTECTED]@unfortunaterelay.com and the MTA at unfortunaterelay.com allowed the message through, then some other tool in the chain might see this as a message to [EMAIL PROTECTED] and re-send the message there. The supposed-closed relay would effectively be an open relay, with all the implications that go with that. Greg > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill > Sent: Monday, August 16, 2004 1:21 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ > signs for register? > > > On Mon, 16 Aug 2004, Olle E. Johansson wrote: > > > James Freire wrote: > > > > > Hi All, > > > I am trying to setup another sip trunk in addition to what I am > > > already using. The sip provider we are using right now gives you > > > your username as your email address. So IE. If my email is > > > [EMAIL PROTECTED] that is my username . Now... When I put this in > > > the sip.conf file I have found that Asterisk is not able to parse it > > > correctly and instantly goes to the email server to authenticate the > > > sip user upon registration > > > > > > Here is the line below in my sip.conf file > > > > > > register => [EMAIL PROTECTED]:[EMAIL PROTECTED] > > > > > > THe error is below > > > > > > Aug 16 11:30:05 NOTICE[114695]: chan_sip.c:3922 sip_reg_timeout: > > > Registration for '[EMAIL PROTECTED]@sip.voipamericas.com' timed > > > out, trying again Aug 16 11:30:06 NOTICE[114695]: chan_sip.c:6575 > > > handle_response: Failed to authenticate on REGISTER to > > > ';tag=as1c528b93' > > > > > > > That's obviously an error. Please add it to the bug tracker and we'll > > solve it. > > > I disagree.. although the sip rfc (I'm looking at 3261, June 2002) > doesn't specifically state that the user field cannot have the @ > character in it, the language there suggests that '@' is supposed to be > the separator between the user string and the host string. In addition, > it is stated that the sip URI scheme follows RFC 2396, which states that > all of [;/?:@&=+$,] are reserved characters. See section 2.2: >The "reserved" syntax class above refers to those characters that are >allowed within a URI, but which may not be allowed within a >particular component of the generic URI syntax; they are used as >delimiters of the components described in Section 3. > > I think Asterisk's behavior is correct and the syntax > '[EMAIL PROTECTED]@sip.voipamericas.com' is debateable at best. It's > possible that replacing the @ in the intended user portion with %40 may > allow it to slip through Asterisk and get un-escaped by the server on > the far end.. Anyway, the issue may warrant some more dialogue before > declaring it a defect. > > Greg > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?
On Mon, 16 Aug 2004, Olle E. Johansson wrote: > James Freire wrote: > > > Hi All, > > I am trying to setup another sip trunk in addition to what I am already > > using. The sip provider we are using right now gives you your username > > as your email address. So IE. If my email is [EMAIL PROTECTED] that is > > my username . Now... When I put this in the sip.conf file I have found > > that Asterisk is not able to parse it correctly and instantly goes to > > the email server to authenticate the sip user upon registration > > > > Here is the line below in my sip.conf file > > > > register => [EMAIL PROTECTED]:[EMAIL PROTECTED] > > > > THe error is below > > > > Aug 16 11:30:05 NOTICE[114695]: chan_sip.c:3922 sip_reg_timeout: > > Registration for '[EMAIL PROTECTED]@sip.voipamericas.com' timed > > out, trying again > > Aug 16 11:30:06 NOTICE[114695]: chan_sip.c:6575 handle_response: Failed > > to authenticate on REGISTER to > > ';tag=as1c528b93' > > > > That's obviously an error. Please add it to the bug tracker and we'll solve it. I disagree.. although the sip rfc (I'm looking at 3261, June 2002) doesn't specifically state that the user field cannot have the @ character in it, the language there suggests that '@' is supposed to be the separator between the user string and the host string. In addition, it is stated that the sip URI scheme follows RFC 2396, which states that all of [;/?:@&=+$,] are reserved characters. See section 2.2: The "reserved" syntax class above refers to those characters that are allowed within a URI, but which may not be allowed within a particular component of the generic URI syntax; they are used as delimiters of the components described in Section 3. I think Asterisk's behavior is correct and the syntax '[EMAIL PROTECTED]@sip.voipamericas.com' is debateable at best. It's possible that replacing the @ in the intended user portion with %40 may allow it to slip through Asterisk and get un-escaped by the server on the far end.. Anyway, the issue may warrant some more dialogue before declaring it a defect. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing out and ringing issue
On Mon, 16 Aug 2004, Info wrote: > Hoping someone might know how to resolve this (probably an easy one). I > have one Asterisk PBX with a single NIC and an FXO card with PSTN line > attached, and one IP phone (Budge Tone 100) on the LAN. Via the phone I > get no dial tone, and dialing 9, doesn't allow me to dial out. > I also need the phone to ring when the asterisk PBX is called. I have > modestly tweaked the sample configs to get this far. > > I can check and retrieve and delete voicemail via the phone however. > Any ideas? from the asterisk CLI, run "sip debug on" and then place your calls (both the call to voicemail and the other call). This shows you what asterisk is actually receiving and should help to track down the problem. For example, you'll notice whether the phone is sending anything at all to asterisk, and if it is, then you'll see what it sends. This will help you get a better idea of what fixes you need in order to get the phone a working sip account, registration, extension configs, etc. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no tones detected
On Sun, 15 Aug 2004, Johnathan Bunn wrote: > maybe this has been covered before but, i can't find it, has anyone > had a problem where outside lines can't use number presses like choose > extensions but inside lines can, > I am using voicetronix hardware with asterisk and when i call from a > station port I hear my greeting and can dial an extension and connect, > but if I call in I can here my greeting and pushing buttons does > nothing, and they dont show up on the console either I have tried > 3 land line phones... > > any help or a point in the right direction would be extremely helpful I don't know anything about your hardware, but it sounds like yours is a DTMF mode problem. There are multiple ways of communicating the DTMF tones in the land of VOIP. SIP, for example, typically uses RFC2833, info, or inband. You set up * (or it defaults) to expect DTMF in a particular format, and it ignores the other formats. So if the far end transmits inband and asterisk is expecting RFC2833 (default on SIP connections) the it won't recognize the caller's button presses. Search google and the wiki for "dtmfmode" or "dtmf mode" for your device's connection method (sip/zap/iax/whatever). Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto remove digits from a called number
On Sat, 14 Aug 2004, administrator tootai wrote:
> Hi list,
>
> I have SIP clients and H323 GK connected through h323 channel (Nufone).
> In h323 conf I gave prefix=09 so all call starting with this prefix are
> send to asterisk. The context is also given their as [fromh323]
>
> But now, in asterisk, I want to have the called number without this 2
> leading digits so the exten variable will be according to my actual
> dialplan. Here's an exemple:
>
> In extensions.conf I have
>
> exten => 100,1,Goto(demo,s,1)
>
> If I call #100 from SIP it's ok. So now, if I want to reach this
> extension from an h323 EP, I have to call 09100. This call will never
> succeed (or I create a new exten line, same as above, with this prefix).
You're right, you will have to create an extension to match the 09xxx
numbers. But you don't have to create one for every "real" SIP extension
you have. Instead, make one that matches all 09xxx extensions and does a
goto:
exten => _09XXX,1,Goto(yoursipextensionscontext,${EXTEN:2},1)
for three digit "real" extensions. Add or remove X's for more or fewer
digits, or just use _09. for _ANYTHING_ that starts with 09 (keep that in
mind.. sometimes that wildcard extension comes back to bite you!).
Greg
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RE: [Asterisk-Users] Asterisk & MyPhoneCompany.com (aka Talk(n))
On Thu, 12 Aug 2004, Travis Conway wrote:
> I signed up for BV and entered everything, to the best of my knowledge,
> in the sip.conf and extensions.conf file, but am having problems getting
> it to connect. If someone could show me an example of their working BV
> sip.conf that would be greatly appreciated. I am certain that I have
> the extensions.conf correct since I can see it in the console (see
> below).
>
> -- parse_srv: SRV mapped to host proxy.lax.broadvoice.com, port 5060
> -- Executing SetCallerID("SIP/loni-5ddf", "4047952206") in new stack
> -- Executing Dial("SIP/loni-5ddf", "SIP/[EMAIL PROTECTED]") in
> new stack
> -- Called [EMAIL PROTECTED]
this could be your problem. They use 10-digit dialing. You may use 7-digit
dialing if you wish, and your account's area code will be assumed. Anyway,
don't dial the 1 in front of the destination number. Then if that doesn't
work, turn on sip debug at the CLI and attempt a call. Sifting through the
pages of debug you get back should help to track down the problem.
Greg
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Re: [Asterisk-Users] BroadVoice Voicemail
On Wed, 11 Aug 2004, Chris wrote: > Set your sip.conf and your phone to inband as BroadVoice requires. Then > simply create an extension for BV Voicemail and use the SIPDtmfMode() > command like this > > exten => *86,1,SIPDtmfMode(rfc2833) > exten => *86,2,Dial(SIP/[EMAIL PROTECTED],60) > exten => *86,3,Congestion() > > This works, but I still think there may be a bug, you shouldn't need to do > this and this SHOULDNT work but it does... Thanks for the fix! I've done battle with this issue a number of times, but never did get it figured out. I'm inclined to believe that there is something buggy in the DTMF code. I thought my config worked to interact with any IVR, but I discovered last week that I can't get IBM's support line (800-426-7378) to recognize my DTMF. So I hang up and call another system, and it works fine..! It sounds like the basic idea in this hack is that we set * to rfc2833/info so that it looks for DTMF anywhere but inband and doesn't notice we've slipped a few digits through in the audio stream. This makes me wonder about the inband DTMF generation in * -- When my xlite softphone generates the DTMF tones, BV's voicemail server is pleased with them. But when * generates the tones, the server doesn't catch them. Rather, it does recognize that something came down the wire, because the greeting message stops playing. But it doesn't actually recognize one or more of the digits. So the fix mentioned above let xlite's DTMF go through and fixed the issue with the BV voicemail server, but it didn't work for the IBM number I had trouble with. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BroadVoice Voicemail
On Thu, 12 Aug 2004, Jay Milk wrote: > Praytell, if you have Asterisk working, why do you even bother with the > BroadVoice voicemail? First order of business when I activated my BV > lines was to disable voice-messaging and set "busy-forward" to my PSTN > number (which in the meantime has been changed to my Vonage Softline) For me the primary value is because I'm currently stuck with a less-than-reliable ISP. That is, I am their network outage monitor: I notice the network is down (it's a WISP), I call them, they say "oh, would you look at that. it IS down!" and then they send somebody to do something about it. So I'd like to let BV keep my voicemails for me during those times when there are network issues between my box and theirs. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF issues
On Tue, 10 Aug 2004, AJ Grinnell wrote: > I hadnt heard of that setting until today either, but it still doesnt work. > I am using dtmfmode-rfc2833 in sip.conf, and I have my spa's set to avt or > auto. The "internal" dtmf used for voicemail or anything within * works just > fine. I hust cant get tones out to the PSTN. The DTMF sounds very distorted > on the other end of the call. I am trying to use rfc2833. If you have any > ideas, please let me know. Thank you. it wasn't until a few minutes after I posted that I realized I should have recommended using 'sip show channel ...' while the call is active. Asterisk will tell you which DTMF mode it's trying to use on the channel. That's how I initially discovered the setting for Broadvoice users to receive dtmf on inbound calls.. I knew it should be inband, but sip show channel told me that asterisk was using rfc2833 for that leg of the call. I went looking for a way to change that setting and found a solution to the problem. If your codec supports inband, you could give it a try.. (I prefer out-of-band too, but if the endpoint only supports inband... then you can't beat 'em and end up joining 'em.) oh, you said "dtmfmode-rfc2833" above. That was a typo, right..? You really have a '=' rather than '-' in sip.conf? as for SIPDtmfMode, try a case-insensitive grep on the source to see if that string comes up anywhere. If it isn't there, then it's not a real option and would probably be ignored by the config-parsing algorithm. Greg > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Greg Hill > Sent: Tuesday, August 10, 2004 1:32 PM > To: Asterisk > Subject: Re: [Asterisk-Users] DTMF issues > > > On Tue, 10 Aug 2004, AJ Grinnell wrote: > > > I am now at a total loss. Using Sipura spa-2000s connected to *, I get > > DTMF working just fine for internal extensions, voicemail, etc. If > > making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I > > get no dial tone. I am working unsuccessfully with Cisco right now on > > this, but they cant find anything wrong. I have tried all suggestions I > > can find from the list and elsewhere. I have added SIPDtmfMode to my > > outgoing extensions, that still doesnt help. Does anyone out there have > > experiance or ideas with this setup? > > I haven't heard of any SIPDtmfMode setting, but there is dtmfmode= (in > sip.conf, not extensions.conf). Which dtmf mode are you hoping to use? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues
On Tue, 10 Aug 2004, AJ Grinnell wrote: > I am now at a total loss. Using Sipura spa-2000s connected to *, I get > DTMF working just fine for internal extensions, voicemail, etc. If > making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I > get no dial tone. I am working unsuccessfully with Cisco right now on > this, but they cant find anything wrong. I have tried all suggestions I > can find from the list and elsewhere. I have added SIPDtmfMode to my > outgoing extensions, that still doesnt help. Does anyone out there have > experiance or ideas with this setup? I haven't heard of any SIPDtmfMode setting, but there is dtmfmode= (in sip.conf, not extensions.conf). Which dtmf mode are you hoping to use? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] truncated extensions
On Mon, 9 Aug 2004, Kevin Johnson wrote:
> When dialing 8437624, I get the following output:
> -- Executing NoOp("SIP/office1-b727", ""call for "843762" "43762"
> "6") in new stack
>
> on the following line:
> exten => _8.,3,NoOp("call for "${EXTEN}" "${EXTEN:1}" "${LEN(${EXTEN})})
this is really odd. I've got a similar extension (_8.) that I use for
routing calls to FWD, and of course it works fine.. Which SIP UA are you
using? Maybe you could turn on sip debug to see whether the UA is really
sending all your digits to asterisk.
Greg
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Re: [Asterisk-Users] truncated extensions
On Sun, 8 Aug 2004, Kevin Johnson wrote:
> I'm having a problem with extensions.
>
> Any extension longer than 6 characters gets truncated to 6 characters.
>
> For example,
>exten => _7XX,3,NoOp("call for"${EXTEN})
> results in
>call for 712345
> when given
>7123456
that's ${EXTEN}, not ${EXTEN:1}, right (I mean what's actually in your
extensions.conf file)? And you don't have any other extensions with the
'.' wildcard in them which might be getting matched instead? Next step:
mention which version you're running and maybe include extensions.conf.
good luck -
Greg
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Re: [Asterisk-Users] Asterisk : No Sound No Dial
On Sun, 8 Aug 2004, niko singh wrote: > Thanks for taking a look greg and hank. This seems to be getting bettre > everyday..help please My sjphone is running on the same box as > asterisk...i believe then the red hat firewall should not be a problem. Some of these problems sound like they might be related to having sjphone and asterisk running on the same machine: they both want to bind to port 5060. Whichever program starts first gets the port and the other one doesn't. Maybe you'll get different results if you set one of them to run on another port. Better still would be to run asterisk on one PC and your softphone on a different PC. > [general] > port = 5060 ; Port to bind to > exten => _8.,1,Dial(SIP/[EMAIL PROTECTED],tr) you should put a : in place of - in the above line to strip the leading 8 off the extension number. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk : No Sound Issues
On Sat, 7 Aug 2004, niko singh wrote:
> Thanks greg , for pointing out the valuable resources for reference. I
> tried SJphone in a windows environment to connect to fwd and it worked
> fine(including (audio). Now have to do the same thing for linux(red hat
> 9 ) and hope the nat issue is resolved.
your mention of firewalls below reminded me of a certain "feature"
included with RedHat 9. The installer likes to set up a firewall (using
the ipchains tools) to help protect the machine against attacks. This
could potentially cause problems if the firewall blocks connections when
your softphones try to register with asterisk. A quick-and-easy temporary
fix is to remove the firewall rules entirely by using "iptables -F;
iptables -X" as root. The firewall rules are restored the next time you
reboot. Long term, it would definitely be a good idea to read about
firewalls with ipchains and get yours set up as you need.
> Now i would like to connect asterisk to fwd and instead of the SJ phone
> connecting to fwd directly i would wish to connect through asterisk, writing
> the extensions to transfer all dailled numbers from my SJphone to fwd. At a
> later stage make asterisk accept calls dialled to my fwd number and operate
> thm through the SJ phone
register your box to fwd (for incoming calls to your fwd number): add to
sip.conf in the [general] secion
register => fwdnum:[EMAIL PROTECTED]
calls to your fwd number will be routed to your context specified in the
[general] section.
To make calls to the fwd network, you'll need something like this in
sip.conf:
[fwd]
type=friend
secret=
username=
host=fwd.pulver.com
context=incoming
and then in your extensions.conf something like:
exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
then any number that starts with an 8 will be tried at fwd. This exten
statement would need to be in the same context as your softphones in order
for them to use it.
> How can nat issues be resolved with asterisk.
typically you have to set up port forwarding on your nat device and use
externip= in sip.conf. You may also need to use canreinvite=no in some
contexts of sip.conf as well as nat=yes. Keep browsing and searching,
especially on the wiki but also on google.
Greg
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Re: [Asterisk-Users] Asterisk Dry Run
On Sat, 7 Aug 2004, niko singh wrote: > I just installed asterisk on my system with the purpose of rerouting > calls on sip channels. I don't think i need any hardware for that. you're right, mostly. There are some asterisk features like meetme conferences which require a timing source. This could be from a Digium card or derived in software from some USB chipsets or from some RTC's. But for basic sip channel stuff, you won't need additional hardware. > I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both > of them and to call one from the other on the same machine, however could > not. > I 1-) I could connect sjphone in isolation to freeworld dialup howver i got > no sounds ...the records at free world dialup confirmed i made and recieved > calls. My sound card works absolutely fine..however i got no echo or ring, > yet the record shows i did make a call when i did. not getting sound back through sjphone calling direct to fwd might be a nat problem.. you didn't mention anything about the arrangement of your network, though, so it's anybody's guess. > 2-) With asterisk when i enter "sip show registry" it does not show either > device...however they do appear in sip show peers at CLI. Are the devices > registered or no. "sip show registry" shows what sip servers you have registered asterisk with. In other words, those servers would see asterisk as a client, just as asterisk will see your softphones as clients. So this is the expected behavior. (unless you set asterisk to register with FWD, in that case it should have shown up in this list) > Could someone kindly prescribe a few tests to check my asterisk > installation ( i get the CLI) but now it seems may be sound devices are > a problem...will installing an x100p solve it ?( i know it is an fxo but > probably it uses its own on board sound config) no, an x100p won't change anything. It's just a PSTN interface card. > 3-)Are there any other free sip accounts i could check with..last night > fwd seemed unreachable it isn't sip, but you can try iaxtel. IAX is a protocol developed with asterisk. It's similar to sip but has many improved features. > 4-) I went to the LIPZ4 (zultys) config file in /usr/local/zultys/mp4_*.cfg > and added my internet settings ...what changes do i need to make if i wish > to use it to connect to fwd straight or connect to asterisk. > 5-) I am runnign askterisk and all these on the same machine ..could someone > kindly send me a sample zultys config. and some explanation. I played with the lipz4 for a few minutes and couldn't get it working so I dropped it. Maybe somebody else knows how to make it go. > 6-) Absurd question perhaps but i was just thinking if this is a > possibility : My number ( PSTN ) is 5678 ..now if someone dials 56789056 > will i get the call ? can i get 9056 as extension. that would depend on your PSTN provider. But that's most likely not to work. (your phone number is really only 4 digits long??) However, somebody could dial 5678 and you could answer that and send it to an IVR (menu) where they could dial the 9056 extension. > Lots of questions...new here and would be grateful if someone from the > communtiy could anser them be sure to visit www.voip-info.org, www.asterisk.org, www.google.com, etc. After spending time reading the above sites you'll be better equipped to set up your system, as well as to know what information about your setup is useful to us so that we can help with any problems you might encounter. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Access Providers for Asterisk
On Wed, 4 Aug 2004, William R. Lorenz wrote: > I'm looking for U.S. providers that will provide access to the PSTN and > allow me to easily use my Asterisk box with their services. I would > prefer a provider that supports number portability, so that I can park my > existing cell number on their network and later move it again, but I'm > open to doing some funky stuff with call forwarding if I have to do that. http://www.voip-info.org/wiki-VOIP+Service+Providers > Can anyone provide their recommendations or experience in using a VoIP > provider, as opposed to a LEC, to provide Asterisk with PSTN access? After you go through the providers listed in the page above, reviewing each of their coverage areas, features, use policies, etc, you'll probably have narrowed it down to just a few who could meet your particular needs. At that point you can use google or another list archive search tool to find praises, rants, and probably config examples, for the ones you are interested in. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Get MWI from Telco's voicemail
On Wed, 4 Aug 2004, Scott Petersen wrote: [snip] > What I am seeing is an event every half hour exactly, on each of the two > voice lines. This causes the simple switch to kick in and ring the > extensions. Of course there is no one there. I have put a workaround in [snip] > Since these events happen every half hour and only on the lines that > have voicemail I am very confident that it is the telco sending a > trigger to turn the MWI on the phones either on or off. I really don't > want to have to try and find out from the telco as their support is > much, much, much, less than knowledgeable or helpful. What I am I found that when I'm dealing with a technical issue on POTS, I get the best results by personally visiting my CO and hoping a nice person answers when I ring the bell. For example, I finally got my DSL to work through this approach: it was up and down often, apparently related to ambient outdoor temperature. I got to know the (then US west, now qwest) DSL tech who handled that CO and worked with him directly to resolve the issue. When he could see that he'd need to spend more than just a few minutes on a job, he'd take my phone number and tell me to call the 800-number service people. When they generated the work request and he received it, he'd call me directly to find out what the real problem is and get it addressed. So what I'm getting at is that you might find it advantageous to at least try talking with techs in your particular CO. They're much more likely to understand what you're asking about and to know the answer than a 800-number support rep. Just make sure you're nice and avoid taking more than just a few minutes of their time. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP Packets going over caller and calle !!
On Mon, 2 Aug 2004, Carlos Arnt wrote: > I Have a problem here, if anyone know a method to avoid please tell me. > Using * with the option canreinvite=yes i can in theory tell to my * > box, send RTP Packet directly from one Sip device to another one, then > "In Theory", i will not use my own internet connection. So this mean > that will a lower connection something like "512/512kbps", i can have > lot's of channels connected and using my * just to bridge then . Thats > correct ?? essentially, yes.. Asterisk can send a REINVITE to the SIP devices to tell them to communicate directly between themselves, in a peer-to-peer way. This removes asterisk from the loop entirely, and you're not bridging the calls anymore. The call would disappear from your box (I think). > But if all people is under a NAT ?? Like 2 sip devices using * box > connect over my * box but that two is under NAT ?? Will work anyway ? > All RTP Packets is flow using their connection ? Not mine ? Not likely. The problem is that when the RTP starts flowing directly from one NAT to the other, the receiver NAT is caught by surprise. It wasn't expecting any traffic to come from the sender NAT, because there wasn't any connection opened to the sender first. The receiver, not knowing what to do with the traffic, will most likely just drop it. If you had really smart NATs that would "eavesdrop" on the SIP communication, then perhaps they could be aware of the REINVITE and know what to do when a new RTP stream shows up. I don't know of any that do this, though.. and though there may be one, it's probably more expensive than either of us would like to spend. And that assumes that you have control over the NATs and could replace them if you wanted to. Basically you're in the double-NAT hole. Several people have been working on how to get traffic through double NATs, but there is a difference: they wanted to create the connection, from the beginning, going through both NATs. It sounds like in your situation you want two NATed devices to connect to your public * server and then after a call is established communicate directly between themselves. It'll take some creativity, to say the least (I'm avoiding saying that it's outright impossible, because maybe it isn't, but it's also going to be far from easy). Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RC1 - error message : Request to schedule in the past
On Mon, 2 Aug 2004, Areski wrote: > I have made an update to asterisk RC1 - all works well :) > but I am getting all the time an error message: > > Aug 2 13:34:57 NOTICE[15375]: sched.c:221 sched_settime: Request to > schedule in the past?!?! > Aug 2 13:34:57 NOTICE[18446]: sched.c:221 sched_settime: Request to > schedule in the past?!?! > > > I cannot figure out if this error message is important or not and if > there is a real problem ! > > > I found that in the wiki but it s quite poor for me to understand > http://www.voip-info.org/tiki-index.php?page=Asterisk+Request+to+schedule+in+the+past It means "something can't keep up" -- in other words, your computer is slow (like mine.. I get these messages often!). I don't know; maybe it could be due to network congestion, but I'd say it is most likely because your processor+RAM aren't able to run fast enough. You probably notice (if you're on a call while these show up) that the audio doesn't sound very good. Running the console with fewer v's (asterisk -rc instead of asterisk -rc, for example) could help, or simply not running the console at all could also help (use safe_asterisk to start asterisk without a console attached to it). Transcoding (changing audio from one codec to another) puts an extra burden on the CPU. Using the same codec for everything might help. (unless it is a high compression (processor-intensive) codec like g.729) If the poorer audio quality doesn't bother you, then just ignore the message. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding SIP Based Termination
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote: > Hi, > I've had a look at it and the timeout error is what happens straight after the > phone disconnects: > > Aug 1 04:07:13 WARNING[106511]: pbx.c:922 pbx_substitute_variables_temp: The > use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' > Aug 1 04:07:20 WARNING[5126]: chan_sip.c:673 retrans_pkt: Maximum retries > exceeded on call [EMAIL PROTECTED] for seqno 102 > (Non-critical Request) > Aug 1 04:07:23 WARNING[106511]: pbx.c:1924 ast_pbx_run: Timeout, but no rule > 't' in context 'sip' > > > Aug 1 04:10:01 WARNING[109583]: pbx.c:1924 ast_pbx_run: Timeout, but no rule > 't' in context 'sip' > Reliably Transmitting (no NAT): > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP 219.88.229.122;branch=z9hG4bKb552afc0d1c81034 > From: "Sahil Gupta" ;tag=f7e5481bb929c765 > To: ;tag=as269fa212 > Call-ID: [EMAIL PROTECTED] > CSeq: 28952 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > to 219.88.229.122:5060 > > Any ideas on that error? A quick search on google didn't bring up much. 403 forbidden usually means you didn't authenticate correctly to the other SIP endpoint. IIRC, your sip.conf section for your this provider included host, secret, and username. Sometimes you need to use fromuser and fromdomain as well -- sometimes you're expected to identify yourself as [EMAIL PROTECTED] or whatever instead of using [EMAIL PROTECTED] (this is what asterisk will use by default). You would make asterisk identify itself the other way by using fromuser=12345 and fromdomain=siptermination.com in the appropriate section of your sip.conf. Give that a try and let us know what happens.. Another thing you could try would be to make a softphone like x-ten lite, msn messenger, or one of the linux varieties connect to your provider. Sometimes they're a little easier to get working because they don't have so many little things you can tweak. After you have a known good configuration there, you could do a sip debug or network packet dump to see the communication it's making to the provider, and then compare that with what asterisk says when it talks to the provider. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding SIP Based Termination
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote: > exten => _34.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) The r at the end of this line tells asterisk to generate a ringing sound for you to hear. In other words, the ringing you're hearing isn't coming from the far end SIP device. Taking the r out will probably help you get a little closer to the solution.. another thing you can do is turn sip debug on while you try to place the call and see what happens. Watch for responses from the far end sip server, etc. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to start asterisk sourcecode
On 29 Jul 2004, ShanKutti wrote:
> I would like to study the asterisk source code(Program). I dont' know
> from which file i've to start reading the code. can anyone helpme.
depends on what you're trying to do, I guess.. if you want to start at the
entry point of the asterisk binary, then 'grep "main(" *' indicates that
asterisk.c might be a good place to begin.
Greg
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[Asterisk-Users] using round-robin dns for sip registrations
I finally decided to get a little source code dirt under my fingernails tonight and dig through chan_sip.c to understand how registrations are currently implemented. The hope is to perhaps at least seed some ideas about how to make registrations to a server name, which resolves to multiple IPs, either attempt each IP in the order they're returned by dns, or, simply attempt to register with them all. This would be a good place for somebody to chime in: which approach would be better? If the servers communicate among themselves, a successful registration at one server might cause another server to forget the registration previously made there. In such a scenario, registering to all the servers would be a waste of time. I'm leaning toward attempting to register with each server in sequence until we are successful. In reading chan_sip.c, I found the function sip_register(..) which reads register=> lines from sip.conf and copies settings from there to a sip_registry struct. The function does a gethostbyname() to look up the hostname immediately. The IP returned, as well as the hostname, are stored in the struct. Later on, in transmit_register(..), is where the data stored in the sip_registry struct gets pulled out and a registration attempt is made. It looks like the function exits after transmitting the registration, and some other function must get called to either 1: notice that the registration was successful or 2: notice that it timed out. I haven't found a function that does this yet. Anyway, my proposed solution would be that when we handle a timed-out registration, we should do another dns lookup on the hostname. The first idea I hit was that if the IP that comes back differs from the one we already had, then try re-registering at that IP. But with a round-robining dns, we could possibly get the same IP back a second time. We'll probably need to find all the IPs, ala dig, and keep them in a list so that we can cycle through them. But then there's the problem of when to flush the list and regenerate it (so that we don't cache out-of-date IP numbers). I guess this could happen everywhere that a gethostbyname() is now found.. Okay, it's late (here); I'll try to give this some more attention tomorrow. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite to Asterisk through NAT?
On Wed, 28 Jul 2004, programmer_ted wrote: > I have an X-Lite phone on my box and I'm trying to register it with a > remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I > know it's a pain to do because of SIP not working well with NATs, but I > know there are ways to do such a thing...moving the Asterisk box outside > the NAT is not a possibility at the moment. One thing we tried was mmm, a double-natted sip session. Now that's more fun than a person should be allowed to have in a single day. You didn't mention whether you have control over the NATs.. Everybody's favorite, port forwarding, may come to your rescue. It seems that x-lite always uses the same port for rtp (can't remember/find the number just now). Set the xp-side NAT to forward traffic on that port in to the xp box. You'd have to forward in the sip control port as well, I think. Then maybe do a similar thing on the * side (maybe you have to forward a large range of ports, 1-2 (?) on the * NAT?). I could be way off in the wrong ball field, though, so feel free to point out why this might not work. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem
On Wed, 28 Jul 2004, Bartosz Wegrzyn wrote: [snip sip.conf] > Incomming calls still fails. > NO SOUND AT ALL!!! okay, hang on a minute.. is it the call that fails, or is it the audio stream that fails? If the call connects but no audio comes through then the problems are likely to be in an entirely different basket as compared with the issues that might get in the way of a call even getting set up. If incoming calls from broadvoice to you aren't connecting (they don't show up with 'sip show channels') then that's a sign that something prevents * from registering with broadvoice. If it's the audio stream that doesn't come through (does this happen only when you receive a call, or also when you place a call?) then it's probably a nat related issue. I noticed that you have externip set to your host name, not an IP. I don't know whether * is willing to do DNS on a name set in that field.. but since it says externip, after all, I'd try putting an ip there. (that's the only way I've ever tried it.. if names work there too, please say so) Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Workaround for BroadVoice and possibly others...
On Wed, 28 Jul 2004, Chris Shaw wrote:
> exten => _9NXXNXXNXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60)
> exten => _9NXXNXXNXXX,102,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60)
> exten => _9NXXNXXNXXX,203,Congestion()
>
> Any thoughts? I know, I know... UGLY... but it would work I think...
so if I'm reading this correctly.. step 1 will allow a call to ring
unanswered for 60 seconds, and then roll over to t extension. Or, if the
Dial() returns an error because it couldn't reach the host in
broadvoice-out context, then we enter step 102, which tries another
context, and thus another host (this sounds similar to the situation where
somebody prefers to dial out via SIP, but falls back to zap channel if the
SIP channel failed). Finally, play congestion if step 102 failed.
That sounds to me like it has a fair chance of working.
but wait.. doesn't * add 100 to the step (is it called priority?) upon
failure? So shouldn't the numbers be 1, 101, 201, ...?
Greg
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Re: [Asterisk-Users] Source for 9-911 Labels to attach to phones?
On Mon, 26 Jul 2004, John Fraizer wrote: > > That should be > > exten => 911.,1,blah > > and > > exten => 9911.,1,blah > > You don't want to not catch a call when the user is scared to death and hits too > many 1's. > won't you need _ in it (_911.) in order to make it do pattern matching? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP gateway context?
On Sun, 25 Jul 2004, Rich Adamson wrote: > I just started service with Broadvoice.com and everything seems to work. > However, apparently my understanding of incoming sip contexts is less > then what I thought it was. Could someone point me in the right > direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) > > In my sip.conf I have: > [general] > port = 5060 > bindaddr = 0.0.0.0 > allow=ulaw > tos=0x18 ;sets ip tos bits (=lowdelay and throughput) > context = bogon-calls ; Send SIP callers that we don't know about here > context=from-broadvoice > register=303539:[EMAIL PROTECTED]/539 > > [broadvoice] ;this is referenced for outgoing calls to Broadvoice.com > type=peer > username=303539 > this doesn't address your question (I think the other post did) but it anticipates your next question.. Add dtmfmode=general to BOTH the general and broadvoice contexts in sip.conf. Asterisk seems to make an incorrect assumption about dtmf with broadvoice (on calls inbound to your box, that is) unless you set it in the general section as well. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VSP? Looking for advice.
On Thu, 22 Jul 2004, Jason Hartman wrote: > Has anyone tried using BroadVoice for VSP? I have Asterisk configured > for a home office & I've been trying to decide which VoIP provider to go there are quite a number of BroadVoice users on the list. (myself included) You'll find find a lot of posts regarding setting up * with BV if you search through the archives http://www.google.com/search?q=broadvoice+site:lists.digium.com will get you started. Google doesn't always have up-to-the-minute list archives indexed, so you may want to find an alternate source for searching past posts. > Also I'm not sure if BV will support multiple lines. Any suggestions > would be very appreciated. call and talk to them. They know we're out here; I know at least one of their support guys monitors this list and uses * himself. Don't expect them to set up your * box, because they won't, but as long as you're up to the task of configuring things how you want them, they'll provide the service. > I am currently spending around $50 a month with Vonage. My intention look for posts in the archive from Jay Milk re vonage's softphone. He's got some interesting stuff set up with that. a final note.. while searching through the archives, look for messages regarding starting a new topic as a reply to a message (instead of actually creating a new thread by composing a new message) and similar list expectations.. You can choose not to, but your posts would likely be ignored by many of the more knowledgeable people on the list. good luck! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Server gives 403 forbidden
On Wed, 21 Jul 2004, Preeti Gopalan wrote: > [EMAIL PROTECTED] > type=user ; either "friend" (peer+user), "peer" or > "user" > context=default > [EMAIL PROTECTED]; usually matches the > section title > host=172.16.4.79 ; we have a static but private IP address > nat=no; there is not NAT between phone and > Asterisk > canreinvite=no ; allow RTP voice traffic to bypass > Asterisk Usually things are set up as [EMAIL PROTECTED] Maybe your username should be 678, without the host tacked on the end. Also, host= is a way to tell asterisk the ip of the remote machine, not its own IP. Maybe have one more look at the samples in sip.conf.. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] question regarding Asterisk. X-Lite, and firewall
On Wed, 21 Jul 2004, Michael Wang wrote: > How do I change configuration of Asterisk so that phone B can use > aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address? sounds like * is using reinvite to get itself out of the loop and let the phones send RTP directly between themselves. Because of the NAT, this won't work. To prevent * from sending the reinvite, and to keep RTP traffic flowing through *, try using nat=yes and/or canreinvite=no in sip.conf (you choose which section, general or phone-specific) Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to launch asterisk and connect to console. ?????
run safe_asterisk and then asterisk -rc (add v's to your liking). You probably weren't able to connect to remote asterisk because none was running. safe_asterisk is a script which re-starts asterisk in the event that it segfaults, dies, or otherwise implodes.. Greg On Mon, 19 Jul 2004, James wrote: > Any ideas? > > Thanks. > > > > [EMAIL PROTECTED] root]# asterisk -r > Unable to connect to remote asterisk > [EMAIL PROTECTED] root]# asterisk -vgcd > Parsing /etc/asterisk/asterisk.conf > Asterisk 0.7.0, Copyright (C) 1999-2001 Linux Support Services, Inc. > Written by Mark Spencer <[EMAIL PROTECTED]> [snip] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap PoE switches/injectors?
On Mon, 19 Jul 2004, Kevin P. Fleming wrote: > Scott Laird wrote: > > > So $1600 for 24 ports. That's not *too* bad. HP seems to have a > > similar model (2626-PWR) for a similar price. 3com also seems to have a > > 24-port injector for $800. > > I still don't understand why I can buy single-port injectors for $20, > but multi-port models are $30 per port and up. You'd think that having a > single combined power supply and other bits would reduce the cost, not > increase it. officially, a POE capable switch/etc is supposed to do a discovery routine to detect, when a device is plugged into it, whether that device requires POE. Right? And the single-port POE injectors are usually nothing more than two RJ45 packs with a dc power connector, right? That could be the difference in price there: the detection circuitry. Or am I way off? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] The stories people tell to support.
On Fri, 16 Jul 2004, Jeremy McNamara wrote: > How about top posting and improper editing? > > Jeremy McNamara > yeah, that's a good one too. (I didn't write the paragraph below.. Carlos did!) > > > Greg Hill (DIDN'T) wrote: > > > > One thing I have learned is to document the questions, so people can > > help sooner, etc, ah, and also not to hijack threads.. I did it one > > too.. oops > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] The stories people tell to support.
On Thu, 15 Jul 2004, Jeremy Kenney wrote: > I have a cisco 7960G and would like to run in on asterisk I have an issue > though for some reason it has a problem running thru NAT > > Any assistance would be great! Note that you posted your question as a reply to another thread. This action alone is enough to have your post ignored by a marjoity of the list. Try composing a new message (yes, type or copy the list address rather than using reply and cutting out the text) with an appropriate subject line. Also mention some details about your setup and what you have already attempted in trying to get the phone working. But before doing any of this, check the wiki www.voip-info.org and google in general. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Cards in Boxes without Power Connectors
On Wed, 14 Jul 2004, Nik Martin wrote: > Gabriel Millerd wrote: > >>> Is there a magic 'fan card' that has a power out that people are > >>> using? > >> > >> This may work for you. > >> > >> http://www.thermaltake.com/products/subzero/subzero4g.htm > >> > > you lost me, its a processor cooling device. it doesnt provide > > any power that could be used for a digium card. > > This card is AC powered, and has a drive power output plug, normally used > for a case fan. It's drive output could possibly be used to supply the > Digium card make sure it's got nice clean outputs; you could end up introducing ripple and other noise into your system supplies by connecting a card intended just to drive a fan. Also, if the Digium card is wired such that the +5 on PCI and the +5 on the power connector (for example) are wired together, then you could potentially have problems because the two power supplies could try to drive slightly different voltages onto the supply bus and end up toasting something. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as plain PABX in call centre
On Wed, 14 Jul 2004, jurgen wrote: > > Of course you can do this, but you will need a channel bank for all > > those non-VoIP phones you want. You will probably want to use T1 channel > > banks with the TE4xx cards. One E1 coming in plus 2 or three T1 channel > > banks. > > I guess what I'm missing here is *how*. How does Asterisk know what to > do with all the extra buttons on these phones? How does it know what to > do with the LCD screen? Some of the buttons simply send DTMF into the > PABX, but there are other ones that are specially programmed lines etc > etc. Surely there's some kind of protocol involved right now in the way > the phones talk to the PABX. I'm not even sure of what to look for in > this regard. Hang on a minute.. are these phones a normal analog phone (ie could be plugged into the pstn) or are they Fujitsu phones, designed only to work with the Fujitsu PABX? Or, in other words, can your fax/answering/etc devices connect to any port on the Fujitsu, or are there only certain ports they'll work on (and the other ports are for Fjuitsu phones only)? If the Fujitsu phones are standard analog phones then you'll be able to plug them into a channel bank. I'm not familiar with your phones, so I couldn't say what you would be able to do with the LCDs and extra buttons. But if they are not standard analog phones, then you may be out of luck. In that case, the only option I know of would be to keep them connected to your existing PABX and connect that to asterisk. Then additional phones (analog through channel bank or fxs adapter sipura/cisco/etc, or voip phone) could be connected to asterisk. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
On Sun, 11 Jul 2004, Kannaiyan Natesan wrote: > When I call a SIP user, the phone should ring in more than one > extentions. Also more than one phone should be able to register with > asterisk. Right now it is not the case. The last phone which register will > be receiving the calls. This type of situations might be needed in call > centres. I think I understand now what you're looking for. But under an arrangement like this, how will asterisk know when a phone which had registered from some IP has re-registered itself sometime later on a different IP? Such a situation could happen in a dhcp environment. Automatic time-outs may be able to avoid or minimize the impact of something like this. What other difficulties might come up? Although the idea does have appeal, it seems like the increased potential for problems outweighs any inconvenience incurred by modifying a line in extensions.conf. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] xlite "calls not approved"
On Fri, 9 Jul 2004, Jay Milk wrote: > Try dtmfmode=rtc2833 then "sip reload" er, make that rfc2833 instead. :) Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
On Mon, 5 Jul 2004, hank smith wrote: > how would I do this but do it with broadvoice? > I want to give people the oppsion to call my cell phone but I use a voip > carier stay tuned to see how he gets the thing figured out, then change exten => 2000,2,Dial(Zap/1/5551212,10) to exten => 2000,2,Dial(SIP/[EMAIL PROTECTED],10) or similar. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback over Console
On Mon, 5 Jul 2004, Chris Foster wrote: > I'm trying to setup a primitive announcement-paging system in my house > using the line-out from my * box to a cheap amplifier that runs to > speakers on our first and second floors from the basement. I have a > extension that connects to Console, and console is set to auto-pickup. > I'm using alsa drivers. > > This all works great, except for one thing. I want to play a tone over > the console after the console picks up. What i'm doing right now is > calling Playback after the Dial. However, No playback sound or > background sound is being heard over the console speakers or are any > error messages appearing in the command line. I suspect that after Dial has happened the auto-answer connects you to the console, and the call doesn't reach Playback until after the console hangs up. Watch the CLI while you make the call to see if this is what happens (and try issuing a hangup command at the CLI to see whether you get the Playback sent to whatever device placed the call). As for how to do what you're after.. I dunno! Maybe you can find a way to "pick up" the console as if to dial from the console out to somewhere and issue the Playback then. After it completes, hangup and Dial() the original call to the console, with auto-answer turned on. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC
>From the archive on June 18: Date: Fri, 18 Jun 2004 19:23:57 -0500 From: Brian K. West <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] #asterisk is +r now, meaning register your nick with nickserv How do you register? do this /msg NickServ help or /msg NickServ register [yourpassword] You will be required to /msg NickServ IDENTIFY [yourpassword] before you can join #asterisk. I'm sorry we had to do this but the spambots that join and part 100+ times per hour were getting way out of hand. Thanks, Brian On Sun, 4 Jul 2004, Chris HARIGA wrote: > When did U join chat community last time??? > > Take a look... :) > > Chris HARIGA > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Mail > Sent: Sunday, July 04, 2004 11:43 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] IRC > > If you are in windows > http://www.mirc.com/ > > if you are in linux, I use > http://www.xchat.org/ > in fact it has a windows version too > > all you need to do is download it and install it once it's up and running > all you need to do to enter a room is /join #[room name] > > > On Sun, 2004-07-04 at 11:04 -0400, Chris HARIGA wrote: > > Hi, > > > > Can someone tell me how to register and enter in irc.freenode.net chat? > > > > Thank You for your time, > > > > Chris HARIGA > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Asterisk to automatically dialout
On Tue, 29 Jun 2004, Andrew Elchuk wrote: > I'm trying to get asterisk to auto-dail out. I created a *.call file did you create the file in /var/spool/asterisk/outgoing/, or did you create it elsewhere and then move it to that directory? The docs mention that if the file is created in the outgoing directory, * may read the file before you've finished writing it. But if you create the file elsewhere and then move it, the entire contents are guaranteed to be in the filesystem when * finds it. This might explain the difficulty you're having.. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why? oh why can't I dial out?
On Sun, 27 Jun 2004, Vassilis Konstantinou wrote:
> I have been struggling with my Asterisk setup for 3 days now and I think I
> have done well...apart from the small detail that I cannot dial out on my
> phone (PSTN) line.
> [snip]
> The scenario is: if I dial 9123 (for the UK clock) then output from the
> console is:
>
>-- Executing Dial("SIP/5000-96f1", "Zap/1/123") in new stack
> -- Called 1/123
> -- Zap/1-1 answered SIP/5000-96f1
> -- Hungup 'Zap/1-1'
>[snip]
>
> [incoming]
>
> exten => s,1,SetCallerId(${CALLERID})
> exten => s,2,dial(SIP/5000&SIP/5001,10,tr)
> exten => s,3,Voicemail,u1000
> exten => s,102,Voicemail,b1000
>
>
> exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1})
> exten => _9.,2,Congestion
So, assuming that calls from your SIP device are in the same context as
the above extensions, all extensions beginning with a 9 should be dialled
on ${CONSOLE}. On my box, ${CONSOLE}=console/dsp... the sound card. Is
yours set to something similar (or is it really set to dial the zap
interface?)
Not being from the UK myself, I don't know whether the clock's number is
123 or 9123. If it's 9123, then you should be dialing 99123 in order to
get through your dialplan with the 9123 still intact to send to the PSTN.
Greg
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