[Asterisk-Users] Re: GXP-2000 fw 1.0.1.13 and NTP
Leif Neland [EMAIL PROTECTED] writes: My GS BT101 have also developed problems with sync'ing to my ntp-server. I can see, using tcpdump, that the phone asks my server and gets an answer, but the display is not updated. It used to work, but now it usually doesn't, but strangely, sometime it does... Try power cycling the phone. The Grandstreams seem to get flakier the longer they are up. Normally I notice it when they the phones stop allowing incoming www connections. A power cycle always cures it. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Semi-OT: porting numbers away
trixter aka Bret McDanel [EMAIL PROTECTED] writes: Gotta wonder about a company that puts something like that in their contract. My favorite are the indemnification clauses. I count how many things some large company wants *me* to indemnify *them* against. Don't these jokers have a legal budget? Do they think any money I can chip in is going to amount to a hill of beans? In any case why would I want to agree to pay their legal expenses? (I'm not a lawyer so I might be misreading things a bit, but many of them sure seem to be very open-ended in what they want users to indemnify them against. They way I see it, if the user does something wrong that costs a company money, then the company can always sue the user. Indemnification clauses are simply a way to get money from the user even in cases when no court would agree with them that the user did something wrong.)) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Stay away from Grandstream!
Andrew Kohlsmith [EMAIL PROTECTED] writes: Honestly you said it yourself though... they are turning it up too high and pushing the audio beyond what its design specifications are. This is perhaps the fault of the software guys, as they allow you to go beyond what what the acoustic coupling was good for, but then again I am pretty sure they allowed the volume to be increased due to customer complaints of the phones being too quiet. :-) I wonder if these same phones with a decent in-the-ear earphone and a mini boom-microphone would have the same problems. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can you time limit access to a trunk?
David Tillman [EMAIL PROTECTED] writes: I have an issue where I have an extension (that is powered down and disconnected) still connected to a trunk (Sipura SPA3000) na dhas been for 11 hours. First of all, I don't know how it got in that condition. Secondly, is there a way I can limit the amount of time an extension can stay connect to a trunk? exten = 1234,n,Set(TIMEOUT(absolute)=600) exten = 1234,n,Dial(...) I see the same problem here whenever a phone crashes. Asterisk merrily keeps the call up until I shoot it down manually from the asterisk command line. (This is with cvs-head from Nov-1 and sip phones that all can reinvite.) I'm surprised that folks using asterisk for gatewaying into the pstn haven't raised a stink. Having a trunk stuck on a for-pay call can easily cost folks lots of money. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Teliax billing question
from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html The scam isn't new, and its certainly not limited to home 800 numbers. The same basic principles were used by some of the 900 number folks a few years ago as well. My fear wasn't that someone would stuff phony charges on my bill (like charges for 900 calls that were never made). I was more afraid of the case where someone in bad faith war-dials the 800 number so they can collect the 60-cent (???) per call payphone charge. Will VOIP providers let your dispute this charge because the calls were made in bad-faith or is this simply a grin-and-bear it type situation? I understand that within the PSTN there is a 2-bit value associated with the class of phone that the call is placed from (normal, payphone, prison-phone). If voip/pstn gateways started passing this on it might make it easier for folks to guard against payphone scams by configuring their asterisk to only answer the 800 calls made from normal residential phones. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Teliax billing question
Ryan Burke [EMAIL PROTECTED] writes: Is there any other charges because of the toll free number? I was toying with the idea of getting an 800 number too, but the issue of a substantial per call fee for pay-phones calls has me worried. Hopefully someone here can clarify what the deal is there. I've seen numbers quoted as high as a 60-cents for the payphone settlement. from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html Watch Out for New 1-800 Number Scam - An old scam may be cropping up again for consumers with personal 1-800 numbers. Most long distance companies charge subscribers a per-call fee for calls placed from a payphone to a residential 1-800 number. This fee is then sent back to the owner of the payphone. While this arrangement is perfectly legitimate, in 2002, scammers in Berkeley, California found a way to take advantage of the system. They set up a phony payphone company and connect a bank of payphones to an automatic dialer. The dialer then randomly dialed 1-800 numbers until it hit a residential toll-free number. When the call is picked up, the scammer pocketed the 24ยข fee. Thanks to the auto-dialer, they could quickly rack up profits from the scam. By the time the operation was shut down by police, they had netted almost a half million dollars. Reports of a similar scam are coming in and consumers with residential 800 numbers are urged to check their April and May long distance bills for mysterious one-minute phone calls from Denver, Colorado. If you find such a call, be sure to contact your phone company. For more information on this scam, click herei. (Thanks to ConsumerWorld.org for this tip.) WIRELESS WATCH -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Meetme and Sipura SPA-941 - bad jitter/distortion
Ryan Booz [EMAIL PROTECTED] writes: Now, however, there is a (very) slight echo introduced into any calls made to this extension. So obviously the way that the phone sends packets is causing some issues. Anyone have a resource or guide to point me to on best way to debug packet transmission for good calls? Are you sure the echo isn't acoustic echo from the handset itself? Its older sibling, the SPA-841 was really bad in this regard. On a purely sip call between two SPA-841's, if you bumped the earphone gain past halfway on the display the other side would invariably complain about the echo. I always wanted to fill the Sipura handset with modeling clay and see if that helped things any. (The echo was only a problem on direct sip-to-sip calls. Any calls going into the PSTN seemed to always be processed by an echo-can, so it wasn't noticed there.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipura Vertical Service Activation Codes
Do the Sipura Vertical Service Activation Codes have any meaning to the phone itself? It doesn't seem like they do anything, but that leaves me with the question why are they listed at all? I'm trying to reconcile asterisk's idea of some features like group pickup being on *8 with the sipura's desire to have it on *37. Which one is more common these days? Can I just make an extension and assign the pickup code to it? exten = *37,1,pickup(SOMETHING_TBD); -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Linksys SPA-841 Missing Calls
Dave Morrow [EMAIL PROTECTED] writes: Hi all, I have been plagued by an issue with my SPA-841 phones. The issue is that frequently, usually after a period of inactivity on the phone, an incoming call will be missed by the phone. The call works, cause the caller ends up at voicemail, but the phone never rings. I've managed to trap one of these missed calls in Asterisk, the log is below. Can anyone make sense of it? Might the SPA-841 be crashing and rebooting? With the current firmware (v. 3.1.4) I often see my phone hang and flash all its lights in the reboot pattern if it is the first time I've used in a long time. Often just trying to dial out is enough to push it over the edge. Sometimes on incoming calls it manages to hang in there for a minute or two and then it crashes in the middle of the conversation. (And no, asterisk doesn't clear the call until after the spa841 reboots.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Linksys SPA-841 Missing Calls
Might the SPA-841 be crashing and rebooting? With the current firmware (v. 3.1.4) I often see my phone hang and flash all its lights Really? For me the 841 is a quite stable phone. Out of the 15 we have in the office neither one crashed in the past 3 months. And they are used heavily. The phone has weaknesses, but stability in my opinion is not one of them. Phone info: Software Version: 3.1.4(a) Hardware Version: 1.0.0(1813) Elapsed Time: 50 days and 09:48:10 I only have 1 phone so it is hard to tell if the crashing is a hardware or software problem. I never noticed the phone having problems previous to this. I did resync asterisk to HEAD a month ago. Thats also about the time the phone started crashing (or at least I started noticing it). Come to think of it, I've been running the current firmware in the phone since July 20th. The only think that changed in recently was asterisk. I wonder if there is something the newer asterisk is doing that the phone really hates... Asterisk CVS HEAD built by [EMAIL PROTECTED] on a amd64 running OpenBSD on 2005-11-02 00:58:42 UTC Software Version: 3.1.4(a) Hardware Version: 1.0.0(700b) Elapsed Time: 1 day and 05:54:03 (crashed during a call) People have been reporting a finicky ethernet connector, so maybe that is the reason the phone does not answer to any traffic? Yea, this phone has that problem too. ;-) Some cables just don't work. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sip URL peering
Klaus Darilion [EMAIL PROTECTED] writes: It's not that easy. If you want to have open SIP URIs (just like email is open for everybody) you will receive SPIT calls. E.g. the SPEER group tries to define rules for VoIP peering which allows authentication to enable open SIP URIs. (I won't open acces to my SIP URI if I can not verify the senders URI). Keeping spam in mind seems like a really good idea. I'm also a big fan of keeping a cryptographic paper trail so that one can figure out who spammed. On the other hand, is SPAM / SPIT a big enough problem at this point to warrant scuttling any interconnectivity? It seems a bit premature to worry about a problem that may not develop for 5 years and allow that fear to stop direct sip dialing. As an amusing aside, I inadvertently added a captcha to my phone line when I had the local number go into an IVR that asks the caller to press 1 for person XXX and 2 for person YYY and 3 of they are a telemarketer. I don't think anyone other than my friends has ever pressed 3, but the predictive dialers used by the phone-spammers doesn't seem to pass the turing test and isn't able to press 1 or 2. ;-) I see lots of timeout-hangups in the IVR with caller-id's like CAR PROMO or VOIP CALL. If spam/spit is ever a problem, I'm simply routing previously unseen calls to a turing test of the same type and anyone that has previously called (and/or been called) gets to bypass the turing test. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: sip URL peering
Wolfgang S. Rupprecht writes: If there is enough interest, maybe the greater asterisk community could adopt some semi-official mapping tables. I'd be willing to periodically generate a flat mapping file and an extension.conf dialplan snippet from sipbroker's list or whatever else is deemed more neutral or useful if there was any interest in such. Just to try to get the ball rolling, I put together an asterisk config file that allows folks to direct dial other open sip servers using the same prefix codes as sipbroker. Sipbroker also encourages folks to add listings for their sip servers, so in theory everyone here could join in the fun. I'll update these files periodically, so they should track sipbroker's web page as folks add themselves. http://www.wsrcc.com/wolfgang/ftp/exten-peers.conf (asterisk conf file) http://www.wsrcc.com/wolfgang/ftp/sip-peers.txt (raw mapping file) http://www.wsrcc.com/wolfgang/ftp/dial-out.conf (dial-out macro) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: sip URL peering
Klaus Darilion [EMAIL PROTECTED] writes: There is a new ietf WG to come which deals with peering issues. It's called SPEER (formerly VOIPEER) The list archive is at http://darkwing.uoregon.edu/~llynch/voipeer/ minutes from last ietf meeting: http://www3.ietf.org/proceedings/05nov/minutes/voipeer.html It looks interesting, but these things always seem to be scuttled or reduced to glacial progress by the telecom interests. VOIP peering isn't something that should require years of meeting to make happen. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: sip URL peering
Patrick [EMAIL PROTECTED] writes:
Shouldn't the last line in exten-peers.conf be:
exten = _**XXX.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])
^^^
Similar to the previous line sipbroker line:
exten = _**999.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])
Thanks for looking this over. Extra eyes always help.
The last line is a bit different from the ones above it. In the
normal case the ${EXTEN:5} was meant to strip the 5 chars in the
routing prefix **999 and only pass on the base number to the
remote sip server.
The catch-all sipbroker line is meant to have the 4 of those 5 chars
passed off to sipbroker so that they can examine the routing prefix
and route the call. This should only happen for the prefixes added
between the time I last updated the file and whenever a new prefix was
added to sipbroker. The reason the first * needs to be stripped is
that sipbroker wants to see the prefix codes as *999, with only one
*. Asterisk along with my Sipura phone seem to use *XX codes for
their own purposes and I didn't feel comfortable enough putting the
dial prefix codes in potentially clashing real-estate. (Comments
suggestions are very welcome. I've got very little telco/telecom
experience and am just winging it.)
The one thing I think I do have a minor error on is in the dial-out
macro. I copied it from somewhere, but the last s-. line looks more
wrong the longer I look at it. I think it should really be _s-. and
not s-..
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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[Asterisk-Users] sip URL peering
One thing I haven't seen get much airtime on the digium lists is sip
URL-based peering. I imagine many of us have far more asterisk
extensions than PSTN numbers. It would be really nice to be able to
do something like call [EMAIL PROTECTED] from [EMAIL PROTECTED] It
looks like all or most of the pieces are in place, but I don't see
folks discussing it much. Is no-one else interested in this?
One group that seems to have an ever growing list of sip servers that
accept direct incoming sip calls is sipbroker. Using their service
doesn't really buy the average asterisk admin much, but they do have a
nice list of sip servers and they do assign a unique prefix code to
each server which might be useful to snarf into an asterisk database.
http://www.sipbroker.com/sipbroker/action/providerWhitePages
extensions.conf:
;; send everything else with a ** prefix to Sipbroker
;; strip one of the stars since they only want one in total.
;; http://www.sipbroker.com/sipbroker/action/providerWhitePages
exten = _**XXX.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])
sip.conf:
;;; outgoing to Sipbroker
[sipbroker-out]
type=peer
host=sipbroker.com
Is sipbroker just a well-kept secret from the asterisk crowd, or is
everyone else using asterisk for phone spamming from call centers and
the last thing they want is folks to be able to call them back and
give them an earful over disturbing their dinner?
-wolfgang
--
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[Asterisk-Users] Re: sip URL peering
Chris Hills [EMAIL PROTECTED] writes: Perhaps you would be interested in TRIP (telephony routing over ip)? Each organisation can apply for an ITAD number, just like a domain. TRIP numbers take the form extension*itad, for example, 1234*222. As you can no doubt surmise, TRIP numbers can be dialled from a regular telephone handset. For more information, please see the following documents:- http://www.iana.org/assignments/trip-parameters http://www.ietf.org/rfc/rfc3219.txt Thanks. I'd entirely forgotten about that. Having the routing numbers coordinated by some benign central authority like the IANA is certainly preferable to some enthusiast web site which might or might not be around in a year. Having just skimmed RFC 3219, it seems to add quite a bit of hair to what is essentially just assigning an N-digit prefix (or suffix) to every cooperating SIP server. I'm sure that must have some advantages for whatever situations they where concerned about, but I don't really get it. SIP already does all the routing and redirect internally so having the redirecting done at the top level seems redundant. Ideally, for me at least, would be a simple ascii list in the style of /etc/services or /etc/protocols that had an official mapping of dialing-prefix and sip-server name (or domain name with _sip._udp SRV entries). If there is enough interest, maybe the greater asterisk community could adopt some semi-official mapping tables. I'd be willing to periodically generate a flat mapping file and an extension.conf dialplan snippet from sipbroker's list or whatever else is deemed more neutral or useful if there was any interest in such. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: Linksys SPA941
Patrick [EMAIL PROTECTED] writes: Too bad they have only one Ethernet port. I read on the review at http://voipspeak.net/index.php?option=com_contenttask=viewid=41Itemid=27 that the next 942 model will have a second Ethernet port and PoE support. Other than that it looks like quite a nice phone. Polycom's IP301 featureset beats the SPA941 on some points as it does have dual Ethernet ports and PoE support but it is limited to 2 lines. Anyone with first hand experience with both phones want to share their thoughts? The thing that bothers me the most about the spa941 is that it still doesn't have any wideband codec. Even the low-end Grandstreams can do 16kz audio (which sounds really nice when you talk between two of them with reinvite). -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 alternatives
[EMAIL PROTECTED] (Bruce Komito) writes: If you're going to promote your product, you might consider making sure your web site is up, before giving out the URL. And he could also lose that flash animation when promoting to an opens-source/linux audience. The fordvoice web site has a big blank blotch where I assume some information presented in flash format would go. Not exactly effective marketing... -wolfgang ___ 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] Why even have set CallerID option?
[EMAIL PROTECTED] (Matthew Boehm) writes: Why even have the ability to set callerid name/number if end offices don't honor it? VOIP is bigger than just PSTN-gatewayed calls via some specific company. The end goal is to connect the VOIP islands directly. That is already happening at some large companies where they call their supplier directly on a purely voip link. For a concrete example look at the sip-edu program. It is a growing group of universities that exchange SIP calls directly. (Some even have their asterisk and SER config notes on line.) In all cases the caller's calling-number and calling-name stuff will get passed to the callee. http://voip.internet2.edu/SIP.edu/ -wolfgang ___ 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] SIP callid
[EMAIL PROTECTED] (Chuck Ramirez) writes: Looking at the source code I noticed that rand() is used four times to get a callid. Is that safe enough? RAND(3) OpenBSD Programmer's Manual RAND(3) NAME rand, srand - bad random number generator ... Is there some reason asterisk can't just read some bytes out of /dev/urandom? -wolfgang ___ 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] Registration issues with Sipura SPA-841
[EMAIL PROTECTED] (GliTcH) writes: I'm trying to investigate going to a different manufacturer, but I don't like the Cisco ATA-186's very much and they're too pricey, so I don't know where to go next. voipsupply has a pretty big collection, maybe I'll order 1 of each for testing. Grandstream? While my Grandstream BT-101's may need a periodic reboot to stay happy, you can't beat the sound quality of two grandsteams talking g722-wideband between themselves. It is the setup I use to show folks how good a phone call can sound once you get away from 8khz-ulaw. -wolfgang ___ 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] VoIP service through Asterisk?
[EMAIL PROTECTED] (dean collins) writes: You will be very disappointed at the call quality if you try and run other software on an asterisk box, pc interrupts and processing glitches just don't 'play well' with voice. This isn't that much of a problem if you structure your phone system to be reinvite clean. In that case you can let asterisk set up the talk path to be directly between your phone and the upstream PSTN gateway. Minor scheduler delays then aren't going to cause any audio hickups. -wolfgang ___ 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] Registration issues with Sipura SPA-841
[EMAIL PROTECTED] (Iassen Hristov) writes: Anyone having problems with registration to * from a SPA-841? I have a spa-841 (firmware 3.1.1a) on my desk right next to me and it registers just fine w. ~current asterisk (CVS-HEAD-03/16/05-08:43:40). The one problem I do notice that the phone is very touchy about the cat-5 ethernet cable one uses. The one that came with the phone works just fine (of course). Any of my 5 longer store-bought cat5e cables don't work at all. Tcpdump shows the phone registering and asterisk answering, but the phone never hears the reply. Might you be seeing something like this? -wolfgang ___ 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] About the weather..
[EMAIL PROTECTED] (Steve Prior) writes: The recorded prompts by Allison are more in line with the very language structured text forecasts typically seen by pilots There are home weather stations with computer interfaces that simply tell you the current stats (temp, pressure, humidity, wind direction, wind speed, rainfall rate). Converting this information to something that could use the asterisk sound snippets wouldn't be that hard. -wolfgang ___ 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] adding to asterisk db from a script
It looks like asterisk isn't honoring EOF on stdin. file add-phonelist: database put cidname 200551234 name 1 database put cidname 200551235 name 2 database put cidname 200551236 name 3 database put cidname 200551237 name 4 asterisk -rn add-phonelist What I see is an infinite stream of prompts as asterisk is banging onto the EOF. How is one supposed to add a bulk list of clid names to asterisk? -wolfgang ___ 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 E911?
[EMAIL PROTECTED] (Kevin P. Fleming) writes: Well, incoming call handling on SPA-3000 kind of sucks at the moment... but I don't see how it could be configured to ring a bunch of phones anyway. At best it can deliver the call to a single gateway/proxy, and even it really wants to answer the line first and present a second dial tone to the caller before doing so. If the call is just going to go into an Asterisk server for routing to ring all the phones, then IMO it'd just be easier to skip the SPA-3000 completely. It isn't hard to make an incoming POTS call to the spa-3k ring all the phones. 1) set up an extension that rings all the phones. exten = ,1,Dial(sip/6001sip/6002sip/6003sip/6004..sip/N) Have the spa3k use an S0 dialplan: PSTN Line: ... Dial Plan 8:(S0 : ) ... PSTN Caller Default DP: 8 Too bad the spa3k software lacks a way to save a dialplan or to diff it against factory defaults. It makes it kind of hard to post a definitive recipe. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ ___ 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: DISREGARD!![Asterisk-Users] Broadvoice outgoing problems
[EMAIL PROTECTED] (Brian Dingman) writes: I doubt that was the problem. I would be interested in hearing what else you did besides that to get it working. If he had a bad entry in the /etc/hosts file that could have been a problem. The hosts files does require periodic maintenance. Automatically checking the entries in /etc/hosts against dns once a day and mailing gripes to the admin wouldn't be a bad idea. -wolfgang ___ 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-841 Problems
[EMAIL PROTECTED] (LES.NET 1996 INC.) writes: Yes, I upgraded some prior to the problem. it seems to affect both versions of the firmware. But you cannot upgrade them after they lock up. I don't know if this is related, but I couldn't get my sipura spa-841 working using any of the half-dozen store-bought cat-5 patch cables I had laying around. It just refused to register. Tcpdump confirmed that packets were coming from it, and we answered, but it never heard us. Just out of randomness I tried the shorter enclosed cable that came with the spa-841 and would you believe that it started working? As far as I can tell, the rj-45 socket on the phone is just a bit non-standard and the wires just don't make reliable contact to the spades on the cable. It isn't a case of some of the wires in the socket being bent, they are all straight and look normal. All I can think is that the contact wires have a slightly higher than normal angle and end up hitting the plastic lip of the rj-45 plug instead of resting on the gold spade contacts. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ ___ 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] VoipJet Terms of Service
[EMAIL PROTECTED] (John Goerzen) writes: I've heard good things about VoipJet here, so I was going to set up an account. Then I noticed their Terms of Service here: https://www.voipjet.com/tos.php Ignore them. There are plenty of players and you don't need to deal with one that has NDA clauses or indemnification clauses in their contracts. (My favorite is the latter, where they ask you to pick up their cost of fighting off lawsuits even though they may be their fault and/or not even involve you. Thats real chutzpah.) -wolfgang ___ 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 configuration changes for Outbound
[EMAIL PROTECTED] (Dan Weber) writes:
On Sat, 5 Mar 2005, Wolfgang S. Rupprecht wrote:
Does broadvoice participate in e164.{arpa,org,info}?
Yes
Does this change mean that non-customers can't call broadvoice
customers with a pure SIP call by routing the call to
sip.broadvoice.com?
Calls can be made to broadvoice phones by phonenumber@sip.broadvoice.com
(From a security standpoint what is the difference between calling the
BV customer directly vs over the TELCO lines? Perhaps I'm missing
something, but better/cheaper/faster to cut out the telco middleman.)
Much cheaper over internet vs. telco.
That's great news! I had a sinking feeling when I heard the words
authenticated invite.
Unfortunately some large voip companies (cough cisco) are locking down
their sip servers to only talk to established peers. Perhaps I'm
missing something crucial, but these companies still have DID numbers
for their employees, so locking down the sip server just forces the
call to go out via the PSTN.
So are BV customers listed in the in e164.org dns zone (or some other
publicly accessible routing database)? I would love to have some way
to bypass the telco when calling friends without having to put a
by-hand entry into asterisk for each person that can accept direct
calls via some voip proxy.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
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Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
[EMAIL PROTECTED] (Gabriel Gunderson) writes: As an early adopter kinda guy, I'm happy to tweak stuff to make things work. I can't however explain to my wife why the phone doesn't work *again*. I'm going to hang in there a bit longer in hopes that things will get better, if they don't, it's off to another VSP. You might want to get a backup, pay-as-you-go, provider to help cover this situation. (I use both teliax and gafachi. When dialing out, if one is down asterisk transparently rolls over to the next one. They are both 2cents/minute so the cost is the same and the ordering in the dial-plan is mostly determined by their ping times.) Too bad not many providers do a database dip into e164.org. If they did, I'd still be able to get some calls when my DID provider was down. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ ___ 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 configuration changes for Outbound
[EMAIL PROTECTED] (Dan Weber) writes:
I was doing the best I could to get the situation under control. This
really was out of my hands, but I'm trying to repair as fast I could.
If you haven't received the email regarding the change yet, please
notify me.
Does broadvoice participate in e164.{arpa,org,info}?
Does this change mean that non-customers can't call broadvoice
customers with a pure SIP call by routing the call to
sip.broadvoice.com?
(From a security standpoint what is the difference between calling the
BV customer directly vs over the TELCO lines? Perhaps I'm missing
something, but better/cheaper/faster to cut out the telco middleman.)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
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Re: [Asterisk-Users] e164.org and FWD now have peering arrangement
[EMAIL PROTECTED] (Duane) writes: There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990fwd number FYI: FWD shows a different inbound prefix: **164 e164.org8781039311 Is there enough spare numbering space there for you to assign e164.org dialable numbers to people in the asterisk community too? While it might be nice for asterisk home users to have their single DID listed, it strikes me that the real utility would be to have a blocks of 100 or 1000 numbers assigned to folks, so they could have each of their voip phones directly dialable from anyone that queries your db. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ ___ 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] List tips for new subscribers
[EMAIL PROTECTED] (Andrew Kohlsmith) writes: You did type it yourself, but you replied to a message in a thread and erased everything, thus screwing up the threading. I think that's what he was referring to. Wouldn't it be nice if the mailinglist software were hacked to enforce some rules? * reject all HTML email * reject any mail with more quoted text than original text * reject any mail that starts a totally new subject but threads to a different unrelated one. eg. has references, but new subject with no (was: oldsubject) * reject any mail that has re: or the same subject line as other msgs, but no references. (This one needs to be done very carefully.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Microsoft xbox power cords: Finding innovative new uses for our blue-screen technology. ___ 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] Re: Encrypted VOIP?
[EMAIL PROTECTED] (Randy Bush) writes: Just run point to point encryption over a vpn. Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious. classic problem. how do you know, in a way that the application and user can see it, that the data are on a crypted channel? this is a problem in general with all the rfcs which say for privacy, run it over ipsec. there is no signaling from the transport to the app. Isn't this just an API problem? Shouldn't the kernel be able to tell the user app that a socket is associate with an ip/esp encapsulation? (And yes, I know that one of the common ipsec implementations strips the ipsec headers and then sends the unwrapped packet back through the IP machinery a second time.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Hate software patents? Sign here: http://thankpoland.info/ ___ 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] Encrypted VOIP?
[EMAIL PROTECTED] (Steve Blair) writes: Some SIP phones support sRTP. I know Zultys claims to but I have no real experience in this area, yet. Our installation is at the point where this is very likely the next issue to be addressed. In theory, the Sipura line supports SRTP. I've got both a spa-841 and a spa-3000 that have config areas for loading the srtp rsa keys. Unfortunately there isn't enough information given by sipura as to how to generate these rsa keys. (eg. can one use an openssl generated key?) It would be great to have asterisk interoperate with the sipuras. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Hate software patents? Sign here: http://thankpoland.info/ ___ 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] Grandstream stops working after Register
[EMAIL PROTECTED] (Andrew Braae) writes: I was hoping someone can help with a problem with my GrandStream Budgetone hanging after a while. Well, they aren't called Bugtones for nothing... Under some of the early firmware loads the phones needed to be power-cycled every few days. If you haven't updated your firmware to the current version, you might want to do that first. It might save you quite some time. http://gs-firmware.gratissip.dk/firmwares/latest/ -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Hate software patents? Sign here: http://thankpoland.info/ ___ 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] RE: Answering Machine Function?
[EMAIL PROTECTED] (Jason Kawakami) writes: Is it in principle possible to create a dialplan that allows prefix-free dialing to an outside line, and move all the PBX-like features behind some special prefix? i.e. recognize 3, 7 and 11 digit numbers as phone numbers and dial them without further ado, and put voicemail and every other PBX-ish feature behind, say #? I don't know if you even need to work that hard to hide the pbx numbers. I just grab 6XXX as pbx-local numbers and pass all the rest to the outside world. The slight downside is that if someone is dialing a 7-digit or 10-digit outside number that starts with 6 and they dither a bit after dialing 4 digits, it will end up calling the corresponding inside number. That hasn't been a problem in practice. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Hate software patents? Sign here: http://thankpoland.info/ ___ 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] Re: Best Grandstream firmware to use?
[EMAIL PROTECTED] (Aldo Bergamini) writes: [EMAIL PROTECTED] is believed to have said: http://fm.grandstream.com/gs/ Thanks! And thanks from here too! I've been annoyed at the non-working message button from 1.0.5.16 for about a month. It is nice to have that working again. In case anyone else is trying to use the version 1.0.5.16 HTTP method of upgrading the firmware, don't bother. It doesn't work. I couldn't go from *.16 to *.20 until I went back to using TFTP. The files get loaded by the grandstream, but it never seems to burn them into eeprom. As soon as I set up the tfttp server again and changed the GS's config, it loaded and burned just fine. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Hate software patents? Sign here: http://thankpoland.info/ ___ 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] Re: VoIP SPAM, what's next ?
[EMAIL PROTECTED] (Loek Gijben) writes: hank [EMAIL PROTECTED] wrote: voip spam? I have never gotten any yet. It's is just waiting for the first one to arrive.. The mechanics are just too appealing for spam-like businesses. I got one the other day, but it turns out it was a buddy trying out his skills at generating UDP from a shell script. I figure if voice spam gets to be a problem I'll simply use a whitelist arrangement where some aspect of the caller is looked up in an asterisk DB. Callers in good standing get to ring the phone. Others go to a voice-menu tree that asks them to press a certain key if they are a telemarketer, or calling for a political party, or collecting for a charity. They will then get a canned message to please put us in their do not call list. All other callers are encouraged to press a different key to ring through to me. Unlike email, phone calls are interactive and sorting the robo-caller from the real people shouldn't be hard. The only thing bugging me is, is there a law that would prevent a telemarketer from lying and pressing the key for I am not a telemarketer. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] Interesting catalog: Viking Electronics
[EMAIL PROTECTED] (Jay Milk) writes: Make a sign -- I've been trying train my mail-carriers to use the DOORBELL and not just knock. Geez, people, what does it take?? Why not get one of those dual PIR (Passive Infra Red) / microwave alarm sensors? They cost ~$50, run off of a 12v supply and have a relay closure for output. Just wire it to the dial button on your door phone and you'll have a CDR record of when the delivery people dropped off a package even if they have an aversion to using the doorbell (which mine also have). Come to think of it, maybe a cool but simple hack would be an asterisk driver that took 8 relay closures via the parallel port and could initiate a different pre-canned phone message for each of them. Eg. Water has been detected in the computer room, There is smoke in the computer room etc. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] Problems loading chan_h323 on Opteron 64 bit
[EMAIL PROTECTED] (Roger Schreiter) writes: I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. ... Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi I run asterisk on my amd64 under openbsd and don't see this. I take it this is something that only pops up when you actually try to connect with an H323 client? What version of gcc and ld are you using? Could it be there are were some symbol-table bugs that got fixed in the mean time? gcc version 3.3.2 (propolice) GNU ld version 2.14 20030612 In case you are curious, these are my openbsd/amd64 patches: http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ ___ 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] FCC Rules VoIP Must Be Tappable
Me raises his hand. All in favor of IAX with native encrypted tunneling say Aye :-) Now I'm likely in the target rings of Big Brother :-) If the voice data passed through a service provider run asterisk system, I'd imagine they'd just get a court order to force IAX encryption to be turned off. (Or try to pull some strings if the service provider was in a foreign country.) The question I have of this ruling is does this make end-to-End RTP encryption illegal? Ditto for re-invites that cut out all the middlemen? How are they planning in getting the two endpoints to stop encrypting things without tipping off the same two endpoints? What about VPN tunnels? Are they illegal now by the same logic? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] FCC Rules VoIP Must Be Tappable
[EMAIL PROTECTED] (Greg Blakely) writes: Does the FCC honestly expect that criminals are going to stop using encrypted point-to-point VOIP connections just so that they won't be breaking the law? Yeah, right. I'm sure they'll all erase their encrypted IM clients so that the FCC will be happy. The way I see it, just like the phone and cable companies the Three-Letter-Agencies really hate doing truck rolls because of the cost (and risk). They would much rather punch a few keys in the privacy of the sub-sub-basement of the J. Edgar Hoover Building. This is nothing other than a cost-saving measure. The hard jobs will still have to be done by sending people out into the field. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] VoIP experiences with Cable and DSL
[EMAIL PROTECTED] (Chris) writes: The thing that really kills you on the ISP end is RED... it may be great for large traffic but it just KILLS voip... and there's not thing 1 you the customer can do about it... :( Interesting and somewhat disheartening. RED was really meant to put back-pressure on the protocols that understand a delicate touch, such as modern TCP. Trying to push back on UDP seems a bit pointless. I wonder if collective cry of the voip users can get the RED implementors to avoid whacking UDP packets until things get really dire (say when drop rates go past some magic number like 5% or 10%). -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] [RANT] Today's possible problems with Broadvoice????
[EMAIL PROTECTED] (Andrew Kohlsmith) writes: That's not asking ofr help. He knows his system hasn't changed. He knows that BV is up and down like a bride's nightie. Hey, without this thread you wouldn't have gotten to use that great line. Thanks for the chuckle! The thread is no longer about help. It's a bitch and moan and status report on BV. Actually there are a few asterisk issues mixed in there along with a VIOP service provider that has technical pains. The fact that asterisk doesn't periodically re-check or deal with multiple IP's handed out by DNS and register with all of them is an asterisk problem brought to light by the BV configuration. The other problems are more VOIP provider issues, but one that asterisk newbies might not realize would be problems before they put their money on the table and try to get asterisk running with a particular provider. Here is my checklist that I will use to evaluate future VIOP providers. * Does the provider give you the raw settings so you can use a generic SIP device (such as asterisk)? * Can the VOIP provider assign you a local telephone number or will people be forced to call long distance to reach you? * Does the VOIP provider have an 8-bit clean ulaw path or is one forced to use their choice of compression? * Does the VOIP provider try to hide their ping times by filtering ICMP echo-requests and/or echo-replies. # Does the VOIP provider filter all ICMP's at the border router because of some security issue involving copious hand-waving. * Does the VOIP provider have SRV records setup for _sip._udp.company.topdomain ? (Shame, shame on any provider that can't take the minute or two to add those DNS records!) * Does the VOIP provider have forward and reverse DNS entries in place for all machines that send packets to customers? * Does the VOIP provider have low delay times (say 20ms - 40ms) for SIP-pings to their server? (Asterisk issues: Asterisk could really have a yellow/orange/red alarm system for indicating when delay times are heading upwards and user's will notice and complain.) * Does the VOIP provider force you to mung your native SIP address and make it impossible for your SIP device to re-invite and cut out the delays associated with forwarding all packets through their server? * Does the VOIP provider allow you to inject your SIP name and number on a per-call basis? (Eg. Can family members have the sip name aka caller id, indicate the real calling party's name?) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
[EMAIL PROTECTED] (Rich Adamson) writes: Like *, it also has an internal dialplan, however understanding the various interactions requires some experimentation, as each of the interfaces seem to be considered a gateway, and part of the dialplan directs calls to gw0, gw1, gw2 (etc) which correspond to physical interfaces in most cases. I felt some pangs of guilt turning all that stuff off, but I couldn't think of any time I'd want two dialplans in series. The box was truly targeted for the residential user where existing phones interface on one side, the pstn line on the other side, and the default call is sent to the voip interface. Disconnected (or failed) ethernet results in a relay flipping, tying the fxs directly to the fxo. Same with power failure. Nice. I think the cut-through from the fxs to the fxo (and backwards) is via a digital connection. In normal use you appear to end up getting hit by the digitization delays. As far as I can tell the relay cut-through is only used for power failure. Initial tests did not show any signs of echo, very good volume and audio quality, and would probably be a good choice for small quantities of pstn lines (particularily soho and residential users). I still notice some low-volume problems with FXO-asterisk-grandstream-bt101 even though I bumped the FXO incoming (and outgoing) gains to +12dB. (To keep calls from the FXO-asterisk-FXS a reasonable volume I needed to drop the gain of the fxs port to -15 (from the factory of -3). Somebody with a real phone VU meter needs to have a look at the Sipura-3000 FXO. I can't believe it is off that much. Might the Grandstream BT-101 be really low in volume and I'm just mistakenly blaming the volume problem on the Sipura? The only downside I've seen thus far (not much experience as yet) is that * calls to the pstn line are cut through immediately, so one hears the initial dialtone from the pstn and the sending of the dtmf tones on all outgoing calls. Kind of annoying, but there might be some config option to handle it; I've just not found it as yet. (If anyone knows how to handle that, sure would appreciate a suggestion.) Given the choice between hearing dead air and hearing the tones, I think I'd rather hear the tones. At least I know something is happening. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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 problems again
[EMAIL PROTECTED] (Olle E. Johansson) writes: The easiest first-level hack would be to randomly choose on of the SRV records provided they have the same weight. One of the other posts mentioned their ATA that simply registered with all the addresses. I don't think it would be a big or difficult change to have asterisk register with all the addresses also. I'm not sure what the right thing for outgoing is, or if it is even possible to have asterisk try all the sip servers in parallel, and then blow off the ones that are late in replying. That sounds like a much more involved hack. (I'll try to hack the registration issue here and post some GPL-ed patches if I get it working.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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 problems again Attn: James
[EMAIL PROTECTED] (James Jones) writes: you can not ping that address because ICMP is turned off. Do you mean *all* ICMP is turned off or just icmp-echo-request / icmp-echo-reply? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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 problems again
[EMAIL PROTECTED] (Rich Adamson) writes: However, for the last hour or so their site has been unreachable with an icmp destination unreachable coming from 199.232.42.62, which belongs to Cambridge Entrepreneurial Network in Quincy MA. Would guess either someone upgrading hardware or a failure near broadvoice. I am having the same problem with sip.broadvoice.com. My asterisk /var/log/asterisk/messages has over 1200 lines of gripes about them starting at Jul 25 16:49:40 (PDT) and continuing to the present. Does broadvoice have a status page somewhere with real info on it. (Like what is causing this extended outage?) Ob-asterisk. I should really see how hard it would be to hack asterisk to register with all addresses if a hostname has multiple aliases. It seems that some of these outages could be weathered if asterisk were to keep tabs on all the sip servers a provider offered, and then actively uses whichever one was up and had the lowest round-trip-delay. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] spa-3000 review?
(Dameon D. Welch-Abernathy) writes: Wolfgang S. Rupprecht wrote: Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? What I discovered via tcpdump was that the Asterisk box wasn't responding to the authentication request for whatever reason. I couldn't get it to work until I upgraded to the latest CVS release. Once I did that, I could do it with authentication. Interesting. I'm at -current +/- a day and do see a NAK/retry-with-md5 exchange when I do a sip debug. The md5 authentication is also NAK-ed. My fear was that it was expecting the calling user to use their own username in the validation instead of asterisk using the shared secret with a shared user-id. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] spa-3000 review?
[EMAIL PROTECTED] (William Suffill) writes: Seems quite interesting. Any suggestions of where to order one and about how much? Mine was $125 from www.voxilla.com. I ordered it on Sunday and had it in my hands on Tuesday. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] spa-3000 review?
[EMAIL PROTECTED] (Tom Neville) writes: ; FXO port - Line from our office PBX. [40] ... secret=NOPE Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? I can only dial out when I tell the SPA-3000 to use no authentication. Eg: admin-PSTN Line-VoIP Caller Auth Method-None Changing it to the following doesn't work (adapting the example to use your values from above): VoIP Caller Auth Method: HTTP Digest(their name for MD5 digest) ... VoIP User 1 Auth ID: 40 VoIP User 1 Password: NOPE Turning on sysloging on the sipura wasn't informative at all. (All I got was a bunch of lines like this: Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Etherdump also showed quite a few invalid syslog lines coming from the sipura. Mostly they were missing the local0.debug. Some went to local2. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] Re: OT: saving/restoring sipura config
[EMAIL PROTECTED] (Dameon D. Welch-Abernathy) writes: On Tue, 2004-07-13 at 19:10, Wolfgang S. Rupprecht wrote: I'd settle for just a way of restoring it from a file. I just got my Sipura-3000, and it would be nice to keep the master config on disk and under CVS. If you generate your own configuration files and have the device download those configuration files from a tftp or web server, then you could easily do that. You need a copy of the Sipura Profile Compiler. Is the Sipura Profile Compiler some perl script that massages the data? Where can I ftp/http it from? (Or is it some ms-binary, in which case it wouldn't be all that useful to me.) I'd really prefer just finding a spec describing the config file syntax/format and the list of variables that I could set. It is perfectly fine if the downloaded file needs to be binary. It just needs to be well-enough defined that one can write a BSD or linux program for. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] spa-3000 review?
[EMAIL PROTECTED] (Rich Adamson) writes: Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. I got it yesterday afternoon. It is a very cute unit that is surprisingly small. (When I saw the size of the package I was at first afraid they'd mistakenly only sent me a power supply!) The fxo and fxs are indeed separate and show up as two peers and users. bonnet*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 9757/9757192.83.197.10D 255.255.255.255 5061 OK (29 ms) 6003/6003192.83.197.10D 255.255.255.255 5060 OK (22 ms) bonnet*CLI sip show users Username Secret Accountcode Def.Context ACL NAT 9757 XXX from-untrusted- No No 6003 YYY from-trusted-in No No The biggest problem with the unit is that it doesn't come with the slightest scratch of documentation. Not even a URL to download a preliminary manual. Setting it up is apparently meant to be a test that only the true followers of the Polynesian god Sip-Ura will be able to undertake, If one is used to the Grandstream one-page does-it-all http configuration, this baby is going to be a real shock. It goes on for pages and pages and has multiple views where the harder to explain features are not shown, apparently in an attempt to not scare every last person away. It is quite evident that Sipura put quite a bit of work into the code and intent is clearly to provide a mini firmware-based gateway/server that can be used standalone to do much of what we use asterisk for. From paging through the configs it is clear it can do PSTN-VOIP, VOIP-PSTN, VOIP-analog-phone, analog-phone-VOIP, analog-phone-PSTN and PSTN-analog-phone routing, all under the control of touch-tone passwords and/or md5 passwords or RSA certificates. This is all without involving any outside SIP server. I can see that it is going to be a while before I expose this to an outside IP address lest some kiddie that understands the passwords better than I do notices that he can make free PSTN phone calls because I missed filling in filling in one of the dozen or so passwords. Sorry, no detailed HOW-TO's yet. This thing can obviously be made to do what I want of it, but it will be a while figuring it all out. This thing really needs a wiki devoted to it. ;-) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] spa-3000 review?
[EMAIL PROTECTED] (Mike Benoit) writes: Have either of you experienced echo when making a call from the FXS port to the FXO port on the SPA-3000? There is some echo on LD PSTN calls when the two ends mistakenly talk over each other. I believe they have some VOX that attempts to enforce a ping-pong talk path (eg. the amps in one direction are always set for a gain of 0 while the other direction is a 1.0). Now the first thing that I noticed in making a PSTN call is that the remote side is very hard to hear. The gain between the Sipura-3000/PSTN and a Grandstream BT-100 is much less than between two BT-100's. I'm going to have to bump the gain up a bit in at least the PSTN-VOIP direction. Perhaps I need to do the VOIP-PSTN direction too. This is going to make echo even worse. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] Re: OT: saving/restoring sipura config
[EMAIL PROTECTED] (Randy Bush) writes: Sorry for this OT but I bet someone here knows if there is a way to save a Sipura 2000 current config and restoring it after a reset. hard as this is to believe, there isn't. major bummer, eh? I'd settle for just a way of restoring it from a file. I just got my Sipura-3000, and it would be nice to keep the master config on disk and under CVS. -wolfgang PS. I'm starting to feel nostalgic for the Moringstar Router and its text config files that one could ftp to/from the unit to update its config. -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] permission problem
[EMAIL PROTECTED] (Cyprien Simons) writes: Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? (/var/run/, /var/log/asterisk/messages) any help will be appreciated, Grab my patches below. It does both chroot and setuid to user asterisk. (You might need to back out one or two of the obvious Openbsd fixes.) I've been running chroot and as user asterisk for a few weeks now on this sip-only server. There are still few loose ends (like music on hold not running correctly, but part of that appears to be an asterisk scheduler problem under OpenBSD that happens even with no chroot etc.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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 external SIP
[EMAIL PROTECTED] (Jon Lawrence) writes: codec's are set to allow all. Thats your problem. I tried this too as an experiment and asterisk appears to take all to mean all codecs you can think of, not just the ones you have converters for. Instead of all you may want to try listing the codecs asterisk actually has (this is from -current): ; ; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw ; disallow=all allow=ulaw allow=alaw allow=gsm allow=adpcm allow=g726 allow=ilbc ;; allow=lpc10 (robotman) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] Intermittent SIP 404 Not Found response?
[EMAIL PROTECTED] (Andrew Yager) writes: I believe I'm experiencing the same problem with Grandstream phones, although I haven't had time to track it down yet. When your GS fails, slap a tcpdump on the line and have a look at what it is sending. When my GS fails it forgets how to route stuff on the internet and attempts to ARP for something that is halfway around the world (eg. sends an arp-request for the sip server even if that machine isn't local). I like GS's sound quality and price, but their firmware clearly has some serious corruption problems. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] vonage.ca * integration possible?
[EMAIL PROTECTED] (Brian McSpadden) writes: Your problem with doing this is this line right below...you have no idea what your authentication secret is. This is a closely guarded secret of Vonage. They don't have any interest in letting anyone do this. The closest you could do would be a softphone, unlimited inbound and 500 mins outbound calling. There are sample configs floating around out there to make that work. On Fri, 9 Jul 2004 10:28:06 -0400 (EDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Authorization: Digest username=1905XXX, realm=216.115.25.187, nonce=720170349, uri=sip:bspgroup1.bsp.vonage.net:5061, response=6a2fe5ec7b98a098aaf82a7dfc1340aa, algorithm=MD5 I thought the same thing at first, but then started wondering about a man-in-the-middle attack. Supposed asterisk simply used the Motorola ATA as a dongle and forwarded any tough authentication questions to the ATA and forwarded the ATA's answers back to the remote SIP server? Could that be made to work? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] Re: Re: iax or sip
Consider it backwards compatibility, sure, use [EMAIL PROTECTED] where you can, but I surely know if I told my parents to call me at ... Right now my grandstream bt-100 and asterisk team up to deliver 6001 as the number that I can be reached at to any remote caller. Somehow I don't think that my non-FQTN (Fully Qualified Telephone Number) is going to deliver much joy to folks hoping their return call button is going to do something useful. Would programming wolfgang at wsrcc dot com (damn spam-bots!) as the sip phone number allow a significant percentage of the folks to dial me back? (Assuming I have my _sip._udp SRV crap set right.) Do any commercial SIP providers lookup SRV? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] RE: is srv lookup being done when REGISTERing?
And when can we expect a patch from you for this? :P I'd like to see this too and be willing to do this under GPL. Is that good enough? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd asterisk http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] SPA-2000 and time of day
[EMAIL PROTECTED] (Chris Luke) writes: NTP is time-zone and season agnostic. It always transmits UTC. Yup. This is the answer to the most common FAQ on comp.protocols.time.ntp . Offsets from this are set in the client, including DST stuff. If they can't be set, get a better NTP client. :) I wish Grandstream were listening. The fact that you need to click on one of two buttons to decide whether to apply the -1 hr correction or not to get the right time is pretty lame. Daylight Savings Time: o No * Yes (if set to Yes, display time will be 1 hour ahead of normal time) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd asterisk http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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
