Re: [Asterisk-Users] Has Sixtel gone under?
That's always been the site at that url. On 8/2/05, Tony Hoyle <[EMAIL PROTECTED]> wrote: > Carlos Chavez wrote: > > I have been using Sixtel from the beginning of the year and service was > > getting worse and worse. Yesterday I tried to access the website to get the > > CDR and I got an error saying that the domain no longer exists. I checked > > the > > whois and it says that the domain is on hold. Have they finally folded? > > > http://www.sixtel.net/voip/ doesn't look too promising... > > Tony > > ___ > 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] Sipura SPA-1001: Bad Outgoing Call Quality
Greetings, I have a Sipura SPA-1001. When I make outgoing calls, I have very jittery sound. Incoming calls work fine. This wasn't the case a few months ago, I am running head as of yesterday. Any suggestions? Thanks, Erik ___ 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] Any experience with Sixtel--tollfreedirect--iax.cc?
The Good: I've had sixTel since December 2004. I've had surprisingly good call quality and very short ping times. Overall the service has been good. Response times for support were great when I needed it in the past. I got a $10 rebate for my toll free DID because it was a couple of weeks late, however I was kept in the loop from beginning to end until it was active. I was informed that the provider they used lagging. I was informed how to determine if the DID was routed to them yet or not, etc. The Bad: My buddy, based on my recommendation, signed up for sixTel + toll free DID on May 2005. To this day he has not recieved a response from sixTel or been able to contact them. I have attemped to contact them on his behalf myself, since I was so successful before, with no luck. He is chalking this up to a bad luck and looking for a new provider. The Ugly: My buddy is stuck in a predicament, his custom toll free DID is currently in the possesion of sixTel and he can't move it to a new provider. In addition, during the few times I have attempted to communicate with them I have observed the following: 1) All calls to sixTel's toll free number lead to a voiceprompt that tells you to use the web site, no one is ever available. 2) sixTel's own toll free number has disappeared on more than one occasion, where you don't even get the voice prompt telling you to use the web site. 3) The web site's emergency support that supposedly pages a tech doesn't get a response. In short, I myself will be looking for a new provider once the money I have deposited in my sixTel account runs out. I'd like to have a cheap per minute rate similar to sixTel, and I'd like to be able to LNP my current numbers away from sixTel. If anyone knows of a provider that meets this criteria please shoot me a note. Thanks, Erik On 7/26/05, Jay Milk <[EMAIL PROTECTED]> wrote: > There was a thread about sixtel not too long ago. Quality is ok, when > it's working. DID requests can take weeks or months, some are never > answered. Termination usually works, and with decent quality. Looks, > feels and smells like a one-pony show -- and the pony needs a vet. > > Try this: http://www.google.com/search?q=sixtel+site%3Adigium.com > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: Tuesday, July 26, 2005 8:31 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Any experience with > Sixtel--tollfreedirect--iax.cc? > > > Appears to be a diversified company (Google of it shows that they do web > design, VOIP, etc.). But this link caused me concern. > > http://www.ripoffreport.com/reports/ripoff145784.htm > > Thoughts? Experiences? > > ___ > 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] Best VoIP provider
All I'm saying is avoid sixTel. They have great sound quality, good prices, good ping times, and a pretty control panel. However if you ever need support anything done that doesn't happen automatically on their web site (number doesn't ring, custom toll free did, refund, support in general) forget about it. It's been three months since i ordered my custom toll free did, and it's not active yet! Calling their support number just tells you to go to the web site. Adding tickets, even emergency tickets, seem to go to /dev/null. Also if anyone knows of a provider that does LNP, would be able to move my numbers away from sixTel if they don't respond after a certain date and has decent prices please let me know. Erik On 7/19/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Tuesday 19 July 2005 13:38, Kanuri, Seshu (Company IT) wrote: > > It does not look like Nufone is still in business, judging from the > > content on their site, which is very little. There is not even a > > configuration document to download, to connect to their network. > > Hahahaha... Considering I put over 5000min/mo through them, I think that's > kind of funny. > > Nufone seems to have always been a DIY type of VOIP provider. Their new > members page works very well and shows connection information and so on... > maybe their email was blacklisted by some spam filter on your side? > > -A. > ___ > 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] Business Edition
I once tried to call in support for digium for 4 IAXy's that I purchased ($400). They told me to e-mail this mailing list. I appreciate all the hard work that they did to produce asterisk, I just don't trust this company to support anything. Just my $.02. Erik On 7/18/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Monday 18 July 2005 13:20, Kevin Walsh wrote: > > Absolutely not. If you find that you need $995 worth of support, some > > time in the future, then I'm sure that you can obtain it from one of > > several providers. I don't think it's worth paying up-front for > > something you probably won't need, but that's really for you to decide. > > To some the stability and repeatability of ABE is well worth the money. > Contractors can be good, and with a contractor I'm sure you can get the same > level of stability and repeatability as ABE. I think that is particuarly WHY > Digium's trying to create it... it's a successful service and they want in > on it. And honestly, Digium's got the resources to pull it off very nicely. > Your regular contractor (me, for instance) can't realistically provide 24/7 > support on his own. Digium can. Your regular contractor can't realistically > fix any problem that may happen on his own. Digium can. Kudos to them, I > say. > > I dunno... people seem all up in arms about this but honestly I fail to see > the problem. Digium is doing what they can to make money and provide > services while keeping Asterisk as free and openly developed as possible. I > have (small amounts of) code contributed to Asterisk and I am working on > more. Digium and Asterisk have given me a lot of newfound freedom and > flexibility and power in my phone system. I appreciate that, and I don't > feel that this dual-licensing or granting of a nonexclusive perpetual license > to the bits and pieces of my code is too much to ask. My bits and pieces > would be worthless without the bits and pieces and chunks and slabs of code > that others have provided, and it'd all be useless without the framework that > Digium came out with. > > If you don't want or don't like ABE, don't use it. Nobody is cramming it down > your throat. > > -A. > ___ > 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] Memory leak in asterisk CVS
Hi Walter, I had high load and extreme memory usage on my machine. My machine wasn't running on SMP. My point was that the cvs version you were using contained some bad patches, and it was probably a good idea to upgrade or move to stable. Thanks, Erik On 7/18/05, Walter Klomp <[EMAIL PROTECTED]> wrote: > Hi Erik, > > You put me to a page which refers to high load on CPU on SMP. Nothing to do > with memory leak. Furthermore I am not running SMP. > > Any other suggestions in which direction to look? Am I the only one > experiencing this ? > > Do you mean if I update to the today's CVS the memory leak issue will be > resolved ? > > Thanks > Walter > > --- Original Message below --- > > Message: 20 > Date: Sat, 16 Jul 2005 21:42:44 -0700 > From: Erik Espinoza <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Memory leak in asterisk CVS > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Known issue. This was reverted later. > > Check the thread on the mailing list > > http://lists.digium.com/pipermail/asterisk-users/2005-July/116246.html > > Thanks, > Erik > > On 7/16/05, Walter Klomp <[EMAIL PROTECTED]> wrote: > > Hi, > > > > My Asterisk CVS is apparently not doing much (other than keeping SIP & > > IAX2 registrations alive and doing some ZAP calls (without > > echo-cancellation), but slowly the memory is filling up, so much so that > > 100m virtual memory is used up within 12 hours and I have to restart the > > asterisk application every 48 hours to make sure I have enough memory... > > > > How can I help resolve this problem? > > > > Problem occurs on both Sangoma and Digium installed systems. Fedora Core > > 3 and Centos 4.1 don't make a difference either. > > > > My version is Asterisk CVS-HEAD built on a i686 running Linux on > > 2005-07-11 16:29:02 > > > > I have removed the mailbox entries in my sip.conf which greatly reduced > > this problem. So, I suspect it may be in the sip or iax channel > application. > > > > I also run quite a bit of agi scripts but none of them were "alive" when > > these memory-usage increases as shown below over a 1 minute interval > > with only 4 zap channels alive (2 calls) occured: > > > > ps -AF output... using this script: > > n=1;while [ 1 ]; do i=`ps -AF|grep ast|grep sbin|grep -v grep`; m=`echo > > $i|cut -f 6 -d\ `;if [ `echo $m` -ne `echo $n` ]; then echo $i; n=`echo > > $m`;fi;done > > > > root 15875 26881 0 15727 46240 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46248 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46256 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46268 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46280 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46288 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > > > Hope we can fix this somehow. > > > > Walter Klomp > > Singapore. > > > > > > > > > ___ > 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] Memory leak in asterisk CVS
Known issue. This was reverted later. Check the thread on the mailing list http://lists.digium.com/pipermail/asterisk-users/2005-July/116246.html Thanks, Erik On 7/16/05, Walter Klomp <[EMAIL PROTECTED]> wrote: > Hi, > > My Asterisk CVS is apparently not doing much (other than keeping SIP & > IAX2 registrations alive and doing some ZAP calls (without > echo-cancellation), but slowly the memory is filling up, so much so that > 100m virtual memory is used up within 12 hours and I have to restart the > asterisk application every 48 hours to make sure I have enough memory... > > How can I help resolve this problem? > > Problem occurs on both Sangoma and Digium installed systems. Fedora Core > 3 and Centos 4.1 don't make a difference either. > > My version is Asterisk CVS-HEAD built on a i686 running Linux on > 2005-07-11 16:29:02 > > I have removed the mailbox entries in my sip.conf which greatly reduced > this problem. So, I suspect it may be in the sip or iax channel application. > > I also run quite a bit of agi scripts but none of them were "alive" when > these memory-usage increases as shown below over a 1 minute interval > with only 4 zap channels alive (2 calls) occured: > > ps -AF output... using this script: > n=1;while [ 1 ]; do i=`ps -AF|grep ast|grep sbin|grep -v grep`; m=`echo > $i|cut -f 6 -d\ `;if [ `echo $m` -ne `echo $n` ]; then echo $i; n=`echo > $m`;fi;done > > root 15875 26881 0 15727 46240 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > root 15875 26881 0 15725 46248 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > root 15875 26881 0 15725 46256 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > root 15875 26881 0 15725 46268 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > root 15875 26881 0 15725 46280 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > root 15875 26881 0 15725 46288 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > Hope we can fix this somehow. > > Walter Klomp > Singapore. > > > ___ > 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] SpanDSP rxfax, no tiff.
Hello Rob,
I don't really know the answer to your woes, but I have a queston.
Where are you downloading the rxfax and spandsp? I can't seem to hit
any of the sites that claim to have it.
Erik
On 7/13/05, Rob Danz <[EMAIL PROTECTED]> wrote:
>
>
>
> Hello,
>
> Let me start by saying I have checked the wiki and the archives and did find
> some relative information. I tried the suggestions in those threads, but
> still have the same problem.
>
>
>
> I'm using the CVS Asterisk from July 11, 2005.
>
> Redhat FC2
>
> SpanDSP 0.0.2pre18
>
> Libtiff 3.5.7
>
> Digium PCI card 1 FXO, 1FXS.
>
>
>
> I have a single POTS line coming, but I have 2 numbers and am using
> distinctive ring detection in *.
>
> When you call my "fax" number, the ring detection does work, and does send
> it to the fax context correctly.
>
>
>
> The debugs show the call is answered, rxfax is invoked and it is trying to
> write to the fax file. After the sending party hangs up, it tries to
> execute a script that will ultimately mail me the fax file. But since the
> tiff file isn't there to begin with, that fails. The permissions on that
> folder are 777 for now… so permissions aren't the problem.
>
>
>
> I saw a post by Steve Underwood from last year on a similar problem, but it
> was looking like timing slips on the T1/E1 for that user … I'm just using a
> POTS line though. I've also done ztmonitor to look at the Rx and Tx levels.
> Rx is a little hotter than Tx, but they're both well on the right hand side
> of the scale.
>
>
>
> Any help is appreciated. Debugs & extensions.conf excerpt are below.
>
> Thanks,
>
> Rob
>
>
>
> Debug output ---
>
> Jul 13 10:04:34 NOTICE[7975]: chan_zap.c:5759 ss_thread: Got event 2
> (Ring/Answered)...
>
> -- Detected ring pattern: 93,0,0
>
> -- Distinctive Ring matched context fax
>
> -- Executing Answer("Zap/4-1", "") in new stack
>
> -- Executing Macro("Zap/4-1", "faxreceive") in new stack
>
> -- Executing Set("Zap/4-1",
> "FAXFILE=/var/spool/asterisk/asterisk-fax/1121267067.12.tif")
> in new stack
>
> -- Executing RxFAX("Zap/4-1",
> "/var/spool/asterisk/asterisk-fax/1121267067.12.tif") in
> new stack
>
> -- Executing System("Zap/4-1", "/usr/local/sbin/mailfax
> /var/spool/asterisk/asterisk-fax/1121267067.12.tif ") in
> new stack
>
> Jul 13 10:05:03 WARNING[7975]: app_system.c:75 system_exec_helper: Unable to
> execute '/usr/local/sbin/mailfax
> /var/spool/asterisk/asterisk-fax/1121267067.12.tif '
>
> -- Hungup 'Zap/4-1'
>
>
>
>
>
> Extensions.conf section ---
>
> [fax]
>
> exten => s,1,Answer
>
> exten => s,2,Macro(faxreceive)
>
> exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE}
> ${EMAILADDR})
>
>
>
> [macro-faxreceive]
>
> exten =>
> s,1,Set(FAXFILE=/var/spool/asterisk/asterisk-fax/${UNIQUEID}.tif)
>
> exten => s,2,rxfax(${FAXFILE})
>
> exten => s,3,Set([EMAIL PROTECTED])
> ___
> 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
>
>
___
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Re: [Asterisk-Users] Gizmo: Skype done right?
On 6/30/05, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 15:30, Thu 30 Jun 05, Erik Espinoza wrote: > > Agreed. IAX2 would have been a much better way to go. Regardless i > > don't see how an open source, standards based softphone will compete > > with Skype. Skype has a few things going for it: > > > > 1) Hype, lots of it. It's no coincidence that that the two rhyme > > 2) Built in traversal of firewalls - p2p style (have I mentioned I > > hate sip + nat) > > 3) Encryption, Encryption, Encryption > > huh?? why is this a pre ? > All my connections (cept to our ITSP) are encrypted. > Even the IAX2 and SIP links between the different asterisk > machines. > In my opinion encrypting the whole network stack is a better > way to go then just encrypt one protocol while leaving a lot > of other sensitive stuff flow unencrypted. > Encryption is higly overrated in voip world. Did you ever > try to evedrop a call ??? Skype emulates an stun type system using third parties, which is where the p2p aspect of it comes into play. Having encryption when your data is travelling through untrusted third party hosts is a good thing. As far as I know SIP w/ Asterisk is not encrypted, in fact there is a bounty for it: http://www.voip-info.org/tiki-index.php?page=Asterisk+Bounty+SIP+encryption Using your own encryption with a vpn is nice, when you have full control over everything but unfeasible when you are dealing with other peoples networks. For example, ssh works great behind nat, but some vpn's need extra configuration. We are talking about a client for the unwashed masses, not for a group of system folks who know what they are doing. > > > > An open source, standards based free implementation does not win over > > users. There needs to be more, just ask the Ogg folks how MP3's doing. > > I have around 10 GB of mp3 and "only" 64 GB of .ogg files. > More and more ppl adept .ogg cause it's simply better. > A better example would be Video2000 vs VHS. That's where the > best hyped version (read, the best lost) won. Really? That's good for you, Ogg Vorbis is a great audio codec. However you must realize that you are the exception and not the rule. I mean face reality dude, the biggest selling digital audio players don't support Ogg! I stand by my example for the same reason. Ogg has some market share, but it's puny. I expect the same to happen with Gizmo. ___ 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] Gizmo: Skype done right?
Agreed. IAX2 would have been a much better way to go. Regardless i don't see how an open source, standards based softphone will compete with Skype. Skype has a few things going for it: 1) Hype, lots of it. It's no coincidence that that the two rhyme 2) Built in traversal of firewalls - p2p style (have I mentioned I hate sip + nat) 3) Encryption, Encryption, Encryption An open source, standards based free implementation does not win over users. There needs to be more, just ask the Ogg folks how MP3's doing. Also it's worth noting that both are free, however Skype has a Linux version! Skype = Win, Mac, Lin x86, PocketPC Gizmo Beta = Mac, Win (Coming Soon: Linux?) Erik On 6/30/05, Matt Fredrickson <[EMAIL PROTECTED]> wrote: > On Thu, Jun 30, 2005 at 01:31:55PM -0700, Jerry Glomph Black wrote: > > I've just submitted this as a Slashdot story, too. > > I have absolutely no connection with any of the principals, I just think > > they > > are doing the right thing. This could have a major impact on the Asterisk > > community, and VoIP usage in general. > > > > Michael Robertson, of mp3.com fame, has been battling for open standards in > > the > > IP telephony world, in addition to his better-known Lindows (now Linspire, > > at > > http://www.linspire.com) venture to promote Linux on the desktop. His > > sipphone.com VoIP operation works great for me, but Michael has been long > > concerned about the totally closed and proprietary nature of Skype (as well > > as a > > lot of the misleading hype surrounding it). > > > > Today his crew released "Gizmo" (at "http://www.gizmoproject.com) (a > > tentative > > name until a better one is found) which has the main benefits of Skype, > > PLUS it > > is layered upon SIP, DUNDI, and the existing sipphone.com infrastructure, > ^^^ > > Looks like they already messed up... If they're going to redo all of this > anyway, > they might as well use a protocol like IAX where you don't have NAT problems. > > Matthew Fredrickson > ___ > 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] [Fwd: Asterisk Balancing solution]
I can only think of 2 ways to proceed: 1) Set a shorter register interval 2) Set static ip on all phones, and forgo registration On 6/30/05, Mohamed A. Gombolaty <[EMAIL PROTECTED]> wrote: > > > Dear All, > > I am using Linux-High Availability between two Asterisk servers, everything > is fine but I do have one problem with > this, When a server fails and the other assumes the ip address and start > asterisk on server 2, the ip phone must > re-register themselves again, otherwise the phones are dead. > > Does anyone have Ideas of how to overcome this. > > -- > Thx > MAG > > > > > > > > ___ > 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] SixTel?
Nope, sixTel has been acting up big time recently. I've put in requests for support online. Calling has gone to a message that says go to the web site. This really blows, the prices were good but unless they shape up I'm switching providers. Thanks, Erik On 6/27/05, JD Austin <[EMAIL PROTECTED]> wrote: > I was just checking out the dids for all of my fail over providers and > noticed that neither DID that I have with SixTel work. > Both pause for a long long time > The local number gives a recording: 'The number you have dialed is not > in service or is assigned in a different area code. Please check your > number and dial again'. > The 800 number just rings busy. > Anyone else having this issue or am I a lone data point? > > JD > > -- > JD Austin > Twin Geckos Technology Services LLC > email: [EMAIL PROTECTED] > http://www.twingeckos.com > phone/fax: 480.288.8195 > > ___ > 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] Welltech 4 Port FXO - Asterisk
Hey does anyone know how to configure the 4 port fxo to work with Asterisk? I have the updated firmware. All ports register, however incoming calls are never handled properly by the fxo. I even set hotline. Does anyone have any info, or perhaps a web site? ___ 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 firmware was 'why even use SIP'
They probably don't want to deal with the snickering and laughter that this code will ensue. Digium has a good rep. in the open source community for Asterisk, they don't want to release this mockery on anyone! On Wed, 23 Mar 2005 15:38:06 -0600, Chris Wade <[EMAIL PROTECTED]> wrote: > I've noticed a few recent messages regarding if/when a fixed firmware > for the IAXy will become available that deals with everyone's issues > regarding the little device. > > After some thought on this subject, I wonder what Digium's stance on > open-sourcing the firmware for the device? Is there anything in the > IAXy that couldn't open-source? Asterisk is moving along quite nicely > due to the oss community around it, why couldn't the IAXy have the same? > > Anyway, just some thoughts, > -chris > > ___ > 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] why even use SIP
> Never had a problem with mine. I set my DHCP server to hand out a > specific IP to the IAXy, too. Works on some, but not all. It fails on Microsoft and Cisco DHCP Servers. Just because it works for you doesn't mean it's implemented correctly. > What's the advantage over unplugging the unit and plugging it back in? That's not resetting, that's rebooting. There is a difference. If I forget the password to my linksys, I can hit the reset button to reset to defaults. Besides there are many things wrong with the IAXy. 1) Price, twice as expensive as the SIPURA equivalent 2) Codecs, only supports ulaw/pcm 3) Doesn't properly support DHCP 4) Security, there is none. If it can be communicated with, then it can be zapped without a password 5) MWI, Call Waiting, 3-way calling missing 6) Configuration requires Linux, as opposed to a web browser or something more standard. Let's face facts there, the IAXy sucks by any definition. Digium support is pretty much go on IRC or e-mail this list and ask here. After my experience with the IAXy, I never thought I'd be happy to give Linksys my money (PAP2-NA's) On Tue, 22 Mar 2005 22:02:43 +0100, Jens Vagelpohl <[EMAIL PROTECTED]> wrote: > > - proper DHCP and possibility of static IP > > Never had a problem with mine. I set my DHCP server to hand out a > specific IP to the IAXy, too. > > > - a 'reset' button > > What's the advantage over unplugging the unit and plugging it back in? > > > > And my IAXy doesn't work with my european phone (no tone) it's kind of > > a drag :( > > My IAXy works perfectly fine with a cheapo Panasonic cordless I bought > here in Germany. > > jens > > ___ > 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] IP PHONE with chip PA1688 and IAX2 Authentication
remove auth=md5 from your iax.conf and try again. On Tue, 22 Mar 2005 21:38:15 +0100, Androtech <[EMAIL PROTECTED]> wrote: > Dear All, > I bought one IP PHONE from Integrated Networks which was showed to wiki too: > http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks > I have problems with the Asterisk authentication. It does't want to LOG IN > to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and > all paramters seems to be correct. > Does somebody use the same IP PHONE with Asterisk and IAX2? Can someone tell > me the right configuration for it? > > My actual configuration is: > > locale ip: 192.168.0.75 > subnet mask: 255.255.255.0 > router ip: 192.168.0.1 > dns: 192.168.0.1 > > protocol: iax2 > service type: common > use service marked > service address: 192.168.0.1:4569 > nat trasversal: disabled > phone number: 103 > account: ipphone > pin: test > register port: 4569 > signal port:1701 > control port:1721 > rtp port:1721 > local type: auto > > the phone firmware is V.1.38.009 > > My Asterisk extension is: > > [ipphone] > type=friend > username=ipphone > secret=test > auth=md5 > host=dynamic > context=fullaccess > mailbox=103 > callerid="ipphone"<103> > > Thanks for any help > Regards, > > ___ > 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] Wireless LANs and Asterisk
The ulaw codec is the heaviest codec there is. Have you tried lighter codecs such as the gsm codec? Erik On Thu, 10 Feb 2005 22:01:01 +0100, Michiel van Baak <[EMAIL PROTECTED]> wrote: > Hi, > > My experience: > A handfull of concurrent calls, all works fine on 54mbit. > Dont try to go beyond that. Specially when your link is not > totally 100%. We tried to do 10 calls on a dedicated > Conceptronic AP and all fell down. even with the ulaw codec > it was not doable for normal conversations. Even disabling > web was not the answer, so we took the good old wires again. > > And another concern: privacy. > As you know, WEP is not that strong. And as long as there is > no solid encrypted RTP stream everyone with a laptop is able > to monitor/record your calls. > > just my 2 cents > -- > Michiel van Baak > http://lunteren.vanbaak.info > [EMAIL PROTECTED] > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > "Two of the most famous products of Berkeley are LSD and BSD. I don't think > that this is a coincidence." > > ___ > 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] Multiple mailbox on the same SIP extension
Create two extensions on the phone, the 7960 allows this. Then associate the second one with the sales mailbox. Erik On Thu, 03 Feb 2005 13:56:33 -0500, Martin Roy <[EMAIL PROTECTED]> wrote: > I'm wondering if there's a way it will show on the phone when there's a > new message. Here's what I'm trying to do : > > in my extensions.conf when someone call from a PSTN line on my TDM04B > card they have a choice. When someone press 1 for sales then I have 3 > phones ringing at the same time. Each phone as already there own mailbox > because if someone know there extension instead of pressing 1 they can > enter there direct extension and then if they don't answer they are > transfer to the voicemail of that extension. When someone press 1 all > the 3 phones rings if no one answer then it transfer it to another > voicemail not related to any of the phones let's call it sales main > voicemail. > > If someone leave a message on that voicemail there's no way (the way my > things are setup currently) that the staff in sales will know there's a > new message for them... > > Currently on my Cisco IP phones 7960 when someone leave a message for a > SIP extension it will light up on the phones and tell the user there's a > new message on the server for them. > > I'm wondering can I configure 2 mailbox on the same phone? > > here's a sample of my extensions.conf : > > exten => s,1,Answer > exten => s,2,DigitTimeout(10) > exten => s,3,ResponseTimeout(20) > exten => s,4,Wait(1) > exten => s,5,Background(Intro_EN) > exten => 0,1,Dial(SIP/221,15) > exten => 0,2,Voicemail2(u221) > exten => 0,102,Voicemail2(b221) > exten => 0,103,Hangup > exten => 1,1,Dial(SIP/231&SIP/237&SIP/239,15) > exten => 1,2,Voicemail2(u7253) > exten => 1,102,Voicemail2(b7253) > exten => 1,103,Hangup > exten => t,1,Dial(SIP/221,15) > exten => t,2,Voicemail2(u221) > exten => t,102,Voicemail2(b221) > exten => t,103,Hangup > > Here's a sample of one of my extension : > > exten => 231,1,Dial(SIP/231,15) > exten => 231,2,Voicemail2(u231) > exten => 231,102,Voicemail2(b231) > exten => 231,103,Hangup > > sample of sip.conf (with some info removed) : > > [231] > type=friend > username=231 > secret="not written for security purpose" > context="the context in extensions.conf" > callerid="removed for this post" <231> > qualify=1000 > host=dynamic > canreinvite=no > mailbox=231@"the context in voicemail.conf for my mailbox" > disallow=all > allow=ulaw > allow=alaw > > I haven't tried it yet but I have also another concern, since my > extensions start with the number "2" if I add a choice that when someone > press 2 it dial some SIP extensions will it take the 231 if I type it or > it will say it as a 2 and forget the other digits? > > Thanks > > Martin > ___ > 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] Multi Asterisk Server Transfers
Sounds like you need a SIP Proxy, which just relays calls to the destination, rather than Asterisk which handles all aspects of a call. I'd recommend Sip Express Router, http://www.iptel.org/ On Thu, 27 Jan 2005 12:58:56 +1100, Mike Sander <[EMAIL PROTECTED]> wrote: > Hi, > > We are in the business of setting up * servers for businesses, attached via > IAX trunks to our VoIP provider (also using *). > > I have a client with a head office * server, who wants a number of remote > offices, with just 1 SIP connection to each. I can arrange this no probs > with our providers, but there are issues with transfer. > > I don't want the remote offices making their direct SIP connection to the > head office, because bandwidth is limited and then for them to make an > outgoing call, the head office has both an incoming and outgoing connection > - or double the bandwidth. This is the same for an incoming call to head > office that gets transferred to the remote, the call stays with the head > office * server, and the server makes another outgoing call to the remote > office. All these calls are free, but use double the bandwidth. > > The question: > > The remote offices can make direct SIP connections to our provider. If the > head office * transfers a call, then the server releases the call entirely > back to the providers * server and calls from there. > > I.E. call in to head office from PSTN through the provider. Call gets > transferred to the remote office. Head office could then unplug/burn/blowup > their asterisk server without disrupting the call between the remote office > and the PSTN network. > > Is this possible? Companies with multiple * servers in many remote office, > surely have this system, to conserve bandwidth? How is the transfer made? > Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] > basic > release. > > Thanks > Mike > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 > > ___ > 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] IAXy's apparantly failing in the field
I have one of these phones. I bought it off of eBay. Not sure where to get them direct. You will need to load the proper image, in that I believe it ships with SIP by default. Each protocol has its own image. Erik On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote: > Michael Giagnocavo wrote: > > >>Yes, the IAXy has faults, but until other IAX2 devices ship, it's the > >>only game in town. I know that the Farfon device will be out soon and > >>we'll be able to judge its quality at that time. > >> > >> > > > >Or any PA168 phones, which are already out, and support IAX2, SIP, H323, > >MGCP and N2P. (I've got one on my desk here as do a few others, and it works > >great.) > > > > > I want one of them! > > Which model is it? Did you have to do any software upgrade? How much > does it cost? > > Cheers, > Jean-Michel. > ___ > 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] IAXy's apparantly failing in the field
Ouch, sorry to hear that you bought 20 of those damn things. I'd recommend you provision them up with a static. The IAXy doesn't really use dhcp, it uses a mix breed of bootp and dhcp which works on only certain implementations of dhcp with varying degrees of success. Since it is acting more along the lines of bootp, it never renews a lease. Perhaps your server releases the address and gives it to dhcp user causing an ip conflict and knocking your IAXy off the network. My only advice is to avoid the IAXy like the plague. My advice, get a few Sipura's. They are cheaper and work great with Asterisk. As much as I want to help out with the development of asterisk, I can't justify using the horribly designed IAXy with all it's shortcomings. On Fri, 21 Jan 2005 15:49:28 -0700, Brent Goran <[EMAIL PROTECTED]> wrote: > I am not sure if this is the place for Digium user-to-user discussion, > but... > > We have deployed many (20+) IAXy's in the field. At a couple of locations, > the IAXy's have just stopped working after 1 or 2 days use. No lights go on, > no DHCP lease is renewed as far as we can tell, and of course no dialtone > and no registration with the server. > > Has anyone else experienced high failure rate with these devices? > > > > ___ > 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] Can IAXy be setup for PPPoE ???
No, it only does bootp On Wed, 19 Jan 2005 15:24:23 +0800, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: > Can IAXy be setup for PPPoE ??? > > If so, how? > > bye > > Ronald > > ___ > 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] Is an unregistered phone busy?
What exactly do you want an unregistered phone to do? Ring nothing and then go to voicemail? What is it that you are trying to accomplish? On Wed, 19 Jan 2005 00:25:08 +0100, Rob Scott <[EMAIL PROTECTED]> wrote: > Asterisk seems to regard an unregistered phone to be busy. > Is that correct? Is not an unregistered phone unavailable? > > It is odd to me that if someone dials an unregistered extension, then > the dialplan jumps to busy and voicemail kicks in saying that the person > is on the phone, when clearly they can't be if the phone hasn't > registered. > > Any way around this? > > ___ > 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] Packet8 DTA310 and Asterisk
Whoops, I meant auth=plain for a packet8 dta. Erik On Fri, 14 Jan 2005 16:52:06 -0800, Erik Espinoza <[EMAIL PROTECTED]> wrote: > Under your sip.conf change to this: > > [8006] > type=friend > host=dynamic > auth=md5 > secret=1234 > dtmfmode=rfc2833 > context=sip > callerid=8006 > [EMAIL PROTECTED] > > The key is auth=md5 > > I have set a few of my buddies who use to have packet8 on my asterisk > box just fine. > > Erik > > On Fri, 14 Jan 2005 19:08:52 -0500, Olson, Dana <[EMAIL PROTECTED]> wrote: > > I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware > > version (Application Code Version: DTA version 1.0 US (8x8 00)) onto it > > via TFTP, so I could access the SIP configuration. > > > > Under the SIP config, I put the IP of my * system, the 5060 port, and for > > Domain Name, I put default (is that right?). I checked off the Send > > Registration Request box. Dial Plan I left at the default, 1xx|x.T > > (is that right?), and Transport is set to UDP. For Line 1, I have it set as > > follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no > > idea what this is), Username=8006, Password=1234. > > > > In the OOB Signalling page, where I set the RFC2833 options, I haven't > > changed anything from the defaults. Same goes for the VLAN pages. > > > > The CODECS page currently has G711U, G711A, and G729 selected, all three > > with Silence Suppression turned off. > > > > I have other VoIP phones (soft and hard) working. > > > > >From extensions.conf: > > [sip] > > exten => 8006,1,Answer > > exten => 8006,2,Wait(1) > > exten => 8006,3,Dial(SIP/8006,20) > > exten => 8006,4,Voicemail(u8006) > > exten => 8006,5,Hangup > > exten => 8006,103,Voicemail(b8006) > > exten => 8006,104,Hangup > > > > >From sip.conf: > > [8006] > > type=friend > > host=dynamic > > secret=1234 > > dtmfmode=rfc2833 > > context=sip > > callerid=8006 > > [EMAIL PROTECTED] > > > > >From voicemail.conf: > > [default] > > 8006 => ,Packet8,[EMAIL PROTECTED] > > > > Okay, so when I apply those settings and restart the unit, I get a bunch of > > these messages in the * console: > > > > Jan 14 18:03:42 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to > > authenticate user 8006;tag=t2a7aca1ef6g41a7 for > > SUBSCRIBE > > > > And these messages continue once or twice every second until I reset the > > phone or unplug it. During this time, if I pick up the phone, I hear a > > dialtone. If I hang up and pick up again, the dialtone is still there. > > However, if I try dialing another * extension, say 8005, it doesn't do > > anything. If I hang up and pick up again after trying this, the dialtone is > > gone. If I try the same thing but instead dial 8005#, the same thing > > happens. If I wait for a few seconds/minute after hanging up, I'll get the > > dialtone back. If I try to dial a long disatance, the same thing happens. > > If I try without the 1 for long distance, the same thing happens. If I try > > with a 9 in front of 1 and then the number (this works for our other > > phones) then the same thing happens. However, now I'm getting more messages > > in the * console mixed in with the original ones: > > > > Jan 14 19:04:24 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to > > authenticate user 8006;tag=t2c3b0a372ad3g for > > SUBSCRIBE > > > > Maybe the dialplan is set wrong in the web config of the DTA310? I don't > > know what I would set it to though. > > > > If anyone can assist, then that would be appreciated. > > __ > > Dana Olson > > > > Disclaimer: The information transmitted in this message is intended only > > for the person or entity to which it is addressed and may contain > > confidential and/or privileged material. Any review, retransmission, > > dissemination, or other use of or taking of any action in reliance upon > > this information by persons or entities other than the intended recipient > > is prohibited. If you received this message in error, please contact the > > sender and delete the material from any system. > > > > ___ > > 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] Packet8 DTA310 and Asterisk
Under your sip.conf change to this: [8006] type=friend host=dynamic auth=md5 secret=1234 dtmfmode=rfc2833 context=sip callerid=8006 [EMAIL PROTECTED] The key is auth=md5 I have set a few of my buddies who use to have packet8 on my asterisk box just fine. Erik On Fri, 14 Jan 2005 19:08:52 -0500, Olson, Dana <[EMAIL PROTECTED]> wrote: > I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware > version (Application Code Version: DTA version 1.0 US (8x8 00)) onto it > via TFTP, so I could access the SIP configuration. > > Under the SIP config, I put the IP of my * system, the 5060 port, and for > Domain Name, I put default (is that right?). I checked off the Send > Registration Request box. Dial Plan I left at the default, 1xx|x.T > (is that right?), and Transport is set to UDP. For Line 1, I have it set as > follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no > idea what this is), Username=8006, Password=1234. > > In the OOB Signalling page, where I set the RFC2833 options, I haven't > changed anything from the defaults. Same goes for the VLAN pages. > > The CODECS page currently has G711U, G711A, and G729 selected, all three with > Silence Suppression turned off. > > I have other VoIP phones (soft and hard) working. > > >From extensions.conf: > [sip] > exten => 8006,1,Answer > exten => 8006,2,Wait(1) > exten => 8006,3,Dial(SIP/8006,20) > exten => 8006,4,Voicemail(u8006) > exten => 8006,5,Hangup > exten => 8006,103,Voicemail(b8006) > exten => 8006,104,Hangup > > >From sip.conf: > [8006] > type=friend > host=dynamic > secret=1234 > dtmfmode=rfc2833 > context=sip > callerid=8006 > [EMAIL PROTECTED] > > >From voicemail.conf: > [default] > 8006 => ,Packet8,[EMAIL PROTECTED] > > Okay, so when I apply those settings and restart the unit, I get a bunch of > these messages in the * console: > > Jan 14 18:03:42 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to > authenticate user 8006;tag=t2a7aca1ef6g41a7 for > SUBSCRIBE > > And these messages continue once or twice every second until I reset the > phone or unplug it. During this time, if I pick up the phone, I hear a > dialtone. If I hang up and pick up again, the dialtone is still there. > However, if I try dialing another * extension, say 8005, it doesn't do > anything. If I hang up and pick up again after trying this, the dialtone is > gone. If I try the same thing but instead dial 8005#, the same thing happens. > If I wait for a few seconds/minute after hanging up, I'll get the dialtone > back. If I try to dial a long disatance, the same thing happens. If I try > without the 1 for long distance, the same thing happens. If I try with a 9 in > front of 1 and then the number (this works for our other phones) then the > same thing happens. However, now I'm getting more messages in the * console > mixed in with the original ones: > > Jan 14 19:04:24 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to > authenticate user 8006;tag=t2c3b0a372ad3g for SUBSCRIBE > > Maybe the dialplan is set wrong in the web config of the DTA310? I don't know > what I would set it to though. > > If anyone can assist, then that would be appreciated. > __ > Dana Olson > > Disclaimer: The information transmitted in this message is intended only for > the person or entity to which it is addressed and may contain confidential > and/or privileged material. Any review, retransmission, dissemination, or > other use of or taking of any action in reliance upon this information by > persons or entities other than the intended recipient is prohibited. If you > received this message in error, please contact the sender and delete the > material from any system. > > ___ > 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] linphone -> NAT -> * -> NAT -> firefly woes.
Did you enable passthrough for the rtp ports on the asterisk box? I had the same problem until I enabled udp 1:2 on the firewall. On Wed, 12 Jan 2005 22:06:18 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote: > Hi folks > > an issue I don't understand. I'm running * stable 1.0.3 on public > internet, with following iax.conf / sip.conf entries: > > iax.conf > > [100] > type=friend > username=Foo > context=default > auth=md5,plaintext,rsa > secret=secret > host=dynamic > callerid="Foo" <100> > qualify=no > > sip.conf > > [10] > type=friend > username=Bar > context=default > callerid=Bar <10> > host=dynamic > secret=secret > nat=yes > canreinvite=no > > On iax exten 10 I register firefly, on sip exten 100 linphone, > both behind nat. > > Now, calls I can do is e.g. > firefly -> * -> linphone > linphone -> * echo test (copied this from demo and put it on exten 600) > > but what wouldn't properly work is is sip to iax bridging > linphone -> * -> firefly > > More specifically, firefly rings properly, but when I press Accept > it just keeps ringing, and finally * tells me that linphone didn't > send any frames: > > channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: > SIP/10-e8bd > Jan 12 21:55:02 DEBUG[22661]: channel.c:2725 ast_channel_bridge: Bridge stops > bridging channels SIP/10-e8bd and IAX2/100/2 > > Doing my tcpdumps I checked that there's really no data sent by linphone, > while nothing is dropped by firewalls either. > > Did anyone experience similar troubles? A hint about how to resolve or further > debug this would sure be appreciated. > > Another point I'm wondering about is why, in that same connection, the > caller id handed to firefly is just "10", and not the one specified > in sip.conf, i.e. "Bar <10>". > > I tested all that stuff also with iaxcomm, i.e. pure iax bridging > iaxcomm -> NAT -> * -> NAT -> firefly > and here, everything works OK, calls in both ways and caller id > transmission. > > Thanks, Bruno. > > ___ > 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] Sipura SPA-1001 and Tivo Series 1
Oh yeah, for reference try looking at tivocommunity.com on how to configure your tivo for 14.4 or 19.2 kbps On Thu, 6 Jan 2005 19:46:48 -0800, Erik Espinoza <[EMAIL PROTECTED]> wrote: > Most digital devices such as modems, fax machines and tivo's can not > be used without a lot of changes on VoIP. > > I've seen success with TiVo when you use a special code to kick it > down to 14.4 kbps and use g711ulaw as the codec. I think your best bet > is to try to eBay the custom nic for the TiVo series 1. > > Erik > > > On Thu, 6 Jan 2005 20:39:45 -0500, David Ishmael > <[EMAIL PROTECTED]> wrote: > > Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series > > 1 (yes its old). When I do a test call with Tivo, the call always fails (it > > seems to dial the number but never connects). I can pick up the phone line > > and hear the Tivo "talking". I've tried looking around for anything special > > I need to do but its still not working. I can connect a phone to the > > SPA-1001 and can make outgoing calls just fine. I even tried calling the > > Tivo number and can hear the modem pick up. Has anyone done this? Any help > > would be greatly appreciated. > > > > Thanks, > > Dave > > > > ___ > > 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] Sipura SPA-1001 and Tivo Series 1
Most digital devices such as modems, fax machines and tivo's can not be used without a lot of changes on VoIP. I've seen success with TiVo when you use a special code to kick it down to 14.4 kbps and use g711ulaw as the codec. I think your best bet is to try to eBay the custom nic for the TiVo series 1. Erik On Thu, 6 Jan 2005 20:39:45 -0500, David Ishmael <[EMAIL PROTECTED]> wrote: > Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series > 1 (yes its old). When I do a test call with Tivo, the call always fails (it > seems to dial the number but never connects). I can pick up the phone line > and hear the Tivo "talking". I've tried looking around for anything special > I need to do but its still not working. I can connect a phone to the > SPA-1001 and can make outgoing calls just fine. I even tried calling the > Tivo number and can hear the modem pick up. Has anyone done this? Any help > would be greatly appreciated. > > Thanks, > Dave > > ___ > 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] call waiiting and 3 way calling
Greetings, I recently dumped my packet8 line in favor of sixtel + asterisk/sipura spa-1000. It's working great, but I want to enable call waiting and 3 way calling. When i am on a call and get a second call, currently things go straight to voicemail. It looks like sixtel is passing on the extra calls, but my asterisk box doesn't know to route it all the way. Is there a way to set it so that i can have call waiting, and possibly 3 way calling like packet8 does? Thanks, Erik ___ 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 2000 vs 2100
The 2100 has two ethernet ports On Thu, 6 Jan 2005 09:29:48 +0100, Robert Rozman <[EMAIL PROTECTED]> wrote: > Hi, > > I've found approximate same pricing for both. Sipura 2100 seems to have more > features... > > What are differences between those two ?What about their reliability > (specially regarding fact, that they deal with analog phones) ? > > Thanks in advance, > > regards, > > Rob. > > ___ > 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] Cisco 7920 and Asterisk - Anyone got this working?
I'm not entirely sure this phone supports sip. Have you tried building the asterisk extra's and configuring it with skinny? Erik On Thu, 6 Jan 2005 00:37:32 -, Paul A Brown <[EMAIL PROTECTED]> wrote: > I am having all sorts of probs. It just won't connect. Anyone got any > example configs I could look at? > > Thanks > > Paul > > ___ > 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] 8 pstn lines+ on Asterisk supported hardware.
Or just get a couple of these: http://www.ipeya.com/VOIP_Products.htm (Specifically the 4 Ports FXO SIP VOIP-PSTN Gateway) Available from eBay at a discount at: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61839&item=5741966868&rd=1 And do it all without worrying about irq's or the motherboard. Just let the device do it's job. Erik On Mon, 3 Jan 2005 15:48:47 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > I have an Asterisk with 2 TDM with 8 FXO modules and I don't have any > problems. > > One thing to look for is that the cards don't share any IRQ. > > Use a motherboard where you can assign IRQ to the PCI slot. I used an > Intel board. > > Hope this help > > On Mon, 3 Jan 2005 19:43:12 +0200, Hadi Jadallah <[EMAIL PROTECTED]> wrote: > > Hi all, > > > > I have this project that requires me to use 8 PSTN lines and possible more. > > I was thinking 2 TDM cards with FXO modules. > > The I got to read the "Qs about FXO/FXS cards" thread and that scared me. > > Can anybody recommend anything that is known to work ok with no mysterious > > problems? > > I was thinking OpenSwitch12 cards. What do you guys think? > > Any help is appreciated. > > > > Regards, > > Hadi > > > > -- > > No virus found in this outgoing message. > > Checked by AVG Anti-Virus. > > Version: 7.0.290 / Virus Database: 265.6.7 - Release Date: 12/30/2004 > > > > > > ___ > > 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] How to connect two Asterisks as secure as possiblewithout too much additional bandwidth ?
Check this out: http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/ There's an article on how to use openvpn to encrypt data between two Asterisk Boxes. Should help, looks easy enough. Erik On Tue, 28 Dec 2004 16:57:11 -0800, Christopher Dobbs <[EMAIL PROTECTED]> wrote: > This problem is being solved. > See > http://lists.digium.com/pipermail/asterisk-users/2004-November/073666.html > I am currently in pre-testing phase of development. > > Features include: >Optional Secondary Compression >Selectable Encryption Level, from 32bit to 1024bit >Uses UDP >Voice and Data over same Link >Trunking >ADSI Support > > -- > Christopher Dobbs > > ___ > 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] Recommended IAX softphone.
Because it works well and supports a lot of codecs. All the other ones I have used were underdocumented and buggy. Firefly works perfectly, is very easy/customizable and free. Works great with Asterisk in both IAX2 or SIP. Erik On Fri, 24 Dec 2004 01:32:09 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote: > On Thu, 2004-12-23 at 15:50 -0800, Erik Espinoza wrote: > > I'd recommend Firefly by Virbiage. It's free and works on third party > > networks with sip/iax2 support. > > Any specifics as to the why? I browsed their site and it made a good > impression, but I didn't try it yet. The phone itself though seems to > have good codec support and nice looks. > > The latter is of minor importance to me, but might make it more easy to > convince my fellows to install, as some of them are long time Windows > users who tend to think that only software that looks nice is actually > good. > > So, presumed you're using the phone yourself, you are then satisfied > with sound quality, robustness, latency and asterisk interoperability? > > Thanks, Bruno. > > > ___ > 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] Recommended IAX softphone.
I'd recommend Firefly by Virbiage. It's free and works on third party networks with sip/iax2 support. On Fri, 24 Dec 2004 00:27:28 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote: > On Thu, 2004-12-23 at 16:36 -0600, Michael Van Donselaar wrote: > > > iaxComm is Open Source, and currently runs on Win32 and i386Linux platforms. > > Earlier versions run on Mac OSX, but I don't have hardware to compile it, > > and > > have not had any recent reports. > > Thanks Michael > > I've tried it and it seemed a reasonable choice to me, with it's codec > support, clean gui plus being open source. I guess I'll go for it > then ... > > Regards, Bruno. > > ___ > 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] IAXy playing dead again
As much as I appreciate the work done by Digium on Asterisk, it appears as though the IAXy is not ready for prime time. 1) IAXy has no security of any kind, anyone with iaxyprov can reprovision your device without so much as a password!!! 2) The IAXy doesn't work with regular dhcp, it uses bootp (thus never renews an address, which confuses quite a bit of dhcp servers) 3) Supports only two codecs, pcm/ulaw 4) Not configurable via http 5) No default IP (usually not a problem, if the damn thing would do dhcp!!!) 6) Cost almost twice as much as Sipura SPA-1001 I'm mentioning this in the mailing list because when I had issues getting the IAXy to get an ip from a Microsoft DHCP Server, Digium instructed me to ask in the mailing list or on irc. Digium charges quite a heavy premium for their equipment, and gives away their software. Weird how Asterisk is the coolest thing since sliced bread but their hardware is somethin right outta the trash heap. On Wed, 22 Dec 2004 11:12:00 -0600, Jay Milk <[EMAIL PROTECTED]> wrote: > Sounds like a thermal problem -- which most "intermittent" problems are. > Had this happen with a network switch in my home office. Pull it out, > disconnect it and put it in a cool spot for a few hours. If the problem > goes away, see whether you can stabilize the environmentals. > > > -Original Message- > > From: Wilson Pickett [mailto:[EMAIL PROTECTED] > > Sent: Wednesday, December 22, 2004 10:26 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] IAXy playing dead again > > > > > > It's happened before, cleared up and now it happened again. > > > > The IAXy, working for a total of about 6 months. > > > > Symptoms: > > > > Registered with asterisk and even receives calls (the LED shows it's > > ringing) but phones connected to it are dead. > > > > Same phones work connected directly to the phone line. > > Cable swapped out, no difference. > > Endless re-provision (with normal looking output) and power > > recycling. > > > > This really looks like a dead FXS - except - this has > > happened before and it came back. > > > > Comments? Suggestions? > > > > As a TV repairman once told me, "The most obscene word in > > technology is 'Intermittent' " > > ___ > 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] Example config for SPA-1001
try auth=md5 under sip.conf
On Mon, 20 Dec 2004 18:35:53 -0800, Paul Austin
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> Has anyone managed to create a setup with a Sipura SPA-1001 as a client?
>
> Right now I can connect to the device by dialing the extension number
> but when I try to connect from the phone handset to make an outbound
> call it gives an unavailable tone.
>
> I'm using Line 2 on the SPA-1001 to connect to the local asterisk
> server, line 1 is used to connect to my VOIP provider until I can get
> the local asterisk setup to work.
>
> My config is below
>
> Thanks,
> Paul
>
> sip.conf
>
> :
> [201]
> type=friend
> host=dynamic
> username=201
> secret=
> mailbox=201
> context=local
> callerid="Me" <201>
>
> extensions.conf
> ---
> [globals]
> PHONE1=SIP/201
> PHONE1VM=201
> :
> [macro-stdexten]
> ;
> ; Standard extension macro:
> ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
> ; ${ARG2} - Device(s) to ring
> ;
> exten => s,1,Dial(${ARG2},20)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
> :
> [default]
> exten => s,1,Answer
> exten => s,2,DigitTimeout(5)
> exten => s,3,ResponseTimeout(10)
> exten => s,4,Background(pls-entr-num-uwish2-call)
>
> exten => 201,1,Macro(stdexten,${PHONE1VM},${PHONE1})
>
> exten => t,1,Hangup
> exten => i,1,Playback(invalid)
>
> ___
> 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
>
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[Asterisk-Users] Cisco Router FXO / Skinny
Hey, Does anyone know how to use the cisco router with an fxo wic with Asterisk? I don't have enough space on this device to support an IOS that supports sip or h323. Currently the only one signaling in there says Cisco. I assume this is the skinny protocol. Does anyone know how to configure this 2600 with Asterisk? Thanks, Erik ___ 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 Toll-Free
NuFone.Net has 800 toll free IAX termination. They have good quality of calls once everything is setup and running. I'd recommend starting with a low dollar amount into your account first, until you have everything working. I went through about 20 e-mails back and forth via their request tracker, and they kept blaming my configuration when in fact it turned out to be an error on their end. Not sure if this is common, but it's a good step to protect yourself. If only Jeremy McNamara spent as much time working out the bugs on his business rather than blasting every user on this list who doesn't buy digium . . . oh well one can hope. Erik On Mon, 13 Dec 2004 12:14:03 -0800, Mark Halverson <[EMAIL PROTECTED]> wrote: > Sorry if this is the wrong list... > > I need a toll-free number to be delivered to me on IAX. (This is NOT an > existing number need to buy the whole service.) > > Anyone know of a service provider offering this? > > -Mark > 707-735-1038 > > ___ > 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] IAXy Configuration
Did that, and still no. I'm wondering if this just wont work with a windows dhcp server. On Sat, 20 Nov 2004 03:29:42 +0100, Wilson Pickett <[EMAIL PROTECTED]> wrote: > > > I can't seem to get this device to grab an ip from dhcp. We have a > > Look at the list of DHCP on the server, and if possible delete all > leases. Then try to power up the IAXy. > > My own IAXy suddenly stopped working today. In fact, it was due to the > coincidence that its internal ip had changed (DHCP) and it didn't > receive the provisioning when the asterisk server changed its address > due to being behind a NAT router. > > > ___ > 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] IAXy Configuration
Never Got that far. The device wont get an ip from dhcp for me to provision it. . . On Sat, 20 Nov 2004 03:29:03 +0100, BetaTeilchen <[EMAIL PROTECTED]> wrote: > Erik, can you please post your config-file ? > > Erik Espinoza schrieb: > > > > >I can't seem to get this device to grab an ip from dhcp. We have a > >working dhcp server (unfortunately it is on Windows), but I don't show > >any leases requested by the iaxy. > > > >Anyone have any ideas? > > > >The ethernet and phone lines are plugged in before the device is powered. > > > >Thanks, > >Erik > >___ > >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] IAXy Configuration
I can't seem to get this device to grab an ip from dhcp. We have a working dhcp server (unfortunately it is on Windows), but I don't show any leases requested by the iaxy. Anyone have any ideas? The ethernet and phone lines are plugged in before the device is powered. Thanks, Erik ___ 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] Cisco FXO
Hello All, I have a router with an fxo that I would like to tie into Asterisk. I have the sip configuration down and used the example at http://www.voip-info.org/wiki-Asterisk+cisco+FXO to create a config for the Cisco router (3640). Although the router detects the vic, I am unable to set the sipv2 protocol. The only protocol available on this router seems to be "cisco" (I assume that means callmanager). I was wondering if anyone out there knew the type of IOS I should get loaded on this bad boy to support sip. Thanks, Erik ___ 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] Cisco FXO
I didn't see an IP Voice, but I did see an IP Plus and an IP/H323. I thought the IP Plus would have it, but was concerned. I know that Asterisk supports h323 so I was considering that one. Anyways thanks for the feedback. Erik On Wed, 13 Oct 2004 21:53:25 -0700, Erik Espinoza <[EMAIL PROTECTED]> wrote: > I didn't see an IP Voice, I did see an IP Plus however... > > > > > On Wed, 13 Oct 2004 20:57:22 -0500, Henry Devito <[EMAIL PROTECTED]> wrote: > > You need to upgrade to the latest IOS. IP voice or IP plus for full SIP > > features. > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Erik Espinoza > > Sent: Wednesday, October 13, 2004 8:24 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Cisco FXO > > > > Hello All, > > > > I have a router with an fxo that I would like to tie into Asterisk. I > > have the sip configuration down and used the example at > > http://www.voip-info.org/wiki-Asterisk+cisco+FXO to create a config > > for the Cisco router (3640). Although the router detects the vic, I am > > unable to set the sipv2 protocol. > > > > The only protocol available on this router seems to be "cisco" (I > > assume that means callmanager). I was wondering if anyone out there > > knew the type of IOS I should get loaded on this bad boy to support > > sip. > > > > Thanks, > > Erik > > ___ > > 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] Generic X100P's
I'm not looking to deploy a new pbx, I'm just looking to set up a system that will call whoever is on-call that night based on a schedule. This system would need to interact with our existing pbx, and I think the tdm is the better approach. Thanks for the insights, however. They are always welcome. On Mon, 11 Oct 2004 17:49:52 -0700 (PDT), Steve Edwards <[EMAIL PROTECTED]> wrote: > On Mon, 11 Oct 2004, Erik Espinoza wrote: > > > Just want to test with real phone lines to ensure that they > > work with our existing pbx before deploying with 8 lines using the > > tdm400p's > > Consider using a t100p and a channel bank. Here's the breakdown: > > TDM T1 > > 2 x tdm40b $6101 x t100p $495 > 1 x channel bank* $200 > 1 x breakout box$100 > > $610$795 > > * I purchased a used (looks new to me) Adtran TA750 with 4 fxo and 20 > fxs ports for $225 a couple of weeks ago. I've seen loaded TA750's go for > between $150 and $350. An Adtran 600 would also be a good choice. > > For the additional $185, you get 16 additional ports, no echo problems, > and no interupt problems. > > I chose a breakout box for my on-the-road demo kit. You may already have a > 66 block or a 110 block wired up. > > Also, if you can get the fxo lines off your old pbx on a t1, you may shave > off a bit more. > > I've got 1 host with a tdm40b and an x101p and another host with the t100p > and the channel bank. The "T1" host is rock solid. > > > > On Mon, 11 Oct 2004, Erik Espinoza wrote: > > >> If reply all actually responds to the reply-to header and reply doesn't, > >> your MUA is broken. > > > > There is no reply-to header being added in from the asterisk-users > > mailing list, I double checked this by looking through other peoples > > posts. The MUA works fine with mailing lists that actually add the > > reply-to header. > > > >> 2 cards is the highest number recommended. But as I mentioned, it won't > >> be completely representative of your suggested final deployment and may > >> cause you unforeseen trouble. If you are being serious about testing for > >> real deployments, you should go ahead and buy final hardware. If you are > >> testing to deploy for a customer, you need to be very aware of your > >> final hardware. > > > > Thanks for the heads up, I'll keep that in mind. I'm building a system > > that is just going to recieve user calls for help and call the person > > who is supposed to be on call for the night. This will be tested > > amongst our managers for a few days, making sure that calls ar routed > > properly and that the software works as promised before purchasing the > > final hardware. The agi's are all written and in the 'all voip' system > > work fine. Just want to test with real phone lines to ensure that they > > work with our existing pbx before deploying with 8 lines using the > > tdm400p's > > ___ > > 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 > > > > Thanks in advance, > > Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST > Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 > ___ 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] Generic X100P's
> If reply all actually responds to the reply-to header and reply doesn't, > your MUA is broken. There is no reply-to header being added in from the asterisk-users mailing list, I double checked this by looking through other peoples posts. The MUA works fine with mailing lists that actually add the reply-to header. > 2 cards is the highest number recommended. But as I mentioned, it won't > be completely representative of your suggested final deployment and may > cause you unforeseen trouble. If you are being serious about testing for > real deployments, you should go ahead and buy final hardware. If you are > testing to deploy for a customer, you need to be very aware of your > final hardware. Thanks for the heads up, I'll keep that in mind. I'm building a system that is just going to recieve user calls for help and call the person who is supposed to be on call for the night. This will be tested amongst our managers for a few days, making sure that calls ar routed properly and that the software works as promised before purchasing the final hardware. The agi's are all written and in the 'all voip' system work fine. Just want to test with real phone lines to ensure that they work with our existing pbx before deploying with 8 lines using the tdm400p's ___ 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] Generic X100P's
Sorry, I hit reply instead of reply all. Erik On Mon, 11 Oct 2004 16:03:51 -0500, Steven Critchfield <[EMAIL PROTECTED]> wrote: > > > On Mon, 2004-10-11 at 14:07 -0700, Erik Espinoza wrote: > > > Since every zapata card generates 1k interupts per second and needs to > > > be on it's own interupt, your demo with 2 cheap winmodems isn't going to > > > be representative of your final config > > > > Am I to understand this that using two x100p's or more in one box is a > > no no? Or should I just change the interrupts? > > The list is there for a reason. The reply-to header is set to the list > to help keep messages on list and help others that might potentially > actually look for answers on their own. Taking a message off list > reduces the effect of answering the question. you also loose the benefit > of answering community questions such as being seen as a knowlegeable > person. The community doesn't get a chance to refute answers that are > incomplete or just wrong. > > All that is to basically let you know that I don't answer private mails > about asterisk unless there is a paycheck involved. > > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ 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] Generic X100P's
Alright, I'd like to start out by saying that this is just a proof of concept. The final configuration will include the purchase of a couple of TDM400P's so all flames for using cheap Winmodems please divert yourselves to /dev/null and don't waste my bandwidth or your time responding. I've noticed that all documentation online talks about how to configure a single x100p (or generic) in /etc/zaptel.conf and /etc/asterisk/zapata.conf. My question is I need to use two for a proof of concept at work. I have a lot of experience with a purely voip (voicepulse, voipjet, fwd, and iaxtel) so I feel I understand asterisk's flow and configuration really well. My question is, for using two x100p's are there any changes that need to happen from the basic x100p config I've seen online. Here's what I'm using: /etc/zaptel.conf defaultzone=us fxsks=1 /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=recieved channel => 1 Examples would be appreciated. Thanks, Erik ___ 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
