Re: [asterisk-users] IAX2 Registered OK without IP
On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass a...@p2ee.org wrote: we upgraded to 1.8.13.1 and we have much the same problem although after the upgrade I don't seem to find any cases where the qualify value is OK (xx ms) and the IP is gone (like we had in 1.4.29) but the effect is the same: the extension becomes non-reachable pretty quickly. I stand corrected. It's EXACTLY the same behavior as 1.4.29, The Status shows OK (XX ms) and the IP is (null) IMHO this is indicating that the qualify settings are being ignored, I've been experimenting with qualifyfreqok and qualifyfreqnotok and just by specifiying _any_ values for these parameters it makes matters much worse. and the only workaround has been lowering the re-register time to sometimes as low as 3 seconds. Even though several docs say that the A re-registration every 3 seconds seems to do the trick but why can't qualify keep the connection alive?? qualify values can also act as a keep alive, it's not working for us and I still have to reduce the register time to very low values because qualifyfreqok and qualifyfreqnotok don't see to be doing anything... Stand corrected: they just make the problem worse. Any clues?? I get the feeling that IAX extensions are definitively not very popular even though most of the older concerns about IAX should be gone by now. We really want to support and keep investing into making IAX work for us but the little support we get here is really discouraging. Best, -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Registered OK without IP
This has come up before on the list and archives but I don't seem to find a solution for this. On just a few nodes we have this situation where we see the IP disappear from the CLI iax2 show peers list but the status shows OK: 3012/3012(Unspecified) (D) 255.255.255.255 0 OK (89 ms) How can the status be OK a few milliseconds ago and have no IP ?? The strange thing is that the IP does show up once in a while and then disappears once again but the OK is always there. Asterisk 1.4.29 running on FreeBSD 7.0-RELEASE -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another IP address to block
We use complete regional blocks from Wizcraft and blocking at minimum all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We block almost anything that is not our actual customer market and screw the rest. On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavez cur...@telecomabmex.com wrote: Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Alejandro, I removed the registration and tried as like yours, even inbound calls are not landing, anyways let me check with vitelity support. In the Vitel web app you ust set the routing method to the IP of your pbx, maybe that's what's happening I'm pretty sure they check that the outbound calls use the same IP. Hi Stephan, I am not using any SBC. As i said let me check with their support. Thanks for all the views comments. Regards, On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). We had lot's of trouble with the 800 numbers as well but after help from Vitelity's support we were able to determine that the problem was that toll free require _exactly_ 10 digits to accept the toll free call. Regarding call to the Caribean we had a lot trouble with cell phones in Venezuela and it seems they were using pre-paid lines that ran out money but they eventually got around and solved it. So I think that if you insist with their support they usually resolve the issue. Best, -- Alejandro Imass Cheers, Jeff LaCoursiere SunFone On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alexander wrote: Alejandro's setup looks correct; you can also get the correct config using Vitelity's wizard tool for setting up the trunks. The only thing I would add is that if your account is setup with a session border controller you will need to use the SBC's IP address instead of the IP the wizard gives you. If you have an SBC, the fact will be noted in your account including the IP address. I've found Vitelity's tech support to be pretty helpful too, should you need to contact them. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass a...@p2ee.org wrote: On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Setup
On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? Thanks, -- Alejandro Imass Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
On Sat, May 12, 2012 at 5:05 PM, Eliezer Croitoru elie...@ngtech.co.il wrote: On 10/05/2012 11:49, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. VIA Artigo Pico ITX Does anyone have inspiration/experience for/about such a model? thx!! BC i dont remember where i have seen it but i have seen this relio industrial Intel Atom: http://www.sealevel.com/store/computing-hmi/industrial-computers/compact/r1420-relio-1-6ghz-intel-atom-n450-embedded-computer-2gb-ram.html but the main thing you should ask yourself is what do i need it for? if it's for just low power consumption you can get cheaper units. if you need just small then you can get others as well.. this one is meant for a very industrial environment. Eliezer -- Eliezer Croitoru https://www1.ngtech.co.il IT consulting for Nonprofit organizations eliezer at ngtech.co.il -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)
On Sat, Apr 14, 2012 at 7:55 PM, Joseph syscon...@gmail.com wrote: I forgot to add: clinic-amd*CLI iax2 show peers Name/Username Host Mask Port Status home_server (null) (D) 255.255.255.255 0 Unmonitored iaxy-322/iaxy-3 (null) (D) 255.255.255.255 0 Unmonitored The two Asterisk boxes have to register to each other so home server and clinic have to be on the peers list on both sides (or at min on one side) and you must see the IP. Make sure you are resolving the names correctly and that Asterisk is set-up to resolv DNS names or use the IP instead. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just work. We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. Our set-up is fundamentally public and private Asterisk servers running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD 8.2 and Asterisk 1.8 but we are in that process right now. Some Asterisk run in jails so I can understand the NAT issues there may be caused by the server itself. I honestly *love* your OpenVPN idea but I have to find a cheap ATA that could run as an OpenVPN client. Taking the simplest example a simple Asterisk 1.6 server on a public IP running on the base system (not in a jail): We run an operation that spans several countries including Canada, the US and the Latin American Andean region. As examples, with Canadian ISPs such as Rogers and Bell we have always had to redirect the ports and use STUN server for the HT-286 to register to the Asterisk server. In the US we have the same problem with Comcast networks, so I don't understand how you say that you plug a Grandtream SIP ATA to a Comcast router and it just works. However, in a couple of NOLA countries the ISP's routers actually give public IPs, so if the SIP ATAs are connected directly to the ISP router, or in the DMZ then it just works as you say, BUT if the ATA is connected behind the firewall, or to a WiFi router, then we've _allways_ had to redirect ports. In every sigle customer we have had to send instructions on how to redirect ports, and of course to configure firewall if present. I just don't understand how you and other here say that a SIP ATA can just work. On the contrarty, with IAX2 using cheap AG-188N from Atcom they are just plug and play when shipped with a standard conf, and we have none of the quality issues you are referring to. We do have some call drops however, and some hangup problems but they don't affect our clients as much as having to deal with NAT issues. We may not run 15K extensions like you but I think we have a pretty good testing ground and have dealt with a fair share of NAT problems with SIP, that you and others here apparently don't have, so I am as amazed by your likeness of SIP as perhaps you are amazed as our likeness of IAX. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 02/29/2012 08:22 AM, Alejandro Imass wrote: [...] The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. My point of view is that we've had many problems with SIP and NAT and that IAX just works great for us, and that in *our* experience IAX has worked better for us. Just to clear my head up a bit: are you supporting the argument that SIP is better for Asterisk than IAX? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 10:34 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] If you can post some SIP debug info from an ATA trying to register without any redirection and also the relevant portions of your sip.conf, I am sure I can help. Do it from a new location with an el cheapo home router, Linksys WRTXXX. Yeah, I think it's time for me to shut up about SIP/NAT problems and, like you Carlos and Kevin pointed out, run a clean un-contaminated test lab to see if we can determine why our current set-up is so problematic with SIP and NAT. If I cannot help you in a few emails, we can take this offline. Thanks for offering to help. I will set-up a test lab but it's gonna take me some time to free a public server to do so. But it is obvious that the problem is on our side after reading all the responses. After all, VoIP is *not* by any means our core business we just use it as a tool, and up until now I thought that *everyone* using SIP ATAs and Asterisk had these NAT woes, so we just assumed it was so, and thought that mostly everyone had to perform particular configurations on the endpoints. It now seems obvious we are wrong. Anyway, my whole argumentative line in this thread is that in our particular case we found that IAX2 works great for _our_ set-ups and we don't share the view that IAX2 is a broken bat, and that in fact for us it just works great. Thanks, -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-29 15:25:49 +, Alejandro Imass said: We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. The original question (mine) was that my sound quality when using IAX was bad; with SIP the sound quality was great. Critically, I mentioned that I wanted to use IAX; I even said I was willing to do some self torture to get IAX working properly. Yeah, I wasn't referring particularly to the original post, just the way the thread turned against IAX like if it's not a viable solution and my point all along has been that for *us* IAX2 endpoints have worked better and easier to configure than SIP ones. Then it turned into a pissing contest, like you say, it happens in every list with the topic this or that. Again, as I pointed out to Steve above, and after reading all of your responses, our SIP/NAT woes seem obviously ignorance on our part, but that doesn't shadow the fact that IAX2 is working great for us with el-cheapo endpoints like Atcom's AG-188N and I would wish that many more manufacturers supported IAX2. We are happy with IAX and honestly never even had the need/curiosity to deal with the many SIP/NAT problems where sometimes it works great, and other times is a real pain in the ass that takes huge amounts of support to fix, and unhappy customers. On the other hand, IAX took some engineering efforts at first, but the support issues are practically non-existent. -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not the truth. - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:19 PM, Carlos Alvarez car...@televolve.com wrote: Perhaps your users live in an internet ghetto where the routers are similar to Yugos with spinners. We haven't run into any routers that don't do NAT properly in a very very long time. Perhaps you should read again and point out where I state that is a router/NAT problem. I said that the configuration of routers and redirecting ports is a pain in the ass for users and creates a lot of support problems that simply don; t exist with IAX2. With a IAX2 ATA you just plug it and works. This cannot be done with SIP and off the shelf cheap ATAs, period. Also, respect netiquette and don't top post and use derogatory remarks and keep your discussion technical. -- Alejandro Imass On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly varies. [...] Unfortunately, top-posting has become normal on this list. I'm just fitting into how others were quoting the conversation. If you find my Top posting seems to be more popular due to use broken smart phone MUAs that can't reply in-line. But if you have the means it should be avoided for future reference and direct people to read the archives and find useful information. remarks derogatory, I don't particularly care. You should care. Words like ghetto, your users, and using name brands like Yugo in a pejorative way are all derogatory and may direct the discussions on a personal level which should always be avoided. I don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. We are all here to share our knowledge and our valuable time so to make it worthwhile we should all care about conserving netiquette. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. What you are saying seems impossible and makes no sense unless the router is assigning a public IP or is SIP aware and knows how to read the routing data contained inside the SIP packets, and none of the consumer routers are SIP aware AFAIK, especially not the WRT-54G. The other option is that the SIP ATA has WAN and LAN ports and the SIP device is being assigned a public IP. SIP does not NAT by itself because it can't, because there is no routing info, it's simply impossible. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual Server
On Fri, Feb 10, 2012 at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I use Asterisk on FreeBSD Jails and works great: http://www.freebsd.org/doc/en_US.ISO8859-1/books/handbook/jails.html I heard, about proxmox, but I don't know if works fine. Regards Carlos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Registration Hijacking
On Thu, Jan 19, 2012 at 8:36 PM, eherr email.eherr9...@gmail.com wrote: I have a honey pot box with extensions that are not just numbers ie ) 100-MySipUserName I have the same problem and I use contactpermit with specific ip blocks! I know for a fact I'm getting hijacked by sip vicious on extension 100 but I can't understand how because I don't even have an extension 100 declared anywhere. I would like to know how to block this MF because he makes calls at 1-2 AM -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Registration Hijacking
On Fri, Jan 20, 2012 at 11:17 AM, eherr email.eherr9...@gmail.com wrote: I always thought Sip Vicious only does numbers ( 0 - 100 ) not Numberic-Alpha ( 100-MySipUserName ). To make my situation more interesting is that I also have fail2ban installed banning after 5 failed attempts. I too have fail2ban and running a relatively updated version of FreeBSD. BTW my install is plain Asterisk -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balance with 2 wan connections
On Sat, Dec 25, 2010 at 1:18 PM, dave george dgeo...@teletoneinc.com wrote: Need some advise or paid help on running asterisk on two WAN connection. I need load balancing and failover support. WAN: 1 DSL + 1 Cable ISP. There are _many_ issues. First outgoing and incoming traffic is completely different for what you want to do. Second SIP is hard enough to NAT and route with a single IP let alone 2 or more and probably dynamic! Third, load balancing/fail-over is not a simple matter even doing by hand with Linux or BSD, there will still be issues with static routing and such. There are some cheap hw that may claim it does, but most probably it will not be meant for VoIP, SIP or IAX. Depending on your budget and needs, if you need reliability and high bandwidth, probably a better solution is to host your main pbx in a reliable server on a fixed and public IP and then route the calls to a local Asterisk using IAX and even SIP. If local bandwidth is limited IAX is a better bet. By having a public box routing calls to local box(es) on your private LAN, you could load balance with multiple local Asterisk servers (easy balance by dialplan, for example). To save on hardware, you could use virtualization or FreeBSD Jails for example. Dunno how the telephony hw works with virtualization or jails (yet, thoug I do have a single Asterisk running on a FBSD jail). Good luck, Alejandro Imass Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
On Sun, Nov 21, 2010 at 8:14 PM, Daniel Bareiro daniel-lis...@gmx.net wrote: Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. You have to be a bit more specific. For example is your Asterisk box behind a router/nat? Or does your asterisk box have two NICs one for the public and/or natted IP and one for the LAN? You need to specify your exact setup. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? It's probably not a bug. Maybe you are registering by name and the name resolves to the public IP, and if you are in a DSL cable connection you public IP will change and perhaps you don't even have a public IP. Another possibility is that your ISP does not in fact give you public IPs (like most in the USA) and you have your LAN in the same network definition as theirs. I mean there are so many possibilities but you need to specifiy the exact network setup (IPs, masks, routing, etc.) Thanks in advance for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
On Mon, Nov 22, 2010 at 11:44 AM, Daniel Bareiro daniel-lis...@gmx.net wrote: Hi, Alejandro. A few days I have problems connecting to the Internet on my house [...] It would appear that the server for some reason was 'locked'. For example, when I try to register from Twinkle softphone, I get the following: - lun 13:41:56 Daniel, registration failed: 503 Service Unavailable - I have had a similar problem when we have some sort of network disruption, but it _never_ affects clients on the LAN, it only affects my SIP registrations on the public network. I have 2 NICs one on the public network with a public IP (but dynamic), and one on the LAN. I also have a cron to a dyndns service that updates the name of this server so other PBX can register to it. Anyway, sometimes, but very rare, something happens and the is no way that it re-registers to external SIP sources, and no other external SIP can register with it either. Nothing works except to reboot the server a-la Windoze. I have Asterisk 1.6 on FreeBSD 8. I have always attributed this problem to my set-up or a quirky NIC but maybe it's related to your problem (although it has _never_ happened to us in the LAN extensions). Unable to find a solution, and since it's really very rare, we have test calls every day to make sure everything is working ;-) Best, Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ekiga can register but not my IP phone
On Sat, Nov 20, 2010 at 5:31 AM, Benoit Chabrier c...@chab.info wrote: Thanks for your help. you were right it also work without a stun server adding to sip.conf: externip=78.47.x.x ; in [general] the IP of the dedicated server nat=yes ; in the description of my peer Exactly. BTW, IAX doesn't have such problems ;-) but sadly not too many devices support it. Especially if you plan to connect your Asterisk boxes, even though you can use SIP, it's _strongly_ advisable you use IAX instead. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ekiga can register but not my IP phone
On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote: Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). I am assuming you mean Asterisk on Ubuntu inside the LAN But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing is that ekiga can connect to the same asterisk server with the same SIP account. Is this outside the LAN? Is there NAT in between? SIP is a pain in the ass with NAT, so it's the only thing I can think of. Usually in my experience it's the other way around! Ekiga is the one that doesn't work and tends to be very quirky (takes a long time to quit, has strange registration quirks, etc.), I mean when compared to HW SIP device. Here is a part of my sip.conf : [general] dtmfmode=auto language=fr ; pour les messages lus par asterisk disallow=all allow=ulaw allow=alaw allow=speex [siemens] type=friend context=interne host=dynamic secret= When i'm doing a sip set debug ip XXX.XXX.XXX.XXX i have some information. It seems that asterisk receives the rengistration request but doesn't answer to it. Here are the logs : http://server.chab.info/Registration_logs_ip_phone.txt Using Ekiga with the same SIP account (name is siemens) and from the same physical location works well : http://server.chab.info/Registration_logs_ekiga.txt I didn't change anything about asterisk config (except sip.conf and extensions.conf). If you have any idea, please share it with me, i really don't to do to fix this problem... Thanks in advance ! The only thing I can think of are NAT issues with SIP. If you are in fact NATing try the Siemens phone to a direct IP to the server (no NAT, firewall, etc.) and see. -- Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ekiga can register but not my IP phone
On Fri, Nov 19, 2010 at 8:23 AM, Benoit Chabrier c...@chab.info wrote: Thanks Alejandro, you were right it was just a NAT problem ! i add a stun server in the phone configuration and it works :) Cool. Also Asterisk SIP conf file has some NAT settings as well that you can play with and perhaps do away with the stun server config in the phone. Here is a great article that explains in detail the issues with SIP and NAT: http://www.voipuser.org/forum_topic_7295.html 2010/11/19, Alejandro Imass a...@p2ee.org: On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote: Hello, I have a Sip phone (Siemens C470IP) which works perfectly with -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many Asterisk PBX operating in the World?
On Wed, Nov 17, 2010 at 12:06 PM, Andrew Latham lath...@gmail.com wrote: 2010/11/17 Sevana Oy sa...@sevana.fi: Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -- John Todd should have a good answer for this. I would start my estimate at 200,000+ if you are including all of the versions and types. Software like BigBlueButton includes Asterisk so it can get confusing real fast. How do you figure this number? I would suspect is a lot higher, especially if you consider just Digium sales in the equation. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many Asterisk PBX operating in the World?
On Wed, Nov 17, 2010 at 12:09 PM, Alejandro Imass a...@p2ee.org wrote: On Wed, Nov 17, 2010 at 12:06 PM, Andrew Latham lath...@gmail.com wrote: 2010/11/17 Sevana Oy sa...@sevana.fi: Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -- John Todd should have a good answer for this. I would start my estimate at 200,000+ if you are including all of the versions and types. Software like BigBlueButton includes Asterisk so it can get confusing real fast. How do you figure this number? I would suspect is a lot higher, especially if you consider just Digium sales in the equation. Just to complement some links we may all find interesting in figuring this out 1) http://www.nojitter.com/showArticle.jhtml?articleID=219401206 From this article, 2009 shipments were 5.5 million units. It also talks about others having 2-3% of this market share. In a rough calculation that is 165 thousand units for the other in 2009 alone. Say that Digium has 3% of the Other market share that's about 5K units for Digium alone in 2009. The article also estimates that OSS provides comprise about 10% of the overall Others share so that alone is 16.5 (this would include the 5K digium Units). Asterisk and Digium were founded circa 1999 and if the Digium sales have been doubling yearly as Digium states on their web page, and we assume that other Asterisk-based telephony systems overall have experience similar growths, collectively they may have sold say 32 thousand units in 10 years, + 2010 sales that would probably be closer to 50K by today. This is probably a low number IMHO, but a good pessimist starting point. So say that for every box sold, there a 3 Asterisks running on custom setups, it totals about 150K total asterisks running. If you accept the number to be 5 per every unit sold, then it 225K. In any case, these numbers tend to agree with the 200K figure provided by Andrew Latham. Of course, my analysis is very crude but I leads me to agree with the number provided by Andrew based on this sole study. 2) http://www.asteriskexpert.co.uk/about-asterisk.php This article states 18% of the _overall market_ and Asterisk being 75% of that. They make no mention of how the reached that number, in fact provide no numbers at all, and if those numbers are product sales or total systems (including non-product custom Asterisk set-ups). So we really can't use the sales figures above as any reference to calculate anything, but if we did, we would wind up with about 2.2 millions Asterisks that when divided by 5 (just for the sake of trying to factor the product/download logic above) we come up with 440,000 Asterisks units running. 3) http://www.telephonyworld.com/news/open-source-pbx-market-growth-fueled-by-repeat-sales/ This report mentions that 18% of OSS-based PBX the overall _PBX market_ (not differentiating from tradiciotnal or IP-based PBX market) and it came from the Eastern management Group a credible source in the area. I'm curious on why asteriskexpert.com failed to cite the source, unless they came up with the same conclusions on their own. Anyway, if Asterisk truly 13.5% of the overall PBX market, then the numbers are much higher than the 200K conservative estimate above. 4) http://www.thefreelibrary.com/The+IP+PBX+Market+is+Predicted+to+Triple+and+Reach+$19.5+Billion+by...-a0142975350 This link mentions that the IP PBX market will be around 19.5 Billion for to 2011, so even the most conservative estimates on Asterisk have to be quite high. My estimate: Although the Linux counter is not a fully reliable source, say that the 29 million user is conservative, just for the sake of argument. Security Space survey in August 2009 checked 38,549,333 publicly accessible Web servers, and they states that roughly 80% were Linux and FBSD (75/5 respectively). I personally think that there must be at least 50 million Linux/FBSD machines in the world in use today, easily (maybe even double that). If 1% of the servers use Asterisk that would be 500K. With the numbers above, I think it must be between 300 and 500K. Best, Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many Asterisk PBX operating in the World?
On Wed, Nov 17, 2010 at 4:04 PM, Andrew Latham lath...@gmail.com wrote: Wouldn't apply to you, Steve, but sooner or later somebody will probably imbed an innocuous phone-home into one of the Asterisk modules and it will take a C person like yourself to point out the Microsoft-ness of this snippet. How many people actually allow their primary PBXes to touch the Internet? Just curious as it sound funny to me. All of ours. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX Options
Hi, Is FAXing with Asterisk a practical option ? Or is it better just to use a plain fax connected to an FXS and just switch with Asterisk. I specifically wanted to know if there was any experience using just the fax scanner to send faxes and receive them via asterisk and the to e-mail. My idea was to take my old fax connect it to an FXS port and send faxes with the fax machine (using the fax mainly as a scanner), but receive them through our existing FXO jack that is connected to the PSTN. the scheme would be something like: PSTN -- FXO - | |Asterisk | FAX -- FXS - I'm using Asterisk 1.4.26.2 on FreeBSD 8.0 TIA, Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Options
On Mon, Aug 2, 2010 at 3:03 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Subject: [asterisk-users] FAX Options [...] TIA, Alejandro Imass IMO, as long as you're using PSTN and nothing fancy like T.38, Asterisk is a solid fax send/receive option. Could you recommend a good starting point? Like a faxing with Asterisk how-to... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Options
On Mon, Aug 2, 2010 at 7:26 PM, Mark Scholten m...@streamservice.nl wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Alejandro Imass Sent: Monday, August 02, 2010 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FAX Options [...] Here we have the following setup, could you say if that is acceptable for you? Thanks! Looks very much like what we're looking for... I am sure it will work for us but we have 1.4 let me test some stuff and get back to you here Outgoing fax: Fax - Linksys pap2t (sip, no t38, for settings see http://www.provu.co.uk/pdf/sipura/ip_faxing_sipura_linksys.pdf) - asterisk - sip trunk provider (this could also be some sip - pstn solution I guess) Thanks again! I will test this by the end of the week and post my results here to follow-up and close the thread. Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk, Skype For SIP
On Sun, Jul 18, 2010 at 7:48 AM, Vieri rentor...@yahoo.com wrote: Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype user. I do not need to call landlines via Skype. I think this is _explicitly_ not supported in the Skype for SIP docs. 2) allow Internet Skype users to call my Asterisk PBX Skype user and route the call to a specific Asterisk SIP extension. Here is how it goes from my experience with Skype: each SIP channel will cost you about $5 a month, regardless if you have a landline number with them or not. Your account will be assigned a special Skype number 99x . With that number a Skype user can call you and it will be free. You _cannot_ call Skype users from your PBX, as I stated above, this is an explicit no-no in the docs. If you want to make calls from your PBX to landlines you have to buy Skype credit just like you would with a regular skype client. If you want land-lines to call your PBX you need to purchase a skype number which about $60 a year. At first, I thought it would be simple and free. However, correct me if I'm wrong but the Skype user I can use within the Asterisk PBX cannot be the standard type (used by eg. desktop Skype applications) but needs to be created by the Skype User Manager for Business Solutions. I believe this has a price although Skype For SIP Open Beta seems to be free until Q4 2010. I think you can associate existing skype users to your Business Solutions manager but I still don't understand exactly how or why this is useful, and I don't think it has to do with you being able to call any of them from your PBX. Then again I haven't paid much attention to that and perhaps you have more insight into this. Has anyone found a way to make pure Internet user-to-user Skype/SIP calls via Asterisk (no PSTN involved) for free? As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. Best, Alejandro Imass Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling DAHDI...
On Thu, Jun 3, 2010 at 5:02 PM, Warren Selby wcse...@selbytech.com wrote: The resolution [1] to this issue was to uninstall and reinstall [2] the kernel headers on the machine...just in case anyone else runs into this issue and would like to know how it was solved. [1] https://issues.asterisk.org/view.php?id=17411 [2] run these commands to reinstall kernel headers: ]# yum remove kernel-devel ]# yum install kernel-devel FYI: I just had the same problem with Asterisk 1.6 on FreeBSD 7.0 RELEASE. Even with correct headers it would complain about som opt_netgraph.h that never existed. Finally, I resorted to try with Asterisk 1.4 and it compiled fine. Conclusion: 1.6 won't work with FBSD 7.0 Best, Alejandro Imass Thanks, --Warren Selby On May 25, 2010, at 9:48 PM, Warren Selby wcse...@selbytech.com wrote: I was at a client site tonight to install OSLEC on his machine running asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I stopped asterisk and DAHDI, downloaded the latest version of DAHDI 2.2.1 (dahdi-linux-complete-2.2.1.2+2.2.1.1) and made the necessary changes to compile OSLEC with DAHDI, but I ran into compilation issues that I had never seen before. So as a test I deleted my /usr/src/dahdi/ directory, re-expanded my tarball (so that I had a vanilla DAHDI package), and tried to compile it again, and I got the same errors. I have not seen these errors before, and I'm not sure what would cause them. Can anyone help shed some light on this? The 'make' output: *snip* -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to login to voicemail with Ekiga
Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says login incorrect. The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual login to the mailbox. I have searched many threads, and most if not all, talk abot the dtmf setiings, but both Ekiga and Asterisk are configured for rfc2833. Here is what I get in the console: [Mar 30 10:30:23] WARNING[1790]: app_voicemail.c:7236 vm_authenticate: Couldn't read username Thanks beforehand! Alejandro Imass sip.conf [101] username=101 type=friend secret=xx qualify=yes nat=no host=dynamic canreinvite=no context=home mailbox=...@home dtmfmode=rfc2833 extensions.conf [home] ...snip... ;internal sip extensions exten = 101,1,Dial(SIP/101,15) exten = 101,2,Voicemail(1...@home) ...snip... ;voice mail exten = 700,1,VoiceMailMain() ...snip... voicemail.conf [home] 101 = ,User Name,u...@domain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to login to voicemail with Ekiga
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria zisha...@gmail.com wrote: The message Couldn't read user name means it is not receiving the DTMF. Do you have an IVR to verify that your system is receiving the DTMF? If not, setup one, call into it and send Dtmf to it and see if it responds at all. If it doesn't, somewhere DTMF settings need to be adjusted. The IVR works fine, and we use it everyday. That's why it seemed to me that it could not be a stmf problem. Any other ideas? Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-03-31 9:15 AM, Alejandro Imass a...@p2ee.org wrote: Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says login incorrect. The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual login to the mailbox. I have searched many threads, and most if not all, talk abot the dtmf setiings, but both Ekiga and Asterisk are configured for rfc2833. Here is what I get in the console: [Mar 30 10:30:23] WARNING[1790]: app_voicemail.c:7236 vm_authenticate: Couldn't read username Thanks beforehand! Alejandro Imass sip.conf [101] username=101 type=friend secret=xx qualify=yes nat=no host=dynamic canreinvite=no context=home mailbox=...@home dtmfmode=rfc2833 extensions.conf [home] ...snip... ;internal sip extensions exten = 101,1,Dial(SIP/101,15) exten = 101,2,Voicemail(1...@home) ...snip... ;voice mail exten = 700,1,VoiceMailMain() ...snip... voicemail.conf [home] 101 = ,User Name,u...@domain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to login to voicemail with Ekiga
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria zisha...@gmail.com wrote: The message Couldn't read user name means it is not receiving the DTMF. Do you have an IVR to verify that your system is receiving the DTMF? If not, setup one, call into it and send Dtmf to it and see if it responds at all. If it doesn't, somewhere DTMF settings need to be adjusted. Zeeshan A Zakaria Is this the right list? or is this for final users? -- Sent from my Android phone with K-9 Mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
