Re: [asterisk-users] How to receive SMS ?
On Sat, 18 Feb 2012 20:50:25 +0100, Andreas Sikkema h...@ramdyne.nl wrote: We're using a GSM gateway to send SMS messages from our network monitoring system. Once you dig through some chipset specs it was suprisingly easy to start sending SMS messages. While we didn't investigate receiving messages fully we did one quick test and that was easy enough. You just need some daemon to monitor the gateway to see if it has received a message and pass it on to Asterisk, sending the other way around is not that different. Thanks for the feedback. Can someone recommend GSM gateways for small businesses? -- _ -- 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 to receive SMS ?
On Sat, 18 Feb 2012 12:21:31 +0100, Administrator TOOTAI ad...@tootai.net wrote: Not true. Some GWs have only a phone port that you connect to an ATA. Good to know. What brands/models would you recommend? -- _ -- 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 to receive SMS ?
On Thu, 16 Feb 2012 19:41:16 +0100, Olivier oza_4...@yahoo.fr wrote: You mean you can receive SMS on a landline in France (or the opposite) ? Supposedly, but I never used it either. www.google.fr/search?q=sms+ligne+fixe+asterisk If a gateway has its own SIM card and GSM stuff, should it receive SMS ? Sure, since it's just a regular cellphone with an Ethernet plug to connect it to the rest of the network. I'd also be interested in learning from anyone who uses a GSM gateway to TX/RX text messages with Asterisk and SIP clients. -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
On Wed, 8 Feb 2012 18:17:46 -0800, Chad Wallace cwall...@lodgingcompany.com wrote: Maybe the release announcements are what you're looking for. e.g., for 1.8: http://www.asterisk.org/node/51444 And you can probably find the same for 1.4, 1.6.x, and 10 without too much trouble. Thanks. It's closer to what I was looking for. I'm just surprised that there's no easy way to know what major features explain why Digium decides to create a new version/branch, which would make it easier to check if it's worth upgrading. -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
On Wed, 08 Feb 2012 20:23:54 -0600 (CST), Richard Mudgett rmudg...@digium.com wrote: The CHANGES file is not just a dump. It is a manually created file that documents each feature addition. There is a ChangeLog file that is a dump of every single commit made to the source file. Sorry about that. Indeed, the CHANGES appears to be a higher-level view of changes brought by a release http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES Does someone of a good site/blog that keeps track of new releases of Asterisk, and explains what the major changes/features when they do occur? Thank you. -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
On Thu, 9 Feb 2012 11:13:38 +, Steven Howes steve-li...@geekinter.net wrote: Why not just use the latest version?.. Because converting Asterisk to run on that non-x86 platform is quite some work, so I need to know what I'm missing by staying with a 1.4.x 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] Major changes between 1.4/1.6.1.8/10?
On Wed, 08 Feb 2012 15:58:43 -0600, Kevin P. Fleming kpflem...@digium.com wrote: No, unfortunately that's not quite correct. The UPGRADE files list *important* changes that users need to know about because they are changes in behavior of existing functionality. New features, even really useful and widely anticipated ones, that don't cause backwards compatibility issues are only listed in the CHANGES files. For example, the addition of T.38 gateway support in Asterisk 10 only appears in CHANGES, not UPGRADE, because if you don't use it, it doesn't affect you. Thanks for the tip. However, the CHANGES fille is just a dump of every single change that was made with each release, so it's hard to tell why a user should upgrade to the next major release (eg. 1.6 to 1.8). Is there really no article on the web that sums up what the major changes were within the four active branches? I'm running 1.4 on a non-x86 platform, and before I spend time trying to cross-compile, I need to know 1) whether I really need to upgrade, and 2) if that's the case, to which version. Thank you. -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett rmudg...@digium.com wrote: The UPGRADE.txt and CHANGES files do just that. They have been a part of the Asterisk source files for a long time. Thanks for the info. The problem is that the ChangeLog files http://downloads.asterisk.org/pub/telephony/asterisk/releases/ are very long to read, and make no distinction between tiny features/bug fixes and major changes, so non-experts are unable to tell them apart. No Asterisk expert keeps track of new releases and blogs about major changes when they occur? At the very least, what is the main difference between the four branches currently under development, so that 1.4 users can tell if it's worth upgrading to another branch (save for the end-of-lifed branches)? Thank you. -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
On Tue, 7 Feb 2012 14:31:31 +, Steven Howes steve-li...@geekinter.net wrote: The upgrade files may be more to your tastes than changes files. Thanks. I downloaded and untarred asterisk-1.8.8.0.tar.gz, and it looks like the UPGRADE*.txt files within tarballs are the closest there is to knowing what major features were introduced in each branch, so as to make an educated guess as to whether it's worth upgrading to a newer release/branch. -- _ -- 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] Asterisk V/s FreeSwitch
On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
On Tue, 07 Feb 2012 06:10:37 -1000, Jean-Denis Girard jd.gir...@sysnux.pf wrote: This link also presents changes between Asterisk versions: http://linuxinnovations.com/applications1.4-1.6.2.html Thanks for the link. -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
Hello Is there a document that sums up the major changes made to the four main releases available (1.4, 1.6, 1.8, and 10), to check if it's worth upgrading? www.asterisk.org/downloads Thank you. -- _ -- 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] Router that support Asterisk
On Wed, 01 Feb 2012 18:47:49 -0500, James Sharp ja...@fivecats.org wrote: The Cisco DDR2200 that I just got from Centurylink for DSL appears to be just that. I haven't tested the FXS ports on it yet, though. Cisco announces the end-of-sale and end-of-life dates for the Cisco DDR2200, DDR2201, and WAG310G ADSL2+ Residential Gateways. www.cisco.com/en/US/prod/collateral/video/ps8611/ps9520/ps9524/end_of_life_notice_c51-694180.html -- _ -- 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] [NAT] SSH vs. OpenVPN?
Hello In case a NAT firewall prevents using STUN to open SIP/RTP ports, a solution is to first connect the phone to the Asterisk server through a tunnel, and then have data go through the tunnel. Are there hardphones that support OpenVPN? If none, what about SSH? Is this a good alternative to use VoIP with SIP? If you've tried either or both solutions, I'm interested in any feedback. Thank you. -- _ -- 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] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 12:54:41 + (GMT), Arthur Stanfield a...@dmcip.com wrote: You can't tunnel UDP through SSH. Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper than the Snom alternatives. Thanks for the infos. So the only way to use SIP through locked-down NAT routers is to use OpenVPN, either with the few hardphones that support it or with a softphone on a computer. -- _ -- 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] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 07:57:22 -0500, bakko asannu...@gmail.com wrote: yeallink T26 and T28 support OpenVPN too Thanks for the infos. If someone tried the Snom, Grandstream, or Yeallink, how good is their OpenVPN connection? -- _ -- 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] SRV record for non-standard SIP port?
Hello To cut down on the number of hackers trying to break into an Asterisk server, I'd like to simply move the SIP port from the standard UDP 5060 to something non-standard. Since this server must be able to receive INVITEs from any SIP UA (server or client), it appears that I must add an SRV record in the DNS so that they can locate the server and the port used to reach it. _sip._udp SRV 0 5060 host.tld. www.voip-info.org/wiki/view/DNS+SRV Are there pitfalls/traps I must pay attention to before going ahead and add that type of record in the DNS? What about internal SIP clients that register with Asterisk: Will they query the DNS to find the SIP port also, or must reconfigure them all to use the non-standard port Asterisk listens on? Thank you. -- _ -- 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] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 10:03:46 -0600, Jeff LaCoursiere j...@sunfone.com wrote: Using Yealink T-28 with OpenVPN works fine - about three weeks now with no issues. Bummed that it seems to only support one tunnel, though. I asked their support team if they could make whatever changes necessary to support multiple, and their response made it sound promising :) Thanks for the feedback. Multiple tunnels are for conference calls? -- _ -- 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] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 10:44:12 -0600, Jeff LaCoursiere j...@sunfone.com wrote: No - the phone allows you to register with multiple servers, and I would like to reach each server over its own tunnel. It won't do that today. Thanks for the info. -- _ -- 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] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 16:25:37 -0600, Dale Noll dn...@wi.rr.com wrote: You can also setup OpenVPN to connect a remote subnet (remote office) and it will route all traffic between subnets. Configure the hard/soft phones on the remote subnet to route through the OpenVPN. This works pretty well for me. Thanks for the info. I was thinking of connecting while on the road/vacation, but it's a good use to connect a remote office to the main office. -- _ -- 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] SRV record for non-standard SIP port?
On Tue, 31 Jan 2012 18:22:41 +0100, Daniel Pocock dan...@readytechnology.co.uk wrote: Something more appropriate for your goal might be a move to TLS, it is definitely needed for any external connectivity [...] As a further safety measure, you could use something like repro or Kamailio as a SIP router to isolate your Asterisk from the public internet. Thanks for the tips. I'll read up on TLS and adding an SIP router in front of Asterisk. -- _ -- 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] allowguest = yes? no?
Hello I don't understand how I should use the allowguest item: If set to yes, callers from the Net should authenticate, but then, how can I allow strangers to call extensions in my system? allowguest If set to no, this disallows guest SIP connections. The default is to allow guest connections. SIP normally requires authentication, but you can accept calls from users who do not support authentication (i.e., do not have a secret field defined).Certain SIP appliances (such as the Cisco Call Manager v4.1) do not support authentication, so they will not be able to connect if you set allowguest=no: allowguest=no|yes (from Asterisk The future of Telephony) Thank you. -- _ -- 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] allowguest = yes? no?
On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito asterisk-users-mailing-l...@devito.cc wrote: What they are talking about is SIP URI dialling. Let say you have extension 1000 the rings a phone on your system. With allowguest=yes I would be allowed to dial SIP:/1...@yourdomain.com and assuming the context defined in your [General] section had access to exten 1000 I would connect to that phone. With alloweguest=no my call would be rejected. Thanks for the clarification. Provided I do want strangers to call extensions through an SIP URI instead of using the PSTN, how can I raise security by requiring that they authenticate? Of do you mean that the choice is between - don't allow SIP URI at all (allowguest=no), so strangers can reach extensions only through the PSTN (but it's a waste of money) - allow SIP URI (allowguess=yes) and make sure the context doesn't allow making calls to the PSTN? -- _ -- 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] allowguest = yes? no?
On Tue, 24 Jan 2012 09:26:26 -0600, Kevin P. Fleming kpflem...@digium.com wrote: By definition this is impossible. If the caller is a 'stranger', that means you have no knowledge of them prior to their INVITE request arriving at your server. If you have no knowledge of them, then you don't have any 'shared secret', and thus they cannot authenticate to your server. Mmm, so if I want to allow strangers to call us over the Net, I must 1. allowgues=yes 2. make sure the context they enter will not allow them to make calls through the PSTN, either directly (through our plug in the wall) or indirectly (through an ITSP). Thank you. -- _ -- 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] Couple of questions: SIP ALG, allowguest=no
Hello I just read this article about an Asterisk server that got hacked to make free international calls through an ITSP: www.rowetel.com/blog/?p=2210 I have a couple of questions: 1. Am I correct in understanding that SIP ALG on a router makes it easier to host an Asterisk server on a private LAN behind a NAT router (no need to map ports for RTP + outgoing packets can be sent directly to the remote SIP client instead of going through the Asterisk server to rewrite the RTP port numbers)? www.voip-info.org/wiki/view/Routers+SIP+ALG 2. If allowguest=no is commented out, it means that any SIP client on the Net can connect to the Asterisk server and make outgoing calls like legitimate SIP clients? Thank you. -- _ -- 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] best softphone for 2012?
On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy seandar...@gmail.com wrote: But what really made us choose linphone was you use it on android/iphone. That has been a huge plus. As a bonus, you can use any degegistered smartphone - that is, one not hooked up to the cellular network,only wireless - as a softphone. I guess you meant de-registered smartphone : what does it mean? -- _ -- 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] best softphone for 2012?
On Sat, 7 Jan 2012 12:34:44 -0500, Sean Darcy seandar...@gmail.com wrote: Yes, I did mean de-registered. I meant a phone that no longer has the ability to use the cellular network - only wifi. For instance, we have a couple of Droids that used to be on Verizon. They work just fine as sip-phones over wifi. Thanks. -- _ -- 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] Installing asterisk on a server vs appliance(e.g digium mypbx)
On Thu, 1 Dec 2011 14:09:29 +0300, James Mutuku listmut...@gmail.com wrote: I have worked with bare asterisk + freepbx before. the mypbx was just an example but my reference to appliances as a whole. The appliances seem to have lower entry costs. Appliances have less RAM + storage, so you'll have to make sure they're OK for what you're trying to do. Also, they usually use non-x86 chips, which means you're restricted to the OS + add-ons available for that platform. www.voip-info.org/wiki/view/Asterisk+Appliances www.astlinux.org www.smallnetbuilder.com/multimedia-voip/multimedia-voip-features/31208-how-to-build-asterisk-appliances-on-the-cheap -- _ -- 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] [1.6.2.9] Echo even when using headset?
On Mon, 19 Sep 2011 12:12:54 +0200, Gilles codecompl...@free.fr wrote: Problem solved: Tried XLite 4.1 on another test, and sound is OK, so I guess it's something in my work PC. s/test/host/ An el cheapo $20 CMedia CMI8738 6CH solved the issue. Great sound :-) -- _ -- 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] [1.6.2.9] Echo even when using headset?
On Sun, 18 Sep 2011 22:28:32 +0200, Gilles codecompl...@free.fr wrote: For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Problem solved: Tried XLite 4.1 on another test, and sound is OK, so I guess it's something in my work PC. -- _ -- 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] [1.6.2.9] Echo even when using headset?
Hello I just set up Asterisk 1.6.2.9 through packages on a test host running Ubuntu 11.04, configured sip.conf/extensions.conf, and launched EyeBeam 1.5.20 to run the echo test. For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Here's a recording: www.megaupload.com/?d=146L0HL6 FWIW, EyeBeam has both Use acoustic echo cancellation (AEC) and Use gain control (AGC) checked. Here are the two files: ;=== sip.conf [general] context=dummy port = 5060 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm nat=no qualify=yes host=dynamic [fred] type=friend context=internal secret=1234 qualify=yes host=dynamic ;=== extensions.conf [dummy] [internal] exten = 600,1,Playback(demo-echotest) exten = 600,n,Echo exten = 600,n,Playback(demo-echodone) exten = 600,n,Hangup() ;=== Are there settings I should(n') use in either Asterisk or EyeBeam to explain/solve this issue? Thank you. -- _ -- 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] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind govoi...@gmail.com wrote: The image you provided didn't open so I'm not sure about the design. Sorry about that. It's a PNG file and it opens in the two browsers I tried. The reason I don't simply get a subscription with a VoIP provider and must go through an Asterisk server + connection to the FXS port is that outgoing calls are free, which is nice when calling cellphones, especially when travelling. If you can send some SIP flow diagram and Asterisk CLI logs maybe it'll help understand the problem. I haven't done it yet, so have no logs to show. I'd simply like to hear what's going on channel #2 while Dahdi is still dialing, instead of simply being kept waiting. Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at home connected to their ADSL modem so that they can make free calls from overseas? Thank you. -- _ -- 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] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes jeroeneeu...@gmail.com wrote: I think this is a very common situation, so I'm not really sure what your problem is. Perhaps it's because I don't use an internal card, but in my situation it works just fine. I dial a number on my SIP phone, Asterisk goes through the dialplan, and puts the call out via the SPA3102. In my ear I hear ringing sounds, busy, wrong number or someone talking to me just like if I had connected a normal phone to the PSTN line. I haven't done this yet, and was looking for information. I was under the (apparently false) impression that Asterisk/Dahdi didn't connect the two legs until the callee had gone off-hook. So it looks like there's really no issue in connecting a remote SIP client with a PSTN number through an FXO port + ADSL VoIP modem to take advantage of free phone calls. Thanks everyone. -- _ -- 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] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 10:54:48 -0500, Kevin P. Fleming kpflem...@digium.com wrote: This is true, but you already answered your own question in your original post: since Asterisk cannot know whether the called party (dialing out via an FXO port) has answered or not, it assumes the outgoing call is 'answered' as soon as dialing has been completed. Because of this, the calling channel is bridged to the called channel as soon as dialing has been completed, and the calling party will hear the progress of the outbound call. Thanks for the confirmation. Too bad Dahdi doesn't provide call supervision so that Asterisk knows if/when the callee has answered. I'll experiment and see how it goes. -- _ -- 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] Monitoring second leg being dialed?
On Fri, 16 Sep 2011 19:35:19 -0400, Eric Wieling ewiel...@nyigc.com wrote: It does on PRI. Unfortunately, this is for an ADSL modem, hence the connection to its FXS port :-/ -- _ -- 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] Monitoring second leg being dialed?
Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the number I wish to dial, and have it dial out through the FXO card and the FXS port on the ADSL modem. Here's the diagram: http://img844.imageshack.us/img844/3308/asterisksippstncallback.png Problem is, Dahdi/Zaptel doesn't provide call progression, so that 1) when Asterisk passes the call to Dahdi/Zaptel, Asterisk considers the call answered although there's no actual phone connection yet, and 2) Dahdi/Zaptel doesn't trigger an event so we know if the call was answered (and if yes, by a live human being rather than an answering machine) or if the line is still ringing. A so-so solution is to simply tell Asterisk to loop through a voice message (This is a call from Joe Allen. Please hit any key and you will be connected), so we know that a human being has answered the call, but I was wondering if there were a better solution. Is it possible for Asterisk to somehow play on channel #1 what's happening on channel #2 while Dahdi/Zaptel is actually still dialing, so that I handle call progression manually from my cellphone and the callee doesn't end up hearing that odd recorded message? Thank you. -- _ -- 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] Asterisk on Android?
On Thu, 08 Sep 2011 14:52:06 -0400, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 08/09/11 02:19 PM, Cobra 2 wrote: I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've gotten asterisk to run on that just fine. I think the question is, can you answer your incoming calls with the Asterisk running on the device? Yes, that's the plan. I'd like Asterisk to run an IVR to screen incoming calls. Cobra: Out of curiosity, what did you use Asterisk for on that Motorola phone if not to handle incoming calls? -- _ -- 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] Asterisk on Android?
On Fri, 02 Sep 2011 16:37:32 +0200, Tamer Higazi th9...@googlemail.com wrote: Do you want to run the entire PBX on the Android client or are you just looking for a IAX programm to be installed for receiving calls?! The entire PBX so I can have an IVR in the phone. -- _ -- 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] Asterisk on Android?
Hello, Out of curiosity, has Asterisk been successfully compiled and ran Asterisk on an Android smartphone? I could use a small IVR on my smartphone to handle incoming calls. Thank you. -- _ -- 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] Asterisk on Android?
On Fri, 2 Sep 2011 13:23:18 +0100, A J Stiles asterisk_l...@earthshod.co.uk wrote: TTBOMK it's been done; but without the necessary Zaptel / DAHDI drivers to interface with the phone line, it's rather less useful than it sounds. I'm looking for a way to an IVR in my smartphone to handle incoming calls, and right it depending on such and such option. Anyone has more information in turning a smartphone (Android and/or iPhone) into a basic IP PBX? Thank you. -- _ -- 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] USB or Ethernet based FXO device ?
On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez cur...@telecomabmex.com wrote: Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports. Thanks for the tip. It looks like the smallest option is 8 FXO ports: www.xorcom.com/telephony-interfaces/astribank-models.html -- _ -- 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] USB or Ethernet based FXO device ?
On Sat, 27 Aug 2011 09:31:12 -0600, linux guy linuxguy...@gmail.com wrote: I'm looking for an FXO device to connect to a POTS line that communicates via USB or Ethernet. For USB, AFAIK, there's only the one from Sangoma. All others are Ethernet-based. www.voip-info.org/wiki/view/VoIP+Gateways -- _ -- 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] Why no traction for Windows version?
On Thu, 28 Jul 2011 12:04:38 +0500, Faisal Hanif fai...@vopium.com wrote: I have tried asterisk on windows XP using Cygwin and it worked fine. Would you mind explaining how to do this? Thank you. -- _ -- 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] Why no traction for Windows version?
On Thu, 28 Jul 2011 12:46:03 -0500, Danny Nicholas da...@debsinc.com wrote: Interrupting - you have to not use DAHDI (SIP Only) and make sure you have the necessary libs downloaded in your Cygwin install. It's OK, I don't mind using a VoIP gateway instead of a PCI card. Has someone written an HOWTO to compile 1.4 or 1.6 for Windows? Does it require patching to Asterisk and/or libraries, or does Cygwin handles the whole thing? -- _ -- 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] Why no traction for Windows version?
On Thu, 28 Jul 2011 13:08:33 -0500, Danny Nicholas da...@debsinc.com wrote: If they have, it would probably be on www.nerdvittles.com It looks like The Incredible PBX runs on CentOS www.nerdvittles.com/index.php?p=740 -- _ -- 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] Why no traction for Windows version?
Hello, Since Asterisk has been ported to exotic platforms like SOHO routers (Linksys, Buffalo, etc.) and non-MMU CPUs (Blackfin, etc.), I was wondering why the Windows port never really took off. As far as I can tell, www.asteriskwin32.com is a one-man effort (Patrick Deruel's) that is not going anywhere (latest version based on 1.2.26.2). Are there just not enough interest and too many, deep, Linux-specific assumptions in the code, that would explain why Asterisk was never officially ported to Windows? Thank you. -- _ -- 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] Why no traction for Windows version?
On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon) soeren.malc...@mcon.net wrote: And asterisk just runs fine on linux why bother ? Because I, for one, would like to run Asterisk on my Windows workstation at home as an enhanced answering machine :-) -- _ -- 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] Why no traction for Windows version?
On Tue, 26 Jul 2011 10:59:22 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Patches are welcomed. Does someone know the kind of changes that were made by AsteriskWin32, and how hard it'd be to apply them to more recent releases of Asterisk? -- _ -- 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] Why no traction for Windows version?
On Tue, 26 Jul 2011 12:07:10 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: There were some later fixes at around 1.6.0 to try to get the code built on cygwin. I would suggest you to try building it on cygwin and see where things fail. Also grep for CYGWIN or such in the source (especially in Makefile-s). Thanks for the infos. -- _ -- 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] [1.4] Minimal installation?
On Mon, 18 Jul 2011 20:59:02 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: /usr/lib/asterisk/modules/ Be sure to only include the ones you need. Finding which exactly may be tricky. Thanks Tzafrir. Actually, since the modules are the biggest files by far, besides the obvious (SIP, Dahdi, etc.), how to investigate which modules I must keep? Does Asterisk report errors explicitely when a module it needs is missing, or does it just crash/malfunction without reporting anything? -- _ -- 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] [1.4] Minimal installation?
On Tue, 19 Jul 2011 09:27:41 -0500, Danny Nicholas da...@debsinc.com wrote: My .02 - FWIW, DAHDI will use almost as much space as the rest of Asterisk, so you could save the space you don't have by forgoing that. Thanks everyone for the feedback. I'll go through the list of modules and see what I can remove, and then do the same for Dahdi. -- _ -- 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] Compact, affordable x86 devices?
Hello I'd like to build a compact, affordable, fanless x86 solution to handle my home landline. I know about the following two platforms: 1. www.pcengines.ch/alix.htm alix1d + case 100 Does Availability 500 mean that it's just not possible to buy just one item? 2. www.soekris.com/products.html?limit=all net4501-30 Board and Case $175.00 Is the net4501 powerful enough to run Asterisk, considering that I'll use an external VoIP gateway to connect it to my landline? Are there other manufacturers I should know about? Thank you. -- _ -- 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] Compact, affordable x86 devices?
On Mon, 18 Jul 2011 08:04:31 -0400, John Novack jnov...@stromberg-carlson.org wrote: Just about any of the HP thin clients, either new or used off eBay, with AstLinux installed do a wonderful job, especially if you are not going to need a PCI card. The older units will need a larger flash. Transcend has several different sizes that are direct replacements Looks like some of the Neoware units will also do the job. Thanks for the tip. I'd like to buy the unit new: Are those devices still manufactured? How easy is it to reflash them to run as a stand-alone Linux host? Which device would you recommend to Asterisk and a couple of other apps (small web server, SQLite, etc.)? Thank you. -- _ -- 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] Compact, affordable x86 devices?
On Mon, 18 Jul 2011 09:03:52 -0400, John Novack jnov...@stromberg-carlson.org wrote: there are other low cost solutions around as well. the ALIX boards I have seen do not impress me. I think they are somewhat overpriced. Jut one opinion Thanks for the feedback. I'll read what HP has to offer. When you mention other low-cost solutions, I assume you mean other thin clients reflashed to run as stand-alone hosts? -- _ -- 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] [1.4] Minimal installation?
Hello, I'd like to run Asterisk on an embedded device, where space is scarce. It should be able to handle calls from a VoIP provider in SIP, calls from the PSTN through Dahdi, and voicemail. If someone's already done this, I'd like to know which directories/files are required for a basic install? Does this look right? = /bin/asterisk /etc/asterisk/ asterisk.conf logger.conf modules.conf sip.conf extensions.conf voicemail.conf /etc/init.d/asterisk /usr/lib/asterisk/modules/ /var/lib/asterisk/agi-bin/moh - /var/lib/asterisk/sounds/moh /var/lib/asterisk/sounds/ /var/lib/asterisk/agi-bin/static-http/ /var/spool/asterisk/ = Thank you. -- _ -- 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] Monitoring connection to VoIP provider?
On Tue, 12 Jul 2011 11:10:28 -0400, Steven Stromer fil...@stevenstromer.com wrote: A quick to implement open source network monitoring tool is smokeping: http://oss.oetiker.ch/smokeping/index.en.html Thanks guys for the tip on qualify=yes and SmokePing. -- _ -- 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] Monitoring connection to VoIP provider?
On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com wrote: Community can help you better if you provide some details about you scenario and requirement. It's a very simple scenario: The Asterisk server is connected to a VoIP provider for calls to the PSTN, and I'd like to have Asterisk (or some other app) monitor the connection so that I can tell how good it is at any time, especially before calling out or receiving a call. The VoIP provides doesn't support any tool, eg. iperf. Is tracert/ping the only tools available in that scenario? Thank you. -- _ -- 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] Monitoring connection to VoIP provider?
Hello I was wondering if Asterisk can be configured to monitor a connection to a VoIP provider, whether someone is currently using it for a call or the connection is idle? FWIW, my VoIP provider doesn't run an iperf server on their side. I don't know if ping/traceroute is a good enough solution to monitor an SIP connection. I'd like this so I can check how good the line is before calling or receiving a call. Thank you. -- _ -- 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] Connect intercom to Asterisk?
Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to have something similar by connecting the intercom end in my appartment to a PC running Asterisk that will dial a phone number through SIP. Does someone know if something like that is available ready to use (Arduino, etc.)? Thank you. -- _ -- 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] Connect intercom to Asterisk?
On Tue, 7 Jun 2011 13:06:23 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Why bother when you can buy off the shelf stuff to do it for you. The trick is that this connector must work with existing interphones, such as this one at home: http://img220.imageshack.us/img220/8334/intercomhome.jpg So after I add a second pair of wires to the existing intercom, I guess the options are - either an ATA which will connect to the two-wire analog signal and turn it into an SIP end-point, or - simply running a phone cable from the intercom all the way to the Asterisk box where it will be connected to some hardware -- _ -- 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] About X100P and TDM400P analog card in China
On Wed, 11 May 2011 01:09:16 +0800, Scott Zhang macromars...@gmail.com wrote: So does this mean no solution when used ZAP/DAHDI with PSTN line? If I installed an E1, will that work? Before getting an E1, maybe ISDN provides call supervision? -- _ -- 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] Orginate not working well with PSTN lines
On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com wrote: I know this thread is dead but: I do not believe this should go into the DAHDI kernel modules. I agree. It's just too bad Dahdi is unable to report how an outgoing call is doing: Still ringing, busy, answered. -- _ -- 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] Orginate not working well with PSTN lines
On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali beaasteriskg...@gmail.com wrote: Anybody can explain me why asterisk is unable to detect ringback tone from PSTN telco ? . I guess it was a lot of work, and nobody bothered adding this to the Zaptel driver. -- _ -- 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] Orginate not working well with PSTN lines
On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali beaasteriskg...@gmail.com wrote: The problem here is that as soon as asterisk dialing on fxo lines it sets channel status as answered although the chennel is getting ring back tone from other party. Anyone can suggest me to solve this issue ? The only solution I know is to have Asterisk play a message in a loop for eg. 1mn, prompting the callee to hit a key to let the server know that the call was 1) answered 2) by a human being. -- _ -- 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] Iptables configuration to handle brute force registrations?
On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Have a look at these: Thanks much Gordon. I'll study the scripts you mentionned. It looks like iptables is good enough and I won't have to install a second tool to watch the logs and reconfigure iptables on the fly. -- _ -- 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] Iptables configuration to handle brute, force registrations?
On Tue, 5 Apr 2011 17:38:15 -0400, Paul Dugas p...@dugasenterprises.com wrote: First, this appears to be working for me though I'm not 100% sure of that and cannot guarantee it will for you in any way, shape or form. With the lawyering out of the way... Thanks a lot, Paul. -- _ -- 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] Iptables configuration to handle brute force registrations?
Hello I'm no expert of iptables, and it seems like it can handle banning IP's that are trying to register and fail too many times. I'd like to use this feature instead of having to install a second tool such as SSHGuard or BFS that parses the logs and reconfigure iptables on the fly. Is there a good iptables configuration that I could use as reference? FWIW, the kernel is uClinux 2.6.13.9, iptables is 1.3.6, ans it's a single-homed host so there's no need to handle the FORWARD chain. Thank you. -- _ -- 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] asterisk and fail2ban
On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias indre...@gmail.com wrote: Just to provide an alternative to sshguard: you could use BFD[1] Thanks Ioan. I'll give it a shot. -- _ -- 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] asterisk and fail2ban
On Wed, 30 Mar 2011 16:54:51 -0500, Darrick Hartman dhart...@djhsolutions.com wrote: One of our developers on the AstLinux team worked out a plugin for Arno's firewall (iptables based) which performs similar to fail2ban, but uses bash. He called it adaptive-ban. You might be able to adapt it for your use, but as it's written, it's integrated with AstLinux. Thanks Darrick. I'll add it to the list of options to check out. -- _ -- 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] Checking status of a cell phone
On Tue, 29 Mar 2011 07:48:08 +0200, magnu...@inputinterior.se wrote: I was a little unclear, it is not the cell phone that does the call-back, it is the cell-phone-network. Makes more sense :-) Thank you. -- _ -- 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] asterisk and fail2ban
On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? Sorry for hi-jacking the thread, but I was wondering if there were a lighter alternative that I could run on appliances? Python uses too much RAM, but I need to find a way to ban hackers from trying to connect to Asterisk from the Net. Thank you. -- _ -- 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] asterisk and fail2ban
On Tue, 29 Mar 2011 07:31:18 -0500 (CDT), Joe Greco jgr...@ns.sol.net wrote: sshguard is *extremely* lightweight compared to most things; it's a very efficient compiled C application that doesn't have (m?)any dependencies. Thanks much for the tip. I'll study how to install/configure iptable and sshguard. -- _ -- 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] asterisk and fail2ban
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. -- _ -- 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] asterisk and fail2ban
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. I agree. Is there a list I could use to check which blocks have been allocated to which countries so I can add them to Asterisk's blacklist? -- _ -- 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] asterisk and fail2ban
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote: On 03-29-2011 19:25, Steve Edwards wrote: Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? after reviewing last week's log i'd say around 25-28k/min :) So it looks like I should check out sshguard instead of relying on blocks of IP's :-) -- _ -- 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] Checking status of a cell phone
On Sat, 26 Mar 2011 14:58:30 +0100, magnu...@inputinterior.se wrote: Celluar Network - E1 - Avaya - OOH323 - Asterisk Thanks for the tip. So here's how it works: 1. The web app calls a script that uses AMI + Originate to send a call to the Avaya PBX 2. Avaya is able to check that a number (cellphone in this case) is busy and calls a different number in Asterisk to indicate the status through a value in the DB 3. The web script reads the value of DS/0733025975 and displays the status -- _ -- 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] Checking status of a cell phone
On Mon, 28 Mar 2011 14:12:09 +0200, magnu...@inputinterior.se wrote: Its not the Avaya that makes the call back, it is mobile. I thought the way you handled things, is that Asterisk would call your cellphone through the Avaya PBX just to check whether the cellphone is in_use/busy. At what point does the cellphone call Avaya or Asterisk back? -- _ -- 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] Checking status of a cell phone
On Sat, 26 Mar 2011 10:50:19 +0100, magnu...@inputinterior.se wrote: I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. I'd like to check I understood: Your Asterisk server is connected to a landline and can call your cellephone (073-302 59 75). When a call comes in from the landline, Asterisk checks whether your cellphone is available and redirects the call; If not available, it calls a landline number (010-602 4975). If this landline number is not available, it tries a third number (010-602 4976)? Is the AMI code below enough to check if the cellphone is available/in-use? Action: Originate Channel: OOH323/00733025975@Avaya\r\nExten: 0106024000 Context: inputinterior.se Priority: 1 Timeout: 1000 CallerID: 106024000 DBPut Family: DS Key: 0733025975 Val: NOT_INUSE -- _ -- 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] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 16:48:28 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
Somehow, I'm guessing that 'failed' means that something failed while
processing the call file or that the call failed to answer, not that
somebody terminated the call.
Thanks guys. After testing with a PCI card + Dahdi, and then with a
Linksys 3102, turns out that neither jumps to the failed or h
extension when the remote number is busy, ie. already engaged (with no
support for callwaiting, ie. two-way calling)
== extensions.conf
[internal]
;call from XLite
;exten = _5.,1,Dial(Dahdi/1/${EXTEN})
exten = _5.,1,Dial(SIP/3102-fxo/${EXTEN})
exten = h,1,NoOp(Called ended with ${DIALSTATUS})
exten = failed,1,NoOp(Call ended with ${REASON})
== CLI
== Using SIP RTP CoS mark 5
-- Executing [5551234@internal:1] Dial(SIP/xlite-000e,
SIP/3102-fxo/5551234) in new stack
== Using SIP RTP CoS mark 5
-- Called 3102-fxo/5551234
#Here, phone is still ringing, but Asterisk wrongly says it has
answered
-- SIP/3102-fxo-000f is ringing
-- SIP/3102-fxo-000f answered SIP/xlite-000e
#Says it bridged calls although remote end hasn't answered
-- Packet2Packet bridging SIP/xlite-000e and SIP/3102-fxo-000f
==
As I no longer have a real landline, it could be due to the way my
ADSL VoIP landline works. Bottom line: I can't use that line to write
a robocall.
Thanks guys.
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Re: [asterisk-users] [1.4] Asterisk doesn't hang up?
On Tue, 15 Mar 2011 14:54:53 +0100, Gilles codecompl...@free.fr wrote: I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: For those trying to do the same thing: Zaptel/Dahdi does detect that the remote party has hung up when using busydetect=yes in zapata.conf/chan_dahdi.conf. -- _ -- 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] [1.4] Failed callfile doesn't jump to failedextension
On Tue, 15 Mar 2011 11:44:20 -0500, Danny Nicholas
da...@debsinc.com wrote:
Don't depend on the tutorials you read to be 100% accurate or up-to-date.
The default action on a failure in Asterisk is usually going to be an s
jump, either to s,1 or s+100. Personally, I would replace failed,1 with
start-NOANSWER,1.
Thanks for the info. After calling out through a call file, Asterisk
plays the MOH and detects that the callee has hung up, but either
doesn't jump to the extension or does jump to h but ${REASON} is
empty:
===
[callback]
;how to wait until callee has answered?
exten = start,1,Wait(2)
exten = start,n,NoOp(${DEVICE_STATE(Dahdi/1)})
exten = start,n,Answer()
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended with ${REASON})
;empty
;exten = h,1,NoOp(Call ended with ${REASON})
;not run
exten = start-NOANSWER,1,NoOp(Call ended with ${REASON})
===
Is this what you had in mind?
Thank you.
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Re: [asterisk-users] [1.4] Failed callfile doesn't jump tofailedextension
On Fri, 18 Mar 2011 10:14:37 -0500, Danny Nicholas
da...@debsinc.com wrote:
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended with ${REASON})
;empty
;exten = h,1,NoOp(Call ended with ${REASON})
;not run
exten = start-NOANSWER,1,NoOp(Call ended with ${REASON})
===
Is this what you had in mind?
Thank you.
That's the ticket.
Unfortunately, it can only jump to h, and ${REASON} is empty.
Based on...
www.voip-info.org/wiki/view/Asterisk+dial+plan+-+working+example
... I also tried this, but Asterisk doesn't jump to any of those
extensions:
= extensions.conf
...
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
;exten = start,n,Goto(${EXTEN}-${REASON})
exten = start,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy ;Only works with SIP calls
exten = s-CHANUNAVAIL,1,Verbose(Not available)
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
= CLI
-- Executing [start@callback:5] Playback(DAHDI/1-1,
manolo_camp-morning_coffee) in new stack
-- DAHDI/1-1 Playing 'manolo_camp-morning_coffee.ulaw' (language
'fr')
== Spawn extension (callback, start, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
[Mar 18 16:41:35] NOTICE[1200]: pbx_spool.c:349 attempt_thread: Call
completed to Dahdi/1/5551234
=
Is there no way to know how a call ended?
Thank you.
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Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
not working as you expect it to, maybe you could read through the source
(/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why.
You could always submit a patch...
HANGUP_CAUSE should be HANGUPCAUSE.
Thanks guys. In which case does Asterisk jump to the failed
extension?
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Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 17:56:12 -0500, Anthony Messina
amess...@messinet.com wrote:
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
Thanks but for some reason, after calling out through a call file,
Asterisk doesn't jump to it although the callee hangs up while
Asterisk is still playing:
===
[callback]
exten = start,1,Wait(2)
exten = start,n,ChanIsAvail(Dahdi/1)
exten = start,n,NoOp(${AVAILORIGCHAN})})
exten = start,n,Answer()
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
;not run
exten = failed,1,NoOp(Call ended with ${REASON})
===
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[asterisk-users] [1.6/Ubuntu] What packages for * + Dahdi?
Hello I'd like to install Asterisk and Dahdi on a Ubuntu host using packages instead of compiling from the source. Are the following packages enough for this? == asterisk - Open Source Private Branch Exchange (PBX) asterisk-config - Configuration files for Asterisk dahdi - utilities for using the DAHDI kernel modules dahdi-linux - DAHDI telephony interface - Linux userspace parts asterisk-sounds-main - Core Sound files for Asterisk (English) asterisk-sounds-extra - Additional sound files for the Asterisk PBX == BTW, I notice dahdi-dkms: Does it mean that when I upgrade the kernel, I'll also need to upgrade Dahdi? Thank you. -- _ -- 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] [1.6/Ubuntu] What packages for * + Dahdi?
On Thu, 17 Mar 2011 13:01:39 +0200, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: BTW, I notice dahdi-dkms: Does it mean that when I upgrade the kernel, I'll also need to upgrade Dahdi? Yes, basically. Good to know. Thanks for the tip. -- _ -- 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] [1.6.2.5] Asterisk can't find MOH file
Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here: http://downloads.asterisk.org/pub/telephony/sounds/ find /var/lib/asterisk/sounds/ -name manolo_camp-mor* /var/lib/asterisk/sounds/fr/manolo_camp-morning_coffee.ulaw extensions.conf ;just in case, but shouldn't be needed exten = ,1,Set(CHANNEL(language)=fr) exten = ,n,Wait(2) exten = ,n,Answer() exten = ,n,Playback(manolo_camp-morning_coffee) exten = ,n,Hangup cat asterisk.conf ... [options] nocolor = yes ; Disable console colors languageprefix = yes ; Use the new sound prefix path syntax [compat] pbx_realtime=1.6 res_agi=1.6 app_set=1.6 cat /etc/asterisk/chan_dahdi.conf [channels] language=fr signalling = fxs_ks ... cat /etc/asterisk/sip.conf [general] language=fr port = 5060 ... CLI -- Executing [@internal:1] Set(SIP/xlite-0002, CHANNEL(language)=fr) in new stack -- Executing [@internal:2] Wait(SIP/xlite-0002, 2) in new stack -- Executing [@internal:3] Answer(SIP/xlite-0002, ) in new stack -- Executing [@internal:4] Playback(SIP/xlite-0002, manolo_camp-morning_coffee) in new stack [Mar 17 15:10:18] WARNING[1888]: file.c:650 ast_openstream_full: File manolo_camp-morning_coffee does not exist in any format [Mar 17 15:10:18] WARNING[1888]: file.c:953 ast_streamfile: Unable to open manolo_camp-morning_coffee (format 0x4 (ulaw)): No such file or directory [Mar 17 15:10:18] WARNING[1888]: app_playback.c:471 playback_exec: ast_streamfile failed on SIP/xlite-0002 for manolo_camp-morning_coffee -- Executing [@internal:5] Hangup(SIP/xlite-0002, ) in new stack == Spawn extension (internal, , 5) exited non-zero on 'SIP/xlite-0002' -- Executing [h@internal:1] NoOp(SIP/xlite-0002, Called ended with ) in new stack Does someone what I missed that would explain the error above? Thank you. -- _ -- 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] [1.6.2.5] Asterisk can't find MOH file
On Thu, 17 Mar 2011 09:23:35 -0500, Danny Nicholas da...@debsinc.com wrote: Moh should be in /var/lib/asterisk/moh not /var/lib/asterisk/sounds or in this case /var/lib/asterisk/moh/custom. Thanks for the tip, but after moving the MOH files to the right location, and even restarting Asterisk, it still doesn't find them, with the same error message: = # ll /var/lib/asterisk/moh/ -rw-r--r-- 1 root root 1954191 2009-12-26 15:57 macroform-cold_day.ulaw -rw-r--r-- 1 root root 1509854 2009-12-26 15:57 macroform-robot_dity.ulaw -rw-r--r-- 1 root root 2232088 2009-12-26 15:57 macroform-the_simplicity.ulaw -rw-r--r-- 1 root root 584771 2009-12-26 15:57 manolo_camp-morning_coffee.ulaw -rw-r--r-- 1 root root 2573886 2009-12-26 15:57 reno_project-system.ulaw = I tried using MusicOnHold() but it doesn't take a parameter, and just plays some other tune: = ;exten = ,n,Playback(manolo_camp-morning_coffee) exten = ,n,MusicOnHold(manolo_camp-morning_coffee) = Actually, how can Asterisk know that a file is MOH and hence, should be found in /var/lib/asterisk/moh/, rather than a regular prompt/sound file located in /var/lib/asterisk/sounds? Thank you. -- _ -- 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] [1.6.2.5] Asterisk can't find MOH file
On Thu, 17 Mar 2011 15:09:18 +, Ishfaq Malik i...@pack-net.co.uk wrote: MusicOnHold() doesn't take a file name as a parameter, it takes a class name or if left blank, plays from the default class Yes, thanks for the tip. Found it: Turns out the Ubuntu package expects sound files to be located in /usr/share/asterisk/sounds instead of the usual /var/lib/asterisk/sounds. asterisk.conf: astdatadir = /usr/share/asterisk Thanks guys. -- _ -- 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] [1.4] Asterisk doesn't hang up?
On Wed, 16 Mar 2011 22:45:35 +1100, John Kosmas batc...@optusnet.com.au wrote: i have the same problem but it doesnt always happen tho from the same caller. im using Asterisk 1.4 - maybe newer version updates have had bug fixes. maybe this could rectify it. Thanks John, but I still get the problem with 1.6. Looks like the VoIP plug on my ADSL modem doesn't provide either polarity reversal or open loop, so Zaptel/Dahdi can't dectect answer/detect. Could be on purpose, to prevent people from hooking up an IP PBX and use this option in ways the ISP wants to prevent ;-) -- _ -- 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] [1.6] Where to put options wctdm opermode?
Hello The Ubuntu Asterisk package doesn't install /etc/modprobe.d/dahdi.conf, so I was wondering where to put the following line: === options wctdm opermode=FRANCE === Should it be in /etc/dahdi/modules? === options wctdm opermode=FRANCE === Thank you. -- _ -- 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] [1.6] Where to put options wctdm opermode?
On Thu, 17 Mar 2011 10:48:07 -0500, Danny Nicholas da...@debsinc.com wrote: You should manually create /etc/modprobe.d/dahdi.conf since /etc/init.d/dahdi start is going to do a modprobe and that's the only way you're going to get this option started correctly (subject to correction). Thanks for the info. -- _ -- 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] [1.4] Asterisk doesn't hang up?
On Tue, 15 Mar 2011 13:45:00 -0400, Paul Belanger pabelan...@digium.com wrote: Is this an analog line? If so, is your CO providing a disconnect tone? Yes, it's an analog line, but it's actually VoIP provided by an RJ11 on an ADSL modem, not a real landline. Is there a way to check how the ADLS/telco provides disconnection, ie. whether it's through polarity reversal, open loop, or by just playing call progress tones? Thank you. -- _ -- 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] [1.4] Asterisk doesn't hang up?
Hello
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
== extensions.conf
;Play MoH for a few seconds, hang up, and
;check ChanIsAvail() able to detect when line idle again
exten = ,1,Answer()
exten = ,n,Playback(/var/tmp/manolo_camp-morning_coffee)
exten = ,n,Hangup
== CLI
;keep line engaged for a few seconds, and hang up from remote end
originate Zap/1/5551234 extension @internal
== extensions.conf
;call from XLite to check line status
;Loop until Zap/1 is available
exten = ,1,Set(INDEX=0)
exten = ,n,While(1)
exten = ,n,ChanIsAvail(Zap/1)
exten = ,n,GotoIf($[${AVAILORIGCHAN} != | ${INDEX}
10]?exit)
exten = ,n,Wait(5)
exten = ,n,Set(INDEX=$[${INDEX} + 1])
exten = ,n,EndWhile()
;how did we exit loop?
exten = ,n(exit),GotoIf($[${AVAILORIGCHAN} = ]?na:ok)
exten = ,n(na),NoOp(Channel still N.A.)
exten = ,n,Goto(end)
exten = ,n(ok),NoOp(Channel OK)
exten = ,n(end),Hangup
==
Even after callee at 5551234 hangs up, Asterisk keeps looping in
extension , and only runs 's Hangup after runs Hangup.
I also tried calling out through a callfile, same result.
Is there another instruction I should use in to have
Asterisk/Zaptel close the channel after the remote end has hung up?
Thank you.
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Re: [asterisk-users] [1.4] Asterisk doesn't hang up?
On Tue, 15 Mar 2011 14:54:53 +0100, Gilles codecompl...@free.fr wrote: I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: It looks like neither Playback() nor Background() check for hangup and will simply play the file all the way to the end, so I simply replaced the long MoH with a short beep: == exten = ,1,Wait(2) exten = ,n,Answer() ;exten = ,n,Playback(/var/tmp/manolo_camp-morning_coffee) ;exten = ,n,Read(key,/var/tmp/manolo_camp-morning_coffee,1,,10,2) ;exten = ,n,Background(/var/tmp/manolo_camp-morning_coffee) exten = ,n,Background(beep) exten = ,n,WaitExten(10) exten = ,n,Hangup == HTH, -- _ -- 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] [1.4] Failed callfile doesn't jump to failed extension
Hello
For some reason, when dialing out through a call file and the remote
line is busy, Asterisk doesn't jump to the failed extension in the
context used by the call file:
== call file
Channel: Zap/1/5551234
Context: callbacktest
Extension: start
Priority: 1
MaxRetries: 1
== extension.conf
[callbacktest]
exten = start,1,NoOp(Status is ${DIALSTATUS})
exten = start,n,Wait(10)
exten = start,n,Hangup
exten = failed,1,NoOp(Reason call file failed is ${REASON})
== CLI
ip04*CLI
-- Attempting call on Zap/1/5551234 for start@callbacktest:1 (Retry 1)
Channel Zap/1-1 was answered.
== Starting Zap/1-1 at callbacktest,start,1 failed so falling back to
exten 's'
== Starting Zap/1-1 at callbacktest,s,1 still failed so falling back
to context 'default'
-- Hungup 'Zap/1-1'
[Mar 15 16:22:11] NOTICE[368]: pbx_spool.c:351 attempt_thread: Call
completed to Zap/1/5551234
==
I followed this tutorial, and don't understand why Asterisk tries to
jump to extension s:
www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Thank you.
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Re: [asterisk-users] [1.4] Reading phone number the French way?
On Tue, 08 Mar 2011 13:22:18 +0100, Gilles codecompl...@free.fr
wrote:
I need to write a script which prompts the callee to type a number,
and then read it back to them as confirmation:
Apparently, the right way to read a phone number back to the user is
not to use SayNumber() (which might be OK for US-style reading) but
rather Playback(prefix:number,say), which will then rely on
say.conf
For instance:
=== extensions.conf
exten = ,1,Set(NBR2CALL=0142928100) ;exten =
,n,SayNumber(${NBR2CALL}) exten =
,n,Playback(phone:${NBR2CALL},say)
===
Using this almost works:
=== say.conf
_pho[n]e:0[1-9] = num:${SAY:0:1}, num:${SAY:1:1},
num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
===
The remaining problem is when a couple starts with a zero, eg. 01
(should be read zero one): In this case, Asterisk ignores the
leading zero and simply pronounces the second digit (one)
Does someone know of a trick so that the pattern handles couples that
have a leading zero?
Thank you.
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Re: [asterisk-users] [1.4] Reading phone number the French way?
On Thu, 10 Mar 2011 13:18:41 +0100, Dave Cotton
dcot...@linuxautrement.com wrote:
Look at the GotoIf statement for example
Thanks Dave for the tip, but I found that I needed to change a pattern
that was already in say.conf:
===
[fr](date-base,digit-base) ;BAD _[n]um:0. = num:${SAY:1}
_[n]um:0X = num:${SAY:0:1}, num:${SAY:1:1}
...
;regular phone numbers : landlines and cellphones
;_pho[n]e:0[1-9] = num:${SAY:0:1}, num:${SAY:1:1},
num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
===
If I got it right, the way say.conf works, is that it reads the whole
say.conf to make a list of the different patterns. Then, when reading
a prefix+number, it reads the patterns on the right side and tries to
find if it furthers matches another pattern.
In the example above, _[n]um:0X will match num:${SAY:6:2}, which
will read the two digits as expected, ie. without ignoring a leading
zero.
Thank you.
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Re: [asterisk-users] [1.4] Reading phone number the French way?
On Thu, 10 Mar 2011 15:30:51 +0200, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: I think you're missing SayDigits(). say.conf does use the syntax of the extensions.conf, but it's not a dialplan. Thanks for the input, but SayDigit() isn't right for what I want to do, since it simply reads a phone number digit-by-digit, which is not the way phone numbers are read in France. SayNumber() doesn't work either in this particular case, but is OK for countries where phone numbers are read digit-by-digit. I figured out how extensions.conf and say.conf work and posted my results in the reply to Dave. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Reading phone number the French way?
On Thu, 10 Mar 2011 14:37:45 +0100, Gilles codecompl...@free.fr
wrote:
I figured out how extensions.conf and say.conf work and posted my
results in the reply to Dave.
Noticed something strange, though: 0800123456 is played OK (ie.
0.800.12.34.56) , but 092123456 is played digit by digit (0.8.9.2,
etc.):
== say.conf
;1-9
_[n]um:X = digits/${SAY}
;10-99
_[n]um:1X = digits/${SAY}
_[n]um:[2-9]0 = digits/${SAY}
_[n]um:[2-6]1 = digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
_[n]um:71 = digits/60, vm-and, num:1${SAY:1}
_[n]um:7X = digits/60, num:1${SAY:1}
_[n]um:9X = digits/80, num:1${SAY:1}
_[n]um:[2-9][1-9] = digits/${SAY:0:1}0, num:${SAY:1}
;100-999
_[n]um:100 = digits/hundred
_[n]um:1XX = digits/hundred, num:${SAY:1}
_[n]um:[2-9]00 = num:${SAY:0:1}, digits/hundred
_[n]um:[2-9]XX = num:${SAY:0:1}, digits/hundred, num:${SAY:1}
;0800XX - 0899XX
;_pho[n]e:08 = num:${SAY:0:1}, num:${SAY:1:3},
num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
== CLI
-- Executing [@internal:4] Playback(SIP/xlite-02a56004,
phone:0810009032|say) in new stack
-- SIP/xlite-02a56004 Playing 'digits/0' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/8' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/hundred' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/10' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/0' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/0' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/90' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/30' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/2' (language 'fr')
-- Executing [@internal:6] Playback(SIP/xlite-02a56004,
phone:0892123456}|say) in new stack
-- SIP/xlite-02a56004 Playing 'digits/0' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/8' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/9' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/2' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/1' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/2' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/3' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/4' (language 'fr')
-- SIP/xlite-02a56004 Playing 'digits/5' (language 'fr')
-- Executing [@internal:7] Hangup(SIP/xlite-02a56004, ) in new
stack
==
Can't figure out why it doesn't use the same pattern to play 0800 and
092 numbers. Any idea?
Thank you.
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