Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
Hi Steve, On 15-01-14 02:44, Steve Edwards wrote: On Tue, 14 Jan 2014, Patrick Lists wrote: ...I guess I'll cook up some dialplan logic that records IP addresses, keeps track of the amount of failed password attempts etc. and block the offending IP addresses... A few iptables rules can protect you from access from China, North Korea, Iran, Iraq, xxxistan, Russia, Nigeria, and any other country you're not expecting calls from. Eliminate 90% of the problem at the front door and you can focus more clearly on the remaining 10%. Yes that's one of the tricks in my bag. Unfortunately it seems that the IP ranges from ip-deny.com are no longer available and even their website has disappeared. Would you mind sharing where you get the per country IP ranges from? Regards, Patrick -- _ -- 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] patlooptest output - errors
Ok, I'm a believer now: shell> chrt -f 99 patlooptest /dev/dahdi/1 -t 300 -v Using Timeout of 300 Seconds Going for it... Timeout achieved Ending Program Test ran 28295 loops of 2039 bytes/loop with 0 errors Still, any hint about this? shell> lsof /dev/dahdi/2 shell> shell> chrt -f 99 patlooptest /dev/dahdi/2 -t 300 -v /dev/dahdi/2: Device or resource busy Thanks. On Wed, Jan 15, 2014 at 12:48 AM, Shaun Ruffell wrote: > On Tue, Jan 14, 2014 at 05:58:57PM +, Rodrigo Borges Pereira wrote: > > > > Running patlooptest following instructions on Digium KB, using dahdi > 2.5.1 > > with a TE205P. I'm getting this kind of output (not full output). Note, > I'm > > using a standard cat5 ethernet cable (30cm), with a Digium T10i > crossover. > > [snip] > > > Event: 29 > > Event: 30 > > [snip] > > > Cause for concern? > > > > I'm testing a card which has been used before without issues. I'm just > > playing with patlooptest for the first time, and wasn't expecting to see > > this on a supposedly good card. > > > > appreciate any input, thx. > > Those events (29 and 30) indicate that patlooptest isn't able to > keep up with the hardware. > > Do you get the same thing when you run patloop test with elevated > permissions? > > $ chrt -f 99 patlooptest > > Cheers, > Shaun > > -- > Shaun Ruffell > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
On Tue, 14 Jan 2014, Patrick Lists wrote: ...I guess I'll cook up some dialplan logic that records IP addresses, keeps track of the amount of failed password attempts etc. and block the offending IP addresses... A few iptables rules can protect you from access from China, North Korea, Iran, Iraq, xxxistan, Russia, Nigeria, and any other country you're not expecting calls from. Eliminate 90% of the problem at the front door and you can focus more clearly on the remaining 10%. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] help with Cepstral 6 and Asterisk 11
On 1/13/2014 6:13 PM, Justin Killen wrote: Another option is to use an MRCP server like UniMRCP along with the Cepstral plugin. One very nice thing about this approach is that there is less 'cepstral version' <-> 'asterisk version' dependency, which is a problem with the current app_swift module (each app_swift version is designed to work on specifically one version of asterisk and one version of cepstral). Cepstral provides details here: http://www.cepstral.com/en/telephony/mrcp Information on the open-source uniMRCP can be found here: http://www.unimrcp.org/ Information for connecting asterisk to uniMRCP can be found here (although it seems to be having issues ATM): http://code.google.com/p/unimrcp/wiki/asteriskUniMRCP -Justin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg Sent: Friday, January 10, 2014 1:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11 Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take more than a couple of minutes to compile, install, and configure the package. Maybe some things need to get adjusted as the author has abandoned the project (at least there is no longer a project web page) and the latest sources are about 2 years old. Building from sources is not that difficult, but if you don't have a proper configure script you are responsible that all prerequisites are met, which can be time consuming if you don't know your distro well enough. Here, there is no configure script and some things, which might be invalid for your machine, are hand coded inside the Makefile. Nothing spectacular, but you could end up asking a lot more questions that have nothing to do with asterisk. jg Justin, Thank you very much for the information, that is great to learn that I have that option as well. Brandon -- _ -- 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] help with Cepstral 6 and Asterisk 11
On 1/9/2014 12:12 PM, Jeremy Kister wrote: On 1/8/2014 9:12 PM, Brandon Coale wrote: However, I am not able to get app_swift to compile. I am running Asterisk 11.6.0 and CentOS 6.4 64-bit. I am wondering if anyone else out there has been able to get app_swift working with Asterisk 11 and could share any tricks they used to get it installed? can you pastie your configure and make ? I don't have Cepstral6 but did submit tweaks to the code that should have made it Cepstral6 compatible. Also since you recently spent money with Cepstral, they'll help you. They've got at least one guy who understands the app_swift code and was working on forking it as an official version. Thanks to Jeremy Kister's advice, I was able to get this compiled. He suggested these two commands: yum update -y yum install asterisk-devel These commands upgraded my Asterisk to 11.7.0 and installed the Asterisk development files that I was missing. After I did that, the make worked beautifully: [root@dialer app_swift]# make ____ (_) / __) _ _ ___ _ _ _ _ _| |__ _| |_ ( | _ \| _ \ /___) | | | (_ __|_ _) / ___ | |_| | |_| | |___ | | | | | | || |_ \_| __/| __/ () |___/ \___/|_| |_| \__) |_| |_| gcc -I/opt/swift/include -I/usr/include -g -Wall -fPIC -D_SWIFT_VER_6 -D_AST_VER_11 -c -o app_swift.o app_swift.c gcc -shared -Xlinker -x -o app_swift.so -L/opt/swift/lib -L/usr/lib -lswift -lceplang_en -lceplex_us app_swift.o * Run 'make install' to install the app_swift module. * When I ran the make install, I did get "install: cannot create regular file `/usr/lib/asterisk/modules': No such file or directory", but I just changed the line SYS_LIB_DIR=/usr/lib in the Makefile to SYS_LIB_DIR=/usr/lib64 and it worked fine. Just wanted to post this resolution in case it helps someone else out there. Brandon -- _ -- 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] patlooptest output - errors
On Tue, Jan 14, 2014 at 05:58:57PM +, Rodrigo Borges Pereira wrote: > > Running patlooptest following instructions on Digium KB, using dahdi 2.5.1 > with a TE205P. I'm getting this kind of output (not full output). Note, I'm > using a standard cat5 ethernet cable (30cm), with a Digium T10i crossover. [snip] > Event: 29 > Event: 30 [snip] > Cause for concern? > > I'm testing a card which has been used before without issues. I'm just > playing with patlooptest for the first time, and wasn't expecting to see > this on a supposedly good card. > > appreciate any input, thx. Those events (29 and 30) indicate that patlooptest isn't able to keep up with the hardware. Do you get the same thing when you run patloop test with elevated permissions? $ chrt -f 99 patlooptest Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] patlooptest output - errors
Thanks for the tips. I forgot to mention, i did try lsof, and it comes out with no result. There is one thing I see different between the spans: shell> head /proc/dahdi/1 Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" B8ZS/ESF LOOP 1 TE2/0/1/1 Clear Master RED 2 TE2/0/1/2 Clear RED 3 TE2/0/1/3 Clear RED 4 TE2/0/1/4 Clear RED 5 TE2/0/1/5 Clear RED 6 TE2/0/1/6 Clear RED 7 TE2/0/1/7 Clear RED 8 TE2/0/1/8 Clear RED shell> head /proc/dahdi/2 Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF LOOP CRC4 error count: 1 E-bit error count: 1 25 TE2/0/2/1 Clear Master LOOP 26 TE2/0/2/2 Clear LOOP 27 TE2/0/2/3 Clear LOOP 28 TE2/0/2/4 Clear LOOP 29 TE2/0/2/5 Clear LOOP 30 TE2/0/2/6 Clear LOOP shell> cat /etc/dahdi/system.conf loadzone=us defaultzone=us span=1,0,0,esf,b8zs clear=1-24 span=2,0,0,esf,b8zs clear=25-48 Note the CRC4 and E-bit lines on span 2. Right now both ports are in loop mode, no cable attached. Maybe you guys @ Digium have seen this before as a symptom of card malfunction? Thanks. On Tue, Jan 14, 2014 at 9:18 PM, Russ Meyerriecks wrote: > On Tue, Jan 14, 2014 at 12:54 PM, Rodrigo Borges Pereira > wrote: > > Interestingly, span 2 goes into loop mode, but patlooptest can't access > it: > > > >> patlooptest /dev/dahdi/2 -t 300 -v > > /dev/dahdi/2: Device or resource busy > > > > And I do not have Asterisk running. > You should be able to use "lsof /dev/dahdi/2" to find out what process > has the file open. > > > So I > > am a bit confused in what's the most correct way of doing this kind of > > tests. patlooptest + physical loopback? patlooptest alone? physical > loopback > > with some other method? > Physical loopback devices are nice because they encompass the cable > into testing. > Digital loopbacks are nice because you don't have to get up from your > desk and plug something in :-) We use them a lot in our automated > testing. > > One other thing to note. Sometimes patlooptest can be starved by the > process scheduler. When this happens, it won't have enough time to > produce or consume all the sequential bytes that it needs to. Then it > will report errors. You can try increasing the scheduler priority > with: > chrt 99 patlooptest ... > > -- > Russ Meyerriecks > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > direct: +1 256-428-6025 > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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] patlooptest output - errors
On Tue, Jan 14, 2014 at 12:54 PM, Rodrigo Borges Pereira wrote: > Interestingly, span 2 goes into loop mode, but patlooptest can't access it: > >> patlooptest /dev/dahdi/2 -t 300 -v > /dev/dahdi/2: Device or resource busy > > And I do not have Asterisk running. You should be able to use "lsof /dev/dahdi/2" to find out what process has the file open. > So I > am a bit confused in what's the most correct way of doing this kind of > tests. patlooptest + physical loopback? patlooptest alone? physical loopback > with some other method? Physical loopback devices are nice because they encompass the cable into testing. Digital loopbacks are nice because you don't have to get up from your desk and plug something in :-) We use them a lot in our automated testing. One other thing to note. Sometimes patlooptest can be starved by the process scheduler. When this happens, it won't have enough time to produce or consume all the sequential bytes that it needs to. Then it will report errors. You can try increasing the scheduler priority with: chrt 99 patlooptest ... -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] patlooptest output - errors
Thanks. The cable is straight, and I was using it together with dahdi:maint. Both spans are set to provide (0). I've now tested span 1 again without cable, and still same kind of result: Timeout achieved Ending Program Test ran 28309 loops of 2039 bytes/loop with 3692 errors Interestingly, span 2 goes into loop mode, but patlooptest can't access it: > patlooptest /dev/dahdi/2 -t 300 -v /dev/dahdi/2: Device or resource busy And I do not have Asterisk running. Mind, I have been told recently by Digium support that a fully conclusive test (in the context of eventual RMA) requires the physical loopback. So I am a bit confused in what's the most correct way of doing this kind of tests. patlooptest + physical loopback? patlooptest alone? physical loopback with some other method? Thanks. On Tue, Jan 14, 2014 at 6:15 PM, Russ Meyerriecks wrote: > On Tue, Jan 14, 2014 at 11:58 AM, Rodrigo Borges Pereira > wrote: > > > Cause for concern? > > For patlooptest ensure that: > - Your ethernet cable is straight through and not crossover > - The span you're testing is set to "provide" timing, not recover > - In dahdi_tools you can "make utils" and just use dahdi_maint to put > the card into loopback without messing with cables. > > > -- > Russ Meyerriecks > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > direct: +1 256-428-6025 > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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] patlooptest output - errors
On Tue, Jan 14, 2014 at 11:58 AM, Rodrigo Borges Pereira wrote: > Cause for concern? For patlooptest ensure that: - Your ethernet cable is straight through and not crossover - The span you're testing is set to "provide" timing, not recover - In dahdi_tools you can "make utils" and just use dahdi_maint to put the card into loopback without messing with cables. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] patlooptest output - errors
Hi there,
Running patlooptest following instructions on Digium KB, using dahdi 2.5.1
with a TE205P. I'm getting this kind of output (not full output). Note, I'm
using a standard cat5 ethernet cable (30cm), with a Digium T10i crossover.
Error 4985 (loop 19898, offset 425, error 214): Unexpected result, Read:
0xff, Expected 0x00.
Offset 422 { ff ff ff ff ff ff ff }
Error 4986 (loop 19898, offset 426, error 215): Unexpected result, Read:
0xff, Expected 0x00.
Offset 423 { ff ff ff ff ff ff ff }
Error 4987 (loop 19898, offset 427, error 216): Unexpected result, Read:
0xff, Expected 0x00.
Offset 424 { ff ff ff ff ff ff ff }
Error 4988 (loop 19898, offset 428, error 217): Unexpected result, Read:
0xff, Expected 0x00.
Offset 425 { ff ff ff ff ff ff ff }
Error 4989 (loop 19898, offset 429, error 218): Unexpected result, Read:
0xff, Expected 0x00.
Offset 426 { ff ff ff ff ff ff 2e }
Error 4990 (loop 19898, offset 430, error 219): Unexpected result, Read:
0xff, Expected 0x00.
Offset 427 { ff ff ff ff ff 2e 2f }
Error 4991 (loop 19898, offset 431, error 220): Unexpected result, Read:
0xff, Expected 0x00.
Offset 428 { ff ff ff ff 2e 2f 30 }
Error 4992 (loop 19898, offset 432, error 221): Unexpected result, Read:
0x2e, Expected 0x00.
Offset 429 { ff ff ff 2e 2f 30 31 }
Event: 29
Event: 30
Error 4993 (loop 24990, offset 0, error 1): Unexpected result, Read: 0x7e,
Expected 0x7a.
Offset 0 { 7e 7f 80 81 }
Error 4994 (loop 24990, offset 428, error 2): Unexpected result, Read:
0xff, Expected 0x2a.
Offset 425 { 27 28 29 ff ff ff ff }
Error 4995 (loop 24990, offset 429, error 3): Unexpected result, Read:
0xff, Expected 0x00.
Offset 426 { 28 29 ff ff ff ff 2a }
Error 4996 (loop 24990, offset 430, error 4): Unexpected result, Read:
0xff, Expected 0x00.
Offset 427 { 29 ff ff ff ff 2a 2b }
Cause for concern?
I'm testing a card which has been used before without issues. I'm just
playing with patlooptest for the first time, and wasn't expecting to see
this on a supposedly good card.
appreciate any input, thx.
rgds.
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[asterisk-users] Asterisk Community Code of Conduct
Hello everyone! Asterisk is a large project and is used all over the world. As a result, the Asterisk project is lucky to have a large and thriving user community that communicates in a variety of places: IRC channels, forums, mailing lists, social media, and more. We in the Asterisk community value that diversity - the Asterisk project is stronger due to our many different experiences and opinions. For a long time, the Asterisk project has had no code of conduct to guide those who wish to participate in the community. While we may wish to believe that we all know the "correct" way to behave when interacting with others, given our wide and diverse backgrounds, this definition of "correct" can vary from person to person. Even people who sometimes mean well can create situations where others feel excluded from the community. This is a situation that we would like to avoid. As such, from today on, people who wish to participate in the Asterisk community agree to adhere to a community code of conduct. The policy, which is available on the Asterisk wiki [1], is reproduced here as follows: Asterisk Community Code of Conduct The Asterisk community is a large group of individuals, representing many nations, ethnicities, ages, technical professions and specialities. Working together on Asterisk can be a challenge with so many differing perspectives and backgrounds. Therefore to ensure the community is healthy, happy, and stress-free, participants in the Asterisk project agree to adhere to the following Community Code of Conduct. Note that by joining and/or participating in the Asterisk community, you are agreeing that you accept and will adhere to the rules listed below, even if you do not explicitly state so. Acceptable Behavior * Be considerate - Experience levels vary. Don't assume that someone can understand your particular explanation. - Keep in mind that English is a second language for many users. * Be respectful - It is possible to strongly disagree without using harsh language or resorting to derogatory comments. If you disagree with someone, disagree with the argument, not the character of the person. - Remember that everyone is entitled to an opinion. * Ask for help - If you don't know how to proceed with an aspect of development or documentation, ask for help! Always read the documentation, but don't be afraid to ask "silly" questions. - When asking for help, take advantage of the resources that are available to you, including the wiki, mailing list archives, and Asterisk: The Definitive Guide. * Take responsibility - If you did something wrong, apologize to the affected. Do your best to fix the issue and, if you can't, ask for help! - If someone does take responsibility, be considerate. * Give credit - Give proper credit to everyone involved In any contribution to the project, be it documentation, tests, code, or anything in between. - If someone fails to give adequate contribution, gently remind them while being considerate. Assume that the omission was accidental, not malicious. Unacceptable Behavior The Asterisk project reserves the right to take action in safe-guarding the community from those that participate in unacceptable behavior. Unacceptable behavior involves: * Flaming - Arguing in a disrespectful way, attacking the character of others, rabidly ranting about things you dislike and refusing to drop the topic. * Trolling - Intentionally baiting others into flaming or heated arguments for the sake of argument or drama itself. * Mean-spirited or offensive talk - This could be combinations of the above, being rude, vulgar, and generally offensive to others. In general, if community moderators and admins are receiving many complaints about your behavior, then you are likely doing something wrong. If you don't have anything nice to say, don't say anything. Consequences to bad behavior may involve bans from communication forums or channels and restrictions on privileges within the community. Complaints about members behavior or appeals in regards to bans or loss of privilege can be sent to asteriskt...@digium.com. Open Community We invite anybody, from any company, locale, or even other projects to participate in the Asterisk project. Our community is open, and contribution is welcome. Diversity makes a project strong, and we are proud to include anyone who wants to collaborate with others in a respectful fashion. Project Leadership The role of project leadership is handled by the founders of the Asterisk project, Digium Inc. As a member of the Asterisk community, Digium develops the project in co-operation with the overall Asterisk community. Community members are always welcome to take positions of leadership as module maintainers within the Asterisk project, particularly when they are the author of the module. In addition to providing development resources for Asterisk itself, Digium provides community resources including the bug tracker, mailing lists,
Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
Hi Steve, On 14-01-14 10:39, Steven Howes wrote: On 14 Jan 2014, at 02:19, Patrick Lists wrote: Thanks for your feedback Paul. The not having outbound trunks is going to be a challenge. Why? it’s what contexts were invented for. Yes that is indeed what they are for but in the case "they" find a loophole or exploit a bug then not having outbound trunks is much safer. Regards, Patrick -- _ -- 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 QOS
On Tuesday 14 January 2014, richard.seg...@marisec.ca wrote: > I asked this on the list over the weekend, and likely missed a few people > inboxes. > > I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes > I have asterisk sending QOS data to the console. It seems I get QOS data > only if the caller hangs up, with the variable being empty if the callee > (or asterisk) hangs up. > > Any idea why I would see this? This is not just Asterisk! "Real" phone exchanges don't know when the callee has hung up, either. It was a limitation of analogue circuit-switching technology that has been slavishly emulated up to the present day, even although a digital phone technically *can* inform the exchange that it has been hung up. If someone calls you, you hang up but they keep their phone off the hook, then the next time you pick up your receiver you will still be connected to them. (Call your landline from your mobile and place the mobile in front of a radio, if you don't believe me.) -- AJS Answers come *after* questions. -- _ -- 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] From: "Unavailable" ; tag=as120a1079.
"*callerid* = : Caller ID information used when nothing else is available. Defaults to *asterisk*." This is the explanation :) 2014/1/14 Nick Cameo > Hello Laszlo, > > That you for your response. Just to confirm callerid=whatever will only > effect the private numbers? And will > not have any effect on FROM headers with valid CIDs, as is intended? > > N. > > > > > -- > _ > -- 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 > -- -- Kind regards, Laszlo Bekesi http://voipfreak.net -- _ -- 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 read IRQs and timing slips values
On Mon, Jan 13, 2014 at 08:42:13PM -0500, Paul Belanger wrote: > > cat /proc/dahdi/2 > > Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" (MASTER) HDB3/CCS > > Timing slips: 175319 > > > > 32 TE2/0/2/1 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 33 TE2/0/2/2 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 34 TE2/0/2/3 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 35 TE2/0/2/4 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 36 TE2/0/2/5 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 37 TE2/0/2/6 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 38 TE2/0/2/7 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 39 TE2/0/2/8 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 40 TE2/0/2/9 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 41 TE2/0/2/10 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 42 TE2/0/2/11 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 43 TE2/0/2/12 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 44 TE2/0/2/13 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 45 TE2/0/2/14 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 46 TE2/0/2/15 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 47 TE2/0/2/16 HDLCFCS (In use) (EC: VPMOCT064 - INACTIVE) > > 48 TE2/0/2/17 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 49 TE2/0/2/18 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 50 TE2/0/2/19 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 51 TE2/0/2/20 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 52 TE2/0/2/21 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 53 TE2/0/2/22 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 54 TE2/0/2/23 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 55 TE2/0/2/24 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 56 TE2/0/2/25 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 57 TE2/0/2/26 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 58 TE2/0/2/27 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 59 TE2/0/2/28 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 60 TE2/0/2/29 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 61 TE2/0/2/30 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 62 TE2/0/2/31 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > > > 3. As shown above, my box has two connections with PSTN (same provider for > > both): one direct, one through an HiPath PBX. > > How can I double check timing slips don't come from "inconsistency between > > both clock sources" ? > > My first thought would be to unplug the link between Asterisk and HiPath and > > compare two /pro/dahddi/1 outputs. > > Thoughts ? > > > > I basically had the same issue as you for one of my sites. I tried > everything under the sun to figure it out, change cables, loop back > test, change out hardware, clocking, etc. > > In the end I had to upgrade dahdi to 2.7+ and the issue went away. > Never did figure out the real problem, but to this day I think the > issue was a delay on the frames from the PCI bus into the software. > > All that to say, try upgrading DAHDI and see what happens. As far as Olivier's concern, I still vote there is some physical cabling issue that is causing problems. However, just for posterity, in my experience if HDLC aborts are occuring and there are timing slips, it does not have anything to do with the card / host communication, but rather the issue has more to do with the framer and connection to provider. This is because the timing slips are reported directly by the framer and that doesn't depend on the host communication. Just FYI... -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- 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 QOS
Generally exten h is not run when the callee hangs up. See also the "g" option
to Dial.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
richard.seg...@marisec.ca
Sent: Tuesday, January 14, 2014 10:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk QOS
I asked this on the list over the weekend, and likely missed a few people
inboxes.
I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes I
have asterisk sending QOS data to the console. It seems I get QOS data only
if the caller hangs up, with the variable being empty if the callee (or
asterisk) hangs up.
Any idea why I would see this?
exten => h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS})
Thanks,
Richard Seguin
--
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[asterisk-users] Asterisk QOS
I asked this on the list over the weekend, and likely missed a few people
inboxes.
I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes I
have asterisk sending QOS data to the console. It seems I get QOS data only
if the caller hangs up, with the variable being empty if the callee (or
asterisk) hangs up.
Any idea why I would see this?
exten => h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS})
Thanks,
Richard Seguin--
_
-- 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:
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Re: [asterisk-users] From: "Unavailable" ; tag=as120a1079.
Hello Laszlo, That you for your response. Just to confirm callerid=whatever will only effect the private numbers? And will not have any effect on FROM headers with valid CIDs, as is intended? N. -- _ -- 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 11.7.0 and dialplan
>> Where is my dialplan? Look for errors in: /var/log/asterisk/messages Doug -- _ -- 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 read IRQs and timing slips values
On 01/14/2014 04:32 AM, Olivier wrote: I'm 100% sure my PBX is configured to use provider's clock (but I won't swear my PBX is currently using provider's clock) I have had to power the server down, UNPLUG the power, leave unplugged for 4 minutes, power up. I had a T1 timing issue this procedure fixed. Adrian -- _ -- 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 11.7.0 and dialplan
I have a fresh install of asterisk 11.7.0 when I run it and do a "dialplan show" The only thing I see is: [ Context 'parkedcalls' created by 'features' ] '700' => 1. Park() [features] -= 1 extension (1 priority) in 1 context. =- extensions.conf has the MANY contexts in it. When I run "asterisk -vc" I see the line that its loading extensions.conf and no errors reported. Where is my dialplan? Thanks, jerry -- _ -- 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] From: "Unavailable" ; tag=as120a1079.
Hey Nick, Just add callerid=whatever to sip.conf (general context). 2014/1/14 Nick Cameo > Correction, and by TO, I mean FROM header :) > > -- > _ > -- 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 > -- -- Kind regards, Laszlo Bekesi http://voipfreak.net -- _ -- 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] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
On 14 Jan 2014, at 02:19, Patrick Lists wrote: > Thanks for your feedback Paul. The not having outbound trunks is going to be > a challenge. Why? it’s what contexts were invented for. Steve -- _ -- 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 read IRQs and timing slips values
I had a late phone call yesterday with provider's level-1 support team. As strange as it seems, the guy said he could "read clock slips in his box logs though his box is supposed to provide clock and not to use my PBX's one". I'm 100% sure my PBX is configured to use provider's clock (but I won't swear my PBX is currently using provider's clock). I think I will insert a new Patton box between my provider's one and my PBX to see what happens. The setup would be: PSTN --> provider's equipment <> Patton GW <> Asterisk Then I'll also replace the card inside my PBX to see if things are changing. 2014/1/14 Paul Belanger > On Thu, Jan 9, 2014 at 12:01 PM, Olivier wrote: > > Hi, > > > > On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), > > users are complaining for bad audio. > > > > My setup is: > > PSTN <--E1/PRI ---> Asterisk <--- E1/PRI---> Siemens HiPath <---E1/PRI > ---> > > PSTN > > > > asterisk -rx "dahdi show version" > > DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC > > > > asterisk -rx "pri show version" > > libpri version: 1.4.12 > > > > > > > > A quick glance at Asterisk logs shows lines like this: > > [2014-01-09 17:19:34] NOTICE[26034]: chan_dahdi.c:3099 > > my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of > > span 1 > > [2014-01-09 17:19:35] NOTICE[26035]: chan_dahdi.c:3099 > > my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of > > span 2 > > [2014-01-09 17:19:49] NOTICE[26035]: chan_dahdi.c:3099 > > my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of > > span 2 > > > > > > I read an old thread inviting an admin to check for shared IRQs and > timing > > slips. > > > > My questions are: > > > > 1. cat /proc/interrupts 's output is: > > # cat /proc/interrupts > > CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 > > CPU6 CPU7 > >0: 90147 0 0 0 0 0 > > 0 0 IO-APIC-edge timer > >1: 2 0 0 0 0 0 > > 0 0 IO-APIC-edge i8042 > >8: 0 0 1 0 0 0 > > 0 0 IO-APIC-edge rtc0 > >9: 0 0 0 0 0 0 > > 0 0 IO-APIC-fasteoi acpi > > 12: 4 0 0 0 0 0 > > 0 0 IO-APIC-edge i8042 > > 14: 93 0 0 0 0 0 > > 0 0 IO-APIC-edge ata_piix > > 15: 0 0 0 0 0 0 > > 0 0 IO-APIC-edge ata_piix > > 16: 3378646209 3378695076 3378691115 3378697362 3378691116 3378706831 > > 3378710635 3378702358 IO-APIC-fasteoi wct2xxp > > > > Can I positively conclude that my Dahdi PRI board IS NOT sharing IRQ > (which > > is good) ? > > > > 2. What would you suggest reading the following output ? > > > > cat /proc/dahdi/2 > > Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" (MASTER) HDB3/CCS > > Timing slips: 175319 > > > > 32 TE2/0/2/1 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 33 TE2/0/2/2 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 34 TE2/0/2/3 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 35 TE2/0/2/4 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 36 TE2/0/2/5 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 37 TE2/0/2/6 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 38 TE2/0/2/7 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 39 TE2/0/2/8 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 40 TE2/0/2/9 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 41 TE2/0/2/10 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 42 TE2/0/2/11 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 43 TE2/0/2/12 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 44 TE2/0/2/13 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 45 TE2/0/2/14 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 46 TE2/0/2/15 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 47 TE2/0/2/16 HDLCFCS (In use) (EC: VPMOCT064 - INACTIVE) > > 48 TE2/0/2/17 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 49 TE2/0/2/18 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 50 TE2/0/2/19 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 51 TE2/0/2/20 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 52 TE2/0/2/21 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 53 TE2/0/2/22 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 54 TE2/0/2/23 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 55 TE2/0/2/24 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 56 TE2/0/2/25 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 57 TE2/0/2/26 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 58 TE2/0/2/27 Clear (In use) (EC: VPMOCT064 - INACTIVE) > > 59 TE2/0/2/28 Clear (In use) (EC: VPMOCT064 - INACT
