Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On Sat, Sep 24, 2011 at 9:35 PM, Bruce B bruceb...@gmail.com wrote: Hi everyone, I don't mean to be rude but honestly which genius comes up with changing the Word to the wise -- if one starts a sentence with I don't mean to be...X your true intentions are to be just that. If you find yourself doing that, please stop. Rethink what you are writing and word it in a more polite manner. You will ruffle less feathers and have a much more constructive dialog. -M -- _ -- 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] Strange network issue
On Thu, Jul 28, 2011 at 4:46 AM, Duncan Turnbull dun...@e-simple.co.nzwrote: On 28/07/2011, at 8:41 PM, Paul Hayes p...@provu.co.uk wrote: On 28/07/11 02:58, Mike Diehl wrote: Any ideas? Mike. I'd go on site if possible and see what actually happens at 19:00. Set up a wireshark trace capturing all traffic through their router. -- I am picking a cleaner plugging a powerful vacuum cleaner in ;-) That's what I mentioned earlier, but thinking about it they must have a German cleaning service to get such precise vacuum timing. -M -- _ -- 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] Strange network issue
On Thu, Jul 21, 2011 at 7:13 PM, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I've got a strange problem with a customer's phones. They've got a bunch of Grandstreams that seem to be rock solid... until 7:00pm. At 7:00, some of the phones become unavailable, and stay down. Call quality is solid almost all the time. But right at 7:00, things go bad. Only some of the phone lines go down and they stay down until the phone is rebooted. I'm not even sure what to look for when I go to the site. Any ideas? Many years ago, in my college days, the network in one building would fail around a certain time every day. The sun would hit the network closet around the same time every day in the summer, causing the equipment to overheat and temporarily fail. I would go there and observe everything which happens at 7:00. Maybe it's something that a cleaning service inadvertently does, like faulty wiring + a vacuum cleaner. -M -- _ -- 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, Jul 18, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote: 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. Where are you going to store the voicemail? Could some of this space be used for asterisk modules? -- _ -- 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] Using Firewall to protect Asterisk
On Fri, Jul 15, 2011 at 12:47 PM, CDR vene...@gmail.com wrote: I need to keep out all connection from 5 countries, which originate most of the Denial of Service attacks. The entries are around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter way to do this by using User Tables in iptables, that will keep the speed equal to LOG(x). I already tried using a straight list and it kills the box. Unless a smarter way us found, there is no way to use iptables. Federico Are you matching on new packets/connections only or all packets? -M -- _ -- 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 Hang up a stale SIP channel?
On Wed, Jul 13, 2011 at 5:35 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're using asterisk 1.8.3.2 and are finding incidences of stale channels remaining after both parties have hung up. We have tried to hang the channel up using channel request hangup But by it's definition, it will not work as it only executes the hangup as soon as the the channel is written to or read from but as the channel is stale, it will not be written to or read from so the command will not instigate the hangup. Does anyone know of any way we can hangup a stale channel via the console? I've had this happen a few times, but with 1.6.2. I ended up writing .call files to the asterisk spool directory instructing it to hang up a particular sip channel. -M -- _ -- 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] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
On Tue, Jun 21, 2011 at 4:12 AM, randulo rand...@randulo.com wrote: On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov abalas...@evaristesys.com wrote: I nominate this for most imaginative use of Asterisk-users of 2011. It's already qualified to win in the grammar and spelling categories. /r My 3 year old. unfortunately, has sent a few messages like this in the past. I guess she watched me unlock the screen enough times to memorize the key code. -M -- _ -- 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 GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x
On Mon, Jun 6, 2011 at 11:55 AM, Silver Thorne szilvertho...@gmail.com wrote: Hello Folks; Perhaps I am chasing my tail here. Before I go any further, is this compatible/supported in Asterisk 1.6x? If so, I would be willing to post any manager.conf or http.conf snippets needed. When I attempt to open the Asterisk Web GUI, I get a 'page not found'. I am sure this is something really minor - something silly that I missed. Any words of wisdom? Glen I would get used to using the command line interface or use PBX In a Flash, FreePBX or something like that if you want / need what they offer. -M -- _ -- 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] AMI buffering event output?
2011/6/2 Örn Arnarson o...@arnarson.net: To clarify; I observe the exact same results no matter how I connect to the AMI on this particular server. I tried connecting FROM this server to an AMI on another server to make sure it wasn't the telnet client or some such, and then it worked perfectly. To answer the question, if I use the external IP address rather than 127.0.0.1 I observe the same results. echo $LANG on each server ? -- _ -- 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 fails when DNS or internet fails
On Mon, May 30, 2011 at 2:44 AM, gincantalupo gincantal...@fgasoftware.comwrote: Hi, it is a known problem, one of the worst. To avoid it: - do not use urls, only ip addresses in sip.conf or put your urls inside /etc/hosts (is what I do especially sip providers urls) or install a dns-cache on your pbx (maybe the best solution) Giorgio Even a dns cache won't help you forever. Once the entry's TTL expires, your cache won't be able to give you an answer until it can contact root servers on the internet. -M -- _ -- 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] click to call with php
On Fri, May 20, 2011 at 4:38 AM, Ishfaq Malik i...@pack-net.co.uk wrote: If you are going to use call files don't write them directly to /var/spool/asterisk/outgoing/ write them in some temp directory and then move them to /var/spool/asterisk/outgoing/ Ish Make sure that your temp file is on the same mounted file system as /var/spool/asterisk/outgoing. If they are on different file systems, mv will do a cp and a rm in this situation and you won't get the atomic operation you were hoping for. -M -- _ -- 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] Restart asterisk destroy all registered SIP peers
On Fri, May 20, 2011 at 2:10 PM, satish patel satish...@hotmail.com wrote: Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks S Shouldn't the phones re-register on their own? Mine do it every few minutes. -M -- _ -- 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] Restart asterisk destroy all registered SIP peers
On Fri, May 20, 2011 at 3:00 PM, satish patel satish...@hotmail.com wrote: We have polycom 501 and i am waiting since last 5 min no registration require appear. -S With Polycom 321 you can poke around the menus -- one of them has a countdown timer which will show you when the next registration happens. -M -- _ -- 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] script to trim sip.conf
On Tue, May 17, 2011 at 4:21 PM, satish patel satish...@hotmail.com wrote: Hey Guys! Sorry i am posting scripting question in asterisk forum but i had no choice. also i am not script expert so i though anyone here might help me. following is my example sip.conf now i want to add accountcode=callerid_name for example accountcode=Katie Wilson in entire file. we have around 200 extension could someone help me to figure out how to do that with perl script or shell would be fine. [100](seb-exten) callerid=Katie Wilson 100 mailbox=100@default [200](seb-exten) callerid=Ramona Minero 200 mailbox=200@default Satish, Give this a shot: cat sip.conf | perl -pi -e s/^callerid=\(.*)\ (.*)/callerid=\\$1\ \$2\naccountcode=\\$1\/ sip.conf.new and compare them. -- _ -- 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-cpu utilization 60 %
On Sun, May 15, 2011 at 4:08 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: Not exactly. Asterisk is multi-threaded. strae traces a specific thread. To see the most active thread, press 'H' (shift-h) in top. Wait for the display to refresh at least twice (on the first time it won't make sense) and now check to see which is the top thread. strace -f -ff ASTERISK_PID traces all threads on my system. -- _ -- 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-cpu utilization 60 %
On Mon, May 16, 2011 at 10:33 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, May 16, 2011 at 10:01:36AM -0400, Mark Deneen wrote: strace -f -ff ASTERISK_PID traces all threads on my system. But do you really want that? Asterisk has many threads generating quite a lot of noise (threads periodically polling something). Probably not. I was merely referring to the statement that strace only traces a particular thread. I would do top -H and then strace the asterisk threads with high CPU numbers. -M -- _ -- 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] question on ways to activate voicemail light on polycom
On Fri, May 6, 2011 at 2:14 PM, Jerry Geis ge...@pagestation.com wrote: Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry I don't think there is a way to do it natively inside of asterisk, but I control it from a shell script. The shell script parses the output of sip show peers, crafts an application/simple-message-summary SIP message and then uses netcat to send it to the appropriate IP address / port. -M -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
On Thu, May 5, 2011 at 4:07 PM, Paul Belanger pabelan...@digium.com wrote: On 11-05-05 12:30 PM, Ira wrote: At 07:56 AM 5/5/2011, you wrote: So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popular? How can the Asterisk project reproduce their success? Well, it's not a lot of people willing to run beta software on their phone system. Phones need to work and for most people they need to work perfectly all the time. I'm one of those oddities that will always run beta software if given the chance but my experience is that quite rare. I am not saying using production servers to test, rather reproducing your production setups in a test environment. You would then create test plans or test cases of the features you use in Asterisk. Once documented, for each and every RC of Asterisk you go through the steps outlined in your test plan / case, confirming this work as expected and then documenting the results. Not everyone has spare dahdi hardware / analog T circuits, but I agree. -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
Satish, You must register your handle with freenode, because the asterisk channel only allows registered people in. http://freenode.net/faq.shtml#nicksetup -M On Fri, Apr 29, 2011 at 11:41 AM, satish patel satish...@hotmail.com wrote: Hey Matt, I have download irc linux base CLI client and connect to irc.freenode.net i can see bunch or channels but i didn't find any #asterisk or #asterisk-bugs name. Am i looking at wrong place ? *** #asterisk You're not on that channel *** #asterisk Cannot join channel (+r) - you need to be identified with services /JOIN #asterisk Date: Fri, 29 Apr 2011 14:26:46 +1200 From: li...@venturevoip.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind? On 29/04/11 1:16 PM, Ira wrote: Well, I've no idea how to do that. I can duplicate the problem every IRC is an online chat system like MSN or Skype except that it's more like a mailing list - you can talk to lots of people at the same time. On Windows you can use a program like mIRC to connect to irc.freenode.net or even a plugin in Firefox. Once you're connected to IRC you can join chat rooms. There are some like #asterisk for discussion about Asterisk and #asterisk-bugs for discussion about Asterisk bugs. Post back here if you have any problems connecting. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot call to my server with SIP
On Fri, Apr 22, 2011 at 11:02 AM, Paul van der Vlis p...@vandervlis.nl wrote: Hello, I cannot call my server over the internet with SIP anymore. Even when I do a maximum logging on my firewall, I don't see packets coming from outside. I've tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is p...@vandervlis.nl. This should connect trought DNS to the machine xen8.vandervlis.nl: When you say that you can't call _anymore_ did it work in the past? How long ago? Here's what I could see from here: * xen8.vandervlis.nl is listening on udp/5060 * there is a srv record published by the authoritative name servers at ns1, ns2 and ns3.vandervlis.nl The only thing that I noticed was that the TTL for xen8.vandervlis.nl was 24 hours, which is why I asked about it working in the past. Is it possible that the ip address on xen8.vandervlis.nl has changed but the old record is still present in some dns caches? -M -- _ -- 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-user] Can't get hostname on asterisk dialplan by ENV()
Rajnikant,
This surely depends on how you start asterisk. How are you starting
the asterisk process?
-M
On Thu, Apr 21, 2011 at 7:20 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote:
Hi Friend,
Can't get hostname environment variable on asterisk dialplan.
Help me about how to get hostname environment variable on asterisk dialplan.
I have written export HOSTNAME in /root/.bash_profile and when i execute
echo $HOSTNAME then get right hostname but not success through asterisk
dialplan.
Get environment variable path right value through below statement.
exten = XXX,n,NoOp(--- ${ENV(PATH)})
I have tried like this:
exten = XXX,n,Set(CDR(hostname)=${System(echo $HOSTNAME)})
exten = XXX,n,Set(CDR(hostname)=${ENV(HOSTNAME)})
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
Call : +91-9737456583
Software Engineer
---
Working On Linux,C/C++,Asterisk Technology
Gandhinagar - Gujarat
--
_
-- 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] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()
On Thu, Apr 21, 2011 at 3:23 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 21 Apr 2011, RAJNIKANT VANZA wrote: Can't get hostname environment variable on asterisk dialplan. 1) Is HOSTNAME in the Asterisk process's environment? What does executing: tr '\000' '\n' /proc/$(cat /var/run/asterisk.pid)/environ This is /var/run/asterisk/asterisk.pid on my system. I use runit to manage the asterisk process, and the chpst program allows fine control over environment and other limits. -M -- _ -- 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-user] Can't get hostname on asterisk dialplan by ENV()
On Thu, Apr 21, 2011 at 4:30 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 21 Apr 2011, Mark Deneen wrote: I use runit to manage the asterisk process, and the chpst program allows fine control over environment and other limits. runit is intended to be a sysvinit (/sbin/init) replacement and is not installed (by default) on CentOS or Ubuntu distributions. Can chpst be used by itself? It seems a useful program except that you need to explicitly name each environment variable you want 'ignored' and it is part of a larger package that may have far reaching implications Steve, runit is actually very unobtrusive. It is capable to replacing init, but I don't think many people actually use it that way. http://smarden.org/runit/useinit.html documents how to use it with init. If I wanted to clear the environment first, I'd just use env and have that call chpst. I like runit because it manages the process without the typical pid-file tracking that most init scripts use. If the process dies, for whatever reason, it is automatically restarted. stdout is captured and redirected to an optional log process which can roll logs, removing the need for logrotate and figuring out what special signal to send the process to tell it that you've truncated the log file. There is a catch, though. Your process has to run in the foreground, and runsv keeps it in the background. So, for programs which auto-detach and background themselves, you have to run them with a switch that says not to run as a daemon. It's not everyone's cup of tea, but I find it to be perfect for my needs, and a very well written utility. -M -- _ -- 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] VoiceMail to text mail
On Wed, Apr 20, 2011 at 4:35 PM, satish patel satish...@hotmail.com wrote: Hey Thanks for that reply after add following option it works but the text output is totally different.. what its totally different is this dictionary problem ? -hmm /var/lib/asterisk/communicator -samprate 8000 In audio file its just: Hello satish this is test message 0: i started is it see no oil you did to less this tonight How many years have you spoken gibberish without knowing? Seriously, though, do you have a bit of an accent (compared to the pocketsphinx developers)? -M -- _ -- 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] R: No Internet, no asterisk
2011/4/19 Niccolò Belli darkbas...@gmail.com: Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Wouldn't a caching nameserver just return NXDOMAIN if it couldn't contact the authoritative server for that domain? -M -- _ -- 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] chan_sip.c: No such host: but I can resolve it from command line ?
dig @193.189.160.13 voip.siol ; DiG 9.6.0-APPLE-P2 @193.189.160.13 voip.siol ; (1 server found) ;; global options: +cmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: REFUSED, id: 35478 ;; flags: qr rd; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0 ;; WARNING: recursion requested but not available ;; QUESTION SECTION: ;voip.siol. IN A ;; Query time: 136 msec ;; SERVER: 193.189.160.13#53(193.189.160.13) ;; WHEN: Sat Apr 16 09:31:24 2011 ;; MSG SIZE rcvd: 27 I can't resolve that from here, but your expected answer (10.253.0.1) falls within a RFC1918 private ip address space so that name server may not be publishing authoritative records. Obviously, no one can check the name server at 10.253.9.6, since it is private as well. -M On Sat, Apr 16, 2011 at 8:41 AM, rob.r...@gmail.com wrote: Hi, I have Asterisk 1.4.10 under LMCE (upgrade is not an option) and have this strange error appearing in full log : [Apr 16 14:35:48] NOTICE[10802] chan_sip.c: -- Registration for 'num...@voip.siol' timed out, trying again (Attempt #22) [Apr 16 14:35:48] WARNING[10802] chan_sip.c: No such host: voip.siol [Apr 16 14:35:48] WARNING[10802] chan_sip.c: Probably a DNS error for registration to num...@voip.siol, trying REGISTER again (after 20 seconds) [Apr 16 14:36:08] NOTICE[10802] chan_sip.c: -- Registration for 'num...@voip.siol' timed out, trying again (Attempt #23) [Apr 16 14:36:08] WARNING[10802] chan_sip.c: No such host: voip.siol [Apr 16 14:36:08] WARNING[10802] chan_sip.c: Probably a DNS error for registration to num...@voip.siol, trying REGISTER again (after 20 seconds) But I can easily solve this URL from command line : host voip.siol voip.siol has address 10.253.0.1 What is wrong ? I have this in resolv.conf : nameserver 10.253.9.6 nameserver 193.189.160.13 And this URL should be resolved by first dns (it’s working from command line) Thanks in advance, Regards, Rob. -- _ -- 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] WARNING chan_sip.c:3115 __sip_xmit
On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Satish, Run dmesg and look for anything funny. This sounds very similar to when I had a netfilter nat helper not helping me at all. -M -- _ -- 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] dialplan is not finding my number asterisk 1.8.3
On Mon, Apr 4, 2011 at 3:20 PM, Jerry Geis ge...@pagestation.com wrote: I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the dialplan show it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] Its telling me it cannot find it. Its there - the dialplan shows its there. When I stop and start it works again for a little while. Matter of fact I just issued dialplan reload and calling into 1105 works again. Whats up? How do I get this to be consistent? Jerry I'm not all that familiar with 1.8 yet but, with 1.6.2, I ran into some similar problems with extenpatternmatchnew=yes. They were similar in that the dialplan was not executed as expected, but the behavior was deterministic. Your experience has things changing over time which is really quite strange. -M -- _ -- 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 Scripting Language
Since my perl skills are pretty much write-only, I've been using Python. I can actually look at it a year later and immediately see what the code is doing. -M On Fri, Apr 1, 2011 at 9:21 AM, Danny Nicholas da...@debsinc.com wrote: I'm going to vote for PERL as well. C is not a scripting language. Also keep in mind that you can compile PERL into C for your hundreds of calls per second box. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Friday, April 01, 2011 8:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Best Scripting Language Do you think C is a scripting language? -- Sent from my iPhone On Apr 1, 2011, at 8:27 AM, Roger Burton West ro...@firedrake.org wrote: On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote: Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Depends on the other parameters. Perl is great for rapid development, but I wouldn't run it per-call on a box taking hundreds of calls per second. (Ditto Ruby and Python.) C will be much faster, but it's more effort to write and debug. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
Look into the ipt_recent / xt_recent module. It's probably what he is using. On Wed, Mar 30, 2011 at 4:25 PM, vip killa vipki...@gmail.com wrote: could you please elaborate on how you have iptables setup to work that way? On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sox and bad quality when converting to 8 kHz
On Thu, Mar 24, 2011 at 4:58 PM, Thomas Winter thowin...@googlemail.com wrote: Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help I have had good results with this: sox in.wav -r 8000 -c 1 out.wav highpass 500 lowpass 4000 resample -ql Play around with the high and low pass numbers because they might need to be changed depending on the properties of your recordings. -M -- _ -- 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 stops responding
On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez cur...@telecomabmex.com wrote: On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote On 22 Jan 2011, at 18:02, Carlos Chavez wrote: Cannot allocate memory Have you tried looking at memory? S The server has 8gb of ram and 8gb of swap. Free indicates that there are at least two free gb of memory and swap remains at 0 use. Just asking the obvious, but, x86-64? How big is the asterisk process? -M -- _ -- 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] Polycom 500 / MWI
On Thu, Jan 20, 2011 at 12:55 PM, Brian C. Huffman bhuff...@etinternational.com wrote: Does anyone know how to setup this phone to work with asterisk so that the indicator light comes on when there's a new message and goes off quickly (less than a minute) after the message is deleted? Thanks, Brian Brian, I'm using Polycom 321 sets, and the MWI works wonderfully. If you look at the asterisk-1.6 source code, in app_voicemail.c, you can see where it calls queue_mwi_event(...) after leaving a message and after deleting a message. If you run a wireshark capture, you should see these in the trace. It also looks like, in most cases, an AMI event of MessageWaiting will be generated. I know it's not much, but it may help you to diagnose the problem further. -M -- _ -- 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] Top Posting
On Wed, Jan 19, 2011 at 2:37 PM, randulo rand...@randulo.com wrote: Slightly OT: why is the Gmail ad server, which is usually all about PBX, Asterisk, etc, now showing me Justin Beiber concert tickets on this thread? Are they seeing it as that childish? /r Also OT: Google combines message context with your personal search history to do ad targeting, so look in the mirror. I just made that up, though. -M -- _ -- 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] Top Posting
On Mon, Jan 17, 2011 at 1:12 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Monday, January 17, 2011 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Sunday 16 January 2011 21:18:54 William Kenworthy wrote: Peoples email clients, work habits and environment mean that people to work the way thats comfortable to them. You want your mails read, you work with them, not get on a soap box and say YOU MUST BOTTOM POST. That was exactly my original point. If the list administrators are the experts, and they say to bottom post, then pissing off the experts is a way Clearly the answer is to compromise and start a new trend of middle posting. to ensure that you get the least help, when asking a question. Follow list etiquette to get the best possible answers. Eqiquette? Can most posters even spell that word, much less define it? Apologies to my fellow list members for opening this round of flame warfare. -- _ -- 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] OpenVPN + SIP configuration?
On Wed, Jan 12, 2011 at 12:08 PM, Gilles codecompl...@free.fr wrote: On Tue, 11 Jan 2011 10:02:48 -0500, Mark Deneen mden...@gmail.com wrote: Using the shared secret will only allow a single point to point connection. That is, you have to use certificates if you want more than one client. Thanks for the tip. I was under the impression that the shared key is just the equivalent of the hashed password in /etc/shadow. Also, when running openvpn --genkey --secret static.key, I wasn't prompted for the hostname or IP address of the client, so I don't understand why using a shared key would limit connections only from a specific client. Or do you mean that once a client is connected, no other client can connect using the shared key? Thank you. From http://www.openvpn.net/index.php/open-source/documentation/howto.html#quick : Static Key disadvantages * Limited scalability -- one client, one server * Lack of perfect forward secrecy -- key compromise results in total disclosure of previous sessions * Secret key must exist in plaintext form on each VPN peer * Secret key must be exchanged using a pre-existing secure channel I honestly do not know what happens if you attempt to connect another client. It's either going to reject that client or disconnect the active one. -M -- _ -- 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] OpenVPN + SIP configuration?
On Tue, Jan 11, 2011 at 9:29 AM, Andrew Latham lath...@gmail.com wrote: On Tue, Jan 11, 2011 at 11:20 AM, Gilles codecompl...@free.fr wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. By any chance, would someone have a working configuration so I can take a look? Thank you. Lazy way would be to use http://www.zentyal.org/ and point and click your way there... * Number one issue with Microsoft Windows clients on OpenVPN is getting the routing right. Verify that you have an end-to-end connection before trying to push any data through it. If you are running windows vista or windows 7 and start the connection with the OpenVPN GUI, you have to run it as administrator or it doesn't have the rights to add a route to the routing table. Don't be afraid of the certificate based method; it's really not hard! Using the shared secret will only allow a single point to point connection. That is, you have to use certificates if you want more than one client. -M -- _ -- 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] VoIP PoE phones for restaurant
On Tue, Jan 4, 2011 at 8:52 AM, Andy Graybeal andy.grayb...@casanueva.com wrote: On 01/03/2011 07:53 PM, cjwstudios wrote: Andy, The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the 2626 can support vlans you can isolate data and voice. Make sure to spec a UPS on the PoE switch. CJW, Awesome. Thanks for the input. For some reason or another I figured EOL wasn't such a bad thing as I could pick up the phones for cheap on ebay or something; but maybe this isn't the best of plans. The IP335 is on average about $10 more than the 501 or 320 new anyway. I thought that the 2610-24/12-PWR had the ability for VLAN as well? Not that it matters, it looks like I can get the 2626-PWR for under $600, and that fills out POE to all the ports. Is it possible that I can run one cable to the phone, then run a cable from the phone to a computer or another device and have those the phone and computer or other device be on separate networks? I'm sorry if this sounds newbish; I'm still learning. The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M -- _ -- 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] Converting asterisk h264 recordings
Torbjörn, I don't have experience with asterisk in this regard, but I would guess that what you have is an elementary stream and not a transport stream. I believe that VLC could play it back for you. Outside of that, I would look at using ffmpeg. It may do what you want. Here is some documentation on dealing with files like I suspect you have: http://processors.wiki.ti.com/index.php/Encoding_and_decoding_DVEVM_clips The link has absolutely nothing to do with Asterisk, but it may be useful. -M On Tue, Dec 14, 2010 at 9:53 AM, Torbjörn Abrahamsson torbjorn.abrahams...@gmail.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, December 14, 2010 3:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Converting asterisk h264 recordings We are setting up an asterisk system for voicemail with video possibilities. We are not using the voicemail app, but rather writing our own dialplan logic. The part of recording, and replaying, the voicemail works, and we receive both an h264 and an wav-file. What I now wonder is how to convert these into one file playable by a (standard) media player. I have not found any real good leads by google:ing, but of course I may have missed it… Any pointers? BR, Torbjörn Abrahamsson Perhaps this would work for you? http://www.linuxjournal.com/article/9005 Thanks, I will take closer look. Although, at a first glance I do not find anything about Asterisk in this article. I had the understanding that the video file which Asterisk saves is not a valid file, I read it’s the rtp packets stored in a file? I may be wrong… Anyway, I will look at the article… -- _ -- 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] TCP port, VPN and resolving the cutting voice problem
Any idea what is it about SIP over IAX2 that made such an improvement? -M On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- 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] Asterisk on smartphone?
But it has a built-in UPS! ;-) On Tue, Nov 30, 2010 at 10:02 AM, Kyle Kienapfel doctor.w...@gmail.com wrote: Sounds like they just need to be told its a hilariously bad idea to host anything important on a cellphone. On Mon, Nov 29, 2010 at 1:20 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Mon, 29 Nov 2010, Gilles wrote: Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? While I can run Asterisk on my Nokia N900, I have to say it's purely for my own benefit and to be a bit geeky... Personally people like this are cheakskates and not worth your time and effort in trying to implment a solution for them unless they're going to spend 1000's with you. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kernel: dahdi: Detected time shift.
On Wed, Nov 24, 2010 at 11:43 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, I'm experiencing a lot of server freezes lately. The server just... freezes. I notice in the log files (/var/log/asterisk/messages /var/log/messages) that logging stops at the time the server hangs. Logging continues when the server has been restarted (which is the only solution). So it is not a proces that hangs, it's the entire server (CentOS5.5 + Asterisk + MySQL). I really have no idea what can be causing these sudden freezes. Memory stays mostly at 250MB of 512 MB total, CPU is 97% to 100% idle... /var/log/asterisk/debug tells me nothing, no lines that indicate something strange before the freeze (debug level 9). I have no core.pid file in /tmp, when I look after rebooting the server. The only thing I have is a high level of mentionning of kernel: dahdi: Detected time shift. in /var/log/messages. What is causing this kernel message ? Could this be the cause of the server freeze ? Thank you for every feedback you can give me. Jonas, Do you have a monitor attached to the server? If the kernel is crashing, you might be able to catch the stack trace there. -M -- _ -- 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] res_musiconhold.c Bug - Patch to solve?
On Mon, Nov 22, 2010 at 9:50 AM, Danny Dias ing.diasda...@gmail.com wrote: 2010/11/22 John Novack jnov...@stromberg-carlson.org Danny Dias wrote: Hello John, What i am asking is if i can apply this patch manually or something like this without making any upgrade of Asterisk, has anyone done this before? I can't answer that question. ummm why not? is something wrong? Or i have to upgrade my Asterisk versions...i don't really want to do this... Why not? MANY fixes have been included in the upgrades. Improved security at the least. There are 10-15 versions between where you are operating and what is current I'm sure that the upgrade will fix this, but if applying the patch without making any upgrade will be better for me, my asterisk servers are working with many calls, realtime, fop etc...and an upgrade could make something happen... I would look at a svn diff between the two revisions and see how different they are. -M -- _ -- 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] FFA (Fax For Asterisk) tif file (size) problem
On Fri, Nov 19, 2010 at 9:42 AM, Michael voip.quest...@gmail.com wrote: Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error FAX handle 0: failed to queue document 'filename.tif', so we set it to 1680x2285, but it's still rejected. Is there a way to debug this further and fix it? We often have tif source files that we prefer to send, without converting to pdf and back to tif. Thank you in advance, Michael I don't know if this is the case or not, but check for differences between the two tiff files. I wonder if one is compressed and the other is not? -M -- _ -- 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] Recommended *WRT router to run Asterisk?
On Tue, Nov 16, 2010 at 9:28 AM, Gilles codecompl...@free.fr wrote: Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: (http://en.wikipedia.org/wiki/List_of_wireless_router_firmware_projects Thank you. Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Most of the open source firmware projects do not support DSL and rely on separate hardware to do the DSL, just like they rely on separate hardware for cable modem internet. -M -- _ -- 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] Recommended *WRT router to run Asterisk?
On Thu, Nov 18, 2010 at 8:52 AM, Gilles codecompl...@free.fr wrote: On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen mden...@gmail.com wrote: Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Sorry about that. Ideally, the unit should be both an ADSL modem + router, but apparently, most of them are just routers so that the user would have to turn their ADSL router into a modem/bridge and connect the *WRT-moded router. If someone's been running Asterisk on that kind of hardware for SOHO use, what would you recommend? Apparently, those are hardware that come up often in forums: http://wiki.openwrt.org/toh/d-link/dir-825 http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h http://wiki.openwrt.org/toh/asus/wl500gp http://wiki.openwrt.org/toh/asus/wl600g I use the Buffalo WZR-HP-G300NH with openwrt here. They are great for remote locations with a few sets -- you can have them hook up to the central server over an openvpn tunnel. I have ~20 sets hooked up to one with no issues at all. That being said, we probably only hit 7 or 8 concurrent calls at any point during the day. Even so, the resources on the router are not close to exhaustion. If you do try it, make sure that you do not have the iptables nat helper module installed. It's not helping you and causes problems when the router is the sip server and not hosting a sip client. -M -- _ -- 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] Recommended *WRT router to run Asterisk?
On Thu, Nov 18, 2010 at 9:26 AM, Darrick Hartman dhart...@djhsolutions.com wrote: On 11/18/2010 07:52 AM, Gilles wrote: On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneenmden...@gmail.com wrote: Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Sorry about that. Ideally, the unit should be both an ADSL modem + router, but apparently, most of them are just routers so that the user would have to turn their ADSL router into a modem/bridge and connect the *WRT-moded router. If someone's been running Asterisk on that kind of hardware for SOHO use, what would you recommend? Apparently, those are hardware that come up often in forums: http://wiki.openwrt.org/toh/d-link/dir-825 http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h http://wiki.openwrt.org/toh/asus/wl500gp http://wiki.openwrt.org/toh/asus/wl600g I never saw the point of spending $100 for something that is so limited. You can spend a little more and get something like an ALIX board that is so much more capable and still fanless/low power. http://www.pcengines.ch/alix.htm The 2d3/2d13 are very nice for the price. If you really want to run on a small router like this, the Netgear WNR3500 is a decent device and can be found for around $90. If you shop around / wait for deals, you can find the Buffalo for ~$70. I'm sure that the ALIX rigs are low power, just like most routers. However, most routers also come with a 4 port switch + plus one WAN interface. The ALIX boards get you at most 3 independent interfaces, but I don't believe that they can act as a switch. A 400 MHz MIPS is fairly close to a 500 MHz Geode. However, you can't get the asterisk g729 module for mips. I can't say I would want to transcode on the ALIX system, though. For a small setup or for a setup at home, it's really not a bad deal. Especially if you want something to do NAT for you. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP calls destroyed after 1:20
On Mon, Nov 15, 2010 at 3:11 PM, Jeremy Kister asterisk...@jeremykister.com wrote: After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? Play around with the session-timers in sip.conf. We had an issue with our sip provider, and this turned out to be a workaround. Their end was okay with supported session timers, but not session timers which were marked as required. -M -- _ -- 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 spontaneous reboot
On Sun, Nov 7, 2010 at 3:58 PM, Jonas Kellens jonas.kell...@telenet.be wrote: On 11/06/2010 09:18 PM, Sherwood McGowan wrote: On Sat, Nov 6, 2010 at 2:45 PM, Jonas Kellensjonas.kell...@telenet.be wrote: On 11/06/2010 07:18 PM, Tilghman Lesher wrote: On Saturday 06 November 2010 11:22:06 Jonas Kellens wrote: Hello, I just experienced a spontaneous reboot of Asterisk. This is my log file /var/log/messages : Nov 6 16:37:37 vps kernel: miniserv.pl invoked oom-killer: First line. Your miniserv.pl allocated more memory than is allocated to the system, so the dreaded OOM killer came into play and killed a selected process. Have you considered enabling swap memory? I have 512 MB real RAM and 1024 of swap. bash-3.2# cat /proc/meminfo MemTotal: 524288 kB MemFree: 23760 kB Buffers: 28564 kB Cached: 348668 kB SwapCached: 6536 kB Active: 193972 kB Inactive: 231216 kB HighTotal: 0 kB HighFree: 0 kB LowTotal: 524288 kB LowFree: 23760 kB SwapTotal: 1048568 kB SwapFree: 949456 kB Dirty: 768 kB Writeback: 0 kB AnonPages: 46652 kB Mapped: 16884 kB Slab: 21000 kB PageTables: 8084 kB NFS_Unstable: 0 kB Bounce: 0 kB CommitLimit: 1310712 kB Committed_AS: 321288 kB VmallocTotal: 34359738367 kB VmallocUsed: 784 kB VmallocChunk: 34359737535 kB miniserv.pl... I have webmin running yes and it was stopped after the restart of Asterisk... So the bad one in this story is WebMin that was eating up all the memory ? Jonas. -- _ -- 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 Yessir, that's the culprit in this case Strange, today I saw this in the logs : Nov 7 17:02:18 vps kernel: crond invoked oom-killer: gfp_mask=0x201d2, order=0, oomkilladj=0 Nov 7 17:02:18 vps kernel: Nov 7 17:02:18 vps kernel: Call Trace: Nov 7 17:02:18 vps kernel: [802bf74e] out_of_memory+0x8b/0x203 Nov 7 17:02:18 vps kernel: [8020f947] __alloc_pages+0x27f/0x308 Nov 7 17:02:18 vps kernel: [802138db] __do_page_cache_readahead+0xc6/0x1ab Nov 7 17:02:18 vps kernel: [802141c7] filemap_nopage+0x14c/0x360 Nov 7 17:02:18 vps kernel: [80208e8c] __handle_mm_fault+0x442/0x1445 Nov 7 17:02:18 vps kernel: [8028866d] deactivate_task+0x28/0x5f Nov 7 17:02:18 vps kernel: [8026769a] do_page_fault+0xf7b/0x12e0 Nov 7 17:02:18 vps kernel: [8025c8ff] hrtimer_cancel+0xc/0x16 Nov 7 17:02:18 vps kernel: [80263b14] do_nanosleep+0x47/0x70 Nov 7 17:02:18 vps kernel: [8025c7ec] hrtimer_nanosleep+0x58/0x118 Nov 7 17:02:18 vps kernel: [8026082b] error_exit+0x0/0x6e Nov 7 17:02:18 vps kernel: Nov 7 17:02:18 vps kernel: Mem-info: snip So this time it is crond that invoked oom-killer... Please read up on how the oom killer works. crond didn't invoke anything, but was rather the unfortunate task chosen to be sacrificed. -- _ -- 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] Short rings for extensions when part of the Queue
On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote: Chad, You are absolutely right on this one. I had setup the Queue time out for agent set to 15 seconds and retry to 2 seconds. So, I think during those two seconds Asterisk for some crazy reason hits another extension and then comes back to the same extension to ring again. So, I have setup the agents to ring for ever for this call center since their agents always have to present or logout if not present. I will see the behavior tomorrow as they test it. My issue might be solved but for those call centers where you want the Queue to move onto the next agent or if you don't want to ring for ever and take a Retry break then it will still remain an issue. I will report back if setting to ring Unlimited doesn't work. Warren, The CLI shows the regular stuff. Nothing out of the ordinary but that it move on to the next agent because the first agent has timed-out for two seconds. Regards, Bruce Have you considered setting the queue timeout to 14 or 16 seconds and retry to 2 seconds? This way the timeout and the retry should line up better. -- _ -- 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] Short rings for extensions when part of the Queue
On Fri, Nov 5, 2010 at 10:38 AM, Bruce B bruceb...@gmail.com wrote: Yeah, I think I had it set to 2 seconds and that creates that short ring on another extension. Thanks, The point was that 14 and 16 are divisible by 2 (evenly) while 15 is not. -- _ -- 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] Funky IAX behavior between 1.4 and 1.8
On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas da...@debsinc.com wrote: Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to Machine 3, I get lags/pauses on Background/Playback commands. I play files and groups of files that last from 1-45 seconds, so I can press keys and proceed, but I don’t expect my end-users to know to do this. Any clues? Do I need to open a tracker issue on this one? Thanks Danny Nicholas https://issues.asterisk.org/view.php?id=18105 From my testing, I have seen it happen in both inside a VM and outside a VM... but it happens far more reliably in a VM. I added some debug code to chan_iax2.c, but I haven't been able to find anything useful yet besides a better idea of how IAX2 works. :) -- _ -- 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] Under heavy attack
On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslak jmas...@antelope.net wrote: If these are mobile users, I hope they never use any public networks (hotels, starbucks) where other subscribers can do things like ARP attacks to do MITM (and steal your calls; it might not be happening today, but it will be happening soon - as the social networking attacks demonstrate). If you do have truly roaming users, I hope you use HTTPS (with validation of certs turned on) or a VPN (likely not an option of connecting to an ADSL site, due to bandwidth concerns). Can you explain why VPN is not an option for ADSL? (Open)VPN overhead is not that high. ~70 bytes per packet if I remember correctly. -M -- _ -- 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] Auto provisioning from public server
On Tue, Oct 26, 2010 at 12:06 PM, Jonas Kellens jonas.kell...@telenet.be wrote: On 10/26/2010 05:52 PM, bakko wrote: Hello, many SIP phones offer you the possibility to provisioning them over a FTP connection (with username and password). Regards - Bakko In this case I will want to use Snom phones. TFTP is available, but no FTP (with indeed then a username and password). FTP would be great... I wouldn't do this unless your connection is encrypted. -- _ -- 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] OpenVPN over TCP 1194 rather than UDP 1194 - Is there an adverse effect when running Asterisk?
On Fri, Oct 22, 2010 at 10:02 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, For some reason a few phones connected to a pfSense box can't make or allow in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP and it works fine. I would like to know if there is any disadvantages to using TCP over UDP for the tunnel when using Asterisk or is just as reliable and solid as a UDP tunnel? Thanks Don't do it. Here is a possibly evil suggestion, though -- almost nobody blocks port 53. -M -- _ -- 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] Remote Unix Connection
I took a look in the source -- it is definitely asterisk -r (or rasterisk) and not AMI. AMI logs something like Manager 'username' logged on from 127.0.0.1. Check the timing between calls and see if a pattern appears. If so, it is some sort of cron/scheduled job. If not, keep looking! -M -- _ -- 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] Audiocodes firmware
On Thu, Oct 14, 2010 at 11:46 PM, Mark Murawski markm-li...@intellasoft.net wrote: Crazy. What do you plan on using for an ATA now? The problems I'm having are getting 500 Server Internal Error on just about every other call placed out of this mp-118. The box has been installed and in use for quite some time and recently started having problems. Reboots, etc don't make a difference. I noticed it had newer firmware than what I had on some other boxes that had no issues whatsoever. I do have a 5.80 firmware I had downloaded a while back and put that on. Now the internal server errors are happening on 70-80% of sip-pstn calls. pstn-sip calls seem to be coming in just fine. Ever since I did the firmware downgrade, now my ssh sessions to the box get disconnected after about 30 seconds with invalid packet errors. I've had problems with earlier firmware as well... once the 5.x firmware started shipping on audiocodes it seemed they were just about DOA. The web interface worked but nothing else worked right. Perfectly working configurations on other boxes that were copied to the new boxes with new firmware would just fail in various ways... disconnect supervision not working, internal routing not working. Finally I managed to get a hold of the 5.80 firmware which got rid of all those problems. Now I'm stuck again. I have a box in service that's having problems and I can't get new firmware. This doesn't sound like something which firmware would fix, especially if it had been running fine before. It sounds like either the mp-118 is failing or you have a damaged cable. -M -- _ -- 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] Remote Unix Connection
On Sat, Oct 16, 2010 at 5:36 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Serious answer: Looks like a process running asterisk -r. Do you have any sort of AGI, cron job or perhaps a nagios check which does this? Not so serious answer: IT IS COMING FROM INSIDE OF THE HOUSE -M -- _ -- 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] drop dead fix
On Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote: The original one is super quiet - obviously not Allison in a studio... Listen to the gsm in Asterisk to see my quandary... What is the end use here? Who will be listening to the recordings? Users on PSTN and mobile phones? -- _ -- 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] drop dead fix
On Fri, Oct 15, 2010 at 11:20 AM, Danny Nicholas da...@debsinc.com wrote: End use is Telephone Banking, so you've nailed the target audience. BTW, the highpass and lowpass filters seem to help, but since I stopped math at pre-calculus, the explanation of the Butterworth filter is beyond my pay grade. Care to offer a better explanation? While, officially, I completed up to calculus 3, the serious lack of use is not helping. You'd be better off taking the highpass number from low to high and listen to the difference, and then do the same for the lowpass number. Your ears will tell you when you have it right (you will definitely hear what each one does), and you can still consider Butterworth inexpensive pancake syrup. -M -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
2010/10/15 Matt Darnell mattdarn...@gmail.com: On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif, Isn't callcounter for 1.6 and not for 1.4? If you are using the Local channel, look into the n option. -M -- _ -- 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] sound file debug
On Tue, Oct 12, 2010 at 4:23 PM, Danny Nicholas da...@debsinc.com wrote: dollars.gsm: data dollars.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Can't be 100% certain on #2, but it must have been right because it works now. Go figure. Isn't WAV wav49 and wav plain old pcm (with the wav header)? -M -- _ -- 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] E1 check with nagios, how to?
Are you monitoring some dahdi hardware or a separate black box? If dahdi, you could write a nagios plugin in shell with something like this: ALARMS=`dahdi_scan | grep alarms | grep -v OK | wc -l` and then set the appropriate exit code if ALARMS is not 0. -M On Tue, Sep 28, 2010 at 9:22 AM, Dario Quiroz darioqui...@gmail.com wrote: We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, --- Dario Quiroz (71) 9275-9080 gtalk: darioqui...@gmail.com --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins d...@cognation.net wrote: Any thoughts on why the lack of traffic? Cheers, Dean Not enough applications to play immature bathroom sounds. Just a guess. -M -- _ -- 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] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens jonas.kell...@telenet.be wrote: warning: exec file is newer than core file. Jonas, I encourage you to read the output. Did you run gdb with a core file dumped from the old build? You need to generate a new core dump with the new executable. Best Regards, Mark Deneen -- _ -- 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] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On Fri, Sep 17, 2010 at 11:51 AM, Jonas Kellens jonas.kell...@telenet.be wrote: On 09/17/2010 05:29 PM, Mark Deneen wrote: On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellensjonas.kell...@telenet.be wrote: warning: exec file is newer than core file. Jonas, I encourage you to read the output. Did you run gdb with a core file dumped from the old build? You need to generate a new core dump with the new executable. Best Regards, Mark Deneen 1. I have re-compiled asterisk 1.6.2 with dont_optimize 2. I have generated the restart/reload of asterisk that I expercience 3. I have built a backtrace with gdb from the resulting core.pid file Don't know what I could have been doing wrong... Jonas. Jonas, What is the timestamp on your asterisk binary and what is the timestamp of the core file? Also, you restarted asterisk after installing the dont_optimize binary? The message suggests that your core file is older than your executable, which should not be possible. Best Regards, Mark Deneen -- _ -- 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] Polycom dhcp boot
On Sun, Sep 12, 2010 at 8:57 AM, colin mcdermott colinjamesmcderm...@gmail.com wrote: Use lowercase for ftp:// . That might be the issue but it should be easy to test. Do your FTP server logs shpw anything? I will double check but I believe that lower case ftp is being used. I do have upper case PlcmSpIp:PlcmSpip as the password. I will see if lower case usernames and passwords work better (although I did try to create user plcmspip with pw plcmspip.) The Sonicwall DHCP logs are not that useful (DHCP granted to client xxx). But maybe I am logging this incorrectly. Thanks for the advice I will post feedback tomorrow. Colin, I believe he meant use ftp://; instead of FTP:// I have a working tftp for some polycom phones I can check for you on Monday, if you are still experiencing problems. -M -- _ -- 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 stop intruder from registering sip?
On Thu, Jul 1, 2010 at 12:53 PM, Tilghman Lesher tles...@digium.com wrote: That would only be true if you used random characters in your 17-character passphrase. In fact, English text has somewhere between 0.6 and 1.5 bits of randomness per letter, whereas an SHA1sum has no more than 4 bits of randomness per letter. Let's assume the higher number of randomness for your English text, which gives us 1.5 * 17, which is 25.5 bits of randomness. Note that the prefix 3 characters have ZERO randomness per character, as they are deterministic from the extension. That gives an even less 21 bits of randomness. SHA1 cryptographic sums have no more than 160 bits of randomness. I say no more than, because, given knowledge of the algorithm used to determine passwords, the sum is reduced to the number of bits of randomness in the source material. You cannot generate randomness by applying a deterministic algorithm. However, given that the source material for the hash sum is of a smaller bit strength than the comparative strength of the hash algorithm, your difficulty of guessing the password is not reduced any by using the hash algorithm for generative purposes. With this in mind, I'll be sure to forge my passwords from Chinese text from now on. -- _ -- 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 1.6 (and 1.4) DTMF problems using RFC2833
On Thu, Jul 1, 2010 at 7:09 PM, Miguel Molina mmol...@millenium.com.cowrote: I've experienced a similar DTMF issue with recent asterisk 1.4 versions (1.4.32, 1.4.33.1), I'm not sure about 1.6.2.X. What happens here is that the DMTF activated features, like disconnect (default *) or blind transfer (default #) stops working after a while. Agents are able to transfer or hangup a few calls and then it stops working. Doing some debugging I could see that asterisk knows (receives) the DMTF too but the features are not triggered... Anyone else has run into this? Miguel, I've tracked it down to a problem with some recent code which was added to detect DTMF RTP frames coming in out of sequence. https://issues.asterisk.org/view.php?id=17571nbn=5 Mark -- _ -- 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 1.6 (and 1.4) DTMF problems using RFC2833
We are experiencing intermittent DTMF problems here, with the following setup: ITSP - PIX - Asterisk (g729, RFC2833 for DTMF). I am running Ubuntu server 10.04, but Asterisk is compiled by us and not installed from the software repository. Essentially, DTMF works for some time, but at some point it simply stops and the point at which it stops appears to be random. Using RTP debug, I can verify that the RFC2833 DTMF is being delivered in the RTP stream, and Asterisk knows of it. Independently, wireshark confirms the same. I can't easily remove the PIX, but as the RTP is showing the DTMF I do not believe the firewall is an issue. Our ITSP is registered as a SIP provider, and we can receive calls just fine. I've attached a file containing portions of the asterisk log, the wireshark log and the dialplan. Has anyone else run into this situation? Best Regards, Mark Deneen [Jun 29 15:31:44] DTMF[26287] channel.c: DTMF begin '*' received on SIP/10.200.10.5-001e [Jun 29 15:31:44] DTMF[26287] channel.c: DTMF begin ignored '*' on SIP/10.200.10.5-001e [Jun 29 15:31:44] DTMF[26287] channel.c: DTMF end '*' received on SIP/10.200.10.5-001e, duration 100 ms [Jun 29 15:31:44] DTMF[26287] channel.c: DTMF end passthrough '*' on SIP/10.200.10.5-001e [Jun 29 15:32:18] VERBOSE[26287] pbx.c: -- Timeout on SIP/10.200.10.5-001e, continuing... (enable RTP debug) Jun 29 15:37:30] VERBOSE[26287] rtp.c: Got RTP RFC2833 from 10.200.10.5:20020 (type 101, seq 017379, ts 5727107, len 04, mark 1, event 000b, end 0, duration 0) [Jun 29 15:37:30] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4) [Jun 29 15:37:30] VERBOSE[26287] rtp.c: Sent RTP packet to 10.200.10.5:20020 (type 18, seq 050557, ts 5669136, len 20) [Jun 29 15:37:30] VERBOSE[26287] rtp.c: Got RTP packet from10.200.10.5:20020 (type 101, seq 017380, ts 5727107, len 04) [Jun 29 15:37:30] VERBOSE[26287] rtp.c: Got RTP RFC2833 from 10.200.10.5:20020 (type 101, seq 017380, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00160) [Jun 29 15:37:30] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to 10.200.10.5:20020 (type 18, seq 050558, ts 5669296, len 20) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP packet from10.200.10.5:20020 (type 101, seq 017381, ts 5727107, len 04) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP RFC2833 from 10.200.10.5:20020 (type 101, seq 017381, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00320) [Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to 10.200.10.5:20020 (type 18, seq 050559, ts 5669456, len 20) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP packet from10.200.10.5:20020 (type 101, seq 017382, ts 5727107, len 04) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP RFC2833 from 10.200.10.5:20020 (type 101, seq 017382, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00480) [Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to 10.200.10.5:20020 (type 18, seq 050560, ts 5669616, len 20) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP packet from10.200.10.5:20020 (type 101, seq 017383, ts 5727107, len 04) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP RFC2833 from 10.200.10.5:20020 (type 101, seq 017383, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00640) [Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to 10.200.10.5:20020 (type 18, seq 050561, ts 5669776, len 20) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP packet from10.200.10.5:20020 (type 101, seq 017384, ts 5727107, len 04) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP RFC2833 from 10.200.10.5:20020 (type 101, seq 017384, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00800) [Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to 10.200.10.5:20020 (type 18, seq 050562, ts 5669936, len 20) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP packet from10.200.10.5:20020 (type 101, seq 017385, ts 5727107, len 04) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP RFC2833 from 10.200.10.5:20020 (type 101, seq 017385, ts 5727107, len 04, mark 0, event 000b, end 0, duration 00960) [Jun 29 15:37:31] DEBUG[26287] rtp.c: - RTP 2833 Event: 000b (len = 4) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Sent RTP packet to 10.200.10.5:20020 (type 18, seq 050563, ts 5670096, len 20) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got RTP packet from10.200.10.5:20020 (type 101, seq 017386, ts 5727107, len 04) [Jun 29 15:37:31] VERBOSE[26287] rtp.c: Got
