Re: [asterisk-users] Ongoing attack from 188.138.100.16
You could traceroute the IP and contact NOC's along the route. They might be interested to hear of flooding/DOS attacks being routed via their equipment. On Tue, Mar 6, 2012 at 4:58 PM, Mike Diehl mdi...@diehlnet.com wrote: route add -host 188.138.100.16 dev lo Good bye. But it shouldn't come to this. On Tuesday 06 March 2012 5:48:26 pm Matt Desbiens wrote: iptables -A INPUT --src 188.138.100.16 -j DROP On Mar 6, 2012 7:29 PM, Mike Diehl mdi...@diehlnet.com wrote: I've been logging sip registrations from this IP address for 2 days now. I've emailed the domain's admin, but nothing seems to come of it. I've routed him into oblivion, but still, I think 50 requests a second for 2 days is a bit much. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- 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 -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA with TCP/TLS support?
Hi List, Has anyone heard of an ATA device that supports TCP TLS? Not much comes up in searching, thought to check here for some device suggestions. TIA, Skyler -- _ -- 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] random digits dialing during call
Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler -- _ -- 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] random digits dialing during call
Hi, thanks for your reply. We're using PAP2T's. I've just recently found that this is most likely 'talk-off', a common issue with PAP2's. A new term for me, but once I found that it was much easier knowing what to search for. I found a few suggestions on changing dtmf to 'inband' so I've done that now and see how it goes. If it continues we'll just have to switch out the devices for another brand...maybe ht286's? couldn't find any talk-off related issues for those so that might be a good next option. S. On Thu, 2011-12-08 at 10:49 -0500, eherr wrote: What are you using for hardware? I have experienced SPA2102s supplying a DTMF tone when someone was talking. This was caused by the talker reaching a certain frequency while talking in which the SPA popped out a DTMF tone. I haven't experienced this behavior on polycoms or anything else. --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Thursday, December 08, 2011 9:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] random digits dialing during call Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
Steve, On Thu, 2011-08-25 at 00:39 -0400, Steve Totaro wrote: ... For fax, I use Hylafax and for text, I use Kannel. These are WAY more powerful than Asterisk apps. With Kannel, I used the Bluetooth GSM modem to send SMS from my cell. Kannel is awesome as is HylaFAX I used the Asteirsk System() app to call lynx with a special URL. The URL contains all the authentication, recipient, and SMS body. Calling that URL via System(), as I said, I like lynx, causes an SMS to be sent. Kannel is extremely customizable. I once had ten cell phones for for SMS modems. My findings with t-mobile were that each phone could send an SMS once a second. With ten, using chan_bluetooth, I could send ten SMS per second using ten phones. Kannel is very well developed. Chan_mobile is incredible. The same is true with HylaFAX. Thanks, Steve T I'm looking at using Kannel for a project here. Would you mind if I contacted you off list with some getting started questions? Skyler -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Hi, On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Not 100% with 1.4 but with 1.6 you don't need to set tcpbindaddr because tcpenable will listen on same IP as udp. No transport either I believe. If you want, set udpbindaddr and tcp will listen on this IP too. tcpenable=yes is all you should need. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bind SIP over TCP port in asterisk 1.4.42.
I have the below in my sip.conf bindaddr = PublicIP tcpenable = yes tcpbindaddr = PublicIP On Thu, 2011-08-25 at 17:22 +0300, Catalin S. wrote: hello, I tried still not working. :( something is wrong. On Thu, Aug 25, 2011 at 4:37 PM, Skyler skchopper...@gmail.com wrote: Hi, On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Not 100% with 1.4 but with 1.6 you don't need to set tcpbindaddr because tcpenable will listen on same IP as udp. No transport either I believe. If you want, set udpbindaddr and tcp will listen on this IP too. tcpenable=yes is all you should need. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On Wed, 2011-08-24 at 08:49 -0600, Linuxguy123 wrote: OK. I'm a 54G guy. I just bought a E4200 the other day for our media network. Nice. The E4200 is what I wanted but it wasn't in stock anywhere when I was on the hunt. I just recently bought an Asus RT-N16 ... my boss is hooked now so I will probably be working on this at work and do this one next ;) Are you using a POTS connection or SIP provider for your phone system ? SIP only. :drool: So you can receive faxes that arrive at home on the road then, as an email attachment, right ? Without having to find a fax machine while traveling and coordinating with the sender ? Yep, arrives as a pdf in my email. If we wanted faxes received on the fax machine, can asterisk recognize a fax tone and route the call to the fax machine ? IIR asterisk can detect fax with nvfax or something like that, I didn't bother and went with a dedicated voxnumber. Will the fax machine send via an analog connection to the asterisk system ? Or does it need its own line directly out ? Fax is receive only. I don't have a physical fax machine, I scan/email. You could easily use a PAP2 or HT286 for the fax machine and register it to the router. What information resources did you use when setting up your system ? Google when I was stuck. I didn't document the sites as I actually couldn't make this work using DD-WRT or Tomato without having to install everything on the USB key. I didn't like this so kept going as I'm familiar with cross-compiling, linux etc. and ended up with my own firmware. You can go with the DD-WRT or Tomato + asterisk-on-usb setup if you like, it does work but not what I wanted. I can tell you to focus on compiling the firmware, getting that to work 100% re-flashing and you're good to go. Then work on the asterisk install. The Asterisk install is minimal (ie: only loads what modules are needed and no more) all extra files etc are removed for space. There's no gui, so I hope you are ok with SSH for config changes. The callback/call-through/fax2email are agi's, those took the longest to get working as I knew nothing about this. Thanks again for the replies. LG -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
I have my wireless router working as a proxy/asterisk system. Its not 100% done yet, config related stuff still lingering, works not so bad so far. I register voip phones or ata's locally and SIP trunk for my Voxnumber(s), also for inbound/outbound. It does callback call-through for mobile, also SIP soft phone ability with local wifi or remote GPRS sip connection for mobile. voicemail, voicemail-to-email, fax2email are working but needs more testing. Of course its still my wifi-N router for devices and PC's as well. is this the kind of thing you were thinking of? On Tue, 2011-08-23 at 17:04 -0600, Linuxguy123 wrote: My original post didn't mention it, but I would like my home system to be Asterisk based. Has anyone figured out how to minimize cell charges when on the road via making calls via the home phone system ? Does anyone have their cell phone forwarded to their home phone system and have it do their messaging ? Is anyone using Google Phone capabilities in conjunction with Asterisk ? Thanks ! On Mon, 2011-08-22 at 14:11 -0600, Linuxguy123 wrote: I'm looking for ideas for building a innovative, powerful home phone system. Something that does voicemail well, integrates cell phones into the house system, etc. I know there are a lot of details that need to be discussed, but lets leave it at that for now. What is everyone doing ? Thanks ! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Hi, On Tue, 2011-08-23 at 21:50 -0600, Linuxguy123 wrote: So you have asterisk loaded on a wireless router ? Linksys 54G by chance ? Yes, Asterisk at the moment. Cisco E3000. 54G is too small for asterisk, not enough flash/cpu. Which VOIP phones are you using ? Which ATA are you using ? I have Aastra 6731i, PAP2T, HT286 a Polycom and an Snom unit. Linphone, Bria, jitsi work as well for PC/Mac/iPhone. Any voip device/software would work. The wife uses call-through on her Blackberry with MY10, she adds contacts with a pause after her voxnumber; like 1NPANXX,personsnumber so it dials in then dials out on the trunk. We have unlimited 60 countries so we can literally call anywhere, from anywhere and never have to think about it. Took me 6 months here-and-there to get it this far. Well worth it though as we save about $180/month in cell phone bills now between us. How big is the system ? (number of lines, users, etc.) Just family and tinkering. I had load tested it with SIPp simulating 10 concurrent calls, sat at a steady 93% cpu. I'd say the E3000 would suffice for home use, 2-3 concurrent users. We stream off the NAS through it also and don't even notice during a call. How does a wireless router handle voicemail ? Ie no hard drive, so where does it store it ? NAS ? It records to memory (flash) and sends a wav to email. Fax works the same way. Storage can be on a USB key, it works but I don't use it that way. Thanks for the reply, I really appreciate it. NP. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??
sip.conf useragent = myasteriskbox sdpsession = myasteriskbox On Wed, 2011-07-20 at 12:45 +0300, Israel Gottlieb wrote: user-agent could be set in sip.conf On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov abalas...@evaristesys.com wrote: On 07/20/2011 05:00 AM, Masood Ahmed wrote: Hello All, Is there any one who can help me to change the From field parameters in option packets, I have seen that in option packtes asterisk sends its own information,If you see the below option packet i have highlighted the asterisk word in from field and in from field tag how can i changed it Please let me know same as in User Agent. These are internally generated, so there is no way to modify them without a source-level change. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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?
Hi, On Tue, 2011-07-19 at 16:14 +0200, Gilles wrote: On Mon, 18 Jul 2011 20:59:02 +0300, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: /usr/lib/asterisk/modules/ Be sure to only include the ones you need. Finding which exactly may be tricky. Thanks Tzafrir. Actually, since the modules are the biggest files by far, besides the obvious (SIP, Dahdi, etc.), how to investigate which modules I must keep? Does Asterisk report errors explicitely when a module it needs is missing, or does it just crash/malfunction without reporting anything? I found this to be helpful: http://www.wains.be/index.php/2008/04/15/slimming-asterisk-for-the-nslu2-under-debian/ also setting full = notice,warning,error,debug,verbose in the logger.conf was helpful to locate dependency errors. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time zone on phones
Hi, On Tue, 2011-07-19 at 14:30 -0500, Warren Selby wrote: On Tue, Jul 19, 2011 at 11:53 AM, motty.cruz motty.c...@gmail.com wrote: Hello All, I have asterisk server running on Centos, some of our users are spreadout throut the states. I want the time zone to reflect our users repective time zones. My questions is how to customize their timez zone accordingly? Is that done in sip.conf? or extensions.conf? Usually this is handled on the phone itself or within one of it's configuration files. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com Do you mean TZ on the devices, TZ on voicemail or ?? In general, usually voicemail TZ is configured in voicemail.conf and handled in realtime DB per user. Devices really depend on how you deploy them. Set GMT for the configs. If you're setup is not realtime, then I'd look to voip-info for examples on extensions.conf S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pops clicks at the end of sound files
Hi, I had a similar issue converting wav files one time. Ended up using sox to convert to .sln as that ended up being the sounding conversion. I used the below command on a directory of files to convert: for a in *.wav; do sox $a -t raw -r 8000 -s -w -c 1 `echo $a|sed s/.wav/.sln/` resample -ql; done S. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: Monday, June 06, 2011 7:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Pops clicks at the end of sound files Hi all, We recently decided to get a professionally recorded set of prompts for our asterisk based IVRs and received these as the following: Bit Rate: 1536Kbps Sample Size: 16bit Channels: Stereo Sample Rate: 48kHz Format: PCM I use Wavepad to convert it to: Bit Rate:64Kbps Sample Size: 8bit Channels: Mono Sample Rate: 48kHZ Format: CCIT-ALAW I copied these files to an asterisk server and then used asterisk -rx to convert the files to g729. The problem I have is that at the end of every file there is a pop / distortion after playback. Anyone have the same issue before? -- _ -- 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] Free CNAM
Hi all, Let's get some feedback going here and see if there is any general support in a user-driven CNAM concept. Assuming that your landline/mobile outbound provider does not push caller-name + number for you with your calling plan. Would you pay $1/yr to have the access to update your own personal CNAM info in a database that you can trust to be correct? One that 1000's or even 100,000's of other voip/pbx owners will use? S. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, June 02, 2011 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free CNAM I just checked several of my numbers and several others known to me, it really isn't much better 2 of them returned names other than mine, and all had the wrong city, though at least the state was correct. All but one also had the wrong carrier. I fear these databases are are so full of errors that they are mostly worthless. John Novack Pascal Bruno wrote: If you can use curl, and can do some text parsing and know regular expressions, you may be able to use this free CNAM service: http://www.numberguru.com/ and integrate into your system. This one appears to have a more complete database. When I tried my number, I have gotten my full name, but when I use the FreeCNAM project below, I just get Florida. On Wed, Jun 1, 2011 at 8:11 AM, Michael R. Wally michael.r.wa...@gmail.com wrote: I've been toying around with the idea of starting some kind of 'Open CNAM' project to destroy the current money hustle BS that dominates this industry. The ever-growing FreeCNAM database may be a good starting point for such a project. I would also like to use Bitcoin (BTC) as the micropayment solution for user-requested updates. Some nominal fee. If anyone wants to get involved, contact me. On 06/01/2011 07:51 AM, Skyler wrote: Hi, The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the carrier's way to isolate their users and another excuse to charge more money for 'the better plan'. In the end, it's the carrier that inputs the info so if it shows FLORIDA with one database I can't see how any other database would be different as the carrier is the only one that controls the outbound CID info. Calling me from POTS to snatch the CID will result in the same. ...unless there were a user friendly CNAM service, where info could be updated by the end-user and queried freely by voip providers. I would update my cellular numbers for sure and know at least a dozen people that would do the same. Everyone is going VoIP so why not? Talking about 'where's the money or angle'... here is one, vanity. Charge $1/yr to a user per DID, if I don't renew then delete it and re-query the original carrier. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pascal B. Personal Web Site http://www.pascalbruno.com/ Twitter: @petchaw -- _ -- 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 -- Dog is my Co-pilot _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1375 / Virus Database: 1511/3675 - Release Date: 06/02/11 -- _ -- 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] Free CNAM
Hi, The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the carrier's way to isolate their users and another excuse to charge more money for 'the better plan'. In the end, it's the carrier that inputs the info so if it shows FLORIDA with one database I can't see how any other database would be different as the carrier is the only one that controls the outbound CID info. Calling me from POTS to snatch the CID will result in the same. ...unless there were a user friendly CNAM service, where info could be updated by the end-user and queried freely by voip providers. I would update my cellular numbers for sure and know at least a dozen people that would do the same. Everyone is going VoIP so why not? Talking about 'where's the money or angle'... here is one, vanity. Charge $1/yr to a user per DID, if I don't renew then delete it and re-query the original carrier. S. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R. Wally Sent: Sunday, May 29, 2011 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free CNAM The system uses real Telco CNAM Dips. Any generic names you get are from the subscriber's carrier itself. We can only provide what we ourselves get. I tried it, but it returns the same kind of junk that some of the databases do. For example, on a Florida number, it just says FLORIDA instead of the proper name (some of the CNAM databases have the right name). -- _ -- 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 - No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1375 / Virus Database: 1509/3666 - Release Date: 05/28/11 -- _ -- 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] Realtime dbase table mods
Hi all, Anyone know if it's possible to force asterisk to use 'my_table' for passwords instead of the 'secret' table? S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk's zombie processes
Hi,
You can run the script below as an hourly cron. Works for me.
#!/bin/sh
# clean-up Asterisk zombies
# file clean_up.sh
# $Id: clean_up all dead parent processes
# use as cron task */30 * * * * root /usr/local/sbin/clean_up.sh
#
##LOG=/var/log/asterisk/agi-cleanup.log
date=`date +%d-%m-%Y_%Hh%Mm`
t1=`cat /proc/stat | grep btime | awk '{print $2}'`
t3=`date +%s`
echo $date - Asterisk Zombies Clean up started. $LOG
echo
for parent in `ps -ef | grep safe_asterisk | awk '$3 == '1'{print $2}'`
do
for ppid in `ps -ef | awk '$3 == '${parent}' { print $2 }'`
do
for i in `ps -ef | awk '$3 == '${ppid}' { print $2 }'`
do
t2=`cat /proc/$i/stat| awk '{print $22}'`
b=$(($t3-$t1));
c=$(($t2/100));
d=$((($b-$c)/60));
if [ $d -gt 30 ] ; then
kill -9 $i
echo Zombie found - killing $i ### $LOG
fi
done
done
done
exit
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, May 18, 2011 1:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk's zombie processes
On Wed, 18 May 2011, vip killa wrote:
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of
too many zombie processes. I eventually had to disable the notification
for the alert but why does Asterisk create so many zombie processes,
I've see more than 30 at times and it generally stays in the 20s... just
seems unusual and wondering if it's harmful, thanks in advance.
Asterisk creates threads, not processes.
Trace back from the PPID of the zombies to see who created them
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Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
_
No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1325 / Virus Database: 1509/3645 - Release Date: 05/18/11
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[asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription
Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It's more of an annoyance, I know but I like to keep my pcap's clean. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription
Really!? Wow, that would be so easy as it looks like qualify=yes is already enabled on each SIP channel. I'll give that a try/test first and report back. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, May 13, 2011 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Friday, May 13, 2011 2:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It's more of an annoyance, I know but I like to keep my pcap's clean. You should be able to turn NAT Keepalive off on the PAP2s. If you need a NAT Keepalive type of service, use the qualify=yes for those peers in Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1325 / Virus Database: 1500/3635 - Release Date: 05/13/11 -- _ -- 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] concurrent call tracking
Many thanks to all that replied. I'm going to test out the suggestions/scenarios and I'll post back with what worked for me. S. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: Thursday, May 12, 2011 6:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] concurrent call tracking On 11-05-11 06:36 PM, Skyler wrote: Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd like to take a look at it for sure. The dial plan example Leif replied with is pretty much what I was thinking, just didn't have a clue how to go about it. ;) You could also look into using LOCK() and UNLOCK() dialplan applications to make sure each insert happens sequentially. Leif. -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1325 / Virus Database: 1500/3632 - Release Date: 05/11/11 -- _ -- 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] concurrent call tracking
Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking into implementing something like this? TIA, Skyler -- _ -- 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] concurrent call tracking
Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd like to take a look at it for sure. The dial plan example Leif replied with is pretty much what I was thinking, just didn't have a clue how to go about it. ;) Haven't figured out how I'm going to display the usage info either so if you don't mind sharing the graph/code as well that would be sweet. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Wednesday, May 11, 2011 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] concurrent call tracking What I do is when ever a call comes in I update a table in MySQL to active = (active +1). On hang up I do active = (active -1). I have a cron that checks once a minute to see how many active and stores it along with epoch in db. I then have a graph that shows channel usage. If you want the code let me know. - Original Message - From: Skyler mailto:skchopper...@gmail.com To: asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2011 19:57 Subject: [asterisk-users] concurrent call tracking Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking into implementing something like this? TIA, Skyler _ -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1325 / Virus Database: 1500/3630 - Release Date: 05/11/11 -- _ -- 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 Avaya SIP Trunking One Way Audio
First, I'm pretty sure avaya peer needs to friend. Try adding the below to sip.conf and do a reload. [general] externip = the.wan.ext.ip localnet = 192.168.1.0/255.255.255.0 If that doesn't work, add nat=yes to avaya peer=friend Skyler From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Thursday, April 07, 2011 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio On 04/07/11 03:00, Shariq Khan wrote: I am facing one way audio problem in sip trunking between asterisk and avaya. +-+ ++ | avaya sip |---| P1 | +-+ ++ | | | +-+ | Asterisk | WAN - | | LAN +-+ | / ++ / | P2 |--+ ++ When P1 dial P2, P2 hears voice clear but P1 could not hear any voice. My sip.conf is [avaya] type=peer fromdomain=xx.xx.xx.xx host=xx.xx.xx.xx disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=yes -- Regards, Shariq Khan 0333-3501125 -- _ -- 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 Turn off reinvite on all extensions and SIP trunks involved and try again. Lyle Giese LCR Computer Services, Inc. -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1498/3523 - Release Date: 03/22/11 Internal Virus Database is out of date. -- _ -- 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
