Re: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

Hi Stefan, > Hi all > > I maintain the above - it was set up by an external party with whom relations > have now been severed by my employer. > > Quite early after the deployment it became evident that all .gsm audio files > produced on this virtual instance at Azure via MixMonitor are

Re: [asterisk-users] Detecting DoS attacks via SIP

Hi Mike, On 15-08-17 21:37, mdiehl wrote: Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load

Re: [asterisk-users] fail2ban Asterisk 13.13.1

On 02-03-17 13:52, Bryant Zimmerman wrote: John V Are you using pjsip? We are have several test servers and I just checked my /etc/fail2ban/filter.d/asterisk.conf and it is not updated for pjsip implementations. Looking at the security log files and the regex I noticed that some items are

Re: [asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?

On 03-01-17 19:06, Joshua Colp wrote: On Fri, Dec 30, 2016, at 05:04 AM, Kevin Long wrote: Hello, I am using asterisk 14.2 and PJSIP, with TLS transport. I’m sure I’m doing something wrong here .. In 2 distinct softphone clients (Bria and Groundwire), I am able to register successfully,

Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

Hi Travis, On 04-03-16 15:23, Ryan, Travis wrote: I start asterisk 13.7.2 and it dies before I can rasterisk into it. I’ve tried getting a coredump, but it doesn’t coredump. I know there are a lot of errors in the log below, but most of those just say it’ll not load a module, and no big deal.

Re: [asterisk-users] cdr_odbc: Error in ExecDirect: -1

On 01/13/16 14:48, Vitor Mazuco wrote: Hi everybody! I'm trying to install a CDR Viewer https://github.com/g613/asterisk-cdr-viewer I'ts work well, but my problem is to make Asterisk store to MySQL using ODBC When I make a call the CLI returns for me See the log: == Using SIP RTP CoS mark 5

Re: [asterisk-users] Signaling ringing on other extension

On 12/30/15 12:24, Luca Bertoncello wrote: Ishfaq Malik schrieb: Do you have a link to the user guide for your exact phone model? Unfortunately not... I have a Thomson ST2022, but I can just find in Internet manual for the ST2030... The administrator manual can be

Re: [asterisk-users] Fritzbox 7490

On 08-06-15 19:00, Christian wrote: Hi, Sorry if off topic, but is anyone here on this list using it? I am currently searching for a good router for my home network wich supports SIP. Many thanks! I use a 7360 and it works ok but if the 7490's firmware is anything like the 7360 then be

Re: [asterisk-users] Asterisk y Ldap

On 16-12-14 14:00, Dario Estupinan wrote: Como integrar asterisk con Ldap.?? https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver Best, Patrick ps this mailing list uses the English language -- _ -- Bandwidth and

Re: [asterisk-users] ICE consuming High CPU

On 05-12-14 08:25, Mayank Kumar Gour wrote: Any help will be appreciated. Help us help you by providing much much much more information then you have right now. http://www.catb.org/esr/faqs/smart-questions.html HTH, Patrick --

Re: [asterisk-users] Asterisk 13 LDAP

On 04-12-14 11:23, Phil Daws wrote: Is there still an LDAP driver as do not see it in the CentOS 6 repository ? AFAICT in EPEL only Asterisk 1.8.32.1 has one: http://koji.fedoraproject.org/koji/buildinfo?buildID=594932 If you need a newer version then you'll have to build it yourself or find

Re: [asterisk-users] Asterisk 11.9.0 crash and restart

On 21-10-14 08:54, 為近 吉摩（情報システム本部）- Tamechika Yoshikiyo - wrote: Hi, My Asetrisk restarted after to output following warning message. [Oct 16 15:59:58] WARNING[17102][C-8e34]: chan_sip.c:4696 update_provisional_keepalive: Unable to cancel schedule ID 738278. This is probably a bug

Re: [asterisk-users] bristuff-0.4.0-RC4-xr7

On 21-10-14 12:36, Ray Image wrote: [snip] Thanks for your response. As far as I can see I have no choice but use Zaptel as that is downloaded as part of the bristuff script. If there is a better way of getting a HFC PCI card to play nicely with Asterisk please let me know :-) The HFC based

Re: [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS

Hi Bryan, On 10/18/2014 11:47 PM, Bryan Burroughs wrote: All, Has anyone seen this before? This appears to be a Swift or app_swift bug. I'm having a difficult time finding any information or support on this. I haven't used app_Swift with Cepstral but iirc it wasn't deemed very stable.

Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13

Hi Rainer, On 15-09-14 09:07, Rainer Piper wrote: Hi, Info !!! not a question !!! the pjsip logger is different: [Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c: Request from '1001 sip:1001@81.20.137.222' failed for '85.25.197.23:5071' (callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc)

Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13

On 15-09-14 17:22, Rainer Piper wrote: Hi Patrick, github done ;-) Thanks! what is HTH ??? Hope this/that helps http://www.internetslang.com/ http://www.urbandictionary.com/define.php?term=internet%20slang HTH :) Patrick --

Re: [asterisk-users] Asterisk secure fine tune - stop attack

On 04-09-14 16:44, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306 http://213.136.81.166:9306) to extension '34422' rejected

Re: [asterisk-users] Setup Own IP PBX Server

On 01-09-14 12:31, Chandran Manikandan wrote: [snip] I have installed Freepbx server and tried to configure sip extension. It's working fine. A better place for FreePBX related questions and to get help is: http://community.freepbx.org/ Or hire their professional FreePBX support:

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 20:18, Khalid Touati wrote: so it seems Asterisk Versions does not support video I guess On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the Bria app on Android and iPhone. With SELinux and the firewall temporarily disabled I couldn't get it to work with either

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

clients you used, the Asterisk version, the OS and the relevant Asterisk config. Thanks, Patrick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 22:52, Eric Wieling wrote: A co-worker was doing video, I dislike video. The phones were Polycom VVX, The settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP Settings / Video section we have Video: Enabled, H.263 and H.263p are the only two video codecs

Re: [asterisk-users] RDNIS with tel: vs. sip: header

On 28-08-14 11:57, Positively Optimistic wrote: Has anyone had success patching chan_sip.c so that Asterisk will recognize the tel: header for RDNIS information? exten = get_in_brackets(tmp); if (!strncasecmp(exten, sip:, 4)) { exten += 4; } else if

Re: [asterisk-users] Understanding local channels

On 25-08-14 17:06, Mitch Claborn wrote: Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but

Re: [asterisk-users] Dahdi CAPI migration

On 11-08-14 11:09, Toney Mareo wrote: Hello The answers to your questions are: 1, OS CentOS release 5.5 (Final) That version is ancient and full of security holes. It is recommended to at least update to CentOS 5.10 + updates. That's assuming there are Trixbox kmod-dahdi-linux* RPMs for

Re: [asterisk-users] Dahdi CAPI migration

On 08-08-14 10:09, Toney Mareo wrote: Hello Thank you for your response. I thought it could be easier moving the old card to the new machine and using the DAHDI driver. Unfortunately my first attempt for this failed. The card shows up in the original machine as: dahdi_hardware -v

Re: [asterisk-users] Dahdi CAPI migration

Hi Toney, Comments inline. On 07-08-14 12:10, Toney Mareo wrote: Hello Folks, I looking to migrate a pbx from one server to another. The original server has this ISDN card: 00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) The new server: 00:00.0

Re: [asterisk-users] Security Architecture or Security Evaluations Docs?

On 28-07-14 12:28, Jeffrey Walton wrote: [snip] Is there anything that includes the development process? I'm interested in the secure development items and testing. Info about the development of Asterisk can be found here: http://asterisk.org/community/developers

Re: [asterisk-users] Security Architecture or Security Evaluations Docs?

On 26-07-14 14:23, Jeffrey Walton wrote: Does anyone know of Security Architecture or Security Evaluations documents that I could read? Searching is turning up no hits. For example, http://www.google.com/#q=security+evaluation+site:asterisk.org and

Re: [asterisk-users] VoIP over 3G/4G Data

On 18-07-14 17:59, Tech Dude wrote: What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain

Re: [asterisk-users] Asterisk and LDAP

On 18-06-14 23:06, Linus Lüssing wrote: Hi, I'm trying to get Asterisk running with LDAP to be able to authenticate sip user registrations. I'm using Asterisk (1.8.13.1~dfsg1-3+deb7u3) on a Debian server. Unfortunately I wasn't successful so far. My res_ldap.conf looks like this (so pretty

Re: [asterisk-users] Asterisk and LDAP

On 20-06-14 15:05, Linus Lüssing wrote: [snip] having [test_phone_120d] in my sip.conf works fine. Ah wait - do I need to have a user both in LDAP and sip.conf and the only thing LDAP can do for me is the authentication/password checking? As far as I know, yes :) Cheers, Patrick --

Re: [asterisk-users] SSL/TLS weakness impact on Asterisk authentication

On 10-06-14 23:44, Michelle Dupuis wrote: After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Asterisk'

Re: [asterisk-users] SIP Softphone

On 09-06-14 08:52, Giles Coochey wrote: On 08/06/2014 22:01, Mark Robinson wrote: Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? Has anyone tried MicroSIP? http://www.microsip.org/ Nope but if it doesn't meet

[asterisk-users] iPhone TLS reg problem: FILE * open failed

Hi, I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to register with TLS to an Asterisk 11.10.0 box. The registration fails and I see this in the Asterisk console: == Problem setting up ssl connection: error:: lib(0):func(0):reason(0) [Jun 8 15:33:39] WARNING[8555]:

Re: [asterisk-users] iPhone TLS reg problem: FILE * open failed

in Linphone. Cheers, Patrick On Jun 8, 2014 6:50 PM, Patrick Laimbock patr...@laimbock.com mailto:patr...@laimbock.com wrote: Hi, I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to register with TLS to an Asterisk 11.10.0 box. The registration fails and I see

Re: [asterisk-users] Get last dialed number in a context?

On 03-06-14 11:31, Stefan Gofferje wrote: Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 = jumps into a context which redials until callresult is not busy Maybe like this:

Re: [asterisk-users] Kernel and DAHDI

On 12-05-14 09:30, Thorsten Göllner wrote: That's correct. When you update the kernel package youhave also to recompile dahdi package. AFAIK that's not true for RHEL6/CentOS6 as the EL6 kernels are ABI compatible so you don't need to recompile DAHDI when there's a new EL6 kernel. When

Re: [asterisk-users] One mailbox for multiple extensions with individual greetings

On 11-05-14 05:10, John T. Bittner wrote: Why don't you use the voicemail copy feature? Create 3 mailboxes 1234, 6789 and 2000 for the shared. VoiceMail(1234@default2000@default,su) VoiceMail(6789@default2000@default,su) Set both 1234 and 6789 to email the voicemail to a fake email address and

[asterisk-users] One mailbox for multiple extensions with individual greetings

Hi, Is there a way in Asterisk 11 to use a single voicemailbox for multiple extensions while still hearing each extension's individual greeting? Use case: someone has 2 numbers and wants all voicemail messages for both numbers to end up in one mailbox. So when dialing 1234 and NOANSWER you

Re: [asterisk-users] AMR installation error

On 30-04-14 12:50, [Digital^Dude] ® wrote: make gives this: IIRC Digium's policy is that there's no support on this list for patented technologies like AMR which are possibly not officially licensed. Obviously to prevent any legal liability. HTH, Patrick --

Re: [asterisk-users] Asterisk support for h.324m

On 29-04-14 20:41, [Digital^Dude] ® wrote: Hello, If anyone has successfully compiled asterisk with: app_rtsp codec_amr mp4_play mp4_save app_transcode h324m_call Please share the versions of OS software, and libraries used. Lets make this thread useful so that all tried and tested video

Re: [asterisk-users] Trunk issue

On 28-04-14 19:49, Haley,Scott A wrote: Now I am getting Permission denied. Have you checked if SELinux is blocking the app? Any blockage should show up as an 'AVC' in /var/log/audit/audit.log You can temporarily set SELinux to permissive with 'setenforce 0' and check if the problem goes

Re: [asterisk-users] Trunk issue

On 28-04-14 20:13, Haley,Scott A wrote: That seemed to fix it. Thanks to everyone. https://bugzilla.redhat.com/show_bug.cgi?id=1092150 HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Asterisk and SRTP

On 04/05/2014 07:56 PM, William Wu wrote: Hi experts, I am trying Asterisk SRTP in my environment, and find that when Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay by Asterisk are local ports on the Asterisk server, media from the two clients out of the NAT (for

[asterisk-users] CLI command to see if SRTP is active?

Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. Thanks, Patrick --

[asterisk-users] Problem with TLS/SRTP with Asterisk 11.8.1

Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI -- Registered SIP

Re: [asterisk-users] Problem with TLS/SRTP with Asterisk 11.8.1

On 24-03-14 21:28, Patrick Laimbock wrote: [snip] == Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Mar 24 21:20:56] WARNING[28467]: tcptls.c:272 handle_tcptls_connection: So others may find the fix: make sure the server

Re: [asterisk-users] dahdi + dlink du128ta

On 03/14/2014 07:53 AM, binary dreamer wrote: hello everyone, I do have a usb ISDN modem that I would like to make it work with dahdi. is it possible? No. I am running debian 7, with dahdi 2.9, asterisk 11.8 dahdi cannot find it at the moment, unless there is something else to be done. Try

Re: [asterisk-users] dahdi + dlink du128ta

On 03/15/2014 10:15 PM, binary wrote: i have tried the misdn from git. my problem is that it needs LCR and it fails to get installed Then you need to fix that. AFAIK there is no other way to use a USB ISDN TA than via mISDN/LCR. HTH, Patrick --

Re: [asterisk-users] Linux call router

On 11-03-14 12:15, binary dreamer wrote: hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes

Re: [asterisk-users] Asterisk crashes at meetme kick all

On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote: Dear Forum, I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk crashed while executing “meetme kick all” CLI command from manager interface. The link says the issue has been closed however I am not able to identify in which

Re: [asterisk-users] IOPS required by Asterisk for Call Recording

On 25-01-14 06:26, Amit wrote: Thanks for response. How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I'm not aware of 400+ calls being recorded succesfully on an Asterisk box. If there is it probably has tons of RAM, enterprise grade SSDs or