similar to: Interesting email project.

I have 8 aastra phones that are loosing registration. On the phone gui it says 408 as the registation error after a minute or say they register. In the cli it eill say the phone is now unreachable then it will show it registering then available. At first they did it every hour all the phones. After messing with the experation it does it every 15 nin. Any ideas on how to troubleshoot this? I tried

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

Hey Guys We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the customer wants to move the callcentre. They are asking for an equiv to the ipview I gather HUD may be or the panel view The problem is that we need to see (a) total calls in the queue (b) calls for specific DID - How can you give 1 DID preference to another DID ie DID 61740410001 = Fred Electrical DID

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

Hello. There are so many sound files in /var/lib/asterisk/en. Is there an easy way to let me play back all of them one by one while I am watching CLI to see the current file name? Thanks for help. -- Jian Gao IT Technician SJ Geophysics Ltd. <http://www.sjgeophysics.com> jian.gao at sjgeophysics.com <mailto:jian.gao at sjgeophysics.com> Tel: (604)582-1100

Hello, all, I have a Linksys 3102 from a VoIP provider. It use SRV record to register to the provider's SIP server. When I configure this line into my Asterisk, the register doesn't work if I use their domain name. So it like this: If I use register => user:pwd at proxy.provider.com then I got: [2010-05-17 11:47:19] WARNING[2366] chan_sip.c: No such host: proxy.provider.com

Hi, No matter I try, I can not register to Voipwise with Trixbox. It is always in "unregistered" state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com fromuser=username host=sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer

Hey all What VoIP networking monitoring, asterisk monitoring tools would you recommend? I started working with an IT company that insists on using DSL with a Sonicwall router. The problem is that the clients are having sound problems. The owner is convinced that it's the Asterisk box. In the 4 yrs I have been doing this I have not had this bad a sound problem and it always came down to a bad

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.....) [custom-inbound] exten => _556,1,answer exten => _556,n,playback(beep) exten => _557,1,answer exten => _557,n,playback(beep) exten => _558,1,answer exten => _558,n,playback(beep) exten =>

hi, I am quite new on using Samba and sorry maybe ask a silly question here. I set up simple Samba server on Fedora3 using the samba rpm package comes with fedora3( version 3.0.10-1.fc3). I use the SHARE security level to make things easier. Everything goes fine so far, except that for some windows user, some times, on browsing some directories, it takes extremely long time to display the

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

(Original question was asked here: http://stackoverflow.com/questions/11506734/routing-error-no-route-matches-when-omniauth-failed-on-registration ) I am using omniauth-identity and configure its "fail on registration". My files: config/initializers/omniauth.rb OmniAuth.config.logger = Rails.logger Rails.application.config.middleware.use OmniAuth::Builder do #... provider

Now that we have voicemail working, people have asked to be able to dial in externally and be able to access their voicemail. My dial plan is simple, after ringing a few extensions for some time, it goes to voicemail. What needs to happen to allow for someone to switch out of this into Voicemailmain in such a fashion that an external inbound caller wouldn't at least hear the option? Can the

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

Hello: Is there a command in virsh to shutdown all domains? I can do one at a time, but that is untenable for a large number of domains. Thanks, Neil -- Neil Aggarwal, (281)846-8957, http://UnmeteredVPS.net CentOS 5.4 VPS with unmetered bandwidth only $25/month! 7 day no risk trial, Google Checkout accepted

Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via:

Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant

Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure