[asterisk-users] asterisk prometheus grafana dashboard
hi, i'm testing https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_prometheus anybody who can share grafana dashboard json ;) ? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [FreePBX 15 and Asterisk 16] Changing/Migrating SIP Trunk Provider from DIDLogic to Hoiio in Singapore
Subject: [FreePBX 15 and Asterisk 16] Changing/Migrating SIP Trunk Provider from DIDLogic to Hoiio in Singapore Author: Mr. Turritopsis Dohrnii Teo En Ming (TARGETED INDIVIDUAL) Country: Singapore Date: 11 Nov 2021 Thursday Singapore Time Type of Publication: Plain Text Document Version: 2021.01 DETAILED INSTRUCTIONS == Reference Guide: Configuring Hoiio SIP provider to work with FreePBX Link: http://www.toughdev.com/content/2015/04/configuring-hoiio-sip-provider-to-work-with-freepbx/ Original Author: ToughDev (year 2015) Login to FreePBX at 192.168.1.9 Click Connectivity > Trunks Delete didlogic_(Teo_En_Ming_Corporation) SIP Trunk. Click Apply Config. Click Connectivity > Inbound Routes Delete the only inbound route there. Click Apply Config. Click Connectivity > Outbound Routes Delete the only outbound route there. Click Apply Config. Login to Hoiio SIP Trunk Provider Portal == Login to Hoiio portal at https://sg.hoiio.com/#/auth/login My SIP Account Information == SIP Username: sip3959347 SIP Password: Caller ID: +656602 Adding a SIP trunk === Click Connectivity > Trunks Click Add Trunk. Click Add SIP (chan_sip) Trunk. Trunk Name: Hoiio Outbound CallerID: 656602 CID Options: Allow Any CID Maximum Channels: 5 Asterisk Trunk Dial Options: T Continue if Busy: No Disable Trunk: No Click sip Settings tab Click Outgoing == Trunk Name: Hoiio PEER Details = host=sip6.b3networks.com username=sip3959347 secret= type=peer Click Incoming === USER Context: 656602 USER Details = host=sip6.b3networks.com type=peer context=from-trunk qualify=yes insecure=invite Register String: sip3959347:@sip6.b3networks.com Click Submit Click Apply Config Login to Asterisk Console == # asterisk -r freepbx*CLI> sip show peers Name/username HostDyn Forcerport ComediaACL Port Status Description 656602175.41.130.108 YesYes5060 OK (81 ms) Hoiio/sip3959347 175.41.130.108 YesYes5060 Unmonitored 2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 0 offline] Adding Outbound Route == Click Connectivity > Outbound Routes Click Add Outbound Route Route Name: all calls Route CID: Route Password: Route Type: Uncheck Emergency, Uncheck Intra-Company Music On Hold? default Time Match Time Group: ---Permanent Route--- Trunk Sequence for Matched Routes: Hoiio Click Additional Settings Call Recording: Don't Care PIN Set: None Click Dial Patterns Dial Patterns that will use this Route === match pattern: 65 Click Submit Click Apply Config Adding Inbound Route = Click Connectivity > Inbound Routes Click Add Inbound Route Click Other tab Call Recording: Don't Care CID Lookup Source: None Language: Default Enable Superfecta Lookup: No Superfecta Scheme: ALL Click Fax tab Detect Faxes: No Click General tab Description: Inbound Route Set Destination: Extensions: 1600 Turritopsis Dohrnii Teo En Ming Click Submit Click Apply Config Creating Phone Extension My phone extension 1600 was created a long time ago. There is no need to create the extension again. Testing with Free Softphone === I will use ZoiPer softphone for Windows. ZoiPer is also available on Mac, Linux, Android and IOS. Download link: https://www.zoiper.com/en/voip-softphone/download/current Launch ZoiPer Click Continue as a Free user Username / Login: 1600@192.168.1.9 Password: Click Login Hostname: 192.168.1.9 Click Next Click Skip Click Next Click Configure Click Finish SUCCESS SUCCESS!!! I am able to make outgoing calls and receive incoming calls with ZoiPer softphone. I must remind myself to buy a mic for my laptop. FURTHER READING MATERIALS == Blog post: Teo En Ming’s Guide to Configuring FreePBX 15 and Asterisk 16 VoIP PBX SIP Server in Singapore with DIDLogic SIP Trunk Provider Redundant links: (1) https://tdtemcerts.blogspot.com/2020/12/teo-en-mings-guide-to-configuring.html (2) https://tdtemcerts.wordpress.com/2020/12/08/teo-en-mings-guide-to-configuring-freepbx-15-and-asterisk-16-voip-pbx-sip-server-in-singapore-with-didlogic-sip-trunk-provider/ Mr. Turritopsis Dohrnii Teo En Ming, 43 years old as of 11 Nov 2021, is a TARGETED INDIVIDUAL living in Singapore. He is an IT Consultant with a Systems Integrator (SI)/computer firm in Singapore. He is an IT enthusiast. -- -BEGIN EMAIL SIGNATURE- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Sus
Re: [asterisk-users] [FreePBX 15 and Asterisk 16] Changing/Migrating SIP Trunk Provider from DIDLogic to Hoiio in Singapore
On Thu, 11 Nov 2021, Turritopsis Dohrnii Teo En Ming wrote: Subject: [FreePBX 15 and Asterisk 16] Changing/Migrating SIP Trunk Provider from DIDLogic to Hoiio in Singapore This may be more useful if sent to a FreePBX mailing list. Redundant links: Agreed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Willing to pay for patch to Asterisk fax detection
Hello, We have a commercial client who wants automatic fax detection, but the existing functionality in Asterisk doesn't quite meet their needs. We're willing to pay for a patch to do the following: 1. Limit the automatic fax detection to the first X seconds of a call. X could be defined system-wide in sip.conf or in an Asterisk dialplan variable. 2. Enable or disable fax detection for individual calls. This could be set with an Asterisk dialplan variable. 3. The patch needs to work with chan_sip on Asterisk 13 and above. If you're able to help with this please let me know so we can discuss pricing and your Asterisk development experience. If anyone has ideas for other places to advertise this request let me know! Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Willing to pay for patch to Asterisk fax detection
On Thursday 11 November 2021 at 22:29:34, David Cunningham wrote: > Hello, > > We have a commercial client > If anyone has ideas for other places to advertise this request let me know! I would suggest http://lists.digium.com/mailman/listinfo/asterisk-biz because that is the commercial list (you have currently posted to the "non-commercial discussion" list), and http://lists.digium.com/mailman/listinfo/asterisk-dev Antony. -- "I find the whole business of religion profoundly interesting. But it does mystify me that otherwise intelligent people take it seriously." - Douglas Adams Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Willing to pay for patch to Asterisk fax detection
Hi Antony, Thanks for the suggestion. I didn't get a response on my request to join the asterisk-dev mailing list. I'll try asterisk-biz as well. On Fri, 12 Nov 2021 at 12:23, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Thursday 11 November 2021 at 22:29:34, David Cunningham wrote: > > > Hello, > > > > We have a commercial client > > > If anyone has ideas for other places to advertise this request let me > know! > > I would suggest http://lists.digium.com/mailman/listinfo/asterisk-biz > because > that is the commercial list (you have currently posted to the > "non-commercial > discussion" list), and > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > Antony. > > -- > "I find the whole business of religion profoundly interesting. But it > does > mystify me that otherwise intelligent people take it seriously." > > - Douglas Adams > >Please reply to the > list; > please *don't* CC > me. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Willing to pay for patch to Asterisk fax detection
Hi Naveen, Yes indeed, Asterisk does have automatic fax detection. The chan_sip version just doesn't have all the features we need, namely the ability to restrict it to the first X seconds of a call, and enabling it on specific calls only. A patch to the existing automatic fax detection code to add the features we need is what we're looking to hire someone for. Thanks. On Fri, 12 Nov 2021 at 13:20, David Cunningham wrote: > Hi Antony, > > Thanks for the suggestion. I didn't get a response on my request to join > the asterisk-dev mailing list. I'll try asterisk-biz as well. > > > On Fri, 12 Nov 2021 at 12:23, Antony Stone < > antony.st...@asterisk.open.source.it> wrote: > >> On Thursday 11 November 2021 at 22:29:34, David Cunningham wrote: >> >> > Hello, >> > >> > We have a commercial client >> >> > If anyone has ideas for other places to advertise this request let me >> know! >> >> I would suggest http://lists.digium.com/mailman/listinfo/asterisk-biz >> because >> that is the commercial list (you have currently posted to the >> "non-commercial >> discussion" list), and >> http://lists.digium.com/mailman/listinfo/asterisk-dev >> >> >> Antony. >> >> -- >> "I find the whole business of religion profoundly interesting. But it >> does >> mystify me that otherwise intelligent people take it seriously." >> >> - Douglas Adams >> >>Please reply to the >> list; >> please *don't* >> CC me. >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > David Cunningham, Voisonics Limited > http://voisonics.com/ > USA: +1 213 221 1092 > New Zealand: +64 (0)28 2558 3782 > -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
