I am Irfan Rifa'i VoIP Engineer Telecomunication Addict

WFH still remaining and have no sign to end, especially in Indonesia as outbreak began on March 2020. Forgot the hot news about Indonesian pandemic issue, we wont cover here as politician.

The challenge is still same, how to support your folks to work remotely without disrupting your bussiness. Old but not old, a conversation with telephone is still psychologically impacted. People feels "humanize" when talk to each other, and telephone still favourite platform to deal with it. 

Then how it will work if your number is FXS line came from ONT modem? A nut problem, while at this pandemic situation you have to sit on your office desk and hold a telephone device.  Your folks need to mobile and carry out your PSTN number with them and ensure your bussiness running well. 

The Problematic

We know in this pandemic situation, your bussnines may be the struggle one. Investment to support your team with remote working condition may became new cost, when you trying to make your folks stand  in with paycheck to paycheck.

The Idea of "lets carry them out and move out to cloud"

Our idea is to carry your PSTN number and move them to our CloudPBX Platform with minimum technical requirement and sure with lower cost as can as possible (hope this so). 

We not only made this idea to simply carry out your number from physical analog line, also to leverage your ability to support your remote working condition. Once you have connect your office number with our CloudPBX platform, then you can use almost all of rich feature of our CPaaS platfrom.

- Callforward

- Voicemail Boxes

- Call Recording (Add On)

- API (Developer welcome)

- Webhooks (Do you know zapier.com ?)

- Real time CDR

- Conference

- Cutomize IVR

- Hot Desking

- Microsoft Teams Direct Routing (Microsoft minded phonesystem)

The Cost

We're realize, sometimes its began sacred when deal with cost. I wont cover how much it cost at quantity. Apart, we can described what the prerequisites you need to deal with moving your PSTN number to our CloudPBX Platform.

Basically, you only need 2 things....

- A Media Gateway

Yeah, same stuff different story. Converting a analog line to digital line only can be done with converter device such as Media Gateway. This major prerequisites is the key of carrying out your PSTN number to CloudPBX platfrom. A fun fact, honestly several telecom provider carry their PSTN line as digital line at the core level then convert them to analog line on user last mile.

- A Internet Connection

Internet, a global connection. A global transport infrastructure which helps your number from media gateway establish with our CloudPBX platform.

If you not a techy folks, seems difficult to understand what it means. But, okay we will cover some simplified explanation. 

We have a user story, a bike store with 4 employees. They have to working remotely due the pandemic

What they have :

- 2 PSTN number and analog phone.

- Lets say 1Mbps internet connection bundled with PSTN number (A broadband services you know)

Their challenges :

- They need to receive a call from their valuable customers, with exsiting published PSTN number from their broadband provider

- They need to make a call also. If using employee personal number, it costly and no privacy gatekeeper

- Boom! their PSTN number locked on their modem, a telephone call cant picked up by agent.

Our breakthrough against their challenges

- Placed our ceritifed media gateway, plug PSTN line and convert their number to digital line

- Connet to internet port, configured  CloudPBX as they request

- Boom! The bike store employee can received and make a call from their home!

- Also, they can review real time who had called, the number, the duration, and customize routing via CloudPBX portal.

Hope you understand bit of them, kidding!

The telephone is a good way to talk to people without having to offer the a drink.

Fran Lebowitz.

Latar Belakang

Mempertahankan Eksisting Number

Perusahaan pada umumnya sudah memiliki sambungan line PSTN dan telpon eksisting. Perpindahan infrastruktur PBX kantor dari onprem menjadi cloud diharapkan tidak merubah nomor DID eksisting dari perusahaan, karena DID atau nomer perusahaan adalah unik dan sudah dikenal sebagai identitas dari perusahaan. Perusahaan akan berusaha mempertahankan agar nomor DID tidak berubah.

Low Rates Outbound Call

Murah adalah sebuah hal wajib bagi perusahaan pada umumnya. Bagaimana melakukan reduce biaya panggilan telpon menggunakan layanan pihak ketiga sehingga telepon murah. Atau perusahaan memiliki koneksi SIP Trunk internal dengan eksisiting IPPBX. Menambahkan outbound trunking menjadi kebutuhan untuk menyediakan konektivitas sesuai kebutuhan perusahaan.

Katakanlah jika anda menelpon ke nomor operator A, anda ingin panggilan melalui jalur operator A sehingga didapatkan biaya yang murah dibandingkan menggunakan line operator B.  

Pilihan Standar SLA Operator

Bagaimana menentukan kualitas layanan dari provider siptrunk? perusahan mempertimbangkan untuk memilih provider/operator dengan standar kualitas sesuai regulatory baik internal maupun eksternal. Hal ini mencakup kualitas layanan, outage, redudansi, dll. 

Katanklah sambungan telpon / PSTN utama dari sebuah perusahaan mengalami gangguan. Lantas perusahaan bergantung pada operator A dan tidak bisa menerima dan melakukan panggilan. Disini lost of bussiness terjadi, diperlukan operator B untuk menjadi backup dari sambungan telpon.

Kebutuhan untuk tidak merubah nomer eksisting, menentukan SLA dari provider, dan mendapatkan tarif ideal untuk setiap percakapan diatas dapat dijawab menggunakan fitur BYOC (Bring Your Own Carrier)

Ofon Cloud PBX - Carrier Manager

Carrier Manager memungkinkan reseller ataupun user untuk mengatur resource outbound trunking (BYOC). Reseller dapat menambahkan trunking dan melakukan kustomisasi panggilan outbound untuk mendapatkan kualitas dan opsi terbaik.

Ofon CloudPBX - List carrier SIP Trunking

Ofon CloudPBX - Menambahkan Outbound Carrier

Ofon CloudPBX - Kustomisasi Rule Outbound Carrier

Ofon CloudPBX - List Carrier terbaru

BYOC memberikan kemudahan dan fleksibilitas bagi perusahaan untuk menentukan biaya, tracing dan kualitas dari sebuah layanan telepon operator.

links : https://www.ofon.co.id/enterprise

 

 

 

 <gateway name="smartpbx">  
                             <param name="username" value="username"/>  
                             <param name="password" value="password"/>  
                             <param name="realm" value="smartpbx.domain.tld"/>  
                             <param name="from-user" value="username"/>  
                             <param name="from-domain" value="smartpbx.domain.tld"/>  
                             <param name="proxy" value="smartpbx.domain.tld"/>  
                             <param name="expire-seconds" value="800"/>  
                             <param name="register" value="true"/>  
                             <param name="register-transport" value="tcp"/>  
                             <param name="retry-seconds" value="30"/>  
                             <param name="ping" value="30"/>  
                             <param name="context" value="public"/>  
                             <param name="caller-id-in-from" value="true"/>  
                             <variables>  
                             </variables>  
                         </gateway>  

 <extension name="smartpbx_to_teams">   
   <condition field="${acl(${network_addr} domains)}" expression="true"/>  
   <condition field="destination_number" expression="^(1017)$">  
    <action application="set" data="domain_name=$${domain}"/>  
    <action application="set" data="sip_secure_media=true"/>  
    <action application="export" data="nolocal:rtp_secure_media=true:AES_CM_128_HMAC_SHA1_80"/>  
    <action application="bridge" data="sofia/gateway/sip.pstnhub.microsoft.com/+622713400062"/>  
   </condition>  
  </extension>  

 <extension name="teams_to_smartpbx">  
   <condition field="${acl(${network_addr} domains)}" expression="true"/>  
   <condition field="destination_number" expression="^(\+62)(11\d{2})$">  
    <action application="set" data="domain_name=$${domain}"/>  
    <action application="bridge" data="sofia/gateway/smartpbx.domain.tld/$2"/>  
   </condition>  
  </extension>  

 

Call Recording atau rekaman panggilan merupakan fitur dasar dan penting yang harus dimiliki perangkat IPPBX. Sebagai contoh kasus, ada kalanya seorang supervisor hendak memutar percakapan seorang staffnya dengan pelanggan untuk memastikan kualitas dukungan layanan terhadap pelanggan. Atau seorang manajer IT yang hendak melakukan evaluasi terhadap kualitas layanan telepon itu sendiri.

Perkembangan dalam transformasi telepon kantor dari on-premise menjadi cloud memberikan tantangan baru dalam memberikan fleksibilitas dan keamanan dari layanan telepon. Lalu bagaimana dengan kualitas dan keamanan dalam layanan telepon kantor berbasis cloud?

Ofon SmartPBX adalah platform teleponi berbasis cloud. SmartPBX menyediakan fitur Call Recording yang dapat terhubung dengan eksisting cloud storage milik pelanggan ataupun dengan cloud storage yang sudah disediakan oleh Ofon.

Dengan integrasi cloud storage. Kapasitas dan ketersediaan ruang penyimpanan dapat diatur sesuai dengan kebutuhan. Keamanan dan ketersedian data rekaman panggilan dapat terjaga karena tersimpan dalam penyimpanan komputasi awan. Didukung dengan web portal yang intuitif untuk memudahkan pelanggan dalam melakukan konfigurasi call recording pada setiap usernya.


Storage Management
Storage Management adalah aplikasi web manajemen untuk menambahkan cloud storage pada portal web SmartPBX. Storage management mendukung integrasi SmartPBX dengan provider cloud storage seperti AWS S3 dan layanan S3 kompatibel lain. Ofon juga menyediakan lokal S3 kompatibel storage

Tertantang dengan kolega yang berhasil menghubungkan Asterisk  dengan Microsoft Phone System. Pada tulisan kali ini saya telah mencoba menghubungkan FreeSWITCH dengan Microsoft Phone System.

 wget -O - https://files.freeswitch.org/repo/deb/debian-release/fsstretch-archive-keyring.asc | apt-key add -  
 echo "deb http://files.freeswitch.org/repo/deb/debian-release/ stretch main" > /etc/apt/sources.list.d/freeswitch.list  
 echo "deb-src http://files.freeswitch.org/repo/deb/debian-release/ stretch main" >> /etc/apt/sources.list.d/freeswitch.list  
 apt-get update  
 #Install dependencies required for the build  
 apt-get build-dep freeswitch  
 #then let's get the source. Use the -b flag to get a specific branch  
 cd /usr/src/  
 git clone https://github.com/signalwire/freeswitch.git -bv1.10 freeswitch  
 cd freeswitch  

 git config pull.rebase true  
 # ... and do the build  
 ./bootstrap.sh -j  
 ./configure  
 make  
 make install  

 <!-- External SIP Profile -->  
 <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>  
 <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>  
 <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>  
 <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>  
 <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/>  
 <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>  
 <X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/>  

 <include>  
  <gateway name="freeswitch.ofon.biz">  
  <param name="username" value="not-used"/>  
  <param name="realm" value="freeswitch.ofon.biz"/>  
  <param name="from-domain" value="freeswitch.ofon.biz"/>  
  <param name="password" value="not-used"/>  
  <param name="proxy" value="sip.pstnhub.microsoft.com"/>  
  <param name="register" value="false"/>  
  <param name="register-transport" value="tls"/>  
  <param name="retry-seconds" value="10"/>  
  <param name="caller-id-in-from" value="true"/>  
  <!--extra sip params to send in the contact-->  
  <param name="contact-host" value="freeswitch.ofon.biz"/>  
  <param name="contact-in-ping" value="true"/>  
  <param name="ping-user-agent" value="Sonus SBC V.5.7"/>  
  <param name="ping" value="10"/>  
  </gateway>  
 </include>  

 send 623 bytes to tls/[52.114.7.24]:5061 at 03:25:41.189516:  
 ------------------------------------------------------------------------  
 OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0  
 Via: SIP/2.0/TLS IP.AD.DR.ES:5081;rport;branch=z9hG4bKDKgetaKpUc4rc  
 Max-Forwards: 70  
 From: <sip:freeswitch.ofon.biz>;tag=4624XDZ9UtypN  
 To: <sip:freeswitch.ofon.biz>  
 Call-ID: e763d8b6-1ee8-1239-0ea5-566fbb48005e  
 CSeq: 20967190 OPTIONS  
 Contact: <sip:gw+freeswitch.ofon.biz@freeswitch.ofon.biz:5081;transport=tls;gw=freeswitch.ofon.biz>  
 User-Agent: Sonus SBC V.5.7  
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY  
 Supported: timer, path, replaces  
 Allow-Events: talk, hold, conference, refer  
 Content-Length: 0  
 recv 365 bytes from tls/[52.114.7.24]:5061 at 03:25:41.238211:  
 ------------------------------------------------------------------------  
 SIP/2.0 200 OK  
 FROM: <sip:freeswitch.ofon.biz>;tag=4624XDZ9UtypN  
 TO: <sip:freeswitch.ofon.biz>  
 CSEQ: 20967190 OPTIONS  
 CALL-ID: e763d8b6-1ee8-1239-0ea5-566fbb48005e  
 VIA: SIP/2.0/TLS IP.AD.DR.ES:5081;branch=z9hG4bKDKgetaKpUc4rc;rport  
 CONTENT-LENGTH: 0  
 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY  
 SERVER: Microsoft.PSTNHub.SIPProxy v.2020.5.27.2 i.ASEA.5  

 <include>  
  <gateway name="netaccess.ofon.biz">  
  <param name="username" value="not-used"/>  
  <param name="realm" value="netaccess.ofon.biz"/>  
  <param name="from-domain" value="customer04.ofon.biz"/>  
  <param name="password" value="not-used"/>  
  <param name="proxy" value="IP.AD.DR.ES"/>  
  <param name="register" value="false"/>  
  <param name="register-transport" value="tcp"/>  
  <param name="retry-seconds" value="10"/>  
  <param name="caller-id-in-from" value="true"/>  
  <param name="contact-host" value="customer04.ofon.biz"/>  
  <param name="contact-in-ping" value="true"/>  
  <param name="ping-user-agent" value="Sonus SBC V.5.7"/>  
  <param name="ping" value="10"/>  
  </gateway>  
 </include>  

 <include>  
  <extension name="teams_to_ofon">  
   <condition field="${acl(${network_addr} domains)}" expression="true"/>  
   <condition field="caller_id_number" expression="^(\+622430000062)$">  
    <action application="set" data="domain_name=$${domain}"/>
    <action application="bridge" data="sofia/gateway/netaccess.ofon.biz/${destination_number}"/>  
   </condition>  
  </extension>  
 </include>  

 <include>  
  <extension name="ofon_to_teams">  
   <condition field="network_addr" expression="^IP\.AD\.DR\.ES$"/>  
   <condition field="${acl(${network_addr} domains)}" expression="true"/>  
   <condition field="destination_number" expression="^(\+622430000062)$">  
    <action application="set" data="domain_name=$${domain}"/>  
    <action application="set" data="sip_secure_media=true"/>  
    <action application="export" data="nolocal:rtp_secure_media=true:AES_CM_128_HMAC_SHA1_80"/>  
    <action application="bridge" data="sofia/gateway/customer04.ofon.biz/${destination_number}"/>  
   </condition>  
  </extension>  
 </include>  

  Import-Module SkypeOnlineConnector   
  $Cred = Get-Credential   

 $sfb = New-CsOnlineSession $Cred -OverrideAdminDomain “ofonio.onmicrosoft.com”  
 Import-PSSession $sfb -AllowClobber  

 Set-CsOnlinePSTNGateway -Fqdn freeswitch.ofon.biz -SipSignallingPort 5081 -ForwardCallHistory $true -Enabled $true -ForwardPai $true -MaxConcurrentSessions 5  

 Identity              : freeswitch.ofon.biz  
 InboundTeamsNumberTranslationRules : {}  
 InboundPstnNumberTranslationRules  : {}  
 OutboundTeamsNumberTranslationRules : {}  
 OutboundPstnNumberTranslationRules : {}  
 Fqdn                : freeswitch.ofon.biz  
 SipSignalingPort          : 5081  
 FailoverTimeSeconds         : 10  
 ForwardCallHistory         : True  
 ForwardPai             : True  
 SendSipOptions           : True  
 MaxConcurrentSessions        : 5  
 Enabled               : True  
 MediaBypass             : False  
 GatewaySiteId            :  
 GatewaySiteLbrEnabled        : False  
 GatewayLbrEnabledUserOverride    : False  
 FailoverResponseCodes        : 408,503,504  
 GenerateRingingWhileLocatingUser  : True  
 PidfLoSupported           : False  
 MediaRelayRoutingLocationOverride  :  
 ProxySbc              :  
 BypassMode             : None  

 Set-CsOnlinePstnUsage -Identity Global -Usage @{Add="Default"}  
 New-CsOnlineVoiceRoute -Identity "CatchAll" -NumberPattern ".*" -OnlinePstnGatewayList ast.ofon.biz -Priority 1 -OnlinePstnUsages "Default"  
 New-CsOnlineVoiceRoutingPolicy -Identity "DefaultVoicePolicy" -OnlinePstnUsages "Default"  

 Set-CsUser -Identity "irfan.rifai@ofon.io" -EnterpriseVoiceEnabled $true -HostedVoiceMail $true -OnPremLineURI tel:+622713400062  

 Grant-CsOnlineVoiceRoutingPolicy -Identity "irfan.rifai@ofon.io" -PolicyName "DefaultVoicePolicy"  

Seringkali masi ketemu sama rekan / mitra yang masih kesulitan untuk set routing incoming routes pada Asterisk base PBX. Jadi sebenarnya untuk routing inbound routes di Astersik base PBX cukup mudah kalau dilakukan melalui GUI web. Untuk konfigurasinya menggunakan CLI mungkin akan saya bahas di postingan yang lain. Secara konsep keduanya tetap masih sama.

Seringkali ketika lagi troubleshoot VoIP nemu obstacle gimana caranya narik log SIP nya yang mudah dan gampang. Dulu awal - awal pas troubleshoot pake tools sejuta umat, wireshark. Cuma lama - lama jadi kesulitan sendiri karena wireshark yang berat dan harus nunggu sesi call selesai utuk troubleshoot.

 yum install ngrep  

 ngrep -W byline -d eth0 port 5060  

 U 185.53.88.40:58777 -> 10.10.10.56:5060 #3  
 INVITE sip:900442038076746@10.10.10.56:5060 SIP/2.0.  
 Via: SIP/2.0/UDP 185.53.88.40:58777;branch=z9hG4bK1763926594.  
 Max-Forwards: 70.  
 From: <sip:61@10.10.10.56:5060>;tag=1590280767.  
 To: <sip:900442038076746@10.10.10.56:5060>.  
 Call-ID: 1842141676-1666533667-1306961472.  
 CSeq: 1 INVITE.  
 Contact: <sip:61@185.53.88.40:58777>.  
 Content-Type: application/sdp.  
 Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH.  
 User-Agent: sdthsrhstuksz.  
 Content-Length: 206.  
 .  
 v=0.  
 o=61 16264 18299 IN IP4 192.168.1.83.  
 s=call.  
 c=IN IP4 192.168.1.83.  
 t=0 0.  
 m=audio 25282 RTP/AVP 0 101.  
 a=rtpmap:0 pcmu/8000.  
 a=rtpmap:8 pcma/8000.  
 a=rtpmap:101 telephone-event/8000.  
 a=fmtp:101 0-11.  
 #  
 U 10.10.10.56:5060 -> 185.53.88.40:58777 #4  
 SIP/2.0 100 Trying.  
 Via: SIP/2.0/UDP 185.53.88.40:58777;branch=z9hG4bK1763926594.  
 From: <sip:61@10.10.10.56:5060>;tag=1590280767.  
 To: <sip:900442038076746@10.10.10.56:5060>.  
 Call-ID: 1842141676-1666533667-1306961472.  
 CSeq: 1 INVITE.  
 User-Agent: VoIP My ID Mediaserver.  
 Content-Length: 0.  
 .  
 #  
 U 10.10.10.56:5060 -> 185.53.88.40:58777 #5  
 SIP/2.0 403 Forbidden.  
 Via: SIP/2.0/UDP 185.53.88.40:58777;branch=z9hG4bK1763926594.  
 From: <sip:61@10.10.10.56:5060>;tag=1590280767.  
 To: <sip:900442038076746@10.10.10.56:5060>;tag=U3gB9U86j2a6Q.  
 Call-ID: 1842141676-1666533667-1306961472.  
 CSeq: 1 INVITE.  
 User-Agent: VoIP My ID Mediaserver.  
 Accept: application/sdp.  
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.  
 Supported: timer, path, replaces.  
 Allow-Events: talk, hold, conference, refer.  
 Content-Length: 0.  
 .  
 #  

 [irontec]  
 name=Irontec RPMs repository  
 baseurl=http://packages.irontec.com/centos/$releasever/$basearch/  

 rpm --import http://packages.irontec.com/public.key  

 yum check-update -y  
 yum install sngrep -y