![\"\"](\"https:\/\/infotechguy.net\/wp-content\/uploads\/2021\/03\/Phone-Final-ORIGINAL-min.jpg\")
<\/a><\/p>\nThis next series of blog posts (Part1<\/a>, Part2<\/a>, Part3) are dedicated to walking through the many aspects related to VoIP(Voice over Internet Protocol) and it’s features.<\/p>\n <\/p>\n Our goal for this Article is to create a small office VoIP network that consists of the following; A few VoIP phones (both Manual and Auto provisioned) running SIP with an extension schema of our choosing, an outside PTSN line where we can call out of and others can call into, an Interactive Voice Response(IVR) for incoming calls, and other features( secure calling, security banning, hold music, etc). Below is our lab where we will accomplish this.<\/p>\n What is SIP? SIP is the life blood of the VoIP network, our transport protocol. SIP allows us to signal a remote VoIP phone to start the call process, SIP transports the audio\/videio payload between VoIP phones to maintain a call session, SIP also handles the registration of our VoIP extensions, and much much more. SIP like most audio\/video network transport protocols uses UDP natively to ensure minimal network overhead and reduces bandwidth requirements. SIP relies on a SDP(Session Description Protocol)<\/a> to negotiate each endpoints capabilities for a call. SIP then relies on RTP(Real-Time Transport Protocol)<\/a> , SRTP(Secure Real-Time Transport Protocol)<\/a>, or TLS <\/a>for media stream transmission.<\/p>\n <\/p>\n **The Excerpt Above is owned by https:\/\/who.rocq.inria.fr\/Philippe.Sultan\/Asterisk\/asterisk_sip_external_authentication.html<\/a><\/em><\/p>\n SIP Request commands<\/strong>: Before we get any further let’s go over the different ones.<\/a><\/p>\n SIP Call Sessions basics:<\/strong> How a call gets setup.<\/p>\n There is another alternative to SIP called Skinny<\/a> which is a Cisco proprietary protocol. You will most like find this running as the Signally protocol in a VoIP network which is predominately Cisco. Keep in mind Cisco equipment can run SIP as well. So if your environment\/equipment is mixed you will want to go with SIP.<\/p>\n FreePBX is a Linux distro that includes Asterisk and a Web Management GUI. Asterisk being the PBX(Private Branch Exchange) software that is used to manage your VoIP logic. A PBX is responsible for your extension numbering plan (x1234), dial plans routing (like when you have to dial *9 to get an outside line), VoIP trunking (merging PBXs together), voicemail boxes, etc. A PBX is also where your VoIP capable phones will register themselves with and receive INVITEs. All in all, just remember the PBX has many features, but it’s primary function is to set up and destroy SIP Call sessions.<\/strong><\/p>\n Let’s get started by installing FreePBX by following the official documentation here(http:\/\/wiki.freepbx.org\/display\/FD\/Installing+FreePBX+Official+Distro<\/a>);<\/p>\n Advanced Options:<\/strong><\/p>\n SIP NAT’ing.<\/strong> If you recall from the SIP registration messages above you can see in the SIP payload sip:192.168.0.2:5060 SIP\/2.0<\/em><\/strong> , so when a call is being setup each phone has been registered with that syntax sip:{address}{port}. Now what happens if my phone is behind a NAT where the real IP address of phone does not match the IP packet that arrives at the PBX server. For example, my phones IP is 192.168.55.5, but my IP packet gets SNAT’d and changed to 1.1.1.10 on it’s way to the PBX server.<\/strong> The IP packet will arrive correctly, but the SIP payload message will contain the line sip:192.168.55.5:5060 SIP\/2.0<\/strong><\/em>. The PBX will then try to send ACK, and INVITEs for calls to 192.168.55.5:5060<\/strong> which it cannot get to. SIP breaks. Setting the PBX server up with the NAT mode to yes<\/span>, tells the server, when this extension registers, ignore IP address in the sip:192.168.55.5:5060 SIP\/2.0 message and use the IP Source in the IP packet instead.<\/strong><\/p>\n There are mostly two types of provision when working with VoIP phones. Manual provisioning and Auto-provisioning. There are pros and cons for each. For example Manual provisioning is easier to get working right away because these phones have a configuration menu on them. I’ve seen most WiFi VoIP phones and VoIP Apps on Smartphones (like UbiPhone<\/a> or Zoiper<\/a> or Linphone for apple users<\/a>) to be this way. You can also find some desktop phones that have manual provisioning, like the Nortel IP1535<\/a>. On the other side of the coin, Auto-Provisioning is automatic! and best used for larger deployments. It is however a more sophisticated setup. Auto-Provisioning deployments are most common in the business or corporate world where you have many VoIP phones that get their configuration from centralized place upon bootup. These are the Cisco or Avaya phones you may have seen.<\/p>\n x86555 to x86888.<\/p>\n So here comes the fun part! Auto-Provisioning of SIP phones! This will be a little more sophisticated than Manual provisioning as more pieces are involved and need to be configured to get it all to work. Each Phone type\/brand may also be different. I am going to use two Cisco phones; 1x Cisco 7941G<\/a> and 1x Cisco 7945G<\/a>, the G stands for Global not gigabit!. The basic process for auto provisioning Cisco 7940s and 7960s series phones is through DHCP boot options and TFTP server housing the specific config files for the phones.<\/p>\n Luckily, our FreePBX server by default installs a TFTP service. Located at \/tftpboot on the server itself.<\/p>\n Important Files:<\/strong><\/em><\/span><\/p>\n The next steps are required so that the Cisco phones will know where the TFTP server is, remember they don’t know anything upon boot and must be told via DHCP Option 150 or 66<\/p>\n If \u00a0you’ve made it this far..Congrats!! Now the fun part of figuring out what went wrong. Like in the next Parts I will only be Troubleshooting what we went over.<\/p>\n The first one denotes authentication failed due to incorrect SIP password for the extension trying to register.PART #1 — Laying the foundation for our VoIP network<\/a><\/h5>\n
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PART #2 — Call routing, Call numbers, SIP Trunks<\/a><\/h5>\n
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PART #3 — Ring Groups, Extension Shortcuts, Call Centers, Voicemails, Secure SIP with TLS, etc<\/h5>\n
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Part#1<\/h2>\n
Our Goal and Network Lab<\/h4>\n
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SIP (Session Initiation Protocol)<\/h4>\n
<\/p>\nREGISTER sip:192.168.0.1:5060 SIP\/2.0\nContent-Length: 0\nContact: ;events=\"message-summary\"\nCall-ID: 1DBDA84B-37E9-4F05-BE8B-E3A0F6BBEE91@192.168.0.2\nMax-Forwards: 70\nFrom: ;tag=220587183498\nCSeq: 3 REGISTER\nTo: \nVia: SIP\/2.0\/UDP 192.168.0.2;rport;branch=z9hG4bK805d2fa50131c9b1434671010000391200000013\n\nSIP\/2.0 401 Unauthorized\nVia: SIP\/2.0\/UDP 192.168.0.2;rport;branch=z9hG4bK805d2fa50131c9b1434671010000391200000013;received=192.168.0.2\nFrom: ;tag=220587183498\nTo: ;tag=as6d76ba1b\nCall-ID: 1DBDA84B-37E9-4F05-BE8B-E3A0F6BBEE91@192.168.0.2\nCSeq: 3 REGISTER\nUser-Agent: Asterisk PBX\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY\nContact: \nWWW-Authenticate: Digest realm=\"asterisk\", nonce=\"4f87b95d\" \nContent-Length: 0\n\nREGISTER sip:192.168.0.1:5060 SIP\/2.0\nContent-Length: 0\nContact: ;events=\"message-summary\"\nCall-ID: 1DBDA84B-37E9-4F05-BE8B-E3A0F6BBEE91@192.168.0.2\nMax-Forwards: 70\nFrom: ;tag=2205872822811\nCSeq: 4 REGISTER\nTo: \nVia: SIP\/2.0\/UDP 192.168.0.2;rport;branch=z9hG4bK805d2fa50131c9b14346710100004e6d00000016\nAuthorization: Digest username=\"sultan\",realm=\"asterisk\",nonce=\"4f87b95d\",uri=\"sip:192.168.0.1:5060\",response=\"fed6890f44712fbaef17c704e6e30eac\"\n\nSIP\/2.0 200 OK\nVia: SIP\/2.0\/UDP 192.168.0.2;rport;branch=z9hG4bK805d2fa50131c9b14346710100004e6d00000016;received=192.168.0.2\nFrom: ;tag=2205872822811\nTo: ;tag=as6d76ba1b\nCall-ID: 1DBDA84B-37E9-4F05-BE8B-E3A0F6BBEE91@192.168.0.2\nCSeq: 4 REGISTER\nUser-Agent: Asterisk PBX\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY\nExpires: 120\nContact: ;expires=120\nDate: Fri, 07 Oct 2005 12:57:55 GMT\nContent-Length: 0<\/pre>\n
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<\/code><\/pre>\nAuthorization: Digest username=\"sultan\",realm=\"asterisk\",nonce=\"4f87b95d\",uri=\"sip:192.168.0.1:5060\",response=\"fed6890f44712fbaef17c704e6e30eac\"<\/pre>\n
<\/code><\/pre>\n<\/code><\/pre>\n<\/li>\n<\/code><\/pre>\nAuthorization: Digest username=\"sultan\",realm=\"asterisk\",nonce=\"4f87b95d\",uri=\"sip:192.168.0.1:5060\",response=\"fed6890f44712fbaef17c704e6e30eac\"<\/pre>\n
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**Owned by Cisco.com<\/em><\/p>\n\n
FreePBX\/Asterisk — Call System Exchange<\/h4>\n
Installing…<\/h6>\n
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Setting up a few extensions…<\/h6>\n
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To NAT or not to NAT, that is the Question…<\/h5>\n
Phone Provisioning<\/h4>\n
Manual Provisioning.<\/h5>\n
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\n\u00a0 \u00a0 \u00a0 <\/a><\/li>\n\n
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\n<\/a><\/li>\n<\/ul>\n<\/li>\n<\/ol>\nTry a Test Call.<\/h6>\n
Auto-Provisioning.<\/h5>\n
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\n<\/a>Firmware Download Mode Screen<\/strong>
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\n<\/strong>Excerpt from my TFTP server upon Cisco 7941G boot.
\n<\/strong><\/strong><\/p>\nAug 27 14:29:44 FreePBX xinetd[1108]: START: tftp pid=8382 from=10.55.55.151\nAug 27 14:29:44 FreePBX in.tftpd[8384]: RRQ from 10.55.55.151 filename SEP001C5BBDDCAA.cnf.xml\nAug 27 14:29:56 FreePBX in.tftpd[8385]: RRQ from 10.55.55.151 filename English_United_States\/mk-sip.jar\nAug 27 14:30:16 FreePBX in.tftpd[8459]: RRQ from 10.55.55.151 filename dialplan.xml<\/pre>\n
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P0S30202<\/code><\/pre>\n<\/li>\n<Default>\n<callManagerGroup>\n<members>\n<member priority=\"0\">\n<callManager>\n<ports>\n<ethernetPhonePort>2000<\/ethernetPhonePort>\n<mgcpPorts>\n<listen>2427<\/listen>\n<keepAlive>2428<\/keepAlive>\n<\/mgcpPorts>\n<\/ports>\n<processNodeName><\/processNodeName>\n<\/callManager>\n<\/member>\n<\/members>\n<\/callManagerGroup>\n<loadInformation115 model=\"Cisco 7941\">SIP41.9-4-2SR1-1S<\/loadInformation115>\n<loadInformation307 model=\"Cisco 7911\">SIP41.9-4-2SR1-1S<\/loadInformation307>\n<loadInformation309 model=\"Cisco 7941G-GE\">SIP41.9-4-2SR1-1S<\/loadInformation309>\n<loadInformation435 model=\"Cisco 7945\">SIP45.9-4-2SR3-1S<\/loadInformation435>\n<loadInformation436 model=\"Cisco 7965\">SIP45.9-4-2SR3-1S<\/loadInformation436>\n<loadInformation437 model=\"Cisco 7975\">SIP75.9-4-2SR3-1S<\/loadInformation437>\n<loadInformation30006 model=\"IP Phone 7970\">SIP70.8-0-3S<\/loadInformation30006>\n<authenticationURL><\/authenticationURL>\n<directoryURL><\/directoryURL>\n<idleURL><\/idleURL>\n<informationURL><\/informationURL>\n<messagesURL><\/messagesURL>\n<servicesURL><\/servicesURL>\n<\/Default><\/pre>\n<\/li>\n
<DIALTEMPLATE>\n<TEMPLATE MATCH=\"*97\" TIMEOUT=\"0\"\/><!-- *97 (*VM for voicemail). Dial immediately -->\n<!--<TEMPLATE MATCH=\"*80,....\" TIMEOUT=\"0\" Tone=\"Bellcore-Stutter\"\/><!-- *80 (Paging and Intercom followed by Ext#). Dial immediately -->\n<TEMPLATE MATCH=\"....\" TIMEOUT=\"0\"\/><!-- 4 digits. Wait 1 second, then dial -->\n<TEMPLATE MATCH=\"*#\" TIMEOUT=\"0\" REWRITE=\"%1\"\/><!-- Dial Immediately After Pressing # -->\n<TEMPLATE MATCH=\"9,1..........\" TIMEOUT=\"0\" Tone=\"Bellcore-Outside\" \/><!-- call using 860.259.1966 line -->\n<TEMPLATE MATCH=\"9,..........\" TIMEOUT=\"0\" Tone=\"Bellcore-Outside\" \/><!-- call using 860.259.1966 line -->\n<TEMPLATE MATCH=\"9,27.........\" TIMEOUT=\"0\" Tone=\"Bellcore-Outside\" \/><!-- call using 27-1-500-4700 line -->\n\n<TEMPLATE MATCH=\"*\" TIMEOUT=\"3\"\/> <!-- Anything else. Wait 5 seconds, then dial -->\n<\/DIALTEMPLATE><\/pre>\n<\/li>\n
<CiscoIPPhoneRingList>\n<Ring>\n<DisplayName>Analog 1<\/DisplayName>\n<FileName>Analog1.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Analog 2<\/DisplayName>\n<FileName>Analog2.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Are You There<\/DisplayName>\n<FileName>AreYouThere.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Bass<\/DisplayName>\n<FileName>Bass.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Call Back<\/DisplayName>\n<FileName>CallBack.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Chime<\/DisplayName>\n<FileName>Chime.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Classic 1<\/DisplayName>\n<FileName>Classic1.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Classic 2<\/DisplayName>\n<FileName>Classic2.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Clock Shop<\/DisplayName>\n<FileName>ClockShop.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Drums 1<\/DisplayName>\n<FileName>Drums1.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Drums 2<\/DisplayName>\n<FileName>Drums2.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Film Score<\/DisplayName>\n<FileName>FilmScore.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Harp Synth<\/DisplayName>\n<FileName>HarpSynth.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Jamaica<\/DisplayName>\n<FileName>Jamaica.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Koto Effect<\/DisplayName>\n<FileName>KotoEffect.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Music Box<\/DisplayName>\n<FileName>MusicBox.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Piano 1<\/DisplayName>\n<FileName>Piano1.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Piano 2<\/DisplayName>\n<FileName>Piano2.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Pop<\/DisplayName>\n<FileName>Pop.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Pulse 1<\/DisplayName>\n<FileName>Pulse1.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Ring 1<\/DisplayName>\n<FileName>Ring1.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Ring 2<\/DisplayName>\n<FileName>Ring2.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Ring 3<\/DisplayName>\n<FileName>Ring3.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Ring 4<\/DisplayName>\n<FileName>Ring4.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Ring 5<\/DisplayName>\n<FileName>Ring5.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Ring 6<\/DisplayName>\n<FileName>Ring6.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Ring 7<\/DisplayName>\n<FileName>Ring7.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Sax 1<\/DisplayName>\n<FileName>Sax1.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Sax 2<\/DisplayName>\n<FileName>Sax2.raw<\/FileName>\n<\/Ring>\n<Ring>\n<DisplayName>Vibe<\/DisplayName>\n<FileName>Vibe.raw<\/FileName>\n<\/Ring>\n<\/CiscoIPPhoneRingList><\/pre>\n<\/li>\n
<CiscoIPPhoneMenu> \n<Prompt>thejimmahknows Directory<\/Prompt> \n<MenuItem> \n<Name>thejimmahknows Directory<\/Name> \n<URL>http:\/\/10.55.55.135\/directory\/directory.xml<\/URL> \n<\/MenuItem> \n<\/CiscoIPPhoneMenu><\/pre>\n<\/li>\n
<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n<CiscoIPPhoneMenu> \n<Title>thejimmahknows Extensions<\/Title>\n<Prompt>Dial<\/Prompt> \n<!--\n<MenuItem> \n<Name>Jim's Phone<\/Name> \n<URL>Dial:86555<\/URL> \n<\/MenuItem>\n--> \n<MenuItem> \n<Name>Jim's Office<\/Name> \n<URL>Dial:86444<\/URL>\n<\/MenuItem> \n<\/CiscoIPPhoneMenu><\/pre>\n
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\n<\/a>.<\/code><\/li>\n<\/ul>\n<\/li>\n<\/ul>\nAlmost There –DNS and DHCP server time.<\/h6>\n
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\n<\/li>\nip dhcp excluded-address 10.55.55.1 10.55.55.100\n!\nip dhcp pool dhcp-pool\n network 10.55.55.0 255.255.255.0\n domain-name local\n dns-server 10.55.55.1 \n default-router 10.55.55.1 \n option 66 ascii sip.local\n option 150 ip 10.55.55.135 \n lease 7\n!\n!\n!\nip domain retry 1\nip domain name local\n!\n!\nip host sip.local 10.55.55.135\nip dhcp-server 10.55.55.1<\/pre>\n<\/li>\n
Troubleshooting:<\/h3>\n
Checking Asterisk and watching SIP messages<\/h5>\n
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NOTICE[24039]: chan_sip.c:28091 handle_request_register: Registration from '<sip:86555@10.22.22.135>' failed for '10.55.55.101:5060' - Wrong password\nNOTICE[24127]: chan_sip.c:28091 handle_request_register: Registration from '<sip:86555@10.55.55.135>' failed for '10.55.55.101:59872' - Device not configured to use this transport type\nERROR[24039]: chan_sip.c:16899 register_verify: 'TCP' is not a valid transport for '86555'. we only use 'UDP'! ending call.<\/pre>\n
\nSecond one denotes the SIP Client isn’t set up with the correct Transport set for the Extension (UDP,TCP, or TLS).
\nCheck FreePBX configuration for this extension
\nThird, you can see why the Transport type failed. Only accepts UDP, not TCP for this extension.
\nGood responses:<\/p>\nNOTICE[24039]: chan_sip.c:23571 handle_response_peerpoke: Peer '86444' is now Reachable. (20ms \/ 2000ms)<\/pre>\n<\/li>\n<\/ol>\n
Checking\u00a0Cisco Phones<\/h5>\n
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