Learning IOS and JUNOS

There are two types of circuits an enterprise can use to route 911 calls to the proper PSAPs (public safety answering points) and deliver the 10-digit caller ID: ISDN PRIs or Centralized Automatic Message Accounting (CAMA) trunks. "ISDN is the wea port of choice from our perspective," says Guy Clinch, Avaya's government solutions director and a member of the National Emergency Number Association's PBX/multi-line telephone system technical subcommittee, because you can fit more phone lines in a PRI and assign each of those numbers to represent a separate ERL. In short, you can map your location and send more granular location information to the PSAP. But as many as 85% of enterprises choose instead to retrofit their IP PBXs using legacy analog CAMA trunks for entry into the public safety system, Clinch says. It's cheaper, because the CAMA trunk is equivalent to on phone number, so they get charged only once. The downside: CAMA trunks cannot deliver a custom caller ...

E&M-FGD The  e&m-fgd setting   allows   E&M interface connections for PBX trunk lines (tie lines) and telephone equipment to use Feature Group D switched-access service.  FGD-EANA FGD-EANA is a Feature Group-D (FGD) signaling protocol of type Exchange Access North American (EANA). This provides certain call services, such as emergency (USA-911) calls. FGD can accept ANI and DNIS for inward calls (Network to CPE), but can only provide DNIS for outward calls (CPE to Network). FGD-EANA can also provide ANI and DNIS for outward calls (Network to CPE), but cannot accept ANI for inward calls (CPE to Network). To provide and accept ANI and DNIS at the same time using CAS we can split the T1 into two different ds0 groups. One group is for inbound calls and one group is for outbound calls.This output shows an example:  ds0-group 0 timeslots 1-12 type e&m-fgd ds0-group 1 timeslots 13-24 type fgd-eana

以下是讓IP Phone註冊到Cisco Call Manager Express最基本的指令。其中有一行非常重要的指令就是 create cnf-files 。如果沒有這個指令，當IP Phone嘗試去註冊到CME時，CME將不會建立xml檔案來發放給IP Phone。這個問題在我之前嘗試要連到TP新加坡Remote Lab時就遇過了，VPN確定通了但是IP Phone就是無法完成註冊，結果請新加坡那邊的負責Remote Lab 的同事協助查了很久還是找不到問題，後來幸好有Chris Yang幫忙，終於找到Remote Lab HQ Router(CME)上缺少了這個config才把這個問題給解決。不過很奇怪的是…經過了幾個月之後，這個星期又請公司admin幫忙再跟新加坡借了一次CVOICE Remote Lab，結果有問題的initial config到現在還是有問題，唉，同時是華人，怎麼台灣跟新加坡的華人工作態度卻是大大不同。 telephony-service max-ephones 2 max-dn 2 ip source-address 172.16.1.1 port 2000 create cnf-files ! ! ephone-dn 1 number 1001 ! ! ephone 1 button 1:1

(USOC Wiring Diagram) Telephone wiring for a phone outlet is typically either 1, 2 or 3 pairs (2, 4, or 6 conductor). Most cable nowadays is UTP (unshielded twisted pair). There may be instances where you may need to connect to or transpose from the old "quad" cable. The diagram below provides the transposition between these standards. Pair 1 (T1 & R1) Usually the primary dial tone or talk circuit is wired to the center two pins (pins 3 & 4) and is the white/blue and blue/white pair (AKA: T1 & R1 - tip 1 and ring 1). A standard single line phone draws dial tone from these center pins. NOTE: The type of wiring shown here is known as USOC (pronounced U-sock). See background below. Pair 2 (T2 & R2) The secondary circuit is wired to the two pins (pins 2 & 5) directly to the side of the center pins and is the white/orange and orange/white pair (AKA: T2 & R2 - tip 2 and ring 2). Depending on the application, the secondary circuit can either be t...

1. 將網路線中斷 (如果不是使用Power Over Ethernet的話，請把電源線移除) 2. 按著 # 鍵不放 3. 接上網路線 (如果使用額外電源變壓器請重新接上電源線) 4. 等待IP Phone畫面出現 “Reset to factoy defaults” 提示 5. 放開 # 鍵並且依序按下以下電話按鍵 123456789*0# 6. 完成IP Phone重設動作

很多人聽過PBX之間交換的標準-QSIG，但是可能比較少人聽過DPNSS，事實上DPNSS也是另外一種跟QSIG類似的PBX交換標準，但是它們之間到底有那些不相同的地方及歷史上演進的結果為何? 請參閱下文Q&A： http://www.pqmconsultants.com/coexist.htm Q. What are DPNSS and QSIG? DPNSS and QSIG are inter-exchange signalling protocols, primarily intended for the interconnection of nodes in a Corporate telecommunication Network (CN). The interconnection of PABXs using leased circuits is a typical application. Both DPNSS and QSIG are common channel signalling systems based on ISDN technology. They are open standards; that is, they permit signalling between equipment from different vendors. Q. How did DPNSS come about? The development of DPNSS commenced in 1981 with the decision by the UK telecomms industry (British Telecom, as was, and a number of PABX manufacturers) to develop a vendor independent private network signalling system. This work resulted in the protocol that is today widely used throughout the UK and elsewhere. The drivers behind DPNSS development are well known. They were...

A method of compression used in speech encoding that transmits a data packet in place of silence at a compressed rate. SID mathematically describes the background noise during a session so that the receiving end can artificially reproduce that noise during silence periods. Transmitting the SID requires almost no bandwidth as transmitting the actual background noise does.

SIP通過類似E-mail形式的資源識別標誌（URI）來標名用戶地址，它通過諸如用戶電話號碼、帳號、主機名等元素來構成SIP URI，其格式為user@domain的表示方式。其中user也可以是傳統電信網路中的e.164電信交換碼。  SIP關鍵元件及其呼叫模式。 SIP的關鍵元件有下列幾項：  用戶代理User Agent：   通常簡稱為UA，是SIP網路環境中的用戶終端設備，其角色相當於H.323 Terminal。在邏輯上包含有User Agent Client （UAC） 以及 User Agent Server（UAS）兩種，UAC負責產生請求，而UAS負責產生依照請求產生應答。每一個UA都同時扮演者UAC和UAS的角色，當它是呼叫別人的主叫端時，就是UAC；當它是被別人呼叫的被叫端時，就是UAS。  目前我們所能看到的各種話機，本質上都是一種SIP UA裝置。有一種USB Phone是配合Soft Phone使用的，它的本質是一種音效裝置，雖然很多人也叫它做網路話機，但是它並不屬於SIP UA的角色。  代理伺服器Proxy Server：   為SIP協議運作的中心，同時具有伺服器端和客戶端雙重角色的中介元件，負責代表SIP UA或者其他的Proxy Server產生請求或將收到的請求代為轉送到另外一個目標SIP元件去。由Proxy Server提供對用戶定位的服務，以轉送到正確的UA位置去，且UA回覆結果也是一樣會經由相反的路由將結果回覆給請求端的UA，這就是Proxy Server的路由功能。  Proxy Server其實就是扮演傳統電信領域中，交換總機的角色。由於它的存在，可大幅簡化UA的設計複雜度（否則UA要能記得所有通訊對象的IP網址），也是VoIP業者營運的中樞。  重定向伺服器Redirect Server：   SIP的其中一個主要特性就是，它將用戶的邏輯位址和實際位址分開，這使得用戶可以定義一個不變的邏輯位置，然後將它映射成別名至一個或多個變化的實際位置。重定向伺服器接受任何SIP元件的請求，並將被呼叫方的SIP位址映射成一個或多個位址並將回應給客戶端。和代理伺服器不同的是，重定向伺服器不會轉遞任何請求到其他伺服器。  註冊伺服器Register Server：   接受註冊請求的伺服器，其目的是記錄用戶在請求中的聯繫資訊，或更...

Tail-End Hop-Off (TEHO) allows a company to reduce its long-distance toll charges. When remotes sites are connected by an IP WAN, you can route calls that are bound to those cities—and those sites—over the WAN. The terminating gateway then routes them out to the PSTN as a local call. This sounds like a good idea in theory, but it can be complex in practice. You need a separate dial peer for each area code/prefix combination in each remote location. Large cities have many prefixes and might have several area codes also. Small cities or regions might have some prefixes within the same area code that are local calls, and some that are long-distance calls. You might need a gatekeeper for an extensive TEHO implementation. Regulatory issues might curtail the use of TEHO, also. TEHO results in a loss of revenue for telecommunications companies, so some countries regulate calls carried across country or regional borders. Careful planning and research on the most recent laws concerning mixed Vo...

Gateway(config)#dial-peer hunt Dial-peer hunting choices, listed in hunting order within each choice: 0: Longest match in the phone number, explicit preference, random selection. (This is the default hunt-order number.) 1: Longest match in the phone number, explicit preference, least recent use. 2: Explicit preference, longest match in the phone number, random selection. 3: Explicit preference, longest match in the phone number, least recent use. 4: Least recent use, longest match in the phone number, explicit preference. 5: Least recent use, explicit preference, longest match in the phone number. 6: Random selection. 7: Least recent use.

Class of Restrictions (COR) is a Cisco voice gateway feature that enables Class of Service (COS) or calling privileges to be assigned. It is most commonly used with Cisco Survivable Remote Site Telephony (SRST) and Cisco CallManager Express but can be applied to any dial peer. The COR feature provides the ability to deny certain call attempts based on the incoming and outgoing CORs provisioned on the dial-peers. COR is required only when you want to restrict the ability of some phones to make certain types of calls but allow other phones to place those calls. COR is used to specify which incoming dial-peer can use which outgoing dial-peer to make a call. Each dial-peer can be provisioned with an incoming and an outgoing COR list. The corlist command sets the dial-peer COR parameter for dial-peers and the directory numbers that are created for Cisco IP phones associated with the Cisco CallManager Express router. COR functionality provides the ability to deny certain call attempts on th...

DTMF tones are the tones that are generated when a telephone key is pressed on a touchtone phone. Sometimes the called endpoint needs to hear those tones, such as when you enter digits during the call in response to a menu. Low-bandwidth codecs can distort the sound, however. DTMF relay allows that tone information to be reliably passed from one endpoint to the other. By default, SIP uses in-band signaling, sending the DTMF information in the voice stream. However, you can configure it to use RTP-NTE, SIP INFO messages, SIP NOTIFY messages, or KPML for transmitting DTMF tone information. RTP-NTE is an in-band DTMF relay method, which uses RTP Named Telephony Event (NTE) packets to carry DTMF information instead of voice. If RTP-NTE is configured, SDP is used to negotiate the payload type value for NTE packets and the events that will be sent using NTE. RTP-NTE can cause problems communicating with SCCP phones, which use only out-of-band DTMF relay. In a CallManager 4.x network with SC...

A Media Termination Point (MTP) software device allows Cisco CallManager to relay calls that are routed through SIP or H.323 endpoints or gateways. MTP, a Cisco software application, installs on a server during the software installation process. You must activate and start the Cisco IP Voice Media Streaming App service on the server on which you configure the MTP device. For information on activating and starting services, refer to the Cisco CallManager Serviceability Administration Guide. Each MTP device that is defined in the database registers with the Media Resource Manager (MRM). The MRM keeps track of the total available MTP devices in the system and of which devices have available resources. During resource reservation, the MRM determines the number of resources and identifies the media resource type (in this case, the MTP) and the location of the registered MTP device. The MRM updates its share resource table with the registration information and propagates the registered infor...

Super Group 3 (SG3) is a standard of fax machines that support speeds of up to 33.6 kbps through V.34 half duplex (HD) modulation and V.8 signaling. Prior to Cisco IOS Release 12.4(4)T, SG3 fax machines could interoperate only over T.38 fax-relay and Cisco fax-relay networks with G3 fax machines, not with other SG3 fax machines, unless the fax machines were specifically configured to work at slower speeds or were configured for modem pass-through. The use of SG3 V.8 fax CM message suppression provides a gateway-controlled solution that enables SG3 fax machines to scale down without end-user interaction and without the extra bandwidth required by modem pass-through. SG3 V.8 fax CM message suppression allows SG3 fax machines to interoperate over a fax-relay network at G3 speeds by blocking the SG3 V.8 CM message, or fax tone, from reaching the called fax machine. This causes the called fax machine to time out on the ANSam tone and scale down to G3 speeds by initiating V.21 negotiations.

Error correction mode (ECM) is an optional transmission mode built into Class 1 fax machines or fax modems. ECM automatically detects and corrects errors in the fax transmission process that are sometimes caused by telephone line noise. The page data is divided into what is known as Octets (small blocks of data). Once the receiver has received all the Octets it examines them (using check-sums) and then advises the transmitting fax of any Octets that are in error. The transmitter then need only resend the blocks in error rather than the whole page. This generally means an ECM coded fax will be more likely to succeed on a noisy line. ECM is the norm rather than the exception. Some fax machines have the capability to enable or disable this function.

The Drop and Insert (D&I) feature allows DS0 timeslots to be taken off one T1 interface and inserted into time slots of the other T1 interface. This feature is available in VIC and WIC applications. Drop and Insert functionality does not support different framing and line coding on the two ports. Therefore, when a tdm-group is configured on the controller T1 or E1, the framing type between the two controllers must be the same. This is only for the tdm-group functionality of the VWIC card. Note:  If you do configure two different framing types, this is the error message that the IOS sends to the console of the router: Voice_Router (config)#connect TDM t1 0/1 t1 0/2 %CONN TDM: Framing type mismatch %CONN TDM: Endpoints are incompatible %CONN: Invalid Command Drop and Insert timeslots do not need to be contiguous. Drop and Insert of timeslots must be on the T1 controllers on the same 2-port VWIC, unless the gateway is Multiservice Interchange (MIX) enabled. When a g...

When a group of PRI interfaces are effectively bundled together, one D-channel can be used for the signaling data of all the combined B-channels, while the redundant D-channels can be used for data transmission. NFAS is only possible with a T1 PRI. ISDN PRI NFAS是一種共用ISDN D channel(64 kb/s)的技術，而且只適用於T1 PRI。在NFAS中至少要設置一個Primary D channel，Backup D channel則是可有可無，如果有的話通常會將Primary & Backup D channel設置於不同的T1 controller上。假設在兩條T1上共有24 * 2 channels，我們可以只使用一條channel當共用的D channel來傳送控制訊號，剩下來的23 + 24 channels就可以當成B channels(64 kb/s)來傳送Data。

當我們在設定VoIP Gaetway時，我們必須要注意Router是如何match撥號內容來決定利用那一個dial-peer(很多人會誤解第一個檢查的是destination-pattern，事實上不然)，甚至inbound和outbound dial peers也是完全不同的match rule： Match Inbound Dial Peers Rule(by full string received in the setup request) 1. incoming called-number ： 首先Router或Gateway會嘗試去match call setup請求中被撥打的電話號碼(called number)，因此會先檢查dail-peer中是否有設定 incoming called-number 參數。 2. answer-address ： 如果dial-peer中沒有 incoming called-number 參數符合被撥打的電話號碼(called number)，Router或Gateway會嘗試去match call setup請求中的來電電話號碼(calling number)是否有符合某個dial-peer中設定的 answer-address 參數。 3. destination-pattern ： 如果dial-peer中沒有answer-address參數符合來電電話號碼(calling number)，Router或Gateway會嘗試去match call setup請求中的來電電話號碼(calling number)是否有符合某個dial-peer中設定的 destination-pattern 參數。 4. port ： 如果incoming call setup請求來自於特定的voice port，利用這個voice port number來match inbound call leg中dial-peer設定的port參數。 5. Fist dial-peer added to the configuration(不是dial peer number最小的喔)： 如果有多個dial peers設定了相同的voice port，Router或Gateway會match第一個加入到configuration中的dial...

G.Clear guarantees bit integrity when transferring a DS-0 through a gateway server, supports the transporting of nonvoice circuit data sessions through a Voice over IP (VoIP) network, and enables the VoIP networks to transport ISDN and switched 56 circuit-switched data calls. With the availability of G.Clear, ISDN data calls that do not require bonding can be supported. In a transit application, because it is possible to have a mix of voice and data calls, not supporting G.Clear limits the solution to voice-only calls. The end-user application is in charge of handling packet loss and error recovery. This packet loss management precludes the use of clear channel with some applications unless the IP network is carefully engineered. In an MGCP environment, the voice gateway backhauls the public switched telephony network (PSTN) signaling channel to the call agent. The call agent examines the bearer capability and determines when a G.Clear call should be established. ----------------------...

Intra-Cluster Communication Signaling (ICCS), which provides the communications with the Cisco CallManager Service process that is at the heart of the call processing in each server or node within the cluster. The intra-cluster traffic between the servers consists of the following: Database traffic from the IBM Informix Dynamic Server (IDS) database that provides the main configuration information. The IDS database is replicated from the publisher server to all other servers in the cluster using best-effort. The IDS traffic may be re-prioritized in line with Cisco QoS recommendations to a higher priority data service (for example, IP Precedence 1 if required by the particular business needs). An example of this is extensive use of Extension Mobility, which relies on IDS database configuration. Firewall management traffic, which is used to authenticate the subscribers to the publisher to access the publisher's database. The management traffic flows between all servers in a cluster....