Cisco Sip Ua Timers

I can make outbound calls but I don't receive inbound calls Here is a copy of my debug voice ccapi all and debug ccsip and run config debug voice ccap 121605. To set how long the Session Initiation Protocol (SIP) user agent (UA) waits before retransmitting a Notify message, use the timers notify command in SIP user-agent configuration mode. Manages SIP-H. com BRKUCC-2932. On the sip side, you may have to use a buffer invite timer to make it appear as though this calling name comes in immediately. mod_event_socket is a TCP based interface to control FreeSWITCH. For more information about defaults and usage guidelines, see the corresponding chapter of the Network Protocols Command Reference, Part 1. Enterprise Video Session for CiscoLive 2013 2 UA – User Agent. But here the call is a direct SIP Call to the Cisco router. These could also be configured on a DNS Server however they are not as flexible. This guide will help you get your Cisco CUBE connected to SIP. As to why there is no ack, you need to check your network and the sip server you are using. They have a tested configuration document using CUBE with their SIP service. Or why not check out our library of useful tips and tutorials on everything you need to know about hair removal and find out how to get the most out of your Braun products. Asterisk 13 Configuration_res_pjsip. ms at the time of this writing. Hi I am having great difficulty integrating EX2k7 SP1 with CME7. Tips and Tricks. You can use Go, ruby, java, nodeJS, or PHP instead as well. registrar 1 dns:gw1. 3 standard is used for H. RFC 3261 SIP: Session Initiation Protocol June 2002 The first example shows the basic functions of SIP: location of an end point, signal of a desire to communicate, negotiation of session parameters to establish the session, and teardown of the session once established. Clone via HTTPS Clone with Git or checkout with SVN using the repository’s web address. Dolly AT&T Labs November 2006 A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML) Status of This Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. I will point out one more thing: it may be that the UAC only wishes to refresh the timer. Personally, I thought they were. OR under the dialpeer we can setup probes so in case CUCM is down, the probe will shutdown the dialpeer. On the sip side, you may have to use a buffer invite timer to make it appear as though this calling name comes in immediately. retry response 3. COMMSCOPE Universal SnapIn Adapter for mounting 1/2"-1-5/8" Snap In hangers to angle or round tower members. With Cisco IOS gateways, dial-peers are used to match phone numbers, and the destination can be a SIP Proxy Server, DNS SRV, or IP address. Steve Blair (May 2005 (November 2004) Overview. In this article, we will see that how to configure two node Redhat cluster using pacemaker & corosync on REHL 7. The Challenge. What is an ASN or AS? An Autonomous System Number (AS number or just ASN) is a special number assigned by IANA used primarilly with Border Gateway Protocol which uniquely identifies an network under a single technical administration that has a unique routing policy, or is multi-homed to the public internet. The following article will show you how to program your remote to control your TV or another device like a DVD player or audio system. The Remote UA initiates\connects to a call using the CUCM. This enhancement allows the network operator to configure the time interval at which a peering session is reestablished by a router when the number of prefixes that have been received from a peer has exceeded the maximum prefix limit. Troubleshooting SIP with Cisco Unified Communications BRKUCC-2932 Paul Giralt Distinguished Services Engineer [email protected] Provided by: Staffan Kerker. 60 expires 3600 refresh-ratio 100 tcp. Cisco CUCM/CUBE SIP Troubleshooting dkuchenski July 21, 2015 0. description **Outgoing Calls to SIP. Configuring a SIP GW - UA. Not all HTTP/1. From advanced smart metering technology to renewables, we have the solutions, services and technology to bring your grid into the modern digital age. Step 2 - specify the parameters for the SIP service and bind to interface session transport [ UDP | TCP]. Clone via HTTPS Clone with Git or checkout with SVN using the repository’s web address. retry invite 2. Each user agent (UA) performs the function of a user agent client (UAC) when it is requesting a service function, and that of a user agent server (UAS) when responding to a request. 18 posts They use shortlesky cloud IP telephony service and use Cisco 7960 IP phones. show sip-ua register status - It will show SIP Registration information show voice dsp - It will show the status of all the DSPs on the Gateway show ccm-manager - It will show information about the active and redundant configured Cisco Unified Communications Manager. Providing Cisco CME Support For SIP : SIP Trunk Features. timers connect 100. session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua authentication username 100002 password 820616 sip-server dns:proxy. What is SIP? Why SIP over T1/PRI? SIP Trunking Cisco SIP End points. retry response 3. Configuration information for a Cisco 2811 running IOS IP Voice 12. Session Timer in the Session Initiation Protocol Speaker Ying Shun Lin Adviser Quincy Wu. Cisco FXO gateway must forward incoming calls to IP22. SIP UA Configuration. The following example shows a Cisco IOS gateway configuration to send calls to a SIP Proxy Server using the SIP Proxy's IP address. RIP Commands. Troubleshooting SIP with Cisco Unified Communications Paul Giralt, Distinguished Services Engineer (@PaulGiralt) [email protected] Combined download should be used where phones may be running pre-4. NET ip name-server primaryDNS ip name-server secondary DNS ip name-server Tertiary DNS voice service pots fax rate vo Help understanding config for Verizon SIP/ IP Trunking (voip) - Cisco - Tek-Tips. The VoIP signaling protocol used is SIP. edu and Configuring Cisco 2620XM PSTN Gateways a Proxy Serve r (draft). 60 expires 3600 refresh-ratio 100 tcp. This document provides typical configuration examples for interoperation between Huawei switches and mainstream IP phones, Cisco ISE authentication servers, Cisco ACS authentication servers, Aruba ClearPass authentication servers, Microsoft NLB servers, multi-NIC servers, and Cisco switches. cloverhound. 8 incoming called-number. Right now my SIP trunk goes to SiSky PE (Skype Gateway) which connects to Skype allowing me to make outgoing and receiving incoming calls. Steve Blair (May 2005 (November 2004) Overview. 18 posts They use shortlesky cloud IP telephony service and use Cisco 7960 IP phones. retry response 3. SIP-GW#show sip-ua service SIP Service is up The show sip-ua statistics command provides statistics on each type of method and response, errors, and total SIP traffic information. Cisco Call Manager Express - SIP/SCCP Configuration. The script finds out the different type of web browsers used by your customers. Our team's dedication to AC/DC power supplies is certain to help you find a reliable power supply quickly from TRC. br no remote-party-id retry invite 5 retry response 3 retry bye 5 retry cancel 5 retry prack 5 retry notify 4 retry register 5 retry options 5 timers connect 100 timers connection aging 30 timers. Router(config)#sip-ua Router(config-sip-ua)#no transport udp Router(config-sip-ua)#end. Cisco CUCM/CUBE SIP Troubleshooting dkuchenski July 21, 2015 0. But here the call is a direct SIP Call to the Cisco router. 40 2 CUCM 8. Ring back works fine on imcoming PSTN calls and on-net calls, but when calling OUT to the PSTN (via sip trunk on voip. 10 authentication username XXXXXXXXXX password XXXXXXXXXX no remote-party-id retry invite 4 retry response 3 retry bye 2 retry cancel 2 retry register 10 timers connect 100 timers register 100 registrar dns: expires 3600. The 86UT640S0UA is a 86” display TV that has SuperSign control with embedded content management. 12, playing ivr-on_hold_indefinitely. sip-ua max-forwards 15 retry invite 3 retry response 3 retry bye 6 retry cancel 3 timers trying 1000 sip-server ipv4: IP. no timers notify. The show sip-ua status command can be useful in troubleshooting, also. After a packet capture, it was very apparent the issue was the NAV timer. Why Use Cron Jobs? Server admins have been using cron jobs for a long time. Creating and Modifying SIP Profiles. You can also change the period that the Cisco IOS SIP gateway waits for a SIP 100 response to a SIP INVITE request by using the command timers trying under the sip-ua configuration. 6 working in a lab environment. RFC 3261 SIP: Session Initiation Protocol June 2002 The first example shows the basic functions of SIP: location of an end point, signal of a desire to communicate, negotiation of session parameters to establish the session, and teardown of the session once established. 6 have SIP trunks connected to provider server as IP Trusted list (not as register SIP trunks) and have internal SIP devices (Apple iphone and ipad) which require registration to this cme as local extensions. TCP starts a retransmission timer when each outbound segment is handed down to IP. There are other aspects of SIP timing that I will address in later blogs, but understanding T1, Timer B and Timer F are crucial to becoming a SIP guru. The SIP Forum’s User Agent Configuration Recommendation for the locating, retrieving and maintaining of SIP User Agents has been ratified and published as RFC 6011 by the Internet Engineering Task Force (IETF). My timers are default: show sip-ua timers SIP UA Timer Values (millisecs unless noted) trying 500, expires 180000, connect 500, disconnect 500 prack 500, rel1xx 500, notify 500, update 500 refer 500, register 500, info 500, options 500, hold 2880 minutes, registrar-dns-cache 3600 seconds tcp/udp aging 5 minutes tls aging 60 minutes. 323 endpoint registered on Cisco VCS-E with H. The Challenge. com expires 60 sip-server dns:proxy. This includes a Supported header field with the option tag 'timer', indicating support for this extensi. Hi, How can a UA publish its capabilities of supporting multiple packetization time support for a particular codec, For example: For codec L8 if it supports both 20 and 30 ms ptime values, For codec L16, it supports 40 ms Is following media line is allowed?. retry register 10. SIP (Session Initiation Protocol) is a signaling protocol, widely used for setting up, connecting and disconnecting communication sessions, typically voice or video calls over the Internet. ip address trusted call-block cause not-in-cug gcid clid substitute name allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip session refresh e911 transport switch udp tcp asserted-id ppi localhost dns:dns. In Delegate and Team-call settings the "ring after" timer can only be set to 0, 5, 10 or 15 seconds, whereas in the Client you can set it from zero to 55 seconds (the maximum value is actually is whatever the unanswered call timer is, minus 5 seconds, which is a maximum of 55 seconds). 7 ! line con 0 transport output telnet line aux 0 transport output telnet line vty 0 4. Here's some reference commands for an Alcatel-Lucent OmniPCX, I just want to share with you, and also because i'm always forgot something that i've learned. !Configure SIP user agent sip-ua. The below example is for a special Ring Tone (ringer14) called Splash to be played on an incoming call when the Alert-Info fields from INVITE requests contain SpecialRing in the SIP Invite. I have a small project coming up and getting refreshed with the Cisco Autonomous Access Points Using the MODE Button. Ensure port 5060 is open to those IP address under 2. The UT640 Series offers superior image quality and colors. Cisco 7960 has problem with music on hold. The Remote SIP UA sends uses the UPDATE method for session timer refreshs. Shop Overstock. Step 3 Use the show sip-ua register status command to show the status of local E. Step 2 - specify the parameters for the SIP service and bind to interface session transport [ UDP | TCP]. That's not a good idea to have that timer that low. Password requirements: 6 to 30 characters long; ASCII characters only (characters found on a standard US keyboard); must contain at least 4 different symbols;. The schedule resides in a configuration file named "crontab". It cannot be set using the CISCO-SIP-UA-MIB. sip-ua authentication username MyUsername password MyPassword no remote-party-id retry invite 2 retry register 10 retry options 0 timers connect 100 registrar dns:voip. This chapter describes the function and displays the syntax for Routing Information Protocol (RIP) commands. com reason-header override ! The IP Host commands at the top are all for configuring DNS SRV for targeting 2 sip proxies. timer receive-rtp 1200! sip-ua retry invite 3 retry response 3. • Configurations specific to sip user agent are under sip-ua. SIP is a standardized protocol with its basis coming from the IP community and in most cases uses UDP or TCP. SPA 900 Series IP Phone pdf manual download. timers connect 100. Cisco SIP firmware doesn't support the MWI RFC; fully. Zoom out and see the bigger picture, or focus in on an unprecedented level of granular data. We want to capture an egress call from Cisco Unified Communications Manager (CUCM) and a pseudo ITSP. Using SIP EO isn't required when setting up a SIP call, we are just using this as a parameter in our setup because, well, it is a more. allow-connections sip to sip sip registrar server expires max 600 min 60 voice class codec 1 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 g711ulaw codec preference 4 g711alaw voice register global system message SRST service active max-dn 200 max-pool 10! voice register pool 1 id network 10. This will cause SIP to only listen on the internal interface, which may assist in limiting the exposure of this vulnerability:. Open a web page to login to CUCM administration using CUCM IP address. SIP FOR TELEPRESENCE AND VOLTE Delivering next-generation unified communication services includes packaging voice and video into rich communication suites (RCS) of application. Or why not check out our library of useful tips and tutorials on everything you need to know about hair removal and find out how to get the most out of your Braun products. com)とB(biloxi. Having a weird issue with outbound ringback on the calling phone. With regards to the following timers. com Page 2 of 11 Pre-Install Checklist If client is using a third party to perform the installs, please provide their contact information below:. 11 specifications. Because of this the mechanism for targeting multiple SIP Proxies is by using DNS SRV records. SIP sets up and manages media sessions (typically RTP for voice) over IP, operating in a request-response model. Troubleshooting SIP with Cisco Unified Communications Paul Giralt, Distinguished Services Engineer (@PaulGiralt) [email protected] After a packet capture, it was very apparent the issue was the NAV timer. I am sending Register invites but am not receiving anything back. Ring back works fine on imcoming PSTN calls and on-net calls, but when calling OUT to the PSTN (via sip trunk on voip. com [email protected] In Understanding SIP Timers Part I, I explained the basics of T1, Timer B, and Timer F. ex host-registrar 3. How to Configure SIP Timer; Information About SIP Timer. But since the target audience of this article is web developers, let's look at a few use cases of cron jobs that are relevant in this area:. sip-ua credentials username N1234567R password ITSPPassword realm exampledomain. sip-ua authentication username password 7 no remote-party-id retry invite 2 retry register 10 retry options 1 timers connect 100 registrar dns: expires 3600 sip-server dns: host-registrar dial-peer voice 1 voip description SIP TRUNK TO PROVIDER. The Session Initiation Protocol (SIP) UPDATE Method Status of this Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. This post will introduce a new type of DMVPN – FlexVPN, unofficially called “DMVPN phase 4”. com)とB(biloxi. 460 traversal capability. You can change the SIP INVITE retry attempts under the sip-ua configuration by using the command retry invite. Versions of browser can also be identified by using this php script. Syntax Description. timers notify time. MagicJack+ Power On sequence SIP and RTP traffic generated by power on the MagicJack+. And before it was done sending the 6 sip invites, the ISDN timers from the service provider had expired so the Cisco voice gateway never got to the point of sending the 6 sip invites before trying to reached the cluster using the secondary dial-peer. 250:5060 ! This is the SIP transaction Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2. 2 Set the Destination Address To send and receive an INVITE between the CUCM and Brekeke SIP Server, set Brekeke SIP. It also keeps the connection to SIPTRUNK alive by sending a registration message about every 60 seconds. com Page 2 of 11 Pre-Install Checklist If client is using a third party to perform the installs, please provide their contact information below:. Our Bulletin 1766 MicroLogix™ 1400 Programmable Logic Controller Systems build upon critical MicroLogix 1100 features: EtherNet/IP™, on-line editing and a built-in LCD panel. Create a Long Distance Dialplan. 8 incoming called-number. credentials username username password your-password realm gw1. I would like to know in case we can delay the Invite sent to the SFB client from a Lync 2013 server. 85 3 Cisco 2800 Integrated Service Router 192. 1 port 8021 and the default password is ClueCon. The SIP Forum’s User Agent Configuration Recommendation for the locating, retrieving and maintaining of SIP User Agents has been ratified and published as RFC 6011 by the Internet Engineering Task Force (IETF). 60 expires 3600 refresh-ratio 100 tcp. Timers buffer-invite 3000. In this case Innovaphone IP22 is used as FXS media gateway with a Cisco 1700 Router to realize an analog PBX extension over an IP network. Ring back works fine on imcoming PSTN calls and on-net calls, but when calling OUT to the PSTN (via sip trunk on voip. Symptom: Cisco Unified Call Manager (CUCM) disconnects SIP calls after a set period of time. This particular configuration is specific to the CUBE, should not be used on CME, and negates the need for any special callerID configuration. What you need to do is get with your ATT rep and have them send you their ATT Cisco CUBE SIP trunking configuration guide. Protecting lives, buildings and assets is our aim. Each user agent (UA) performs the function of a user agent client (UAC) when it is requesting a service function, and that of a user agent server (UAS) when responding to a request. It enables organizations to make the right engineering or sourcing decision--every time. 422 Session Interval Too Small €€€It is generated by a UAS or proxy when a request contains a Session-Expires header field with a duration below the minimum timer for the server [RFC4028] 423 Interval Too Brief. This enhancement allows the network operator to configure the time interval at which a peering session is reestablished by a router when the number of prefixes that have been received from a peer has exceeded the maximum prefix limit. Step 1 - enable sip on GW voice service voip sip. com reason-header override ! The IP Host commands at the top are all for configuring DNS SRV for targeting 2 sip proxies. Adjusting Cisco IOS SIP Invite Timers and Retry Values (sip-ua) for Failure Recovery Posted: 9th February 2011 by Mark in Cisco 1 Cisco IOS contains a configurable object called SIP User Agent (sip-ua) where several different SIP settings may be altered from their default behavior. com expires 60 sip-server dns:proxy. Bosch empowers you to build a safer and more secure world. retry invite 2. timer receive-rtp 1200! sip-ua retry invite 3 retry response 3. 7:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw ! gateway timer receive-rtp 1200 ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10. voice-class codec 1 voice-class sip early-offer forced dtmf-relay rtp-nte!! gateway timer receive-rtp 1200! sip-ua retry invite 2. Here an example of configuration of Cisco VG224 using SIP as signaling protocol and which is connected to a CUCM via a SIP Trunk. CCIE Collaboration Lecture Chapter 4. Conditions: Remote SIP User Agent (UA) is configured to use SIP Session Timer Feature. credentials username username password your-password realm gw1. Replace the IP address shown with that of the DuVoice system. Configure a SIP-User Agent (SIP-UA) Account to register with Asterisk: -Configure CME to act as a SIP-UA and set the account username/password that will be used to register with the Asterisk Server. timers notify time. It enables organizations to make the right engineering or sourcing decision--every time. Posts about Uncategorized written by ngageby. 23 context=default insecure=yes canreinvite=yes As you can see I have identified it by its IP address and here is the message I get when I debug sip messages from to and from the Ciscol. Configuration information for a Cisco 2811 running IOS IP Voice 12. How to reset a Cisco AP using the MODE button. 120 voice-port 2/0/10 supervisory disconnect dualtone mid-call input gain 10 cptone AR timeouts call-disconnect 1 timeouts wait-release 1 timing hookflash-out 50 connection plar 8000 impedance complex2 description COPACO-495211 caller. SIP is a standardized protocol with its basis coming from the IP community and in most cases uses UDP or TCP. edu and Configuring Cisco 2620XM PSTN Gateways a Proxy Serve r (draft). 23 context=default insecure=yes canreinvite=yes As you can see I have identified it by its IP address and here is the message I get when I debug sip messages from to and from the Ciscol. Password: cisco user name : 3006 Domain> 10. Scenario 1: SIP-to-SIP Configuration Network System Configuration - Sip / Sip Configuration Network Addresses Device # Device Make, Model, and Description Device IP Address 1 OpenText RightFax 192. You can also change the period that the Cisco IOS SIP gateway waits for a SIP 100 response to a SIP INVITE request by using the command timers trying under the sip-ua configuration. [cisco] type=friend port=5060 host=192. Considering a CUBE SIP integration was a task I had performed many times with service providers in the US, I thought it would be a walk in the park. Content Library - - Click on the file types below to dowload the content in that format. Multiple SIP providers in CUBE forced dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify codec g711ulaw no va realm provider2. Please hold while I try that extension. The script finds out the different type of web browsers used by your customers. Specifically, we want to use Session Initialization Protocol (SIP) "Early Offer" (EO) to establish the call. Even your approximate physical location can be looked up by anyone or any website if they know your IP address. The network elements that use the Session Initiation Protocol for communication are called SIP user agents. sip-ua timers connection aging 60 registrar 1 ipv4:10. 2 expires 3600 port 5060 transport udp notify telephone-event max-duration 3000!!. My SIP trunk is configured with a destination pattern of 555. Configuring Cisco Media Gateway. Session Timers has nothing to do with SDP, it has to do with re-INVITE's > and UPDATE's. Adjusting Cisco IOS SIP Invite Timers and Retry Values (sip-ua) for Failure Recovery Posted: 9th February 2011 by Mark in Cisco 1 Cisco IOS contains a configurable object called SIP User Agent (sip-ua) where several different SIP settings may be altered from their default behavior. South Carolina Cisco User Group. com expires 3600 sip-server dns: mycompany. One of the biggest problems with SIP clients soft or hardware based , involves with the SIP registrations. Our Bulletin 1766 MicroLogix™ 1400 Programmable Logic Controller Systems build upon critical MicroLogix 1100 features: EtherNet/IP™, on-line editing and a built-in LCD panel. SIP-UA — Configure SIP Dial-peer timers. In this case Innovaphone IP22 is used as FXS media gateway with a Cisco 1700 Router to realize an analog PBX extension over an IP network. 2 set poll-interval 10 set cost 0. timer receive-rtp 1200! sip-ua retry invite 3 retry response 3. Here an example of configuration of Cisco VG224 using SIP as signaling protocol and which is connected to a CUCM via a SIP Trunk. I am thinking that the gateways times out waiting for the ack back for the 200 OK after the called party picks up. Note In these scenarios, no RTP packets or RTCP packets are received for a call. The office has a local SIP gateway with PRI for calling and to support SRST On the Cisco Unified Communications Manager, a device pool is created for this site with media resources that are local to Cisco Unified Communications Manager A SIP trunk that requires an MTP connects to the gateway Users. retry response 3. This is a mini Howto, to configure Nfsen in OSSIM server, to monitor Cisco Routers. Also for: Ucm6102, Ucm6104, Ucm6108, Ucm6116. RFC 3261 SIP: Session Initiation Protocol June 2002 The first example shows the basic functions of SIP: location of an end point, signal of a desire to communicate, negotiation of session parameters to establish the session, and teardown of the session once established. Posts about Uncategorized written by ngageby. The IP Host commands at the top are all for configuring DNS SRV for targeting 2 sip proxies. session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua authentication username 100002 password 820616 sip-server dns:proxy. cloverhound. Program Remote to TV or Other Device. Scenario 1: SIP-to-SIP Configuration Network System Configuration - Sip / Sip Configuration Network Addresses Device # Device Make, Model, and Description Device IP Address 1 OpenText RightFax 192. On Friday 30 August the University of Sydney will recognise Wear it Purple Day for the 6th time with the goal of promoting a safe and inclusive environment for all of our ‘rainbow’ staff and students. This includes a Supported header field with the option tag 'timer', indicating support for this extensi. That would be easier to read. 11 specifications. session target sip-server dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua credentials username 100001 password 1357924680 realm sip-ua. Cisco CUCM/CUBE SIP Troubleshooting dkuchenski July 21, 2015 0. Actual data throughput and wireless coverage will vary. SIP-UA — Configure SIP Dial-peer timers. ” —Jessica. We will go through the basic building blocks of Cisco FlexVPN DMVPN and some of the design best practices for a typical enterprise WAN network. In Understanding SIP Timers Part I, I explained the basics of T1, Timer B, and Timer F. This page provides an overview of the main steps that are required to configure Cisco Media Gateway. I will point out one more thing: it may be that the UAC only wishes to refresh the timer. Service Providers use Session Initiation Protocol (SIP) to converge these applications into simple, cost-effective RCS. The "Supported" keyword indicates the SIP features we support, in here we have timer refreshes and a few other bits and bobs. NET ip name-server primaryDNS ip name-server secondary DNS ip name-server Tertiary DNS voice service pots fax rate vo Help understanding config for Verizon SIP/ IP Trunking (voip) - Cisco - Tek-Tips. The T46S is simple/straightforward to configure and some of its most notable features include HD Audio, Gigabit Ethernet, PoE (Power over Ethernet) support, and a color touch screen. NOTE: This depends on your SIP Server's ability to modify the SIP INFO Header!