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ATL IP300S Скачать руководство пользователя страница 65

IP SIP Phone v2

User’s Guide

Mar. 2005

[65/100]

The phone will adjust the audio frames carried per RTP packet (voice packetization) based

on this setting:

Voice Packetization

G.723.1

G.729A

G.711

10/30(G.723.1)-ms

LAN

1 Frames (10 ms)

20/30(G.723.1)-ms

xDSL, ISDN

2 Frames (20 ms)

30-ms

xDSL, ISDN, Cable

3 Frames (30 ms)

40/30(G.723.1)-ms

56 kbps MODEM

1 Frame (30 ms)

40/60(G.723.1)-ms

Limited bandwidth

2 Frames (60 ms)

4 Frames (40 ms)

The default is “20/30(G.723.1)ms”: 1 audio frame (30 ms voice) per RTP packet for G.723.1

CODEC, and 2 frames (20 ms voice) per packet for all other CODECs such as G.729A and G.711.

Before changing this setting, please refer to the following table “Packet Rate and VoIP

Bandwidth Consumption” to find out the optimal value fit into your environment. We suggest a
reasonable packetization should NOT longer than 40 milliseconds. Pick the longer packetization
time unless you have a very limited bandwidth access at the expense of more latency.

Note:

IP SIP Phone

will dynamically adjust its voice packetization on a per call basis if the

“ptime” attribute is specified in the peer’s SDP. For example: if “a=ptime:10” is specified, the

IP SIP Phone

will ignore the global voice packetization and adjust to the negotiated

packetization time:

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.3.51:5060;branch=abc7801

From: 7751 <sip:[email protected]>;tag=22516

To: <sip:[email protected]>

Call-ID: [email protected]

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Содержание IP300S

Страница 1: ...IP300S User Guide www atltelecom com...

Страница 2: ...EYPAD 14 3 OPERATION 15 3 1 KEY DEFINITIONS IN MENU MODE 15 3 2 ENTER ALPHABETS AND NUMBERS 16 3 3 ADDRESS OF RECORD SIP AOR 16 4 STARTUP 17 4 1 PREREQUISITE 17 4 1 1 NETWORK 17 4 1 1 1 DHCP 18 4 1 1...

Страница 3: ...30 8 1 DIAL SCHEME 31 8 1 1 GUARDING TIME 34 8 1 2 ENUM SAMPLE 35 8 2 REDIAL 37 8 3 ADDRESS BOOK 37 8 4 CALL HISTORY 38 8 5 SPEED DIAL 39 8 6 CALL RETURN 41 8 7 CALLING 41 8 8 CALL FAILURE 42 8 9 AUT...

Страница 4: ...ENABLE PERSONAL PREFERENCE 67 10 11 COMFORT NOISE GENERATION 67 10 12 REGISTRATION ON DEMAND 68 10 13 MULTI DOMAIN REGISTRATION 70 11 VOICE VOLUME ADJUSTMENT 72 11 1 RINGER 72 11 2 HANDSET 72 11 3 SPE...

Страница 5: ...IP SIP Phone v2 User s Guide Mar 2005 5 100 APPENDIX B TROUBLE SHOOTING 96 APPENDIX C TONES 100...

Страница 6: ...up call transfer blind transfer consultative transfer semi supervised transfer and take back call forward call reject do not disturb DND z Call preferences include call waiting auto answer server side...

Страница 7: ...during the last 72 hours or since system startup 1 2 Technical Specifications 1 2 1 Call Control Capability z Fully complies with RFC 3261 SIP with RFC 2543 backward compatible z Fully complies with R...

Страница 8: ...y local conferencing z Voice activity detection VAD and comfort noise generation CNG z Voice and ringer volume control z Real time acoustic echo canceller z IP Type of Service ToS bits set for RTP RTC...

Страница 9: ...2 for network management MIB2 RFC1213 Get and Set operation for internal state Proprietary Enterprise MIB for system configuration access Trap System startup System shutdown by command SNMP Image upgr...

Страница 10: ...ne v2 User s Guide Mar 2005 10 100 2 Layout 2 1 Hardware 2 1 1 Front View 2 1 2 Rear View RJ 45 Ethernet switch to PC RJ 45 Ethernet Jack to LAN Power adaptor Reset SW 2x16 LCD Microphone Handset Spea...

Страница 11: ...calls at most Review the calling information on this channel during conversation Service Realm Display the registration status of each active service domain on idle switch target service domain ISP wh...

Страница 12: ...ter The LED indicates the registration status of each active service domain Green LED On Successfully register to all active service domains Red LED On At least one service domain could not be registe...

Страница 13: ...ss to various phone features The default mappings of these function keys are F1 Forward menu shortcut to activate incoming calls forwarding menu F2 Channel info show information of the last call on ea...

Страница 14: ...m menu 3 2 DSS Functions LCD 2x16 Ring Lamp 1 2 abc 3 def MWI 4 ghi 5 jkl 6 mno MUTE 7 pqrs 8 tuv 9 wxyz FUNC XFER Re Dial Vol Down Vol Up 0 oper SPK HOLD Flash SPD A Call B Call Service Realm Reject...

Страница 15: ...Delete the current character or the previous character if the cursor is positioned at the end FUNC Return to upper level menu SPK Exit the menu Circle through the selected menu items and adjust volum...

Страница 16: ...onsists of any ASCII characters except for and If the Display is present the following address must be enclosed in a paired and z Protocol Usually in lower case such as sip tel or sips Note sip tel an...

Страница 17: ...he max concurrency is 4 IP Range will not apply the changes until user presses Ctrl s to apply the modifications or the client disconnected by Ctrl c z Auto provision on phone startup Please refer to...

Страница 18: ...t disable MENU 6 Network 1 General 4 Use static DNS by picking 2 DHCP Note If DHCP option 42 is present it will overwrite the SNTP server in menu 7 3 2 Server IP Note DHCP option 66 will overwrite the...

Страница 19: ...is LAN dialing Refer to section 8 1 Dialing Scheme on this document If the call could be set up correctly then your network configuration is fine otherwise please refer to B 1 on Appendix B Trouble Sh...

Страница 20: ...m will try to register to those activated SIP domains The flashing green LED of Registration key indicates that the registration is undergoing Once the green LED stops flashing you could know the regi...

Страница 21: ...P s e r v e r ii Download configuration files from TFTP server T F T P s e r v e r n a m e i p r a n g e r c f g iii Download failed T F T P i p r a n g e r c F T i m e d o u t iv Download successful...

Страница 22: ...or keypad to adjust your time zone otherwise the synchronized time may be several hours late earlier than your local time Please refer to section 4 4 1 Date Time on IP SIP Phone v2 Web Administration...

Страница 23: ...thods please refer to section 8 1 Dial Scheme on this user s guide for detail 6 3 Regular Registration 1 Registering R e g t o r e g i s t r a r s i p f c i c o m t w 2 Registration Done R e g i s t e...

Страница 24: ...caller ID could be found on the address book The mm ss keeps track of the time elapsed after the call arrived F r o m m m s s S c o t t S u n 2 The channel LED of this incoming waiting call A B C Call...

Страница 25: ...and the UDP port used for RTP and RTCP session The information will be timely updated in call connection and the unavailable information will be shown as N A 7 2 Reject Call Press Reject on an incomin...

Страница 26: ...d m m s s S c o t t S u n Where mm ss indicates the duration of the call If the user is on speaker phone mode it will return to IDLE state in 5 seconds b User hands up 1 Ringing calls are waiting Bac...

Страница 27: ...ss Redial to dial this number 5 CODEC CODEC employed for the call 6 User agent The phone tool used by the peer for this call 7 Media traffic The media related information will be available only when t...

Страница 28: ...presses Forward key on an incoming waiting call The system forwarding rules will check Do Not Disturb mode first then All Calls Forward Busy Forward finally going to No Answer Forward while no answer...

Страница 29: ...All Calls Forward feature from menu 4 2 All Calls Fwd or on Call Forward web page Forwarded calls are logged in the Missed Calls menu 2 1 If this feature is enabled the corresponding Forward LED will...

Страница 30: ...Reply as busy 2 Record as a received call Forwarding number is available Forward the incoming call to the target forwarding number No Answer Forward Both 1 No answer forward feature is on 2 No answer...

Страница 31: ...seconds the inter digit timed out is programmable Method Rule Example Pick from address book 1 Enter address book 2 Search for entry 3 Press Redial Note The phone will not use the domain you specifie...

Страница 32: ...e the service domain 3 Dial to confirm If the caller is 3100 SIP isp com he could call 3200 SIP isp com by dialing 3200 s i p i s p c o m 3 2 0 0 Dial albert SIP isp com s i p i s p c o m a l b e r t...

Страница 33: ...060 use as 4 Use MUTE to delete previous character on typos 5 69 is reserved for call return Please use 069 instead 6 0 is reserved for server feature access code and will be transmitted as is no tran...

Страница 34: ...as is no translation will be done If your intention is to dial 8863 on address book please enter via punctuation table rather than a when you add such entry into your address book z Multi domain note...

Страница 35: ...proxy for next hop delivery your DNS must have SRV and A records like the following please change those host and IP in red font accordingly ORIGIN SIP isp com Pref Weight Port Target _sip_udp SIP isp...

Страница 36: ...may set it to a proprietary suffix such as e164 net Default is e164 arpa E g if the dial string is 886 3 5639025 the ENUM query string send to DNS server to resolve will be Strip off all non digits R...

Страница 37: ...m INVTTE tel PhoneNumber For example if you dial 88635639025 and IP SIP Phone fails in ENUM resolution it will send INVITE tel 88635639025 3 3 3 In either format you could also configure whether the l...

Страница 38: ...IP SIP Phone v2 Web Administration for detail 8 4 Call History You can pick an entry from the call history either missed calls received calls or dialed numbers press Redial to dial out the specified n...

Страница 39: ...this speed dial digit Press Hold to save 2 Remove a specific speed dial entry z Main Menu 1 Address Book 4 Speed Dials z Go to the digit you want to set for speed dial z Press MUTE to remove the curr...

Страница 40: ...g Position cursor on the text input which you want to clear the speed dial mapping Click Clear collocates with each speed dial entry to remove the mapping 4 Perform Speed Dial IP SIP Phone supports tw...

Страница 41: ...ne must have been configured to use the call return capable default SIP outbound proxy server On the other hand if you mapped the DSS key to the phone set feature Call Return it will find the latest i...

Страница 42: ...er finishing dialing while making a phone call but before connected or hanging up If you press Auto redial on idle mode it is effectively as the same as press R edial follows by a Auto redial After ac...

Страница 43: ...Auto Redial 2 Retry Interval The default redial interval is 15 seconds To specify the activation duration of this auto redial feature once starts measured in seconds please go to Main Menu 5 Preferen...

Страница 44: ...C h a n g H o r a c e F u J a c k y Wa n g J u n D e n g M a x y M a M i c h a e l Wu S c o t t S u n S u n n y S o n g Wi l l i a m G u Enter the 1st character will jump to the 1st entry starts with...

Страница 45: ...akes the voice more clear to the peer and turn on the speaker to make the conversation heard by all by listeners The loud speaker works as follow if you switch to hands free mode with handset lifted o...

Страница 46: ...f you are held by the peer IP SIP Phone will play music on hold 9 3 Mute m m s s M i c H a e l z Set up a connected call z Press MUTE to toggle your voice transmission On muting state the red LED of M...

Страница 47: ...he call from A consultation z If B is unwilling to take this call you may press FLASH to cancel the transfer operation and take back A alternatively you may press A B Channel to cancel transfer and ta...

Страница 48: ...annel to cancel blind transfer and take back A as well Note if you press to finish dialing it behaves just the same as consultative transfer Alternatively you may do a blind transfer as follows z Take...

Страница 49: ...mixing and separate them into two independent calls Note During a conference an auditable tone will be played regularly from the hosting phone to notify all parties that a conference is undergoing The...

Страница 50: ...t to disconnect to hang up first Alternatively you may press FLASH to tear down the conference and separate them into 2 independent calls first then disconnect each of them as your wish z To place a c...

Страница 51: ...into blocking list a Press FUNC to activate menu b Go to submenu 1 Address book 1 Search and locate the party you want to block c Choose screen call to add it into blocking list press HOLD again to to...

Страница 52: ...ries from blocking list a Press FUNC to activate menu b Go to submenu 1 Address book 5 Call Screening and locate the entry you want to remove from the blocking list c Choose Revoke to remove it from t...

Страница 53: ...IP SIP Phone v2 User s Guide Mar 2005 53 100 i s p c o m i s p c o m Select All...

Страница 54: ...the call or the ringing cycle timer expires and the call is given No Answer Forward treatment if applicable When the user has engaged in a call and some new incoming calls are waiting for answer the...

Страница 55: ...t ringing if the handset is placed on hook and there is a call currently on hold Note During a conference an auditable tone will be played regularly from the hosting phone to notify all parties that a...

Страница 56: ...web page IP SIP Phone Preferences Auto Hold on Call Switch to configure it System default is enabled 10 5 Auto Redial You can configure when to stop auto redial once activated by going to MENU 5 Prefe...

Страница 57: ...rections until the target number is reached or a loop ping pong redirect is detected The default is to silently follow the redirections on making calls without user s interference To configure Silentl...

Страница 58: ...DSS Features For example you may F4 as Dial Key by picking Dial Key on F4 Note the DSS dial key will function as well even you pick or FLASH as dial key e Alternatively you may go to web page IP SIP P...

Страница 59: ...page IP SIP Phone Preferences LAN Dial to configure it Default is enabled 10 7 4 Call Command You can configure various call commands such as Calling Line Identification Restriction CLIR or Calling Li...

Страница 60: ...eceived calls which both are kept locally then dial out the latest incoming call number By default F8 is mapped to Call return 10 7 4 2 Anonymous Call CLIP CLIR System default is to enable Calling Lin...

Страница 61: ...remind user You can configure the phone whether the auditable tone should be played or not a Press FUNC to activate menu b Go to submenu 5 Preferences 8 Message Alert 1 E n a b l e d 2 D i s a b l e...

Страница 62: ...menu z Go to submenu 5 Preferences 9 Auto Answer 1 E n a b l e d 2 D i s a b l e d System default is Disabled c Alternatively you may configure this system wide feature from web page IP SIP Phone Pref...

Страница 63: ...th Domain Auto Answer a Once enabled all calls destined to this specific service account will be auto answered on idle mode b This works even when the system wide auto answering is off z auto answer 1...

Страница 64: ...ps G 711 64kbps G 729A G 729AB 8 kbps G 723 1 G 723 1A both 5 3 and 6 4 kbps The default preference is to prioritize them based on their compressed voice quality the higher quality it is the higher pr...

Страница 65: ...is setting please refer to the following table Packet Rate and VoIP Bandwidth Consumption to find out the optimal value fit into your environment We suggest a reasonable packetization should NOT longe...

Страница 66: ...IP kbps PPP kbps Ethernet 802 3 Averaged Band width Utilization Delay ms Mean Opinion Score MOS6 1 35 pps7 16 8 18 4 21 6 28 37 5 2 17 pps 11 2 12 13 6 43 2 67 5 3 12 pps 9 3 9 9 10 9 52 8 97 5 4 9 pp...

Страница 67: ...ed based on their priorities The smaller the value is the higher the priority would be Those disabled voice CODECs which preference is zero will be listed last d Edit the priority value you want to se...

Страница 68: ...flash Based on the registration result the LED will have different indications Green LED On Successfully register to all activated service domains Red LED On At least one activated service domain cou...

Страница 69: ...d SIP service domains and cease regular auto registration scheduling until the Registration key is explicitly pressed again Note reboot the phone set will clear this status and register the SIP addres...

Страница 70: ...v e r c o m X u n k n o w n c o m 10 13 Multi Domain Registration You could register to multiple domains simultaneously such that you could receive calls from those registered domains and make calls t...

Страница 71: ...o matter whether they have registered or not but must before dialing the which finishes the dialing The active service domain will appear on the upper right corner f w d p u l v e r c o m 3 2 0 0 In a...

Страница 72: ...o use when incomng calls arrive Note IP SIP Phone supports the alert info header in the first INVITE message as per RFC3261 alert info header dictates the phone to use an alternative ringing tone whic...

Страница 73: ...SIP Phone v2 User s Guide Mar 2005 73 100 11 4 Ear Phone E a r P h o n e Activate when ear phone is on and the phone set is hooked off or while at least one call is engaged Use and keys to adjust volu...

Страница 74: ...m SIP 2 0 Via SIP 2 0 UDP 192 168 3 50 From John sip 7700 SIP isp com tag 17542c1 To John sip 7700 SIP isp com Call ID 0c1c7a67461 ipr SIP isp com Cseq 281 SUBSCRIBE Contact sip 7700 192 168 3 50 Even...

Страница 75: ...oice Mail z To set up voice mail access number by TELNET or keypad a Press FUNC key to activate menu b Go to submenu 7 Service 1 Voice mail URI c Configure the voice mail number to dial when the MWI b...

Страница 76: ...eceived if this field is absent or is not a SIP AoR the AoR in request is used instead If there are unsolicited out of dialog NOTIFY messages received from different service domains those voice mailbo...

Страница 77: ...lashing messages are for server side notification and they will not be saved thus flashing To activate such feature the received out of dialog instant message must carry a proprietary header P Flash S...

Страница 78: ...to 1024 seconds The default time on system starting up is 00 00 January 1 1970 GMT Unicast Multicast Anycast Sends SNTP request to the specified SNTP server if available Nothing When in multicast mode...

Страница 79: ...uration on IP SIP Phone v2 Web Administration for configuration file format and available tags z To enable auto provisioning on system startup a Press FUNC key to activate menu b Go to submenu 7 Servi...

Страница 80: ...IP SIP Phone v2 User s Guide Mar 2005 80 100 from Batch settings and those read from flash ROM z To configure auto provisioning by web browser a Go to IP SIP Phone Auto Provision...

Страница 81: ...IP SIP Phone v2 User s Guide Mar 2005 81 100 z Auto Provision Flow...

Страница 82: ...ch systems could gain access to these supplant features by dialing special numbers such as 69 or 7 The star sign and the pound sign bear special meaning on IP SIP Phone where a dialing string starts w...

Страница 83: ...access to such server feature For most servers IP PBXs the feature access code is configurable please consult to system administrator for prefix reconfiguration This table summarizes the heuristics ta...

Страница 84: ...IP Settings 4 ENUM E 164 2 Min length which default is 6 digits and only consists of digits Web page IP SIP Phone SIP Settings ENUM E 164 ENUM Minimum Length 1 5639025 tel 5639025 2 3 5639025 tel 3563...

Страница 85: ...public internet or local area network please click IP SIP Phone to show the current Host IP If your host IP is within any of the listed ranges then your terminal resides on LAN otherwise it locates o...

Страница 86: ...When you use a SIP aware router NAT detection should be set to Off as if you were on the public internet and the configuration is the same as Public Internet Configuration set UDP Traversal to be Ful...

Страница 87: ...ls reside under the same NAT their NAT port mappings must not be overlapped since they all share the same NAT resource z Configure RTP ports RTP Port Base 45700 Must be an even number and between 2 an...

Страница 88: ...rvice signaling port Take the scenario above as an example Transport UDP and TCP or UDP you must include UDP anyway SIP Listen port 45706 z Assign static NAT IP s t u n i s p c o m Diagnose NAT option...

Страница 89: ...2 2 NAT Traversal by STUN Setting up the NAT router is impossible in many cases and new equipment may be too expensive For these environments Simple Traversal of UDP Through NATs STUN has come to resc...

Страница 90: ...IP SIP Phone v2 User s Guide Mar 2005 90 100 z Activate STUN Mode s t u n i s p c o m STUN server Enter a functional and reachable STUN server IP for STUN to work UDP Traversal Enable STUN...

Страница 91: ...k Operations 132 239 254 49 ntp ucsd edu CERFNET NSFNET SDSC region and nearby Quincy California ntp1 mainecoon com ntp2 mainecoon com North America Newark DE University of Delaware 128 175 1 3 louie...

Страница 92: ...umbia University sundial columbia edu NYSERnet New York City NY Columbia University Computer Science Department timex cs columbia edu PSINET NSFNET and NYSER region Norman Oklahoma University of Oklah...

Страница 93: ...and any South America Buenos Aires Argentina Network Access Point 200 49 40 1 tick nap com ar 200 49 32 1 tock nap com ar Argentina Buenos Aires Argentina Sinectis S A time sinectis com ar Argentina...

Страница 94: ...Slovenia Hydrometeorological Institute of Slovenia hmljhp rzs hm si Slovenia and Europe Ljubljana Slovenia Academic and Research Network of Slovenia ntp1 arnes si ntp2 arnes si Slovenia and Europe Lju...

Страница 95: ...aland The University of Waikato truechimer waikato ac nz truechimer1 waikato ac nz truechimer2 waikato ac nz truechimer3 waikato ac nz New Zealand Singapore and the Philippines ntp shim org Singapore...

Страница 96: ...n invalid gateway iv Go to IP SIP Phone page to make sure that those Active Network Status matches those configured Specifically if the active DNS is 0 0 0 0 then you may have wrongly configured to St...

Страница 97: ...Map as the way to traverse NAT 5 After setting up a call it always disconnects automatically after 32 seconds This may be due to the fact that either the involved SIP proxy servers and or the terminal...

Страница 98: ...by your ISP or company 7 Sometimes there would be only one party be held successfully on conference mode when the conference master presses HOLD This happens only on some SIP proxy server implementati...

Страница 99: ...ne while picking up handset ii Both of you are under on the same NAT and either one of you employs Enable STUN or Static NAT IP UDP Map to traverse NAT This is largely because some NATs will not loop...

Страница 100: ...usy wav Call transfer failed 600 ms FAST_BUSY_2_TONE FastBusy2 wav 1st digit timeout CALLWAITING_TONE CallWaiting wav Hold recall Conference alerting COVERAGE_TONE Coverage wav Call transfer succeeded...

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