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ATL IP300S User Manual Download Page 74

 

IP SIP Phone v2

 User’s Guide 

Mar. 2005 

 

[74/100]

12. Service 

12.1. Voice  Mail 

Voice mail allows you to access messages left by callers when you are unavailable to take 

their call. Voice mail is an optional feature configured by your system administrator. Your 
particular phone setup might not support accessing voice mail in this way. 

IP SIP Phone

 supports messages waiting indication as per RFC3842. It accepts NOTIFY 

messages with or without prior SUBSCRIBE.   

z

 

MWI Subscription 

If you have configured the voice mailbox link, 

IP SIP Phone

 will SUBSCRIBE to this 

link by sending the SUBSCRIBE message to SIP server. Please refer to the following 
message flow (suppose your SIP AoR is sip:[email protected] and the voice mailbox 
you configured is sip:[email protected]): 

SUBSCRIBE

 sip:[email protected] SIP/2.0 

Via: SIP/2.0/UDP 192.168.3.50 

From: John <sip:[email protected]>;tag= 17542c1

 

To: John <sip:[email protected]> 
Call-ID: [email protected] 
Cseq: 281 SUBSCRIBE 
Contact: sip:[email protected] 
Event: message-summary 
Accept: application/simple-message-summary 
Expires: 3600 
User-Agent: SIP-Phone /1.1 
Content-Length: 0 
  
  
SIP/2.0 

202 Accepted

 

Via: SIP/2.0/UDP 192.168.3.50 
Record-Route: <sip:192.168.3.1;lr=1> 
Contact: sip:192.168.3.1:6060 
Expires: 3600 
User-Agent:   ABS GW v5.1.0 
To: John <sip:[email protected]>;tag=980765 

From: John <sip:[email protected]>;tag= 17542c1

 

Call-ID: [email protected] 
CSeq: 281 SUBSCRIBE 
Content-Length: 0

 

 

z

 

Out-of-Dialog MWI Notification 

IP SIP Phone

 accepts unsolicited MWI NOTIFIY messages as well. The sample 

NOTIFY message is as follows (suppose “sip.isp.com” is one of your active service 
domains and your SIP AoR is sip:[email protected]): 

NOTIFY 

sip:[email protected] SIP/2.0 

Via: SIP/2.0/udp 192.168.0.1;branch=z9hG4bKcfb4 

Content-Type: application/simple-message-summary 

Summary of Contents for IP300S

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

Page 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...

Page 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...

Page 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...

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

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

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

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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...

Page 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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