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ATL IP300S User Manual

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IP300S

User Guide

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  • Page 1: User Guide

    IP300S User Guide www.atltelecom.com...

  • Page 2: Table Of Contents

    IP SIP Phone v2 User’s Guide Mar. 2005 Directory ................................2 IRECTORY ............................... 6 VERVIEW 1.1..............................6 EATURES 1.2........................7 ECHNICAL PECIFICATIONS 1.2.1........................7 ONTROL APABILITY 1.2.2........................8 TIME OICE TREAMING 1.2.3........................... 8 IREWALL 1.2.4.............................
  • Page 3 IP SIP Phone v2 User’s Guide Mar. 2005 7.1..............................24 INGING 7.2............................. 25 EJECT 7.3............................25 ORWARD 7.4............................25 NSWER 7.5............................... 26 ONNECTED 7.6............................26 ISCONNECTED 7.7. DND ..........................28 ORWARD AND 7.7.1. (DND) ......................... 28 ISTURB 7.7.2.
  • Page 4 IP SIP Phone v2 User’s Guide Mar. 2005 10.1............................54 AITING 10.2............................55 IMEOUT 10.3............................55 ECALL 10.4........................56 OLD ON WITCH 10.5............................56 EDIAL 10.6......................57 ILENTLY OLLOW EDIRECTION 10.7..............................57 10.7.1..........................
  • Page 5 IP SIP Phone v2 User’s Guide Mar. 2005 B – T ........................96 PPENDIX ROUBLE SHOOTING C – T ............................100 PPENDIX ONES [5/100]...

  • Page 6: Overview

    IP SIP Phone v2 User’s Guide Mar. 2005 1. Overview 1.1. Features DHCP or PPPoE for host IP, gateway, network mask, DNS (optional, 2 DNS at most), TFTP server, NTP server, and TTL; all those settings could be static assigned as well. If provided by DHCP server, it could use DHCP to get NTP server.

  • Page 7: Technical Specifications

    IP SIP Phone v2 User’s Guide Mar. 2005 Regular alarm and one-time alarm. Prompt user on call diversion for better security support (Configurable) Menu driven configuration by keypad, Web browser or TELNET. Use of Simple Network Time Protocol (SNTP) to synchronize time with network time server and adjust to time-zone (configurable) and daylight saving time (configurable).

  • Page 8: Real-Time Voice Streaming

    IP SIP Phone v2 User’s Guide Mar. 2005 Support REGISTER, INVITE, ACK, CANCEL, BYE, OPTION, REFER, MESSAGE, SUBSCRIBE, NOTIFY, INFO methods Support “alert-info” header for distinctive ring. 1.2.2. Real-time Voice Streaming Fully complies with RFC 1889 (RTP / RTCP), RFC 1890 (AVT profiles), RFC 3551 (RTP Profile for Audio and Video Conference with Minimal Control) and RFC 3555 (MIME Type Registration of RTP Payload Formats).
  • Page 9 IP SIP Phone v2 User’s Guide Mar. 2005 SNMPv2 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 upgrade) SIP Registrar availability Call-Channel Status. [9/100]...

  • Page 10: Layout

    IP SIP Phone v2 User’s Guide Mar. 2005 2. Layout 2.1. Hardware 2.1.1. Front View 2x16 LCD Speaker Keypad Handset Microphone 2.1.2. Rear View Power adaptor RJ-45 Ethernet switch to PC Reset SW RJ-45 Ethernet Jack to LAN [10/100]...

  • Page 11: Back View

    IP SIP Phone v2 User’s Guide Mar. 2005 Back View 2.1.3. RJ-11 Earphone Jack Wall mount RJ-11 Handset Jack 2.2. Keys Service Realm A/B Channel Reject MUTE FUNC FLASH SPK/Hands-free HOLD XFER REDIAL Volume¡ i −,¡ Ï ¡ j ¡ i ¡...

  • Page 12: Call History

    IP SIP Phone v2 User’s Guide Mar. 2005 ¡ i Reject¡ j ¡ G Reject incoming waiting calls ¡ i MWI¡ j ¡ G Message Waiting Indication, MWI: Access to voice mail system ¡ i MUTE¡ j ¡ G Mute | Delete character ¡...
  • Page 13 IP SIP Phone v2 User’s Guide Mar. 2005 ¡ i Forward¡ j ¡ G Forward incoming waiting calls ¡ i URL¡ j ¡ G Use keypad to enter alphabets and numbers (red LED on). ¡ i Address Book¡ j ¡ G Access to address book (search an entry or list all entries). ¡...

  • Page 14: Keypad

    IP SIP Phone v2 User’s Guide Mar. 2005 2.3. Keypad LCD 2x16 A Call B Call Reject Service Realm Addr. Book Auto-Redial Registration Fwd Menu Speed dials Channel info Call detail(CDR) Call Return Packetization Messaging Network info Forward Conference Call History 2(abc) 3(def) 4(ghi)

  • Page 15: Operation

    IP SIP Phone v2 User’s Guide Mar. 2005 3. Operation 3.1. Key Definitions in Menu Mode ¡ HOLD ¡ Enter the selected menu item or confirm the modification. Delete the current character or the previous character if the cursor is ¡...

  • Page 16: Enter Alphabets And Numbers

    IP SIP Phone v2 User’s Guide Mar. 2005 3.2. Enter Alphabets and Numbers Circular input by pressing the same key Alphabet & Number 2->a->b->c->A->B->C 3->d->e->f->D->E->F 4-> g->h->i->G-> H->I 5->j->k->l->J->K->L 6->m->n->o->M->N->O 7->p->q->r->s->P->Q->R->S 8->t->u->v->T->U->V 9->w->x->y->z->W->X->Y->Z 0->[Space] Punctuation Table: @ - * # _ ? & $ / \ , : ; + ( ) ‘...

  • Page 17: Startup

    IP SIP Phone v2 User’s Guide Mar. 2005 Mike Jackson <sip:3200@SIP.isp.com> “Voice Mailbox” <sip:8888@vms.SIP.isp.com> Display with enclosing ‘”’. sip:300@SIP.isp.com AoR without display sip:192.168.3.100 AoR without user part (in dotted IP) tel:+886-3-5639025 ENUM AoR sip:+88635639025@SIP.isp.com;user=phone ENUM AoR with SIP proxy support 4.

  • Page 18: Dhcp

    IP SIP Phone v2 User’s Guide Mar. 2005 PPPoE. 4.1.1.1. DHCP Pick¡ i 1.Mode¡ j \¡ i 1.DHCP¡ j Disable¡ i 4.Use Static DNS¡ j by choosing ¡ i 2.DHCP¡ j Note: if you want to assign a different domain name server instead of using those obtained by DHCP, you should choose¡...

  • Page 19: Verify Network Configuration

    IP SIP Phone v2 User’s Guide Mar. 2005 3. Service Name = Optional, some ISP requires it (modify this as necessary). 4.1.1.4. Verify Network Configuration Press ¡ i F7¡ j (which default is a shortcut to menu¡ i 8.Advanced¡ j \¡ i 4.System Status¡ j \ ¡...

  • Page 20: Configure Nat And Firewall

    IP SIP Phone v2 User’s Guide Mar. 2005 Alternatively, you mayo to ¡ y Main Menu¡ z =>”6.Network” / ”2.SIP settings” / ”1.1 Realm” to configure these information by keypad. If you have applied for more than one service domains, please repeat step I and II until all active domains are properly configured.

  • Page 21: Initialization

    IP SIP Phone v2 User’s Guide Mar. 2005 network, please refer to chapter 13-“NAT Traversal” on this user’s guide if your phone-set is behind network address translator (NAT) and / or firewall. 4.2. Initialization (a) Startup: S I P P h o n e V e r s i o n (b) Check for auto-provision.

  • Page 22: Registration

    IP SIP Phone v2 User’s Guide Mar. 2005 User entry any digit for Time & Date. It must press HOLD Key to be Idle ¡ i ¡ j Ready Mode Display. The phone will synchronize its time by Simple Network Time Protocol, SNTP, with network time server regularly if SNTP is enabled.

  • Page 23: Idle

    IP SIP Phone v2 User’s Guide Mar. 2005 6. Idle 6.1. Registered F r i M M / D D h h : m m M i c h a e l 6.2. Not registered yet or registration expires F r i M M / D D h h : m m M i c h a e l...

  • Page 24: Take Calls

    IP SIP Phone v2 User’s Guide Mar. 2005 7. Take Calls In menu mode, you cannot receive calls. If there is incoming calls when you are configuring your phone, the call will be silently rejected with a 486 busy response. Besides, if you enter menu by DSS keys (see menu-3.2 DSS features), the phone will start ringing when incoming calls are waiting.

  • Page 25: Reject Call

    IP SIP Phone v2 User’s Guide Mar. 2005 1 . C A l l e r I D ( A o R ) 2 . F r o m ( C O n t a c t ) 3 . C O D E C 4 .

  • Page 26: Connected

    IP SIP Phone v2 User’s Guide Mar. 2005 S c o t t S u n The “mm:ss” keeps track of the time elapsed after answered. (A) On Idle State: a. Press ¡ i A / B / ¡ j C Call to answer the call b.
  • Page 27 IP SIP Phone v2 User’s Guide Mar. 2005 . C A l l e r I D ( A o R ) 2 . C A l l e r I D ( A o R ) 3 . C A l l e r I D ( A o R ) You may press ¡...

  • Page 28: Forward And Dnd

    IP SIP Phone v2 User’s Guide Mar. 2005 S e n t : 1 , 2 3 4 , 5 6 7 R c v d : 1 , 2 3 4 , 5 6 7 7-3 Bytes: Received / Sent bytes, including IP / UDP / RTP header and payload S e n t : 1 , 2 3 4 , 5 6 7 K B R c v d : 1 , 2 3 4 , 5 6 7 K B 7-4 Packet lost: Received / Sent RTP packet lost ratio.

  • Page 29: All Calls Forward

    IP SIP Phone v2 User’s Guide Mar. 2005 By web browser: Go to ¡ y Call Forward ¡ z page, then click the “Contacts” on the right panel to pick an entry from the address book to set it as “Target Number”. Delete the number in the text input to remove it.

  • Page 30: Make Calls

    IP SIP Phone v2 User’s Guide Mar. 2005 Forwarding number is Forward the incoming call to available the target forwarding number. ¡ is on ¡ 1. Reply as 480 Temporarily Forwarding number is unavailable. All Calls unavailable 2. Recorded as a missed call Forward Both Forward the incoming call to...

  • Page 31: Dial Scheme

    IP SIP Phone v2 User’s Guide Mar. 2005 T r y a g a I n l a t t e r 8.1. Dial Scheme A. Press ¡ ¡ to activate alphabets typing via keypad on which time the corresponding LED will be on, or type numbers directly via DTMF.
  • Page 32 IP SIP Phone v2 User’s Guide Mar. 2005 4. Valid ENUM dial strings must be longer than 6 (configurable) and containing only digits, optional ‘-‘, spaces, ‘(‘ or ‘)’, such as “#886-3 5639025”, “+86 (3) 5639025” or “#8863”. 5. Those not recognized as valid ENUM dial string will be dialed “as is”...
  • Page 33 IP SIP Phone v2 User’s Guide Mar. 2005 IP Dialing 1. Use ‘*’ as dot, ‘.’. 1. Call 192.168.10.200 (Anonymous Call) 2. Use “**” as ‘:’ then follows peer’s port 1 9 2 * 1 6 8 * 1 0 * 2 0 0 (optional, but must assigned 2.

  • Page 34: Guarding Time

    IP SIP Phone v2 User’s Guide Mar. 2005 numbers should send “as is” by disabling this feature from menu “5.Preferences” -> “7.Dial plan” -> “3.LAN dial”. To facilitate “Contact Dialing”, “IP Dialing” and “LAN dialing” (where most users forget to dial the SIP signaling port of the peer, and end in no responses if the peer doesn’t listen on the standard UDP port 5060 for SIP signaling), IP SIP Phone always listens on UDP-5060 for SIP signaling in addition to the user configured SIP service port.

  • Page 35: Enum Sample

    IP SIP Phone v2 User’s Guide Mar. 2005 digits on expiry as well. To speed up the dialing process, press ¡ i ¡ j key whenever finishing dialing. User does not hook up yet: The default inter-digit timeout is 4 seconds and the phone will dial out the collected ¡...
  • Page 36 IP SIP Phone v2 User’s Guide Mar. 2005 Alternatively, you may go to ¡ y IP SIP Phone ¡ z / ¡ y SIP Settings ¡ z page by web browser and configure “ENUM & E.164”. Below illustrates how to configure them by keypad (TELNET). 3.1.

  • Page 37: Redial

    IP SIP Phone v2 User’s Guide Mar. 2005 follows at least "Min length" digits. If this rule failed does not comply, the dial string will be dial "as is" without any attempt to carry out an ENUM resolution. Default is 6 digits. 3.3.

  • Page 38: Call History

    IP SIP Phone v2 User’s Guide Mar. 2005 any input to go to the first entry on address book. On listing mode, press ¡ i 2 ¡ j twice to first entry prefixed with an ‘A’ (or press ¡ i 8 ¡ j consecutively for 3 times will jump to the first entry prefixed with a ‘U’. etc.) Press ¡...

  • Page 39: Speed Dial

    IP SIP Phone v2 User’s Guide Mar. 2005 go to page ¡ y IP SIP Phone ¡ z / ¡ y Call History ¡ z => ¡ y Missed Calls ¡ z , ¡ y Received Calls ¡ z or ¡...
  • Page 40 IP SIP Phone v2 User’s Guide Mar. 2005 i s p . c o m To assign a speed dial entry: Position cursor on the text input which you want to assign a speed dial mapping. Click the “Contacts” on the right panel to pick an entry from address book to map to a specific speed dial entry.

  • Page 41: Call Return

    IP SIP Phone v2 User’s Guide Mar. 2005 + ¡ i ¡ j 1 + ¡ i # ¡ j . Description ¡ +’0’, ‘0’+ ¡ ¡ SPD , ¡ ¡ SPD +’00’ or ‘00’+ ¡ ¡ ¡ Dial the 0 speed dial number.

  • Page 42: Call Failure

    IP SIP Phone v2 User’s Guide Mar. 2005 C a l l i n g m m : s s M i c h a e l Callee reached T r y i n g m m : s s M i c h a e l Ringing R i n g i n g...

  • Page 43: One-Touch Dial

    IP SIP Phone v2 User’s Guide Mar. 2005 . R i n g i n g 2 . C o n n e c t e d Ringing: only when the peer starts ringing back will auto-redial process stops (Default). Connected: Only when the peer picks up will auto-redial process stops.
  • Page 44 IP SIP Phone v2 User’s Guide Mar. 2005 Press ¡ ¡ FUNC +¡ i #¡ j to enter menu. Go to menu “3.Phone Settings” / ”2.DSS Features” Pick the function key which you want to act as one-touch dial, and press ¡ i HOLD¡ j to choose the mapped feature.

  • Page 45: Call Processing

    IP SIP Phone v2 User’s Guide Mar. 2005 9. Call Processing 9.1. Handset, Speaker-phone, ear-phone and Loud-speaker Press ¡ i SPK ¡ j to switch between speaker-phone and handset. The red LED of ¡ i SPK ¡ j indicates speaker-phone is on or off. However, if you are in ear-phone mode, then press ¡...

  • Page 46: Mute

    IP SIP Phone v2 User’s Guide Mar. 2005 Hold Recall” by keypad or web configuration page ¡ y IP SIP Phone¡ z / ¡ y Preferences¡ z => “Hold Recall Alerting Timeout (s) “): If you are in on-hook state: The phone will ring.

  • Page 47: Consultative Transfer

    IP SIP Phone v2 User’s Guide Mar. 2005 disconnect from the other party. Consultative Transfer -With consult transfer you consult the other party before transferring the call. To consult transfer a caller, you would press the ¡ i XFER¡ j button to place the caller on hold.

  • Page 48: Blind Transfer

    IP SIP Phone v2 User’s Guide Mar. 2005 transfer from call-A to call-B, you should press¡ i Conference¡ j to conference both parties in. 9.4.2. Blind Transfer Follow the following steps to do a blind transfer: transfer the call without asking the transferring target first.

  • Page 49: Conference

    IP SIP Phone v2 User’s Guide Mar. 2005 S t a t i o n l o c k e d ! U n l o c k r s t . 9.5. Conference IP SIP Phone supports Ad Hoc 3-way local conference. The ¡...

  • Page 50: Conference Tips

    IP SIP Phone v2 User’s Guide Mar. 2005 on Case 2: you have two calls at hand already, and you press ¡ ¡ Conference on either active channel: a. The green LED of ¡ Conference will be on if IP SIP Phone can host a conference ¡...

  • Page 51: Block Calls

    IP SIP Phone v2 User’s Guide Mar. 2005 9.6. Block Calls IP SIP Phone can add entries on address book into blocking list and filter out calls from those parties. On receiving calls from those parties, IP SIP Phone will drop them silently with a response “480 Temporarily unavailable”...
  • Page 52 IP SIP Phone v2 User’s Guide Mar. 2005 2. View call records of screening parties a. Press ¡ FUNC + ¡ i ¡ j ¡ to activate menu. b. Go to submenu “1. Address book” / “5. Call Screening”, and locate the entry you are interested in from the list.
  • Page 53 IP SIP Phone v2 User’s Guide Mar. 2005 Select All i s p . c o m i s p . c o m [53/100]...

  • Page 54: Call Preference

    IP SIP Phone v2 User’s Guide Mar. 2005 10. Call Preference You can configure personal preferences by pointing web browser to your terminal’s IP and go to ¡ y IP SIP Phone¡ z / ¡ y Preferences¡ z . Subsections in this chapter illustrate how to configure it by keypad (TELNET). 10.1.

  • Page 55: Dial Timeout

    IP SIP Phone v2 User’s Guide Mar. 2005 System default is enabled. 10.2. Dial Timeout You could configure the dialing timeout if no response (no ringing back) from the peer. If the timer expires before the peer starts ringing back, it will play disconnect tone and prompt the user “Dial timeout”.

  • Page 56: Auto Hold On Call Switch

    IP SIP Phone v2 User’s Guide Mar. 2005 Set the interval. A l a r t I n t e r v A l : [ 1 0 - 6 0 0 ] s e c Alternatively, you may go to web page ¡ y IP SIP Phone¡ z / ¡ y Preferences¡ z => “Hold Recall Alerting Timeout (s)”...

  • Page 57: Silently Follow Redirection

    IP SIP Phone v2 User’s Guide Mar. 2005 A c t i v a t i o n t i m e : [ 3 0 - 8 6 4 0 0 ] 1 8 0 To avoid overflow the network with redial retries, the phone will wait a gap between each redial (default is 15 seconds).

  • Page 58: Dial Key

    IP SIP Phone v2 User’s Guide Mar. 2005 c. Enter the preferred inter-digit timeout, measured in seconds. I n t e r - D i g i t T i m e [ 3 - 9 ] s e c o n d s d.

  • Page 59: Call Command

    IP SIP Phone v2 User’s Guide Mar. 2005 reserved by ISP for server feature access). Please refer to section 8.1 - “Dial Scheme” for LAN dialing, and section 12.5 - “Soft-Switch (PBX) Feature Access”. To disable this LAN dial feature: ¡...

  • Page 60: Call Return

    IP SIP Phone v2 User’s Guide Mar. 2005 10.7.4.1. Call Return Configure the code to access soft-switch feature: “call return”. Whenever your dial string matched the specified call-return string, IP SIP Phone will send the dial-string “as is” (with SIP domain appended) to the SIP proxy server, such as “sip:*69@SIP.isp.com”, and depends on proxy server to keep the latest call history for each user to cover the phone off-line interval.

  • Page 61: Message Alert

    IP SIP Phone v2 User’s Guide Mar. 2005 o=m000dc300051 0 0 IN IP4 192.168.3.5 To configure the access code: ¡ i ¡ j Press FUNC to activate menu. ¡ ¡ b. Go to submenu “5. Preferences” / “7.Dial Plan / “4. Call Command” / “2. Anonymous Call”.
  • Page 62 IP SIP Phone v2 User’s Guide Mar. 2005 Otherwise, auto-answer this incoming call. P-Auto-answer: urgent Dropped an inactive call if all lines are occupied. Put any on-going calls to hold and auto-switch to an available line if not idle. auto-answer the newly arrived call.
  • Page 63 IP SIP Phone v2 User’s Guide Mar. 2005 Realm¡ z /¡ y Auto-Answer¡ z . . E n a b l e d D i s a b l e d Default is Disabled. Alternatively, you may configure this account-specific feature from web page ¡...

  • Page 64: Codec Preference

    IP SIP Phone v2 User’s Guide Mar. 2005 Otherwise, proceed as normal incoming calls Incoming call processing rules (by precedence): i. Check for the server-side invoked auto-answer feature, if “P-Auto-answer: imperious” is present, auto-answer it. ii. If the phone is engaged in do-not-disturb mode, then DND wins iii.
  • Page 65 IP SIP Phone v2 User’s Guide Mar. 2005 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 1 Frames (10 ms) 10/30(G.723.1)-ms 2 Frames (20 ms) 20/30(G.723.1)-ms xDSL, ISDN 1 Frame (30 ms) 3 Frames (30 ms)
  • Page 66 IP SIP Phone v2 User’s Guide Mar. 2005 CSeq: 1 INVITE Contact: "7751" <sip:7751@192.168.3.51:5060> User-Agent: SIP-Phone/1.1 Content-Length: 171 Content-Type: application/sdp o=m000dc300051 0 0 IN IP4 192.168.3.51 m=audio 5102 RTV/AVP 0 8 18 a=ptime:10 VoIP Bandwidth Packet rate is especially important for sizing a network against a router because routers are not only constrained by bandwidth but the number of packets per second (PPS) that they can process.

  • Page 67: Enable Personal Preference

    IP SIP Phone v2 User’s Guide Mar. 2005 2 (50 pps) 25.2 27.6 32.4 28.2% 8 kbps 3 (35 pps) 19.6 21.2 24.4 36.8% 4 (25 pps) 16.8 20.4 43.5% 5 (20 pps) 15.1 16.1 48.8% 6 (17 pps) 14.8 16.4 53.1% 1 (100 pps)

  • Page 68: Registration On Demand

    IP SIP Phone v2 User’s Guide Mar. 2005 switch to the corresponding CNG-capable CODEC, even you have disabled “CNG”. To configure Voice Activity Detection a. Press ¡ + ¡ i ¡ j ¡ FUNC to activate menu. b. Go to submenu “8. Advanced” / ”3. CNG”. c.
  • Page 69 IP SIP Phone v2 User’s Guide Mar. 2005 Alternatively you may issue the following HTTP Get command from your web browser to go online: http://terminal_ip_address/unregister where terminal_ip_address is the IP address of your terminal. Those command web pages are password protected. Please refer to section 4.11.3.1- “Issue Commands by HTTP Get”...

  • Page 70: Multi-Domain Registration

    IP SIP Phone v2 User’s Guide Mar. 2005 pages are password protected. Please refer to section 4.11.3.1- “Issue Commands by HTTP Get” on “IP SIP Phone v2 Web Administration” for detail. You could view registration status of each service domain by pressing ¡ ¡...
  • Page 71 IP SIP Phone v2 User’s Guide Mar. 2005 That is, the default service domain of ¡ B Call is 2 ¡ active service domain if available; otherwise, it will use the 1 active service domain as default. Moreover, you could change the target service domain while making calls by pressing ¡...

  • Page 72: Voice Volume Adjustment

    IP SIP Phone v2 User’s Guide Mar. 2005 11. Voice Volume Adjustment 11.1. Ringer You could adjust the ringing volume either during the phone set is ringing or from the menu. Press ¡ i and ¡ i to adjust the ringing volume while the phone set is ringing. R i n g e r v o l u m e : █...

  • Page 73: Ear Phone

    IP SIP Phone v2 User’s Guide Mar. 2005 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.

  • Page 74: Service

    IP SIP Phone v2 User’s Guide Mar. 2005 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.

  • Page 75: Set Up Voice Mail

    IP SIP Phone v2 User’s Guide Mar. 2005 Contact: sip:192.168.0.1:6060 User-Agent: ISP Soft-Switch Event: message-summary Subscription-State: active To: <sip:1234@sip.isp.com> From: <sip:1234@192.168.0.1>;tag=d8370cb Call-ID: d07b59da8e CSeq: 224493566 NOTIFY Content-Length: 39 Max-Forwards: 70 Messages-Waiting: yes Message-Account: sip: albert@sip.isp.com Voice-Message: 4/8 (1/2) 12.1.1. Set up Voice Mail To set up voice mail access number by TELNET or keypad: a.

  • Page 76: Access Voice Mail

    IP SIP Phone v2 User’s Guide Mar. 2005 12.1.2. Access Voice Mail To access voice mail, press the ¡ i MWI¡ j button and follow the voice instructions. The red LED adjunct to the ¡ i MWI¡ j button also will be flashing whenever you have unread and/or new voice messages (Message Waiting Indication) in your voice mailboxes.

  • Page 77: Synchronize Time

    IP SIP Phone v2 User’s Guide Mar. 2005 By default, DSS key ¡ i F5¡ j is dedicated to ¡ i Messaging¡ j function key, and cannot be re-assigned. The LED of ¡ i F5¡ j will be flashing whenever there are unread messages in your INBOX, and will be off after you press ¡...
  • Page 78 IP SIP Phone v2 User’s Guide Mar. 2005 IP SIP Phone The Simple Network Time Protocol is used to synchronize time with . If you set SNTP server to Anycast mode, the phone will send SNTP query to LAN broadcast address. Otherwise, it sends a request to the specified SNTP / NTP server, extracting the reported time from the reply, and overwrites the phone’s time.

  • Page 79: Auto Provision

    IP SIP Phone v2 User’s Guide Mar. 2005 SNTP servers. broadcast address from broadcast request. any server on the network. The default multicast address is 224.0.1.1. a. Assign the IP of NTP / SNTP server. 2 1 1 . 1 7 9 . 1 9 . 1 3 3 Assign the SNTP server.
  • Page 80 IP SIP Phone v2 User’s Guide Mar. 2005 from “Batch settings” and those read from flash ROM. To configure auto-provisioning by web browser: a. Go to ¡ y IP SIP Phone¡ z /¡ y Auto Provision¡ z [80/100]...
  • Page 81 IP SIP Phone v2 User’s Guide Mar. 2005 Auto-Provision Flow [81/100]...

  • Page 82: Soft-Switch (Pbx) Feature Access

    IP SIP Phone v2 User’s Guide Mar. 2005 We suggest that you use the default configuration file to define values for SIP parameters that are common to all phones. Doing so will make controlling and maintaining your network easier. You can then define only those parameters that are specific to a phone in the phone-specific configuration file.
  • Page 83 IP SIP Phone v2 User’s Guide Mar. 2005 ¡ y ¡ z IP SIP Phone / ¡ y SIP Settings¡ z => “ENUM & E.164” / “ENUM Minimum Length”), which default is 6-digits, and only consists of digits (optionally a leading ‘+‘). Therefore, any string starts with a ‘#’...
  • Page 84 IP SIP Phone v2 User’s Guide Mar. 2005 2. This is a short-hand to 3. *069 => sip:192.168.1.69 directly dial the IP of a SIP 4. *50**3000 => sip:192.168.1.50:3000 CPE residing on the same 5. *1*20**9999 => sip:192.168.1.20:9999 LAN segment. 1.

  • Page 85: Nat Traversal

    IP SIP Phone v2 User’s Guide Mar. 2005 13. NAT Traversal If this terminal locates within a local area network and you want to place a call to public internet, you must configure your terminal to traverse the NAT and firewall it currently behind. To learn determine whether you resides on public internet or local area network, please click ¡...

  • Page 86: Lan Configuration To Traverse Nat And Firewall

    IP SIP Phone v2 User’s Guide Mar. 2005 s t u n : i s p . c o m UDP Traversal: Full Access (Public Host IP) 13.2. LAN Configuration to Traverse NAT and Firewall There are basically two options for CPE to traverse NAT and Firewall: Option 1: Set up a static route in the NAT gateway (Recommend)¡...
  • Page 87 IP SIP Phone v2 User’s Guide Mar. 2005 The network administrator has mapped 7 consecutive UDP ports, 45700 ~ 5706, from NAT to your terminal, which the terminal IP is 192.168.3.57. Note 1: Since the network administrator has to configure the NAT/firewall to map those UDP ports to your terminal statically, thus you should use static IP instead of DHCP as your network configuration.
  • Page 88 IP SIP Phone v2 User’s Guide Mar. 2005 Configure SIP service signaling port: Take the scenario above as an example: Transport: UDP and TCP (or “UDP”, you must include UDP anyway). SIP Listen port: 45706 Assign static NAT IP: s t u n : i s p . c o m Diagnose NAT (optional): you may detect your NAT IP by clicking ¡...

  • Page 89: Nat Traversal By Stun

    IP SIP Phone v2 User’s Guide Mar. 2005 Note, this diagnosis utilize STUN server, you must have assigned a valid/viable STUN server first. Static NAT IP: Fill in the acquired NAT IP from network administrator, such as 218.81.107.51 mentioned above. UDP traversal: Static NAT IP/UDP Map Note, if your NAT equipped with no fixed IP, such as those NATs dial into WAN by PPPoE, then you must synchronize the NAT IP currently set into IP SIP Phone...
  • Page 90 IP SIP Phone v2 User’s Guide Mar. 2005 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 [90/100]...

  • Page 91: Appendix A - Available Ntp Servers

    IP SIP Phone v2 User’s Guide Mar. 2005 Appendix A - Available NTP servers Service Level 2 NTP Server Service Area North America Ontario, Canada: University of tick.utoronto.ca Eastern Canada Toronto tock.utoronto.ca Quebec, Canada: Canadian ntp1.cmc.ec.gc.ca Eastern Canada Meteorological Center ntp2.cmc.ec.gc.ca Ontario, Canada: National time.chu.nrc.ca;...
  • Page 92 IP SIP Phone v2 User’s Guide Mar. 2005 dept. timeserver.cs.umb.edu Minneapolis / St Paul, MN: ns.nts.umn.edu; CICNET region University of Minnesota nss.nts.umn.edu Columbia, MO: University of 128.206.206.12: MOREnet Missouri-Columbia everest.cclabs.missouri.edu Omaha, NE: Radiks Internet 205.138.126.83: allison.radiks.net Midwest U.S. Access Las Vegas, NV: University of 131.216.1.101: cuckoo.nevada.edu NevadaNet, NSFNET, and Nevada System Computing...
  • Page 93 IP SIP Phone v2 User’s Guide Mar. 2005 Medicine region College Station, TX: Texas A&M 165.91.52.110: ntp5.tamu.edu NSFNET, SESQUI region, University THEnet, and TAMUSDSN Plano, TX: Greyware Automation tick.greyware.com; South-Central U.S. Products tock.greyware.com Blacksburg, VA: Virginia Tech ntp-1.vt.edu; Southeast U.S. Computing Center ntp-2.vt.edu Arlington, VA: Center for...
  • Page 94 IP SIP Phone v2 User’s Guide Mar. 2005 Budapest, Hungary: KFKI 148.6.0.1: time.kfki.hu HUNGARNET Research Institute for Particle and Nuclear Physics Italy: Net4u Srl, Vercelli, Italy 195.32.52.129: ntps.net4u.it Italy Oslo, Norway: University of Oslo 129.240.64.3: fartein.ifi.uio.no NORDUnet Oslo, Norway: Alcanet time.alcanet.no Europe International...
  • Page 95 IP SIP Phone v2 User’s Guide Mar. 2005 for Theoretical Physics Russia: Pushchino (near Moscow) 194.149.67.130: ntp.psn.ru Service area: Russia Chernogolovka, Russia: 193.233.46.10: sign.chg.ru Russia Chernogolovka Scientific Center (near Moscow) Far East and Pacific Rim Tokyo, Japan: Cyber Fleet, Inc. 203.139.30.195: ntp.cyber-fleet.net Japan and East Asia Seoul, Korea: Inet, Inc.
  • Page 96 IP SIP Phone v2 User’s Guide Mar. 2005 Appendix B – Trouble shooting To verify your network, please go to Advanced , then type a domain name to ping its ¡ ¡ reachability and/or aliveness, like “yahoo.com”, “iptel.org” or “fwd.pulver.com”. If the response is “Host unreachable”: i.
  • Page 97 IP SIP Phone v2 User’s Guide Mar. 2005 c. Proxy Server d. Registrar Configure SIP domain applied from service provider as appropriate. To apply for a public domain account, you may go to www.freeworlddialup.com or www.iptel.org. Check for NAT and Firewall settings: Go to ¡...
  • Page 98 IP SIP Phone v2 User’s Guide Mar. 2005 whether they convey private IPs (alternatively, you may activate by alternatively, ¡ i ¡ j you may press DSS key , which default maps to “Channel Info”, to activate ¡ ¡ Channel Info ): a.
  • Page 99 IP SIP Phone v2 User’s Guide Mar. 2005 failure cause is due to the mutual exclusion of both parties’ CODEC capabilities. For example, if you specify explicitly to use only G.723.1 for voice stream whereas the peer is only capable of G.711, then the conversation cannot proceed.
  • Page 100 IP SIP Phone v2 User’s Guide Mar. 2005 Appendix C – Tones TONE Usage NULL_TONE Stop tone playing Dialtone.wav DIAL_TONE Dial tone Re-call Dial tone.wav RECALLDIAL_TONE Dial tone on call transfer Alert Ring Tone.wav ALERT_RING_TONE Ringing on incoming calls RINGBACK_TONE RingbackTone.wav Ring back Busy.wav...

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