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Comrex VH2 Product Manual

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Summary of Contents for Comrex VH2

  • Page 1 Product Manual...

  • Page 2: Table Of Contents

    II. settIng uP VH2 HardWare attacHMents audIo connectIons contact closures MakIng connectIons III. confIgurIng VH2 IP InforMatIon IV. confIgurIng tHe VH2 coMPanIon PHone tIMe settIng VH2 settIngs V. telePHone connectIons IntroductIon to sIP settIng uP a sIP ProVIder or Pbx...
  • Page 3 New SectioN - extra settIngs - confIgurIng for a Pbx gateWays - general settIngs - - account InforMatIon - - sIP settIngs - adVanced gateWay settIngs dIal PrefIx outgoIng enabled sIP trunks lIne assIgnMents VI. systeM beHaVIor auto-ansWer studIo audIo I/o about MIx-MInus caller MIx caller duckIng...
  • Page 4 (cH 1 and 2) Hold/xfr (cH 1 and 2) IndIcatIons general oPeratIon IncoMIng calls outgoIng calls transferrIng to Handset endIng calls toolbox call controls xI. about sIP xII. InforMatIon for It Managers about VH2 IncoMIng serVIces outgoIng serVIces xIII. contact and suPPort...
  • Page 5 xIV. WHy do I Hear Hear Myself Myself? MIx-MInus and elIMInatIng ecHo xV. softWare lIcenses gnu PublIc lIcense VersIon 2 gnu general PublIc lIcense gnu general PublIc lIcense gnu PublIc lIcense VersIon 3 lesser gnu PublIc lIcense VersIon 2.1 gnu lesser general PublIc lIcense MozIlla PublIc lIcense VersIon 1.1 strace xVI.

  • Page 6: Introduction

    A block diagram of VH2 audio connections is shown in the following figure. In this simple configuration, a feed of the station’s program audio (pre-delay) is fed to the HOLD AUDIO CH 1 port on the VH2. A studio “mix-minus” feed is attached to the INPUT CH 1 port, and the caller(s) audio (for delivery to a single console fader) is available on the OUTPUT CH 1 port.
  • Page 7 Many applications require that multiple callers be put on-air simultaneously. For this reason, VH2 contains two separate hybrids* that allow each caller to be presented on a different output from the VH2. This allows you to balance each caller on a separate console fader.
  • Page 8 *The term Hybrid is a legacy from analog telephony products, which had the task of separating send and receive audio on the phone line. Since VoIP calls don’t mix this audio, a Hybrid is technically not employed, but rather a conference is built for each output. For simplicity, we’ll continue to use the legacy term for an on-air telephone channel.
  • Page 9 Finally, VH2 is capable of behaving like two, completely independent products for use in two, unrelated studios. This is shown below.

  • Page 10: About Mix-Minus

    MIx-MInus Comrex support spends a lot of time discussing mix-minus, as it is not an obvious concept. In studio telephony integration, there is a golden rule: People connecting from outside the studio must never be sent their own audio back to themselves.

  • Page 11: Setting Up Vh2

    VH2 HardWare attacHMents INPUTS OUTPUTS HOLD POWER SERIAL AUDIO IN CH 2 CH 1/AES CH 1/AES CH 2 VH2 rear Panel CH 1 INPUT - In analog mode, this XLR connector should be sent a balanced, 0dBu signal that is heard by callers when they are “on-air”.

  • Page 12: Audio Connections

    audIo connectIons All analog XLR audio inputs have a nominal level of 0dBu (full scale +20dBu). When the input mode is changed (via DIP Switch #1) to AES3 mode, a 48KHz signal will be outputted. 48KHz is the only supported sampling rate. Analog input and output pinouts: AES3 input and output pinouts: Pin 1...

  • Page 13: Contact Closures

    MakIng connectIons At a minimum, VH2 will need two audio connections and a network connection. Levels of all analog audio I/O is 0dBu (0.775V) nominal. This level will provide 20dB headroom before the clipping point. Input audio is reflected on the front panel LED based peak meters as indicated in .

  • Page 14: Iv. Configuring The Vh2 Companion Phone

    “off-air”. VH2 supports this function with the approved VoIP companion phone (currently the Polycom IP 331). The VH2 can emulate the function of sharing a call between the hybrid and the phone, and “bouncing” a call between them with a single button push. The phone will require special settings (as outlined in section IV.

  • Page 15: Configuring Vh2 Ip Information

    III. VH2 is shipped from the factory set to DHCP mode, so it will find an address on your network if possible. The easiest way to find what that address is to use the Comrex Device Manager software, available on our website for both Windows and Mac platforms.
  • Page 16 If the 5 minute timer has not run out, a pop-up menu will appear allowing you to reconfigure the IP address, netmask, Gateway and DNS information and save it away. You will need to restart the VH2 for these changes to take...

  • Page 17: Time Setting

    VH2 coMPanIon PHone If you have the Polycom 331 IP phone for use with VH2, you’ll need to set it up to be an extension of the VH2. The phone needs to be on the same physical LAN as the VH2, but it doesn’t need a static IP address. You’ll only need to program the static address of the VH2 into the phone.
  • Page 18 Under Server 1: Address-><VH2_ip_address> (remove <>) Port->5170 Transport->DNSnaptr Expires->100 Register->1 Retry Timeout->0 Retry Maximum Count->3 Line Seize Timeout->30 (leave the rest of the fields at default) Your Line 1 Page should read like this (but with your VH2 IP address)
  • Page 19 Click Submit on the bottom of the page, then go to line 2 On the Line 2 page, set the following fields: Under Identification: Display Name -> ‘Line 2’ Address-> ‘1200@<VH2_ip_address>’ (remove ‘ and <>) Authentication User ID -> ‘1200’ Authentication Password ->...
  • Page 20 Line Seize Timeout->30 (leave the rest of the fields at default) Your Line 2 Page should read like this (but with your VH2 IP address) Click the Submit button on the bottom of the page. You are finished with the telephone setup.

  • Page 21: Telephone Connections

    IntroductIon to sIP SIP (Session Initialization Protocol) is the standard used by VH2 to talk to virtual phone lines. These lines must be created in some way externally before they are “applied” to VH2. “Applying” SIP lines to VH2 involves configuring the unit with certain information about the lines and the location of the server that delivers them.

  • Page 22: Setting Up A Sip Provider Or Pbx

    Along with account credentials, you’ll need a Direct Inward Dial (DID) number associated with your account. This is the “old fashioned” phone number users will dial to reach you. VH2 does not need to know this number - translation to the proper SIP channel happens behind the scenes at the SIP provider (although often the DID and SIP account name are the same).

  • Page 23: Advanced Sip Settings

    Under most circumstances this is all you should need. Setting these parameters, clicking Back then clicking Restart should start the process of VH2 registering with the provider or PBX. However, SIP registration can be tricky in some systems, and if registration fails you should check the required SIP settings carefully and use the Advanced SIP settings.

  • Page 24: Under Sip Settings

    Many providers will ignore this. Provider Binding Port - This port is assigned by VH2 based on the number of providers you have assigned. Unless required, you should leave the default setting as is.

  • Page 25: Configuring For A Pbx

    In the case where you wish to set up incoming lines as extensions of an upstream PBX, the instructions are very similar. Your PBX will deliver channels to VH2 in the same way a SIP provider does, and you will need to set up the PBX and retrieve the proper credentials to program into the SIP Provider fields in VH2.

  • Page 26: General Settings

    The figure above shows the settings for gateways (Line Configuration > VoIP Providers > Add Providers > SIP Gateway Device). Many of the settings are populated automatically by VH2, but can be changed to any value you wish. - general settIngs - Name - Give your gateway a unique name.

  • Page 27: Advanced Gateway Settings

    adVanced gateWay settIngs dIal PrefIx Some Gateways require a certain prefix be dialed in order to select a particular legacy port for outgoing calls. If this is required for your Gateway (e.g. 991) enter it here. outgoIng enabled Allow outgoing calls on channels using this Gateway using the VoIP companion phone. sIP trunks With regard to STAC VIP, we refer to SIP Trunks as provider accounts that don’t require registration with a provider’s server.
  • Page 28 Destination Number field of incoming calls and accept only calls with literal matching text. This field can also be set as a “regular expression” for a dial plan, which is a more complex topic and can be handled by Comrex support. As an example, entering the following value in the field: ^(1\d{3}555210\d)$ Would match the sequence “1xxx555210x”...

  • Page 29: Line Assignments

    Once you’re registered with at least one provider, gateway, or PBX, you can assign the VH2’s “channels” to that provider. Important: Simply registering is not enough. VH2’s channel(s) must be assigned to your provider(s) before you can use it.

  • Page 30: System Behavior

    MIx-MInus When we refer to “send” audio to the caller, we’re talking about the feed that is attached to the VH2 CH 1 INPUT and CH 2 INPUT. This is the audio that the caller hears when “on-air”. It is essential that these feeds are specially mixed so that the caller output is not part of that mix.

  • Page 31: Agc

    Selects whether VH2 applies AGC to the caller outputs, helping minimize large level changes between calls. caller duckIng Selects whether VH2 applies an algorithm to the caller audio to reduce it when the host voice is detected. This allows the host to “dominate” the conversation. Ducking, if enabled, has three choices (Low, Medium, High) that allows selection of how much the caller is reduced when the host speaks.

  • Page 32: Test Modes

    These options are used for diagnostics or demo purposes only and will interfere with normal operation when enabled. Make sure all test modes are off before operating VH2 normally. Audio Test - These options provide for specific audio paths to be enabled (e.g., CH 1 •...

  • Page 33: Network Configuration Page

    The default settings allow for a pool of public servers to be used. - IP settIngs - As described in the previous section “Configuring VH2 IP Information”, this is where you set the static IP address of the VH2’s main Ethernet port.

  • Page 34: Primary Network

    Ethernet port. Note that once this is turned off, no further configuration of VH2 is possible. To change any setting you must first apply a factory reset to the VH2, which will wipe all its settings, including all your VoIP account info and static IPs. Turning services off also disables your VH2’s ability to sync with the Comrex Device Manager (until a factory reset...

  • Page 35: Security And User Accounts

    The default password is “456”. This is also the password used to gain access to the full configuration of the phone via the web or the keys on the phone. If you wish to change this password from the default, you can let VH2 know the new value in this field.

  • Page 36: Factory Reset

    VH2 can be restored to factory settings, clearing all IP, behavior and provider info, two ways: If the password is known, use the Device Manager software to login to VH2. Select Device- >Reset to Factory Defaults to issue the reset command.

  • Page 37: Operating Vh2

    This is a toggle that places an active call “on-hold” from the “on-air” state. The caller will be removed from the main audio ports, and hear only the audio presented to the VH2 “on-hold” input. Pressing this button while an incoming call is ringing will send the caller directly to “on-hold”...

  • Page 38: Indications

    IndIcatIons The ON/OFF and HOLD/XFR buttons are lit to indicate the state of each particular channel ON/OFF HOLD/XFR Function Idle-registered and ready for call Blink Green Ringing Green On-Air Green Green Hold Blink Green Blink Green On Handset Auto-Answer No Provider Assigned Blink Red Provider Assigned, but not registered The “Ready”...

  • Page 39: Ending Calls

    endIng calls From “on-air” or “on-hold” state, press the ON/OFF button to end the call. From the handset, simply hang up. toolbox call controls When logged into the web-based Toolbox utility, the last item listed is Control. The status of both Channel one and Channel two are presented here, as well.
  • Page 40 This menu gives you the ability to control each channel either by placing the calls on-air, on-hold, transferring to the handset or dropping the call. This allows you to control the VH2 channels without having to physically push the buttons.

  • Page 41: About Sip

    VoIP telephone, or software running on a PC or mobile device that performs the same functions. The Comrex STAC VIP is a sample of a device designed to interface with VoIP service. It can handle six or twelve calls simultaneously and provide the typical screening, audio processing, and control functions expected of broadcast call-in systems.
  • Page 42 Ports The IP address is the main identifier used to specify a destination to send packets to within a network. But since IP compatible devices can make simultaneous connections for different reasons (e.g. web surfing and email), a scheme is used to designate a specific “port”...
  • Page 43 Known as Network Address Translation (NAT), it’s easiest to use a diagram to illustrate a typical gateway scenario describing a user on a LAN accessing a web page at comrex.com. For this illustration, we’ll ignore the concepts of DNS and URLs (which aren’t particularly useful for VoIP) and live the fantasy that the user is...
  • Page 44 The router will record the source address of the packet (192.168.0.7), change it to the public IP of the router (74.94.151.151), and send it along to the destination IP address. This is so the web site knows the correct address to which to respond. The router will now wait for the response from the web site (it’s smart enough to know to expect something from the destination address of the packet it sent).
  • Page 45 Real Time Protocol A fundamental building block of VoIP is the Real-Time Protocol (RTP). This is a protocol layer that exists within a UDP packet specifically designed to transfer audio (and video) media with low delay. RTP consists of a header that is applied directly after the UDP header in the packet, followed by a media “payload”...
  • Page 46 G.711 actually has two variants, one used mostly in North America (μ-law), and another used elsewhere (a-law). These are defined by the names of the tables used within the encoders to compress. All Comrex codecs and VoIP devices support G.711.
  • Page 47 The 7 KHz spectrum carried by G.722 covers the majority of human voice energy, excluding only the most sibilant sounds in speech. G.722 is the most common encoder for calls that are classified as “HD Voice” in the VoIP world. All Comrex codecs and VoIP devices support G.722.
  • Page 48 SIP connections can be made in two primary ways--registered and unregistered. In unregistered mode, a SIP channel is opened between devices at the time a call is placed. In registered mode, a SIP channel is constantly maintained between a SIP client (like a studio talkshow system) and a SIP server (like that at an Internet Telephone Provider). Most VoIP users will only use registered mode, so that’s what we’ll focus on going forward.
  • Page 49 Sometimes the SIP channel is connected to a server that is removed from the RTP sessions entirely. This would likely be the case when two SIP devices are registered to the same (or sometimes even different) providers. The SIP channel would instruct the devices to create RTP sessions between them, rather than to the provider.
  • Page 50 Challenges with SIP/RTP To summarize the previous sections, most VoIP connections involve a continuously active SIP channel initiated from the user device to a service provider over port UDP 5060. Using this channel, the two ends negotiate calls and create and destroy RTP sessions (each consisting of one RTP and one RTCP) in each direction.
  • Page 51 Don’t block me, bro! Even if the provider gets the correct IP address of the user, there’s plenty that can go wrong. Remember, SIP involves creating extra RTP “channels” in each direction to carry the actual voice. The ports used on each end are negotiated over the SIP signaling channel for each call.
  • Page 52 In reality, an ALG may often take quite a bit of license with your SIP connection. It can rewrite many of the SIP fields in order to comply with its rules, so the IP and port information getting to the service provider may actually be completely different than those sent by the device.

  • Page 53: Making And Receiving Calls

    Registering with a SIP Server or PBX The process of registering a device to a SIP provider, whether it’s in the “cloud” or at your location, is usually simple. Much like registering an email client with a mail server, the VoIP client (the VoIP hardware) must know the location of the server, and a username/password combo with which to register.

  • Page 54: Information For It Managers About Vh2

    Typically, SIP services rely on the presence of a SIP ALG within the firewall to open RTP ports. If Comrex support is required, we may ask for access to the SSH host on the VH2 on TCP 22. SSH service is protected by a private keypair which is not delivered to customers.

  • Page 55: Contact And Support

    Comrex Corporation 19 Pine Road Devens, MA 01434 Technical Support is available Monday-Friday 8:30AM-5PM EST. 1-800-237-1776 (North America) 1-978-784-1776 (International) 1-978-784-1717 (FAX) techies@comrex.com email Product manuals and firmware updates available on the web at: http://www.comrex.com...

  • Page 56: Why Do I Hear Hear Myself Myself

    WHy do I Hear Hear Myself Myself? xIV. MIx-MInus and elIMInatIng ecHo Studio telephone integration is a two-way process. The caller must send his audio to the studio, but also receive a return feed that allows him to interact with other sources, like a host. An important element of voice telephony involves allowing a speaking party to hear his own voice in his own earpiece.
  • Page 57 The telephone network employs digital echo cancellers at various nodes along the path of a phone call to avoid this scenario. And when they malfunction or are “untrained” at the start of a call, the effect is a dramatic echo in the caller’s ear.
  • Page 58 “send” input will meet those needs. Some studio telephone systems, like Comrex VH2, allow telephone callers to appear on one of two outputs (and therefore on two, separate console faders). In this circumstance, you often have a choice of delivering a single mix-minus with neither of the telephone audio sources present, or two distinct mix-minus feeds.
  • Page 59 Finally, if you’re stuck in a studio where you need to create multiple mix-minus feeds with no resources to do so, investigate the Comrex Mix-Minus Bridge product. It’s an easy way to deliver up to six distinct mix-minus feeds (e.g.

  • Page 60: Software Licenses

    Agreement shall otherwise remain in full force and effect and enforceable. The failure of Comrex to act with respect to a breach of this Agreement by You or others does not constitute a waiver and shall not limit Comrexs rights with respect to such breach or any subsequent breaches.

  • Page 61: Gnu Public License Version 2

    gnu PublIc lIcense VersIon 2 linux module-init-tools udev e2fsprogs busybox bash tcpdump alsa-utils ethtool acpid usbutils procps fxload gnu general PublIc lIcense Version 2, June 1991 Copyright (C) 1989, 1991 Free Software Foundation, Inc. 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed.
  • Page 62 You may charge a fee for the physical act of transferring a copy, and you may at your option offer warranty protection in exchange for a fee. 2. You may modify your copy or copies of the Program or any portion of it, thus forming a work based on the Program, and copy and distribute such modifications or work under the terms of Section 1 above, provided that you also meet all of these conditions: a) You must cause the modified files to carry prominent notices stating that you changed the files and the date of any change.
  • Page 63 If any portion of this section is held invalid or unenforceable under any particular circumstance, the balance of the section is intended to apply and the section as a whole is intended to apply in other circumstances. It is not the purpose of this section to induce you to infringe any patents or other property right claims or to contest validity of any such claims; this section has the sole purpose of protecting the integrity of the free software distribution system, which is implemented by public license practices.

  • Page 64: Gnu Public License Version 3

    You should also get your employer (if you work as a programmer) or your school, if any, to sign a “copyright disclaimer” for the program, if necessary. Here is a sample; alter the names: Yoyodyne, Inc., hereby disclaims all copyright interest in the program `Gnomovision’...
  • Page 65 “Copyright” also means copyright-like laws that apply to other kinds of works, such as semiconductor masks. “The Program” refers to any copyrightable work licensed under this License. Each licensee is addressed as “you”. “Licensees” and “recipients” may be individuals or organizations. To “modify”...
  • Page 66 You may charge any price or no price for each copy that you convey, and you may offer support or warranty protection for a fee. 5. Conveying Modified Source Versions. You may convey a work based on the Program, or the modifications to produce it from the Program, in the form of source code under the terms of section 4, provided that you also meet all of these conditions: •a) The work must carry prominent notices stating that you modified it, and giving a relevant date.
  • Page 67 When you convey a copy of a covered work, you may at your option remove any additional permissions from that copy, or from any part of it. (Additional permissions may be written to require their own removal in certain cases when you modify the work.) You may place additional permissions on material, added by you to a covered work, for which you have or can give appropriate copyright permission.
  • Page 68 In the following three paragraphs, a “patent license” is any express agreement or commitment, however denominated, not to enforce a patent (such as an express permission to practice a patent or covenant not to sue for patent infringement). To “grant” such a patent license to a party means to make such an agreement or commitment not to enforce a patent against the party.

  • Page 69: Lesser Gnu Public License Version 2.1

    END OF TERMS AND CONDITIONS How to Apply These Terms to Your New Programs If you develop a new program, and you want it to be of the greatest possible use to the public, the best way to achieve this is to make it free software which everyone can redistribute and change under these terms.
  • Page 70 To protect your rights, we need to make restrictions that forbid distributors to deny you these rights or to ask you to surrender these rights. These restrictions translate to certain responsibilities for you if you distribute copies of the library or if you modify it. For example, if you distribute copies of the library, whether gratis or for a fee, you must give the recipients all the rights that we gave you.
  • Page 71 c) You must cause the whole of the work to be licensed at no charge to all third parties under the terms of this License. d) If a facility in the modified Library refers to a function or a table of data to be supplied by an application program that uses the facility, other than as an argument passed when the facility is invoked, then you must make a good faith effort to ensure that, in the event an application does not supply such function or table, the facility still operates, and performs whatever part of its purpose remains meaningful.
  • Page 72 c) Accompany the work with a written offer, valid for at least three years, to give the same user the materials specified in Subsection 6a, above, for a charge no more than the cost of performing this distribution. d) If distribution of the work is made by offering access to copy from a designated place, offer equivalent access to copy the above specified materials from the same place.

  • Page 73: Mozilla Public License Version 1.1

    PARTICULAR PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE LIBRARY IS WITH YOU. SHOULD THE LIBRARY PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION. 16. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR REDISTRIBUTE THE LIBRARY AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE LIBRARY (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF...
  • Page 74 a.Any addition to or deletion from the contents of a file containing Original Code or previous Modifications. b.Any new file that contains any part of the Original Code or previous Modifications. 1.10. “Original Code” means Source Code of computer software code which is described in the Source Code notice required by Exhibit A as Original Code, and which, at the time of its release under this License is not already Covered Code governed by this License.
  • Page 75 3.3. Description of Modifications. You must cause all Covered Code to which You contribute to contain a file documenting the changes You made to create that Covered Code and the date of any change. You must include a prominent statement that the Modification is derived, directly or indirectly, from Original Code provided by the Initial Developer and including the name of the Initial Developer in (a) the Source Code, and (b) in any notice in an Executable version or related documentation in which You describe the origin or ownership of the Covered Code.
  • Page 76 Once Covered Code has been published under a particular version of the License, You may always continue to use it under the terms of that version. You may also choose to use such Covered Code under the terms of any subsequent version of the License published by Netscape. No one other than Netscape has the right to modify the terms applicable to Covered Code created under this License.
  • Page 77 13. Multiple-licensed code Initial Developer may designate portions of the Covered Code as “Multiple-Licensed”. “Multiple-Licensed” means that the Initial Developer permits you to utilize portions of the Covered Code under Your choice of the MPL or the alternative licenses, if any, specified by the Initial Developer in the file described in Exhibit A.

  • Page 78: Strace

    strlcat() is (c) Todd C. Miller ===== Import code in keyimport.c is modified from PuTTY’s import.c, licensed as follows: PuTTY is copyright 1997-2003 Simon Tatham. Portions copyright Robert de Bath, Joris van Rantwijk, Delian Delchev, Andreas Schultz, Jeroen Massar, Wez Furlong, Nicolas Barry, Justin Bradford, and CORE SDI S.A.

  • Page 79: Programming Your Polycom Ip 331 For Use With Your Pbx, Cloud-Based Phone Service Or Ip Gateway

    Pbx, cloud-based PHone serVIce or IP gateWay dIgIt MaP IntroductIon After you have associated your Polycom IP 331 companion phone to your VH2, you will most likely need to program the Polycom to make calls within your company (office extensions) as well as make outbound calls.
  • Page 80 Polycom has provided a Tech Tip document for understanding digit maps. The following is an excerpt to help in understanding the default digit map for the Polycom IP 331. To read the full technote, visit the following url: http://support.polycom.com/global/documents/support/technical/products/voice/Understanding_Digit_Maps_ Tech_Tip.pdf The default digit map for the Polycom IP 331 is as follows: [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT This digit map shows there are 6 matching options, each separated by the vertical bar “|”.

  • Page 81: Digit Map Time-Out

    PolycoM dIgIt MaP Although it is recommended to have your System Administrator edit your digit map, Comrex understands that is not always a possibility. Here are a couple of steps to make some of the common changes to your digit map.
  • Page 82 The Digit map entry field is where you will make any edits necessary for the Polycom IP 331 to work with your specific PBX. Once you have completed any edits in this entry, you MUST press the Submit button to apply the changes.
  • Page 83 edItIng tHe dIgIt MaP for Internal extensIons The default digit map allows for calls to extensions with 4 digits starting with the numbers 2-9. (Represented by the last entry in the code: [2-9]xxxT ) For internal extensions of 3 digits, overwrite/edit the digit map code with the following: [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxT For internal extensions of 5 digits, overwrite/edit the digit map code with the following: [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxxT...

  • Page 84: Sip Trace

    To perform a SIP Trace, select Trace Target. Select All to capture all traffic to and from the VH2, or select one of your SIP Providers from the drop-down menu to perform a packet capture for one specific provider.
  • Page 85 When you would like to end the capture, simply select Stop Trace. Once pressed, a new button titled Download Trace becomes available. Press this to download the packet capture file (a .pcap extension). This will start an http download to your browser. To review this file, we recommend using Wireshark, a free and open source packet analyser available online.

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