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MarkW S7

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About MarkW S7

  • Birthday 05/15/1987

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    South Texas

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  1. I was curious to know if anyone has had the request or have disabled the DND button on a Polycom handset. I have a client that has several 550's and their employees keep pressing the DND button; on accident or on purpose unknown. They've requested that we disable this feature which I think can be done on the PBX, but also I wanted to check if it can be done on the Handset.
  2. I know this is more of a hardware issue and unrelated to PBXnSIP's side of things, but I need to make a ringer louder. I have a customer that has a P550 in a loud environment: Bar. They can't hear the phone ring during hectic times. Anyone know of a way to boost the ringer louder? I've contemplated getting a sip page horn with nightbell, but I've not had the chance to work with one yet. Any ideas? Thanks
  3. This was addressed in a fix to the PBX and is no longer an issue. I just was clarifying that this is complete.
  4. We're also doing the Faxback as well. Personally I am excited about the inclusion of the software on the AC202's instead of using the FMIS software. Windows updates kills the service continuously as you probably have encountered. I currently have great success with T.38 faxing using both Audio Codes 20x and also Linksys SPA2102. We prefer not to do T.38 faxing to free up call overall capacity.
  5. Are we able to force Authentication though?
  6. Nevermind. I figured it out solo.
  7. Hmm. Sounds like he's not getting authenticated and just passing calls. This is definitely a question for the for our admins. I've always been curious if we can force authentication on our subscribers without the using MAC addresses. In the meantime try locking up the interface and charging a technical fee with a notice of potential fraud .
  8. I have a Snom 370 utilizing two park orbits on buttons 1 (ext 973) and 2 (ext 974). When on an active call there's a few things I'd like to see different or have explained. 1. On an active call I press Park 1, the IVR then says "Your call has been parked on orbit 973" and then parks disconnects. The screen shows *85973. - Can I remove the Park Orbit notification? - Can I not have the star code shown? (this is a very minor question and not a complaint) 2. When I want to retrieve a call that is on one of the park orbits I press the Park 1 Button. - The caller id for the active call shows *601111539 (retrieved park id?) This is confusing my customer's receptionists. Can this be modified or removed? If it helps these park orbits are set up as standard extension accounts with the mailbox enabled as park orbit. Also the extension accounts have the "Explicitly specify park orbit preference" listing both orbits and the "Explicitly specify pickup preference" listing both orbits in attempt to address the above items. Thanks for any help or advice -wheezey
  9. Here's a simplified question. The time that is listed in the fields, are these times when the flag is active or inactive? It would be much easier if the time was listed as inactive as it could input the business hours and not have to start at 00:001 and end at 23:59... Thanks -wheezey
  10. Have you locked the admin interface? To password protect the admin login along with the user login; you can change it's password in the Voice//System page. I also recommend having the spa2102 only accessible on a different http port besides 80. I put my ATA's on a separate port for security and also for remote management if we have the ability to change/request the NAT rules. If this end user is changing his login user id settings along with his proxy address and you use the same SIP password across your accounts, then he's just registering with ease. I'm not sure if you can set up a authentication challenge for these ATA's with pbxnsip. My recommendation is lock down those interfaces if not only the admin access. Also I'd recommend putting a "1" in the Lines field of the registration page of the account's settings on the PBX. I'd recommend putting in the MAC address in the registration page also, but note that if that ATA ever changes you'll need to update that setting or face some negative results.
  11. I'll explain my call flow: Hunt Group Stage1: extensions 12sec Final Stage: 900 (Main AA) Service Flag: 960 Service Flag Number: 900 The goal is to have calls ring extensions in the hunt group during business hours 07:30-17:30. Outside of those hours I want the calls to automaticly route past the hunt group's extensions to the AA directly. So now on to my 960 Service Flag configuration. Mode: Day/Night Monday: 00:00-7:30 17:30-23:59 (repeated Wed-Thur) Friday: 00:00-7:30 17:30-23:59 I feel like I'm going about this backwards, but is this the proper set up for this? Thanks in advance -wheezey
  12. We are seeing a ton of SUBSCRIBE's from only our Polycom customers and it's consuming a TON of CPU resources. Is anyone aware of a bug? We run pbx version 3.3 and the latest Polycom firmware. Any thoughts on how to reduce this?
  13. Ok here's what I would love if possible: Call comes into our AA. Caller presses whatever option that takes them to our support queue.The support queue has 5 accounts in it all ring simultaneously for the entire duration of the caller waiting in the queue. After a certain amount of time let us say 30 seconds, We would like it to include our on call techs' cell phones. So caller here's nice Agent group music with looping statement while all support handset ring and two cell phone are now able to answer. If no accounts or even the added cell's don't answer then the agent group fails out to a voice mail account. Any ideas? I've been trying many different combination's with ring durations, hunt groups, etc.. This call flow would be used in my office so if it can't be done than it's "ok".
  14. Going ahead and making a thread for these devices. If anyone has any questions regarding these let me know and I'd be more than happy to share. Linksys SPA8000 This is an 8 fxs port ATA (Analog Terminal Adapter or Analog Telephone Adapter). You can either connect the lines through the fxs ports or through a RJ-21 (50-pin telco connector). That is great if you want to maintain the carrier demarc style cross connect. Beyond the physical connections, internally the spa8000 offers a whole host of great features and flexible voip settings including offering 4 internal trunks. One huge advantage that I feel that this and the spa2102 have over other ATA (like the Budget Tone or Audio Codes MP's) is that each configurable line is proxy independent. What I mean by this is Line 1 can talk to a completely different proxy or outbound proxy from Line 2. Think of the possibilities in redundancy carriers, a bridge between 2 domains, etc.. The only major thing to note about these devices is that they can only run in a LAN/Host mode, and not as the gateway/router like it's brother the SPA2102 Linksys SPA 2102 This device uses the same OS that the SPA8000 uses and has almost the exact same functionality and features. The main difference is that the 2102 can run as a basic router with dhcp, nat, and limited QoS. Other differences include: 2 fxs ports, device can be set up as a switch and not a router, no trunking features. Both devices support t.38, adjustable DTMF mode/methods, a whole slew of settings that would clutter this board very quickly.
  15. Looks like I have a new customer that will be using the new Audio Codes MP-204 B 4FXS (running 2.6.4_p5_1_build_9) so I can get some lab bench time in. Now this device is almost identical to other audio codes device except of course the additional FXS ports. I've reg'd two accounts (one was for duplication testing) to the mp204 and hooked a plain $10 analog handset to the "line 1" port. I have the CW setting in the Services section checked as enabled and the Call Waiting SIP Reply as Ringing I initiated a call to my cell from the analog handset, and then during that active call I dialed in from another phone. The analog handset gave me a CW tone (during this tone all audio is interrupted; like normal) but immediately after the tone, both the analog handset and my cell pass audio just fine even with additional tones (not during tone but after). I did the same thing with the "line 2" account. I cool thing to note is that when I hit flash the first call gets MoH. I'm trying the Call Waiting SIP Reply as queued and then will also try changing the signaling to connect media on 180.
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