Jordan

How do you uninstall pbxnsip from a Mac? I'm clueless...

That has already been set up. I get into the mailbox using the PIN code, so that can't be the problem. The extension has permission to record the IVR node prompt, so that can't be the problem. When I am in the main menu of the mailbox and I press *, the prompt keeps speaking. I continue to dial the number followed by pound, and I am unable to access the feature...

I'm unable to use star codes in mailboxes. For example, I want to record the greeting for an auto-attendant or an IVR node. I log into the mailbox and am brought to the main menu. I press *9876# and I am told that I cannot place the call. I keep getting this message even though I have appropriate permissions. Am I doing something wrong? Has anybody else been able to make this work?

I keep getting the message that I am not allowed to place the call. I can place the call in the office.

Just to be a little more specific... When I dial that number (189426) from an outside number, I am told that my account does not have permission to dial that number. Before that prompt is finished speaking, I am told that the call has been disconnected by the system and then I hear a series of beeps and then I am disconnected.

How can I record the greeting for an IVR node by dialing into the auto-attendant from a remote location? I tried to set up a number (sort of like a PIN code) to just redirect to the recording extension. I set up 189246 to redirect to *9876. It works perfectly from a phone on the system, but not when dialing in. I did not set any recording permissions so that all accounts can record. Any ideas? Thanks!

I am curious to see what you all think are the best ITSP providers for PBXnSIP in terms of reliability, call quality, even cost. If you've had bad experiences with certain providers, I'd be interested to hear about those as well.

So I did some playing, and the Silence Supression doesn't seem to make a difference. Here's what I think that I have figured out. When echo cancellation is ON, it sounds the way I had described above. When it is OFF, the other party can be heard fine, but the person on our side of the PBX is echoed. Hmm...any thoughts?

Would that be the same as silence detection?

Calls are going in and out fine, however, we're having some issues with call quality. Whenever a person on our side of the pbx speaks, the opposite side is muted for about a half of a second followed by a half of a second of static when they stop speaking. What could be causing this and how can I fix it? The other party does not hear any of this, and they report that the quality of the call is fine. Thanks!

That really doesn't make things any simpler. I'd rather stick to recording blank noise on the IVR node.

Okay, then I propose a feature that will place a recorded message before the auto-attendant message is played. It would be nice if the message can be recorded or deleted by calling into the auto-attendant. If the message is recorded, it is played before the auto-attendant. If no message is recorded (or the last recording was deleted), the call will go straight to the auto-attendant recording.

I did that, but that complicates the system for people looking for a simple solution and aren't that technically savvy. It's especially confusing because the prompt says press * to delete, not hang up after the tone.

Anybody?

I would like to use this functionality, however, I would like it to still forward to the hunt group even if there is no audio file recorded. For example, in a school setting. The IVR node would always be called as the auto-attendant. If a message is recorded, it will play (for example, school is closed today due to snow) and then go to the auto-attendant. If there is no message recorded, it will simply follow the direction (!E!70) and go to the auto-attendant. Currently, nothing happens when dialing an IVR node with no recording.

I was also looking for a broadcast feature. It's very common and often very handy.

We particularly would be interested in using the paging functionality as a type of emergency alert system which would allow pre-programmed messages to be played when triggered. They can be simply pre-recorded audio files, but being able to speak text would make it much more dynamic. For example, we are using 1 key on each of the snom phones as a panic key, that by means of an action URL displays a message on computer screens in the main office. It would be great if we could page those extensions and the system can read "A Panic Button has been pushed at extension ___" That's the way that we would use the system, but there are definitely many other uses for such a feature.

In the next version, I would love to see features that support paging a pre-recorded message by dialing an extension or even an external application trigger. What would be even better is also text-to-speech paging capabilities.

In order to be able to use MultiCast Paging, we would like to upgrade to version 7 of the Snom software for our Snom 320 phones. I upgraded 2 phones to try out the new version, and I'm having some problems. I cannot get the system to function as a key system. On the Version 6 phones, I have co1@btbjpbx, co2@btbjpbx, co3@btbjpbx, and co4@btbjpbx set as lines on the function keys of the phones. I am able to see LED status of the phones as well as pick up calls that have been placed on hold from other extensions. However, on Version 7, I am unable to pick up a call from hold. I can see the LED status, but when I press the key to release the call from hold, I am given only a dialtone and the call remains on hold. I have contacted Snom support for some help with this, and they are telling me that there isn't much they can do for me. Hopefully you guys will be able to help. Thanks!

Thanks. I have the page buttons set up as a speed dial to 88, the paging account. Are you suggesting that I make it a push2talk button? In regards to version 7. How long do you anticipate the wait? And if there is a problem with version 7, can we go back to the current software on the phones? Thanks again.

I'm having an isssue with Unicast Paging and Snom 320 phones. I have a Unicast Paging group going to about 24 phones. The page goes out fine - a bit of a delay or echo, but I'm okay with that. Many of the phones do not hang up after the page is complete. How can we fix this? Should I be using multicast paging? If so, I've been looking into that and trying some things and I am unsure of how to get it to work with the snom phones. Thanks so much for all of your help.

Hi again- Writing this time to let you know that I finally got it to work!! I set the min delay to 750 and the current disconnect to 300 and presto! it works! Not sure why those settings work that way, but that's what grandstream told me to do, and it worked no problem! Thanks again for the help!

I've been continuing to play with this for a little while and inbound calls get into the system after 2 rings and outbound calls are still hit or miss. Hopefully that means something?

Hi Bill. Thanks so much for your help. Here is what I did. I upgraded to the latest firmware. .55 i think? I tried what you said, but it didn't work for me, so here is what I did. I set the trunk as an outbound proxy. Set the Outbound Proxy to the IP of the gateway, set the redirection, and co1...co4. On the gateway, I have Wait for dialtone set to yes, one stage dialing, and 100 s wait before dial. I can make inbound calls and it works great every time, but outbound calls are hit or miss. Sometimes it works, sometimes it doesn't. Do you have any suggestions with what I should try? I'm not sure, but I think the problem is that the outbound call doesn't always make it as far as the gateway. I'm so confused!! Thanks again, I really appreciate it.

Hi everyone. I've been trying for the past 3 days to set up the Grandstream GWX-410x gateway with my installation of PBXnSIP. I have tried everything and I cannot get any incoming or outgoing calls to work. Here is what I have done so far. If anybody can help me out with this, I would really appreciate it. The IP Address of the gateway is 192.168.1.5 The IP Address of the PBX Server is 192.168.1.108 I set up a Trunk called PSTN Gateway. I set the Outbound Proxy to the IP of the gateway. Under CO Lines I have co1 co2 co3 co4. The extension is set to 70 which is our auto-attendant. Everything else I left as default. I'm not sure what to put as the username or password. I left the username blank and tried the administrator password for the gateway and also the user password. I set up a dial plan to use this trunk. The pattern is 9* and the replacement is empty. At the gateway, I left almost everything to the default values. But I changed to one stage dialing. Under profile 1, I set the IP Address of the SIP Server and clicked NO to SIP Registration. Everything else is default. Please Please Please someone help me out. I have tried just about everything and I'm not sure why this is not working. I really would appreciate your help. Thanks, Jordan