samfiller

Thank you, I am running 4.3.0.5021 (Win32)- Windows Thank you again for you help.

My new SMTP server service requires me to verify the domain by placing a file in the root directory. I am unable to find the PBX's root directory. Anyone know where this folder is? Thank you.

Interesting I spoke with Michael D. From Snom November of last year and he explained that i would need to stay on PBXNSIP Build. Anyways who do I contact to upgrade? I am running 4.3.0.5021 Thank you

Any chance PBXNSIP will be updated?

Hello all, I searched the Wiki the Forum and the internet, and cant seam to find a meaning behind this error: DNS: Cannot resolve empty DNS address - your help is appreciated.

Got it, its working now. I guess it should really be called "header Link" . Thanks for the help.

I think I found that it has to do with "Logo Link:" under appearance I am not sure what its looking for.

Hello all, I keep seeing in my logs the following: [3] 2011/11/16 17:08:05: Could not read file http:\\pbx.myserver.com (error code 123) [6] 2011/11/16 17:08:05: Webserver: Could not read file http://pbx.myserver.com Maybe someone can be so kind to tell me, Is it something to be concerned about? And what need to be corrected not to have this error? Thanks!!!

Thank you, I understand that. I just assumed that a registered trunk would be an entity of its own and only "registered traffic" would pass through.... Well you learn something new every day. Thanks!!

I added the call routing in multiple trunks and all seems to work. I guess the question still remains why a registered trunk will allow not registered traffic from the same IP as the registration server.

Thank you for the response, I was hoping the answer was YES ;-) The issue I am having is that I have a trunk that is registered and all the DID's on that registered account are going to extension 800 - via send call to extension. The issue is I have a SIP gateway setup for the same IP as that Registered trunk to except not registered traffic and have set to go to a different extension (850). The issue seems to be that the registered extension is "excepting" the traffic and therefor the calls are going to the wrong extension (800 and not 850). I tried alias and it did not work. I assume if I program the info on the wrong trunk it will work BUT that is for sure not the correct way to do it. And I would have to use some sort of expression of all calls go to 800 except this DID go to 850. A global DID directory would solve this issue But I guess its a bug for a registered extension to except other traffic. Any insight?

My question is inbound handling: From what I understand there are two ways to route an inbound call. One is via an Alias on the extension matching the DID that is being called. The second way is in the trunk setup, setup "Send call to extension:" either putting an extension in there or by using Regular Expressions for a simple example !(8005551212)!800 !(2121234567)!800 I am having some issues that would take a very long time to explain (calls going to the wrong extension..) A solution for me would be to be able to create a document for example DID.xml and in the document list 100 DID in this format: !(8005551212)!800 !(2121234567)!800 etc......... and some how program on all my trunks by Send call to extension: DID.xml . In short have one global list that no matter what trunk this DID hits the system via it will use the global mapping to go to the correct extension. Alias's would probably work but it gets way to messy when you are dealing with lots of DID's. Any help would be greatly appreciated. While I am posting a side question is there a way for the PBX to dip into CNAME? or in order to have CNAME your provider needs to send the DID with the CNAME.

Randoff, I was forwarding a complete extension- First I created a trunk for outbound as a SIP Gateway. for Domain: and proxy Address: I put the second half of the uri (ip or domain) then set up a dial plan to use that trunk for pattern and replacement I put * then I went to the extension on the redirection page by forward all calls I put sip:xxxxxxxxxx@ip.address.com:5080 (they only except traffic on 5080 but you can leave that off if their excepting on 5060). Sorry was in a rush if you need I can try to clarify it a little later.

Thank you again, On doing a search of the word Makecall I pulled up 6 finds. All 6 finds mentioned the word Makecall none actually were on the subject of implantation of Makecall. As to the wiki page you linked to,I dont see Documentation on implementation merely a list of available or will be available features.

Any help anyone can provide on the Makecall CSTA. Thank you,

Snom ONE, Thank you for your insight, that would definitely be useful. But I am trying to integrate the PBX into our CRM. Is there anything else I can try?

Hello, I am trying to initiate a call from my computer. I thought the easiest way would be through HTTP Example: http://my.pbx.com/remote_call.htm?user=123%40my.pbx.com&connect=true&dest=123456789&auth=123%40my.pbx.com%3Asecret Two issues I was having 1) if connect=true is set (which I need) then external calls will not work, my phone rings i pick it up the call starts and dies. I found that from my extension to another internal extension there is no issue. 2) my screen comes up with my extension calling myself. Does anyone know what I could do about issue 1 that is the more important one, and is there a work around for number 2? OR do I need to move to CSTA to use Makecall? I dont know the first thing about CSTA and would not know where to start for this simple project.

Hi again, I made the change and took a back up of the original XML file. IT WORKED!!! I am still concerned about my other providers because they worked fine till now. So the only question I have left can this be done on the trunk level and not on the Global level. Thanks again!!!!!

Thank you so much, I will look into it. Any way to see what they are set to now so I can revert back if necessary? And is the only option a global change any way to do it on the trunk level.

Just for the record, I got it to work based on your incite. Thank you again!

Hello there, I am having trouble with media on one of providers for incoming DID's. The provider engineers believe they pin pointed the issue to this line in the handshake. I did find it in my logs as well - a=rtcp-xr:rcvr-rtt=all voip-metrics anyone have any insight on what this is and how I can change it......

The same would apply to a SIP Registration? and if yes I guess the registration server can be put into the "Explicitly list addresses for inbound traffic:" because where else can it come from. Correct?

I am using Medium Security, when you say no proxy are you referring to "Proxy Address:" Also there are thee types of trunks: Sip Proxy, SIP Gateway and SIP Registration. Do all three have the risk? And not always are we given a proxy.... But most of the trunks are set to out bound only and almost all inbound ones have an IP's in the "Explicitly list addresses for inbound traffic:" Which type of trunk should I check and anything more specific to check for. THANKS AGAIN!!!

It appears from the daily report log that my PBX is being hacked, now I dont know what more I can do. The fire wall on the internet connection is blocking all IPS from China, South Korea etc.. I have everything under Access Blocked except for a few IP ranges. The reason I am saying hacked is is the below call records say these two calls were made from an "account/Extension" called asterisk I do not have any such account on my system. Where could they of created this account how do I block this from happening in the future - and how do I find it to delete it. Time Dir From To Remote Local Duration 5:25 I asterisk 011442070661000 asterisk 02:31 5:31 I asterisk 011442073479999 asterisk 00:27 Any help greatly appreciated!

Thank you for your help, I think I am going to give up for now. I never would of imagined something like this would be so complicated.