Bill H

I am only receiving a Treeview of the Extensions. Can I get the original "Blockview" back? Bill H

It looks like you will need to call Audiocodes on this one. Bill H

If you have a Static Public IP Address (WAN) at the MP114 location, put it in the NAT IP Address on the Quick Setup Page. This will identify where the MP114 (and your extensions) are located. Bill H

You said: "ip phone configired with domain name of pbx as sip proxy and LAN ip address of MP114 as outbound proxy." Enable the SAS feature in the MP 114 and give it an IP Address. The phone should only register to the MP114 on port 5080 as the Registar and you need to create a Proxy Name for the gateway. Once you make these changes try it and let me know if it still does not work. There may be several things you need to do. This is not an easy gateway to setup for the first few times. Bill H

I looked at your traces and some of the characters were like ÿ¨kH— < < h«Þ ’³ E( ( y @õ±À¨ÇÀ¨f Pñâà³Þ Púã n ´ÿ¨kH?Ã Ç Ç h«Þ ’³ E( ¹ z @õÀ¨ÇÀ¨f Pðâ‘=¯ï ÃPýŽPZ From your traces it looks like there was maybe just one call made. Anyway, what exactly is not working? Registration, connections audio??? Bill H

I tried the same thing with the same results. No mater what time I set it for, the cellphone always rings. Maybe we are both doing it wrong???? I asked for the format to use when entering the time of day and was told to use the same method as Day/Night Service. Bill H

If PBXNSIP has only 192.168.1.x as an IP Address, how would he connecting to it remotely? I think your problem with voice is that someone or both parties are not getting the correct IP Address to respond to. With as much bad rap that STUN recieves I still use it and it works OK under certain conditions. Here is what I would look at: Set up Logging in PBXNSIP to see all the SIP Messages. Make a few test calls. Look in the SDP (Session Description Protcol) area of the message (its long and at the end) and see if there are any Local IP Addresses like 192.168.x.x. There should not be any. INVITE sip:2227878@Proxy.ac SIP/2.0 Via: SIP/2.0/UDP 172.215.xxx.xxx:5060;branch=z9hG4bKac710131078 From: "5022" <sip:5022@Proxy.ac>;tag=342231859 To: <sip:2227878@Proxy.ac> Call-ID: 515987274-5062-2@71.190.187.23 CSeq: 10 INVITE Contact: <sip:5022@71.190.xxx.xx:5062> <<< This should be a Public IP Address Max-Forwards: 69 Supported: replaces, path, timer User-Agent: Grandstream HT-502 V1.1B 1.0.0.86 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 433 v=0 <<< SDP Starts here o=5022 8002 8000 IN IP4 xxx.xxx.xxx.xx <<< This should be a Public IP Address s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx <<< This should be a Public IP Address t=0 0 m=audio 5050 RTP/AVP 0 8 4 18 2 97 103 102 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G I use STUN. It solves the NAT problem for me. I don't know if Snom has a STUN feature or not. Use a STUN Server like stun.xten.com or stun.softjoys.com Also, different routers can create VOIP problems. Turn off SPI (Stateful Packet Inspection) Some routers (Netgear) have a hightend NAT control. Set it to Standard or Simple. (something like that) Give your Snom phone a Private Static IP Address to use with the router. Put that IP Address in the DMZ for full exposure. Someone wise (Kevin Moroz at PBXNSIP) once told me that all the answers are in the SIP Traces (Messages). ....... He was right. Bill H

I have never seen anything in PBXNSIP that will Email a conversation that is saved in the Recordings Folder. The only connection with Recordings and Extension Redirect Settings is that they are on the same page. I don't believe that they are interactive. There may have been a plan at some time in the past to do this though. An External Program could look at new recordings as they are created in the Recordings Folder and then Email them to the correct extension user. Bill H

Just a few pointers here. When you use Start and Stop Codes the call is saved in YOUR Mailbox and you are the only person who can listen to it. When you use the "Record incoming calls from Hunt Group etc. the call is being saved in the Recordings Folder. Its location is determined by System Settings Record Location:. Since the recordings are in an easily accessable folder, anyone with access to the computer running PBXNSIP can listen to, copy, save or forward your conversations. Also, recordings start to use a lot of disk space as time goes by. By using the Start and Stop Codes you can control what is and is not recorded. So, use the above information to determin which method is best for your system. Hope this helps. Bill H

Look at the "Block Outgoing Caller-ID" option on the extension settings. Make sure it is set to "No". If it set to "Yes" and your trunk does not support it, the call will fail. ---------------------------------------- Next option: Re-program the remote extension the same extension number as a working phone in your office. This way it will register with working extension assignments. If it doesn't work then your problem is not in PBXNSIP. Then start looking at the settings in the remote Snom. Hope this helps. Bill H

Thank you for the helpful information. How about the ports in the above question?

I am trying to access the database with SOAP. I looked at the PHP example (even though I don't know PHP) and it looks like there are two different Ports used. I see 8080 and 80. I tried both with no success. Does anyone know what port to use to send SOAP to the PBXNSIP database and which port do I listen on? Thnaks, Bill H

This is what I would do. Create a new Extension in System 1 called 555 (as an example) Create a new SIP Registration Trunk in System 2 Create a new Auto Attendant in System 2 called 888 (as an example) In System 2: Register this new SIP Registration Trunk using the System 1 LAN IP Address as the Domain and Outbound Proxy. Use 555 (as an example) as the account. Set Send Call to: 888 Now from System 1 you can dial 555 and be connected to the new SIP Registration Trunk which automatically connects you to the new Auto Attendant 888 in System 2. From there you can call any extension or make an outgoing call on any CO Lines in System 2. To get the same ability from System 2 just replicate this plan in reverse. You can substitue different numbers (555 888) to suite your numbering plan. This is a simple plan. It could be made to be more elaborate with dial plans and managed extension numbers in each system I have not personally tried this, but in theory it should work. If not, let me know and I can dig a little deeper. Bill H

What you are calling "Message 2" isn't a Message, it is a Prompt. You need to re-record the the Prompt. Look in Audio_en folder in PBXNSIP (for English) There is a prompt called aa_enter_first_name.wav Double click on it and listen to it. It should be what you are looking for. Next. Save it with a new name like aa_enter_first_name_original.wav. This way you always have it for future use. Next. Using Windows Sound Recorder or Goldwave record a new prompt with "please enter the first name of the person you would like to call" Next. Save this new prompt with the name aa_enter_first_name.wav in the Audio_en folder. Note: Be sure to record the new prompt with a Bit Rate of 128K, Audio Sample 16 Bit, Sample Rate of 8K, Mono and PCM Do all that and you will be all set.... Bill H

It sounds like you have 2 questions here, both revolving around the Address Book. Names in the Address Book can be displayed on incomming calls when the callers Caller ID matches a telephone number in the Address Book. So I am going to re-write your question a little differently. Q. How can we get a trunk to display a (company) Name ID. We need to enforce a single company name. A. The Address Book doesn't have a "Company Name" area, it only has a "Name" area. To get a single name try this with 2 different Address Book entries: First name: Netfone Last Name: (Blank) First name: Rafeh Last Name: (Blank) This should give you a Single Name when a call comes in. ------------------------------------------------------------------------------------------------------------------------ Q. We have to have then the ability to explicitely ask the caller to key in the first name of the person. We do not have this ability at present. A. You do have the ability by re-recording the Auto Attendants prompt. Have it say "Please enter the first three digits of the company name or the first three digits of the persons first name" Give the caller the exact options they need to complete the call. I hope this helps you Bill H

"Ready" indicates that the extension is Logged In "Logoff" indicates that the extension is Logged Out If you use ACD Groups, it is the extensions ACD status. *64 = Logged In *65 = Logged Out Bill H

Well if PBXNSIP monitors DTMF on the Trunk side where the cell phone came in, a special "Disconnect" sequence like "##**## + PIN" could be entered to disconnect the other side. Then PBXNSIP would return the cell phone to the original greeting "Outbound call press 1, Mailbox press 2, Auto Attendant press 3" It isn't as quick and easy as pressing "##", but it is better then hanging up and calling back in again. Bill H

I can create a 3 Party Conference from my phone (2 outside parties and myself) by using the COnference button. I can not add any more parties to my conference. Pressing the Conference button produces no action. Is it possible to have more people in the conference using the COnference button? Bill H

An additional question: When we call into PBXNSIP from our cellphones the Auto Attendant gives the choices to make a call, go to our mailbox or connect to the Auto Attendant. We press 1 to make an outbound call and it works fine. If we want to make an additional call we have to hang up and repeat the entire procedure from the start. Is there a way to Release the first call and make a Second call easier? Bill H

My settings: Cell phone number: 1xxxyyyzzzz Include the cell phone in calls to extension: Immediately Service Flag: Explicitly specify available times Tuesday: 10:45-10:50 1:00P-5:00P I called my extension before 10:45 today (Tuesday) and the call went to the cell phone. I was not expecting the call to go to the cell phone. Is there another setting to be concerned with or is there a trouble here? Bill H

I looked at http://wiki.pbxnsip.com/index.php/Simple_CDR_Format and I did not see any $String there that would indicate that the call was initially answered by an ACD.

Yes I do realize that it is on a trunk, but the same trunk is also connected to the pbx at the same time. I thought that the Inband DTMF detection: On/Off setting had something to do with this feature. I mean that the PBXNSIP would monitor the call/trunk/extension looking for *77 to initiate a transfer. Since this isn't the case, what then does the Inband DTMF detection: On/Off actually do? Bill H

When a call is transfered from PBXNSIP to a cell phone how can I (if at all) transfer that call back from the cell phone? I tried *77 which is the Transfer Feature Code and that did not work. Is it at all possible? Bill H

From, the extension users web page. What is the format for setting the start times available and not available? Cell phone number: 19992351234 Include the cell phone in calls to extension: Immediately Service Flag: Explicitly specify available times Monday: Tuesday: Wednesday: Thursday: Friday: Saturday: Sunday: Holidays: Bill H

Thank you, That works great. Bill H