asterisk_nicht_mehr

What has been your experience with setting up a system with a Cable provider and Cable modems? Are there problems or not?

Let me clarify my understanding of your instructions: For a point to point ISDN Configuration (PBX exchange) I would normally freely associate to an extention (ISDN Point to Point) (2-5 digit City Code) + (6-7 Main trunk number) + (2-3 digit extension). That I grasp. For a Mulitipoint to Point (Multiple Line Exchange) I should set up the Lines with unique ANIs. That does not sound right to me. If I have one BRI that allows only 2 simultaneous calls over the ISDN line, then the outgoing line could be in use by a number of identified numbers. I need to associate the incoming call to the specific extension. It would seem to me that setting up the tel alias is quite enough to allow the call to be directed to the proper extension. The point I think you were making was about another point than I was trying to clarify. I wanted to determine if it is normal to set up 2 Trunks for every BRI on an ISDN line, or is the system smart enough to use the Line B without being programmed. My question about falling over was framed to clarifiy if one ISDN is busy, will the second caller calling over the ISDN line get a busy if no 2nd Trunk is listed. I was having symptoms of such problems that led me to install a second trunk.

Are you trying to set up a Message before the Phone is answered which is Global? In Germany they call this an "Ansage vor dem Melden" oder "Text vor dem Melden". If the incoming call is directed to an IVR Announcement, it might be possible to set up a Hunt Group behind the announcement. During the time that the caller is listening to the announcement or Text plus music a Hunt Group could start ringing the calls in what ever sequence you decide upon. If you would have to decide who is in the type of hunt group (ie. Secretaries, Sales People, Service). This is an alternative front end to an initial IVR that would allow a person to greet the caller without it having to go immediately to a electronic attendant. If no one in a hunt group is able to pickup, one would need to arrange a redirection to another announcement or extension or mailbox.

I am wanting to understand how a Point to Mulitpoint ISDN configuration should be set up differently from a Point to Point ISDN configuration. I want to avoid mismatching the incoming calls to specific devices when it is a Point to Multi-Point ISDN phone line. It seems to me that a Point to Point Configuration: where the provider gives a main number and an additional range of numbers which may be addressed (usually sequential) is fairly straight forward. Enter the Trunk--->assign the extention within the trunk and one is more or less finishted because the telalias can identifiy the inbound based on both assignments. With a Basis rate configuration that is Point to Multipoint the provider usually gives a list of up to 20 numbers, which most of the time are not sequential, and usually are not logically related. Because they come in over the common ISDN trunk and the numbers are assigned by the provider I have had some small problems with the setup. By chance I found that one must set a trunk for as many ISDN Channels as one has. Is that correct? 1 BRI = 2 SiP Gateway Trunks, so that with 2 BRIs one would require 4 SIP Gateway Trunks. The telalias is then used to provide number assignment (MSN) to the extentions. Is that correct? Otherwise, what is the best method for assigning these numbers when one has several aliases that must be linked to a limited number of extensions (physical telephones)? It is my understanding that with a Point to Point Trunk setup one may arrange for Fall Over to the next Trunk. IS that also possible with a Point to Multi-Point Trunk setup?

What is the cause of a caller to the pbx not receiving a tone on their side that the phone on the pbx side is ringing? Inbound callers are experiencing a long silence until the mailbox picks up. Which files need to be installed and in which file ordeners to correct this problem?

I would like to know who has a successful VOIP supplier in Germany and a link if you have it. I am trying to work out my problems but if that is unsuccessful I will need alternatives.

Before the upgrade to version 3 I was able to receive calls to my DID telephone numbers which are Internet based numbers and not ISDN numbers. Since the upgrade I am not getting those calls and I have entered them as both aliases or with a trunk ANI. I need to coordinate the DID to the extention. The ISP has given 2 3 numbers for internet calling of which 2 are inbound-outbound and one is only outbound. Do I need to set up separate trunks to route the calls inbound or are the aliases enough to do the job. As it is I am not getting the inbound calls on my business line. Ugh.

I am planning to use an ISP for flatrate calling within my home country and for discounted calling to "landed" phones in neighboring countries. This will be the default carrier. I want to redirect calls to the PSTN (ISDN) trunk only for emergencies and for savings on Mobile phone charges. I have set up the redirection to a Call by Call provider over the PSTN trunk for the Mobile calls and that is working very nicely. I simply put in the exact exchange number such as 017* and the replacement value plus the code and it works wonderfully (01017017*). There is a problem though. For calls in my hometown I would like to not have to enter the city code all the time. The problem is that City codes vary in length in Germany. Area codes in Germany consist of 2-5 digits. Usually shorter area codes are used for larger cities, and long area codes are used for smaller towns. On the other hand the telephone numbers are assigned inversely long. The larger cities have seven or eight digits, while those small cities may have three to four digits. To call a number within the area of Germany dial the number only and to make a call outside Germany one has to use the respective country code. I am not so experienced that I know which variables do what and would appreciate suggestions.

Hello I am having a lot of problems getting Strato as an ISP for VOIP telephoning to work. [1] 2008/10/18 23:35:32: Last message repeated 4 times [7] 2008/10/18 23:35:32: Set packet length to 20 [6] 2008/10/18 23:35:32: Sending RTP for 3c284de6d6f9-ocqyjai495w8#bf058bb268 to 192.168.1.106:60634 [5] 2008/10/18 23:35:32: Dialplan Standard Dialplan: Match 08213240@192.168.1.2 to <sip:08213240@strato-iphone.de;user=phone> on trunk Strato 3 [7] 2008/10/18 23:35:32: Set packet length to 20 [7] 2008/10/18 23:35:32: Call c3db4152@pbx#31941: Clear last INVITE [5] 2008/10/18 23:35:32: INVITE Response: Terminate c3db4152@pbx [7] 2008/10/18 23:35:32: Other Ports: 1 [7] 2008/10/18 23:35:32: Call Port: 3c284de6d6f9-ocqyjai495w8#bf058bb268 The Registration works but the handshake falls off. Because of the consolidation of the market here, it is just about impossible to get a tech on the ISP provider side who can understand this issue: Hotlines do not cut it. Everything is set up according to the pbxnsip standard. I am beginning to think that the ISP Strato is having problems with the packet length. I also opened the Ports from the pbxnsip server I have. The carrier has a different set of values for some of their required port openings, but they also sell another Modem that acts as a Pbx (AVM Fritzbox) as standard for hooking up to their service. I am at a standstill over how to troubleshoot this. Does anyone have suggestions? Deutschesfach

That probably means it is not understanding the password or login info. what do you think?

Did you confirm in the documentation that that is how one is supposed to use the Global?

Hello, I keep receiving the code 0x193 on my phone. None of my outgoing calls go thru when I get this code. What does it mean?

Hello, I am having a problem getting several extentions to go through my NETGEAR DG834GBv4 Router. All ports outgoing are open: Outbound Services # Enable Service Name Action LAN Users WAN Servers Log Default Yes Any ALLOW always Any Any Never Inbound Services # Enable Service Name Action LAN Server IP address WAN Users Log 1 sip ALLOW always 192.168.1.2 Any Always I had problems with my Snom370 earlier, but the majority of problems is with 3 Siemens IP470 models. When I try to dial out I receive an message of Voip code 0x193 which is not in their documentation anywhere. Dialing within the network works just fine. The problem is outgoing. It has worked and then stopped for a couple of days runing. I upgraded to V.3 and that did not solve the problem either. I am pretty sure it is not pbxnsip but the firewall. How can I go about locating the source of my problems? Thanks

Excuse me we use no Prefix when dialing out.

Hello, I need to be able to listen to the mailboxes of serveral accounts on my system. We dialout a prefix, and aliases are set up for each user. When I try to call the other person's extention to listen to messages it just routes the call and I do not get to the mailbox. I am obiously not clear on how to log in to the otherperson's account. Can someone help me? D Fach

Hello, Does anyone know where I can look for information that will allow me to authenticate on a Snom370 or similar. I had a number of problems with my Router and the Snom and because I reset it to Factory Default, I am having to reprogram it. Not so difficult, but I have just gotten the unit. Is the authentication required of the telephone with the Extention Registration in pbxnsip? Thanks for your help.

http://wiki.pbxnsip.com/index.php/Prepare_...r_Plug_and_Play I found a page that explains the process of Binding the MACs and extentions.

I read that before and did as instructed.

The only line it is keeping alive is the line to my snom 370. The Siemens are not registering and the PBX is not yet capable of referring phone calls to the other extentions or mail boxes.

Version is 2.1.12.2489 (Win32), Under Extensions Registrations; Field: Bind to MAC Address Is there another way?

Has anyone had the experience that if a *was entered and later removed, that the system will still think it is there and connect incoming calls only to that extention: even after removal. I am having general problems sending the calls to the endpoints where they belong.

Hello, I have been rummaging around and have not found a step by step guide to adding extentions. I am having problems with MAC and ip addresses not registering. Has anyone had a simular experience?

I too am having problems with these IP-Phones. I have been trying to assign a MAC or SIP to register the extentions and the system accepts the input but no connection takes place. The router gives the MAC and IP address but the system just accepts without completing. What do you make of this?

I want to know if anyone has experience with the TELES appliance and the software is a cut down or regular version of the software. I want to make an immediate decision.