I am new to Asterisk. I found a PC running Debian 3.2.54-2 x86_64 acting as t PBE through Asterisk software. I am compiling a list with IP address of each device and its related internal phone number.
I have a full list of IP addresses of all phones but I miss the corresponding phone number.
If I connect via Web to each phone, it is impossible for me to retrieve the associated phone number.
I know that I can use the file “/etc/asterisk/sip.conf” to create a new user but I do not know how to obtain such info.
I use Gigaset DE IP PRO and Cisco SPA IP phones.
Is there anyone who can help me? Perhaps, I have to just run a Linux command line or consult a specific Asterisk file?
Moreover, how to find out current version of Asterisk being installed? This to understand if the software is outdated?
I can’t find any evidence that 3.2 ever existed, and 3.1 is 14 years beyond end of life, so the version of Asterisk will probably be so old that nobody remembers it. “t PBE” appears to be garbled.
Most people make the directory number be the same same as the name configured in the device, and used as the section name in sip.conf, so it would normally exist in the phone.
Most people use DHCP and host=dynamic, so IP addresses may vary with time and only exist in RAM. Phone MAC address will be more repeatable than IP address, but Asterisk does’t use that directly.
If the section name in sip.conf doesn’t match the directory number, most people would put a caller ID setting in sip.conf. They may also do this in other cases.
If the section name doesn’t match the directory number, there would normally be an explicit extension for the directory number, in extensions.conf, which invokes Dial using sip.conf section name.
However, Asterisk is very flexible and there are many alternative ways in which things could be done. In particular, dialplan can be stored in a database, or in alternative syntaxes.
Also many people use GUIs, which may store the master configuration in their own database format, and either construct configuration file on the fly, or load configuration into RAM using management interfaces.
thank you very much for taking your time in replying to my message!
I will consult both “sip.conf” and “extensions.conf” to see if I can find more info.
Can you please provide me some command lines to run at Linux shell for inquiring the system for Asterisks version and perhaps additional info? I fear that 3.2 should refer to Debian Linux OS, could it be? Unless you already gave me the answer “ver.3.2”.
I just run
uname -a and pasted iit output in this post
Any other configuration file to look for in case I fail to find what I need?
Ver. 18 should be the latest version of Asterisk, correct? What I have running is still coming from Opensource project: is this correct?
One quick question, just to think about it: someone expert, in the past, set up the entire PC with Asterisk.
If by any chance I wish to take the risk to build a brand-new PBX, is it easy to set up the new PC and revert existing settings from current PBX to new one? Is there some documentation I could consult? This with little Asterisk knowledge?
At this point, I only wish to understand how difficult and risky it is to do so.
Last question, is it fine/good idea to virtualize the new PBX on a Hypervisor (VMWare with sufficient hardware) or is it always better to use a standalone PC?
The documentation available online indicates that Debian went from 3.1 to 4.0, with no 3.2, but 3.1 and 4.0 are so old that 3.2 could have been missed. There never has been a version 3 of Asterisk.
It’s the latest long term stable version; there is a version 19, but unless yo want leading edge features you should use 18.
Depends on the complexity of your configuration, and there are UPDATE files in the distribution which describe the major changes. It is likely there will be significant work if you are starting from a very old version. You should be using chan_pjsip, rather than chan_sip, which adds additional changes. I don’t think there is any documentation on such a large jump.
Most people seem to virtualise, typically by using a cloud based host, although my personal view is that one should run real time software on real hardware.
You would have to check sip.conf (it’s probably pre PJSIP days) and see if there’s an IP associated with the endpoint’s AOR. If you don’t see any addresses specified…then they were using DHCP on the phones and there is no IP address.
The important thing would be to look at all the logins for the phones. The IP address isn’t important unless you’re specifying IP’s for every endpoint.
IT’s possible the devices didn’t have “phone numbers”. Unlike the old days where each phone was a PHYSICAL extension; that’s not the case. The phones have identifiers; but whatever number they respond to depends on the dialplan. For example…my phone has 5 different “numbers” assigned to it. What’s important is it’s PJSIP login; that’s how Asterisk finds the endpoints.
I think the old days may be much older than you think. For many PABXes, and for that matter PSTN local exchanges, there was a disconnect between equipment number (asterisk resource name) and directory number (asterisk extension), long before VoIP existed, and in fact probably decades before the start of eternal September.
I seem to recall that the SL/1 addressed lines by cabinet/shelf/card/port.
I suspect that Strowger was the last technology where they tended to be related.
On the other hand, only power users of Asterisk tend to make SIP user names differ from extensions, so it is unusual to find Asterisk VoIP configurations that separate the two.
The one physical PBX I ever messed with was while tearing it out to salvage parts; but extensions were physical extensions off the PTSN prefix. Phones had internal extensions…but you could call (555)555-0001 like “most people were supposed to” and get the receptionist; or you could call (555)555-1234 and get extension 1234.
I for one felt it just made more sense to give my extension more human names; so all my endpoints are names.
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