Excuse my newbieness :-(. I want to setup a VOIP system that converts a POTS to VOIP but I dont need or want to use the Internet. I want this all setup to operate over a LAN.
Ideally what I would like is the exsting POTS system to continue to operate the way it does not but converting to IP and back again, and Im sure the only way to achieve this is using a POTS to VOIP adaptor. To that end I purchased a couple of cheap Linksys PAP2T NA PAP2T units, and so far bugger all works on them, presumably becuas ethey need configured to talk to a VOISP which is where I am stuck
Asterisk can mimic an ITSP (I think some small ones actually use it).
It looks like you will need DHCP, on the network, and may need a DNS server. Modern routers tend to provide these services, but they will run on the Linux platform used for Asterisk.
ha, I am having fun. I am used to all types of tech but this is my first delve into PABX. I have asterisk installed and running, and Ive tried everything to get the Linksys units to register but they just wont connec.
I have a special route issung IPs, and have fixed the IP on both the Linksys units, with their gateway set to the linux Asterisk server.
Im just trying now to figure out why they wont register
I understand, Im just not entirely sure what to give.
Regards the asterisk config I have done next to nothing in it. Followed a guide that has me adding 2 users into the users config. Regards the manager one I have made no changes other than enabling the web gui which does not work anyway
Unfortunatley instructions I have found have either been specific to a type of unit, or full of massive assumptions that mean I immediatley need to halt and figure out what the assumption is.
I had hoped for a simple step by step guide, but seems no such guide exists, and the help on Asterisk is a confusing mass of assumption, presumably their for people alrady experienced in PABX systems, which I am not.
I had assume dwhat I wanted to do would be relatively simple, since no outside network was needed, just a LAN, 2 PAP2 boxes, both with an analogue phone connected and a linux machine running Asterisk
That’s not how you would describe configuring Asterisk. I think you are using a GUI, in which case this is the wrong forum. What I would be expecting here is an extract from pjsip.conf, assuming you followed the general advice that new systems should use chan_pjsip, rather than chan_sip.
The information I found on the internet for the PAP2T says that it supports two telephones per box. Why are you only using one? (It also suggested to me that they were overspecified, providing features that would normally be handled by Asterisk in an Asterisk based system. However, I’m not an expert on the ATA market, an it is possible that the primary market is for use for ITSPs, and therefore the devices have some PABX type functions included.
I couldn’t find a description of the full configuration menu structure, so I can’t work out what would be needed for Asterisk.
No gui, not that I can find anyway. The Asterisk is all via terminal windows in Linux.
I think the issue I am struggling with is the many so called helpful step by step guides, that are all for something different or are all again making certain assumptions.
This is for a proof of concept. Yes the PAP2 has 2 handset option but I only purchased these because they were cheap, and take the analogue phone in, which I needed. A client is using a POTS system over Fibre, no internet or external systems, just POTS, and rather than convert everything they have to IP, which would be a very difficult task (I cant say why)I figure converting to IP woudl be easier. Essentially imagine a ton of analogue tech that ends with a pOTS conneciton - then my own special router tech, operating over a secure private network, then back out to a POTS system. So I have POTS → PAP2 → IP ← PAP2 ← POTS is what I am aiming for. They only need a single handset per location on the network, no public access needed.
So I ended up with an install of Asterisk, and these 2 units, both with a single handset connected and neither working
Tonight I am starting from scratch but I cant find a single set of instructions that help in any way at all.
The material I read implies that the PAP2T only provides FXS access (sources ring current, source battery, receives line seizes), but your PSTN access requires FXO access (sinks ring current, sinks battery, and initiates line seizes).
Im not entirely sure, but I decided to start over, wipred, and did a clean install, during which I did notice several things I did not do qst time around, so much so that I now have a miuch better looking system.
I configured the config files with static IP rather than dynamic since thats what I am using, and programmed the PAP2 accordingly, only this time, unlike before I do get some errors on the Asterisk server, which is actually promising
This is the message 193.163.35.9 is the server, 35.7 is the PAP2
user config in asterisk is type=friend, username=craig, secret=1234, host=193.163.35.7, context=tutorial
[Feb 16 21:20:22] NOTICE[43067]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘REGISTER’ from ‘“craig” sip:1000@193.163.35.9’ failed for ‘193.163.35.7:5060’ (callid: 642743c2-d657f222@193.163.35.7) - Failed to authenticate
type=friend is only valid is a sip.conf option, but the log message shows that you are actually listening with PJSIP. Best advice is to disable sip.conf and configure pjsip.conf, which has a very different format.
When you say I am listening, do you mean the server is waiting for a PJSIP or that the remote points are looking for a PJSIP?
If its the serve,r then how do I go about simply switching PJSIP off and leaving SIP as it is?
I mean that chan_pjsip is bound to the port, on the machine running Asterisk, that is the destination of the requests from the Patton device. Asterisk acts as both server and client, often in the same call.
The preferred way of resolving this is to remove sip.conf and/or modify modules.conf to not load chan_sip.so, and port your configuration to pjsip.conf. However, if you don’t mind not being able to get bugs fixed, you can swap sip and pjsip, in the above.
right, ok so I have renamed the sip.conf file so its no longer used. Looked in the PJSIP and modified 2 of the existing demo extensions to match what I want for testing, and antered this new data into both the PAP2 units.
I looked into modules.conf and could only see 2 channels loaded, chan_pjsip.so and chan_bridge_media.so, so it seems no sip is being loaded at all.
Now I get zero errors but still no registrations and still the PAP2 unit saying unable to connect
