On 02 Sep 2024, Gamgee said the following...
bbsing wrote to Gamgee <=-
Gamgee, how did you configure your system to use your modem over
I've been searching for a solution for days.
I've got a basic asterisk freepbx system with two pjsip exention my cisco spa122 ata, but my modems can't complete the handshake. RING then beep but they never connect.
can you send me some knowledge nuggets?
Here is my .ini file, hope it helps and good luck!
<SNIP>
Thank you Gamgee. That is some great information.
Where you using an ATA device to hook up your modem?
Yes, a Grandstream HT-802. I did change a few default settings, that I can't actually even remember now, but it wasn't too hard, and the info was find-able online.
My problem is that the modems don't seem to connect, even they are se to ATS0=1. That handshake doesn't seem to be working. The modem picks up the line, then they screetch a bit but no completion of the handshake.
So I think I'm having some issues with the ATA or Asterisk, maybe bot
The ATA can be quite finicky and sensitive to mis-configuration. I
don't have much knowledge on any of that any more.
Sorry I can't be more help... Hopefully you can figure it out. Seems like it might be a modem initialization string issue.
Good luck!
Thanks Gamgee,
I sent out a message in the echos to anyone I searched who had information. Once I figure out who game the the link to the info I'll grab it.
But here is the info that got things going for me. ------------------------------------------------------------------
URL:
https://gekk.info/articles/ata-config.html#Troubleshooting ===============================================================================
VoIP Setup
Once you have web access to the SPA as above, you can configure the ports. We will use the bogus number 9095551010, but if you want to use another one, just replace it (in yellow) below.
1. Log into the web interface
2. Go to the Voice section, then Line 1
1. Set Make Call Without Reg and Ans Call Without Reg to yes
2. Set User ID to 100
3. Scroll down and find Dialplan, and replace its contents with the following: o (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|<9095551010:101>S0<:@127.0.0.1:5061>|)
4. Under the Audio Configuration section, set everything that says Fax to no
5. Click Submit and wait about two minutes, then click on the Voice tab again if it doesn't redirect
3. Go to Line 2
1. Set Make Call Without Reg and Ans Call Without Reg to yes
2. Set User ID to 101
3. Scroll down and find Dialplan, and replace its contents with the following (it's different, so don't just reuse the first one!):
o (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|<9095551010:100>S0<:@127.0.0.1:5060>|)
4. Under the Audio Configuration section, set everything that says Fax to no
5. Click Submit and wait about two minutes, then click on the Voice tab again if it doesn't redirect
4. Configuration is complete!
===============================================================================
Troubleshooting
This should just work, but here are a couple things you can do if it doesn't: 1. Test basic dialing functionality:
1. Get a plain, basic telephone and plug it into one port
2. Try to dial 9095551010. Regardless of what's plugged into the other port, you should hear ringing. If you hear busy signal or dead air, you missed a config step.
3. If you have a second phone, plug it into the other port. Test dialing 9095551010 from either one; it should ring the other set and you should be able to pick up and talk.
4. If all of the above works then there's nothing wrong with the ATA dialing
2. Apply data optimization settings:
1. The instructions given earlier include the necessary step of disabling fax detection, but if that isn't enough, you can do this too.
2. In Line 1 and Line 2, apply the settings below. They will tell the ATA not to try to "help" and should cause it to just pass through audio unmodified.
1. After applying the settings to Line 1 and hitting Submit, make sure you wait for the page to reload before moving on to Line 2.
3. In the Network Settings section:
o Network Jitter Level: Extremely high
o Jitter Buffer Adjustment: No 4. In the Audio Configuration section:
o Preferred Codec: g711u
o Second and Third Preferred Codec: Unspecified
o G729a Enable: No
o Silence Supp Enable: No
o Echo Canc Enable: No o Everything that says Fax: No
o Modem Line: Yes
3. If you're not getting any dialtone, check that the SPA has an active Ethernet link on the blue port. If it doesn't have a connection and a valid IP, it'll shut off the voice module.
4. You will not get a 56k connection speed no matter what you do - the V.90 and V.92 specifications explicitly state that the modems you have ("analog modems") are only capable of originating a 33.6 connection. You need special ISP equipment to originate a 56k connection.
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... If a pig loses its voice, is it disgruntled?
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