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Audiocodes Media Pack 20x ATA's


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Hello,

 

I was wondering if anyone has any experience with using the audio codes media pack line of ATA/routers. We have used them with previous systems and also with on site Faxback ATA CPE's. What I'm having trouble with now though, is that we'd like to use them with the PBX but after I reg the device to the pbx I can't make outbound calls (I receive a busy). I know this is a dial plan issue especially since I can call in just fine. Anyone out there use or looking to use these devices?

 

On another note, I've been working with the Linksys Pap2t, SPA2102, SPA8000 as ATA's and had success with all. The SPA2102 and SPA8000 support T.38 but I am not 100% on it's full operation. The Audiocodes MP202, MP202B, MP204 all support T.38 and actually have some unique settings compared to the Linksys products. Also the Audiocodes provide traffic shaping, better firewall, and on some newer series even wireless. (I'm operating as an ITSP)

 

Anything would be helpful, even if you're in the same boat as me, let's get these ATA's under wraps.

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I tried using the Audio Codes

here is what happened, when I registered, it was acting like a trunk and when I dialed an extension number, it would actually dial extensions from a different domain

I also had a problem with letting the PBX set the outbound caller id, instead every time I made a call the CID was 0000000001

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I've made progress on this. I'll detail how I registered the audio codes to the pbx (I'm in a hosted multi tenant environment) and how I was able to send calls out.

 

Go to Voice Over IP\Signaling Protocol\Signaling Protocol

1.All settings default here except "Gateway Name - User Domain". Put in your domain/realm that you are pointing to, example: customerdomain.com

This is huge so that outbound calls have appropriate header info

 

Go to Voice Over IP\Signaling Protocol\SIP Proxy and Registrar

1. Check "Use SIP Proxy"

2. Enter your domain/realm that you are pointing too, example: customerdomain.com

3. Leave proxy port and max # of authentication retries default (unless you want to fiddle)

4. Check "Use SIP Proxy IP and Port for Registration"

5. Leave Register Expires and SIP security default

6. Check "Use SIP Proxy IP"

7. Enter your PBXnSIP's IP, example: 68.142.234.143

8. Leave the port 5060 default

 

Go to Voice Over IP\Line Settings

1. Select "Edit" for either Line Number 1 or Line Number 2

2. For User ID, input your Account Extension that your are registering to, example: Account: 100 (extension) would be the user id

3. Don't block CID unless you want to

4. Display Name could be whatever you want (more for naming purposes) does not effect your header info

5. SIP Proxy:

a. Authentication User Name: the Account that you're registering to, example: 100

b. Authentication Password: The SIP Password you specified when setting up the Account originally on the PBX

6. Leave everything else default

 

 

These steps should get you registered and passing audio to the FXS ports. I don't claim to know this 100%, but this is what I've found so far and that has worked. Oh and I have t.38 signaling working, and that configuration takes place on the Voice Over IP\Voice and Fax section.

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It worked excellent!

Now how do we correct the issue that when call waiting comes in, there is silence for 5 - 7 seconds

 

Where does the silence occur? Are you describing a one way audio issue, a pause in audio after the user has flashed over to the CW caller for both parties, or silence on the active call as the CW caller is calling in?

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Silence on the active call as the CW caller is calling in

If I am on the phone (using the Audio Codes) and I get another call coming in, I won’t hear the person I was talking to for a few seconds (its right after I hear a tone letting me know I have a call waiting, and its until I hear the 2nd tone letting me know I have a call waiting)

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Silence on the active call as the CW caller is calling in

If I am on the phone (using the Audio Codes) and I get another call coming in, I won’t hear the person I was talking to for a few seconds (its right after I hear a tone letting me know I have a call waiting, and its until I hear the 2nd tone letting me know I have a call waiting)

 

I'm wondering if I need to play with the 180 Ring or 182 Queued on this. I'll fiddle with this some today if I can. We don't generally mess with CW since almost all of our FXS users are either integrated lines into analog phone systems or for faxing.

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We would like to use the MP20x for faxing and for using plain old cordless phones with the PBX

we have the MP203B for use with faxing

and the MP202 for the cordless phone, unless if a MP202 can be used with faxing as well

you seem to be a Audio Codes Expert!

 

Expertise based off of immediate and pressured requests from customers :) . Actually though, I'm more familiar with the Linksys SPA ATA's and PAP2's. I recommend giving the SPA8000 and SPA2102 a shot. I know for certain that the SPA2102's handle t.38 very well (except we did encounter a signaling issue with a Sangoma media gateway at one point).

 

Anyhow, what f/w are you running the audio codes MP202 in? I haven't used the MP203 yet.

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Silence on the active call as the CW caller is calling in

If I am on the phone (using the Audio Codes) and I get another call coming in, I won’t hear the person I was talking to for a few seconds (its right after I hear a tone letting me know I have a call waiting, and its until I hear the 2nd tone letting me know I have a call waiting)

 

Looks like I have a new customer that will be using the new Audio Codes MP-204 B 4FXS (running 2.6.4_p5_1_build_9) so I can get some lab bench time in. Now this device is almost identical to other audio codes device except of course the additional FXS ports. I've reg'd two accounts (one was for duplication testing) to the mp204 and hooked a plain $10 analog handset to the "line 1" port. I have the CW setting in the Services section checked as enabled and the Call Waiting SIP Reply as Ringing

 

I initiated a call to my cell from the analog handset, and then during that active call I dialed in from another phone. The analog handset gave me a CW tone (during this tone all audio is interrupted; like normal) but immediately after the tone, both the analog handset and my cell pass audio just fine even with additional tones (not during tone but after). I did the same thing with the "line 2" account. I cool thing to note is that when I hit flash the first call gets MoH.

 

I'm trying the Call Waiting SIP Reply as queued and then will also try changing the signaling to connect media on 180.

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  • 5 years later...
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