Logiciels Libres Appliqués

Configuration avec un HT-503 (fxo/fxs)

http://aelintra.com/docs/cgi-bin/view/Main/DocChapter253

Notes:

Create a SIP trunk. Use General SIP for the trunk template. Use an ip address of "dynamic". This way, asteris will expect the trunk to register with it.

type=peer
host=dynamic
qualify=3000
canreinvite=no
username=myusername
secret=mysecret
disallow=all
allow=alaw
allow=ulaw

Give it a user-id of something like soundwin1 and give it a password. IN the ATA set the user-id to soundwin1 and set the password. It should now register with SAIL and you can create a route to send numbers to it which you want it to dial.

Pour fonctionner comme un trunk, on choisira dial stage 1  et on forwardera inconditionnellement les appels voip vers l'extension ou groupe de son choix.

Unconditional Call Forward to VoIP -> there you put where your call should be forwarded. Unconditional Call Forward to VoIP: ->called number must be set same as you normaly call from this sip server.

PIN for VoIP calls -> EMPTY

FXO:
Number of Rings 0 (for not waiting with forward, and ringing phone)

Essentially turning the 503 into a VoIP gateway.

On the HT503 "Basic Settings" page, "Unconditional Call Forward to VOIP" field, enter the VoIP account you want to forward to. In my case, that's a VoIP ring group. You can enter yours with format account@provider and their port (5060 by default). In my case that's RingGroup@MyPBX.

In the "FXO" page, "PSTN Ring Thru FXS" option, disable forwarding to FXS by selecting "No". On the field above that, set # of rings to 2. Save, reboot.

Call the PSTN and the call will ring twice (you won't hear it though) and then start ringing on the forwarded VoIP account/ring group. There are other options like having FXS ring for the given number of times and then, if you don't answer there, forward to VoIP.

You want to select at least 2 rings so the 503 has a chance to get the CID information to pass along. I have a call into Grandstream support about this behavior. You essentially lose 2 rings while the HT503 gets the CID. Another issue is that after the 503 has forwarded the call to VoIP, no amount of hanging up the caller phone will get it to stop ringing the extensions or account. I believe the PSTN side of it does hang up correctly but the VoIP forwarding does not. In my case, the ring group continues to ring until Asterisk voicemail picks up and senses a steady dial-tone that is being sent by the 503. At that point it forces the call to terminate but you always end up with a voicemail consisting of one or two seconds of steady dial-tone. Grandstream has been alerted of this issue as well and I'll post here their reply on both instances. -- JRoque 

Exemple de fichier de configuration: http://www.creasol.it/telefonia/gsconf/gsconf_503.txt

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SIP Trunk Settings 
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Outgoing Settings 

Incoming Settings 

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HT503 Settings 
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Status (of what your device should say after completion of this setup) 

Basic Settings 

FXO Port 

Caller ID Minimum RX Level (dB) may need some tweaking based on your Landline provider.