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Grandstream HT503 as PSTN Gateway


Digital Pimp
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Hi Guys

 

I'm looking for some advice as to how to get my Grandstream HT503 to work as a PSTN gateway with Snon One, I’ll confess before I start that I’m new to Snom One and still finding my feet but so far much better than 3CX for what I need. I understand from reading other posts that the HT503 is not the best device to be using as a gateway but it’s what I’ve got to hand to tinker with.

 

If any of you have managed to actually get this device to work could you let me know the settings you have used for both the trunk setup and the HT503?

 

So far my tinkering has yielded it picking incoming calls up after two rings but immediately dropping to a dial tone, I haven’t worked out if the call is actually making it to Snom One – settings used below

 

HT503 – 192.168.0.250

Snom One – 192.168.0.249

 

HT503

FXO

SIP Server: 192.168.0.249

User ID: 123

Authentication: 123

Password: Set

SIP Registration: No

 

SnomOne Trunk

Type: SIP Gateway

Account: 123

Domain: 192.168.10.250

Username: 123

Password: Set

Proxy Address: 192.168.0.250

Send call to extension: 70 (tried both to the automated menu and a direct extension)

 

Any advice on other settings to check?

 

Thanks

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Hi Guys

 

I'm looking for some advice as to how to get my Grandstream HT503 to work as a PSTN gateway with Snon One, I’ll confess before I start that I’m new to Snom One and still finding my feet but so far much better than 3CX for what I need. I understand from reading other posts that the HT503 is not the best device to be using as a gateway but it’s what I’ve got to hand to tinker with.

 

If any of you have managed to actually get this device to work could you let me know the settings you have used for both the trunk setup and the HT503?

 

So far my tinkering has yielded it picking incoming calls up after two rings but immediately dropping to a dial tone, I haven’t worked out if the call is actually making it to Snom One – settings used below

 

HT503 – 192.168.10.249

Snom One – 192.168.10.250

 

HT503

FXO

SIP Server: 192.168.10.250

User ID: 123

Authentication: 123

Password: Set

SIP Registration: No

 

SnomOne Trunk

Type: SIP Gateway

Account: 123

Domain: 192.168.10.249

Username: 123

Password: Set

Proxy Address: 192.168.10.249

Send call to extension: 70 (tried both to the automated menu and a direct extension)

 

Any advice on other settings to check?

 

Thanks

 

Can you post a siptrace of the event? Thanks

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Hi Mr X

 

Thanks for the reply, below is the extract from the systems log file of an attempted call in - it's passed through the correct mobile number that I was using to test, just the call doesnt actually go any where. BTW I made a few errors on the IP addresses in my first post, now updated.

 

I'm presuming the 404 messages below are playing a part in it failing to work.

 

 

[5] 2012/07/16 20:06:42:

 

SIP Rx udp:192.168.0.250:5062:

 

 

INVITE sip:2000@192.168.0.249:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport

Route: <sip:192.168.0.249:5060;lr>

From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997

To: <sip:2000@192.168.0.249:5060>

Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA

CSeq: 10 INVITE

Contact: <sip:123@192.168.0.250:5062>

Max-Forwards: 70

User-Agent: Grandstream HT-503 V1.1B 1.0.6.13 chip V2.2

Supported: replaces, path, timer, eventlist

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE

Content-Type: application/sdp

Accept: application/sdp, application/dtmf-relay

Content-Length: 388

 

v=0

o=123 8002 8000 IN IP4 192.168.0.250

s=SIP Call

c=IN IP4 192.168.0.250

t=0 0

m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100

a=sendrecv

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:8 PCMA/8000

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 G726-32/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=20

a=rtpmap:102 G729E/8000

a=rtpmap:100 AAL2-G726-16/8000

 

 

[5] 2012/07/16 20:06:42:

 

SIP Tx udp:192.168.0.250:5062:

 

 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport=5062

From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997

To: <sip:2000@192.168.0.249:5060>;tag=f791a28ac7

Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA

CSeq: 10 INVITE

Content-Length: 0

 

 

[5] 2012/07/16 20:06:42:

 

SIP Tx udp:192.168.0.250:5062:

 

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport=5062

From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997

To: <sip:2000@192.168.0.249:5060>;tag=f791a28ac7

Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA

CSeq: 10 INVITE

Contact: <sip:2000@192.168.0.249:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1075 Delta Aurigids

Content-Length: 0

 

 

[5] 2012/07/16 20:06:43:

 

SIP Tr udp:192.168.0.250:5062:

 

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport=5062

From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997

To: <sip:2000@192.168.0.249:5060>;tag=f791a28ac7

Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA

CSeq: 10 INVITE

Contact: <sip:2000@192.168.0.249:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1075 Delta Aurigids

Content-Length: 0

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