SPA-3102 with a GSM adapter

Asterisk Asterisk Support
1

Hi,

I’m trying to allow my asterisk box to make cell phone calls via our external Nokia GSM adapter. I’m using an SPA-3102 to connect the adapter’s analog line to my asterisk.

Problem is, i can’t make outbound calls. If i plug an analog phone in the phone plug on the SPA i can make the call, so the SPA is comunication perfectly with the GSM adapter.

From asterisk, i receive the following in the console after dialing the numbe,r and waiting a 30 seconds.

-- Executing [0630xxxxxxx@unrestricted-phones:1] Dial("SIP/zzzzzzzz-0000000a", "SIP/gsm-adapter/0630xxxxxxx,Tt") in new stack
-- Called gsm-adapter/0630xxxxxxx
-- SIP/gsm-adapter-0000000b is ringing
-- SIP/gsm-adapter-0000000b answered SIP/zzzzzzzz-0000000a
-- Packet2Packet bridging SIP/zzzzzzzz-0000000a and SIP/gsm-adapter-0000000b

== Spawn extension (unrestricted-phones, 06307476553, 1) exited non-zero on ‘SIP/zzzzzzzz-0000000a’

SPA debug puts this in syslog:

06-10-2010 10:48:35 Local7.Debug 10.10.2.18 SIP/2.0 200 OK
To: sip:0630xxxxxxx@10.10.2.18:5061;tag=24920b01cb529de7i1
From: “zzzz zzzz” sip:zzzzzzzz@10.10.2.91;tag=as52c86e3c
Call-ID: 59dd0c216f4b19c80379560a60a1a6ee@10.10.2.91
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.2.91:5060;branch=z9hG4bK18888829
Contact: GSM sip:0630xxxxxxx@127.0.0.1:5061
Server: Linksys/SPA3102-5.1.10(GW)
Remote-Party-ID: GSM sip:gsm-adapter@10.10.2.91;screen=yes;party=called
Content-Length: 247
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 287349 287349 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 16458 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

06-10-2010 10:48:35 Local7.Debug 10.10.2.18 BYE sip:zzzzzzzz@10.10.2.91 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-bd704cf7
From: sip:0630xxxxxxx@10.10.2.18:5061;tag=24920b01cb529de7i1
To: “zzzz zzzz” sip:zzzzzzzz@10.10.2.91;tag=as52c86e3c
Call-ID: 59dd0c216f4b19c80379560a60a1a6ee@10.10.2.91
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0

06-10-2010 10:48:35 Local3.Debug 10.10.2.18 [0]FM Alert Stop RxTx (c=002550b0;a=0)
06-10-2010 10:48:35 Local2.Debug 10.10.2.18 [1:0]AUD Rel Call
06-10-2010 10:48:35 Local2.Debug 10.10.2.18 CC:Failed w/ Answering
06-10-2010 10:48:35 Local2.Debug 10.10.2.18 AUD:Stop PSTN Tone

06-10-2010 10:48:36 Local7.Debug 10.10.2.18 BYE sip:zzzzzzzz@10.10.2.91 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-bd704cf7
From: sip:0630xxxxxxx@10.10.2.18:5061;tag=24920b01cb529de7i1
To: “Zzzz Zzzz” sip:zzzzzzzz@10.10.2.91;tag=as52c86e3c
Call-ID: 59dd0c216f4b19c80379560a60a1a6ee@10.10.2.91
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0

06-10-2010 10:48:36 Local7.Debug 10.10.2.18 SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-bd704cf7;received=10.10.2.18
From: sip:0630xxxxxxx@10.10.2.18:5061;tag=24920b01cb529de7i1
To: “Zzzz Zzzz” sip:zzzzzzzz@10.10.2.91;tag=as52c86e3c
Call-ID: 59dd0c216f4b19c80379560a60a1a6ee@10.10.2.91
CSeq: 101 BYE
Server: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

sip.conf

[zzzzzzzz]
type=friend
secret=****
qualify=yes
nat=no
host=dynamic
context=unrestricted-phones

[gsm-adapter]
port=5061
type=peer
secret=********
qualify=no
host=10.10.2.18
nat=no
authuser=gsm-adapter
dtmfmode=rfc2833
context=gsm-out

extensions.conf:

[unrestricted-phones]
include => gsm-out

[gsm-out]
exten => _0630XXXXXXX,1,Dial(SIP/gsm-adapter/${EXTEN},Tt)

Any ideas?

2

You could try changing the Linksys config. I’d start with upping PSTN Dialing Delay from 1 to 2 (it’s an option in the Admin mode “PSTN” tab).

3

I changed every timeout parameter, still no luck.

Actualy during the 30 sec timeout i can see on the SPA status page the following: VoIP status: answering. So maybe it can’t even connect to the asterisk?

Also, i have (xx.) set to the dialplan. Could this cause the problem? (i don’t intend to receive inbound calls on the gsm adapter, only outbound)