GSM gateway setup SIP for scripting?

As previously discussed, I am trying to use OpenVox products to setup a IVR (robo caller) to confirm appointments. I realized I may have been trying to go about it the wrong way and I should be treating the OpenVox differently and stop trying to get the box itself to do the work, and script it elsewhere using something like Elastix or this: github.com/theintencity/rtclite which allows me to communicate with the OpenVox hardware using SIP.

I don't know a lot about this, so I tried buying a Cisco phone to troubleshoot with and I can see it connecting in the logs, but it doesn't let me make an outbound call. This is supposed to work right? Here are my logs:

<--- SIP read from TCP:192.168.1.237:51384 --->
GET / HTTP/1.1
Host: 192.168.1.55:5060
User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:53.0) Gecko/20100101 Firefox/53.0
Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8
Accept-Language: en-US,en;q=0.5
Accept-Encoding: gzip, deflate
Cookie: PHPSESSID=a12hu9absjgg8ef4ob9s6u8bt2
Connection: keep-alive
Upgrade-Insecure-Requests: 1

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:[email protected]:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001794dffcb5>";+u.sip!model.ccm.cisco.com="7"
Content-Length: 0
Expires: 120

<--- SIP read from UDP:192.168.1.207:50859 --->
REGISTER sip:192.168.1.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK7ebb9b7e
From: <sip:[email protected]>;tag=001794dffcb500020540a7be-1680874e
To: <sip:[email protected]>
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
Max-Forwards: 70
CSeq: 101 REGISTER
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:[email protected]:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001794dffcb5>";+u.sip!model.ccm.cisco.com="7"
Content-Length: 0
Expires: 120

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--- (11 headers 0 lines) ---
Sending to 192.168.1.207:50859 (NAT)

<--- Transmitting (NAT) to 192.168.1.207:50859 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK7ebb9b7e;received=192.168.1.207;rport=50859
From: <sip:[email protected]>;tag=001794dffcb500020540a7be-1680874e
To: <sip:[email protected]>;tag=as5a8937a2
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
CSeq: 101 REGISTER
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10a2730a"
Content-Length: 0


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Scheduling destruction of SIP dialog 'This email address is being protected from spambots. You need JavaScript enabled to view it.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.207:50859 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK620e6f83
From: "1234" <sip:[email protected]>;tag=001794dffcb5000319d5ac32-26556e60
To: <sip:[email protected]>
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:[email protected]:5060;user=phone;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 277
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 8612 0 IN IP4 192.168.1.207
s=SIP Call
t=0 0
m=audio 18386 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.207
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
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--- (16 headers 13 lines) ---
Sending to 192.168.1.207:50859 (NAT)
Using INVITE request as basis request - This email address is being protected from spambots. You need JavaScript enabled to view it.
peer=(null).
No matching peer for '1234-192.168.1.207' from '192.168.1.207:50859'
user_and_ip=1234-192.168.1.207, res=-100, sipmethod=5.
peer=1234.
Found peer '1234' for '1234' from 192.168.1.207:50859
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.207:18386
Looking for 18005554027 in sipinbound (domain 192.168.1.55)
list_route: hop: <sip:[email protected]:5060;user=phone;transport=udp>

<--- Transmitting (NAT) to 192.168.1.207:50859 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.207:5060;branch=z9hG4bK620e6f83;received=192.168.1.207;rport=50859
From: "1234" <sip:[email protected]>;tag=001794dffcb5000319d5ac32-26556e60
To: <sip:[email protected]>
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
CSeq: 101 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


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Audio is at 18590
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

Hi,

Can you give me remote access for a check?It is more convenient to find problem by remote.I need more details about that

You can also add my skype for contact.My skype: This email address is being protected from spambots. You need JavaScript enabled to view it.