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8 posts

Wannabe Geek


# 128608 16-Aug-2013 14:16
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Hello

I have asterisk 1.6.2.11 and trying to set it up to work with WXC and VFX.

Is this possible? I know WXC don't support asterisk. I have everything working with callcentric and my ip phones but I cannot make calls in or out with WXC/VFX?

I can register to WXC/VFX with asterisk but I get no further. Before I spend anymore times on this is it even possible?

Thank you


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3594 posts

Uber Geek

Trusted
WorldxChange

  # 879541 16-Aug-2013 14:18
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yes it is possible, do you have any traces or details Jeff




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications



8 posts

Wannabe Geek


  # 879569 16-Aug-2013 14:40
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Thank you for the reply I think I may have an authentication error, can you confirm the following is correct for my SIP trunk?

Hostname=pan.wxnz.net
username=99xxxxxx (phone number)
password=xxxxxxxxxxxxxxxxxx
from domain=pan.wxnz.net
from user=99xxxxxx
authUser=xxxxxxxxxxxxxxxxxxx
insecure=port
Outbound proxy=pan.wxnz.net

Everything works just using my cisco phones to connect direct but I would like asterisk to do this then use the ip phones as extensions.

Thank you

 
 
 
 




8 posts

Wannabe Geek


  # 879591 16-Aug-2013 15:01
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Things seem to progress further using a different realm/domain and username which is supplied as MyVFX username


Hostname=pan.wxnz.net
username=99xxxxxx (phone number)
password=xxxxxxxxxxxxxxxxxx
from domain=xport.co.nz
from user=joe.bloggs
authUser=xxxxxxxxxxxxxxxxxxx
insecure=port
Outbound proxy=pan.wxnz.net

comments?

Thank you



8 posts

Wannabe Geek


  # 879600 16-Aug-2013 15:24
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<--- SIP read from UDP:192.168.x.x:5061 --->

INVITE sip:0210221xxxx@192.168.x.yy SIP/2.0

Via: SIP/2.0/UDP 192.168.x.x:5061;branch=z9hG4bK-1955328d

From: "6001" <sip:6001@192.168.x.yy>;tag=6e0f387ebc8d6065o1

To: "Jeff Mob" <sip:021xxxxxx@192.168.x.yy>

Call-ID: 71bd3932-715e3741@192.168.x.xx

CSeq: 101 INVITE

Max-Forwards: 70

Contact: "6001" <sip:6001@192.168.x.xx:5061>

Expires: 240

User-Agent: Cisco/SPA504G-7.5.2

Content-Length: 397

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

Supported: replaces

Content-Type: application/sdp

 

v=0

o=- 1653836 1653836 IN IP4 192.168.x.xx

s=-

c=IN IP4 192.168.x.xx

t=0 0

m=audio 16430 RTP/AVP 18 0 8 2 9 96 97 98 101

a=rtpmap:18 G729a/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:9 G722/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

 

<------------->

--- (14 headers 18 lines) ---

Sending to 192.168.x.xx : 5061 (NAT)

Using INVITE request as basis request -
71bd3932-715e3741@192.168.x.xx

Found peer '6001' for '6001' from 192.168.x.xx:5061

 

<--- Reliably Transmitting (NAT) to 192.168.x.xx:5061
--->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.x.xx:5061;branch=z9hG4bK-1955328d;received=192.168.x.xx

From: "6001" <sip:6001@192.168.x.yy>;tag=6e0f387ebc8d6065o1

To: "Jeff Mob" <sip:0210221xxxx@192.168.x.yy>;tag=as596d4d9c

Call-ID: 71bd3932-715e3741@192.168.x.xx

CSeq: 101 INVITE

Server: Asterisk PBX

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5,
realm="pms-nz.no-ip.biz", nonce="2e957d09"

Content-Length: 0

 

 

<------------>

Scheduling destruction of SIP dialog
'71bd3932-715e3741@192.168.x.xx' in 32000 ms (Method: INVITE)

 

<--- SIP read from UDP:192.168.x.xx:5061 --->

ACK sip:0210221xxxx@192.168.x.yy SIP/2.0

Via: SIP/2.0/UDP 192.168.x.xx:5061;branch=z9hG4bK-1955328d

From: "6001" <sip:6001@192.168.x.yy>;tag=6e0f387ebc8d6065o1

To: "Jeff Mob" <sip:0210221xxxx@192.168.x.yy>;tag=as596d4d9c

Call-ID: 71bd3932-715e3741@192.168.x.xx

CSeq: 101 ACK

Max-Forwards: 70

Contact: "6001" <sip:6001@192.168.x.xx:5061>

User-Agent: Cisco/SPA504G-7.5.2

Content-Length: 0

 

 

<------------->

--- (10 headers 0 lines) ---

 

<--- SIP read from UDP:192.168.x.xx:5061 --->

INVITE sip:0210221xxxx@192.168.x.yy SIP/2.0

Via: SIP/2.0/UDP 192.168.x.xx:5061;branch=z9hG4bK-334ecd6d

From: "6001" <sip:6001@192.168.x.yy>;tag=6e0f387ebc8d6065o1

To: "Jeff Mob" <sip:0210221xxxx@192.168.x.yy>

Call-ID: 71bd3932-715e3741@192.168.x.xx

CSeq: 102 INVITE

Max-Forwards: 70

Authorization: Digest
username="6001",realm="pms-nz.no-ip.biz",nonce="2e957d09",uri="sip:0210221xxxx@192.168.x.yy",algorithm=MD5,response="e58fd7d71d2a96a39962011b91a43cee"

Contact: "6001" <sip:6001@192.168.x.xx:5061>

Expires: 240

User-Agent: Cisco/SPA504G-7.5.2

Content-Length: 397

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

Supported: replaces

Content-Type: application/sdp

 

v=0

o=- 1653836 1653836 IN IP4 192.168.x.xx

s=-

c=IN IP4 192.168.x.xx

t=0 0

m=audio 16430 RTP/AVP 18 0 8 2 9 96 97 98 101

a=rtpmap:18 G729a/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:9 G722/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

 

<------------->

--- (15 headers 18 lines) ---

Sending to 192.168.x.xx : 5061 (NAT)

Using INVITE request as basis request - 71bd3932-715e3741@192.168.x.xx

Found peer '6001' for '6001' from 192.168.x.xx:5061

Found RTP audio format 18

Found RTP audio format 0

Found RTP audio format 8

Found RTP audio format 2

Found RTP audio format 9

Found RTP audio format 96

Found RTP audio format 97

Found RTP audio format 98

Found RTP audio format 101

Found audio description format G729a for ID 18

Found audio description format PCMU for ID 0

Found audio description format PCMA for ID 8

Found audio description format G726-32 for ID 2

Found audio description format G722 for ID 9

Found audio description format G726-40 for ID 96

Found audio description format G726-24 for ID 97

Found audio description format G726-16 for ID 98

Found audio description format telephone-event for ID 101

Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer -
audio=0x101d0c (ulaw|alaw|g726|g729|ilbc|g722|h263p)/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.x.xx:16430

Looking for 0210221xxxx in DLPN_DialPlan2 (domain 192.168.x.yy)

list_route: hop: <sip:6001@192.168.x.xx:5061>

 

<--- Transmitting (NAT) to 192.168.x.xx:5061 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.x.xx:5061;branch=z9hG4bK-334ecd6d;received=192.168.x.xx

From: "6001" <sip:6001@192.168.x.yy>;tag=6e0f387ebc8d6065o1

To: "Jeff Mob" <sip:0210221xxxx@192.168.x.yy>

Call-ID: 71bd3932-715e3741@192.168.x.xx

CSeq: 102 INVITE

Server: Asterisk PBX

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

Supported: replaces, timer

Contact: <sip:0210221xxxx@192.168.x.yy>

Content-Length: 0

 

 

<------------>

Audio is at 118.90.114.xxx port 14262

Adding codec 0x100 (g729) to SDP

Adding codec 0x8 (alaw) to SDP

Adding codec 0x4 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 58.28.20.150:5060:

INVITE sip:0210221xxxx@58.28.20.150 SIP/2.0

Via: SIP/2.0/UDP 118.90.114.xxx:5060;branch=z9hG4bK37daae17;rport

Max-Forwards: 70

From: "asterisk" <sip:joe.bloggs@xport.co.nz>;tag=as283d252e

To: <sip:0210221xxxx@58.28.20.150>

Contact: <sip:joe.bloggs@118.90.114.xxx>

Call-ID: 071a495c2d4c3116576b5f410d228e2c@xport.co.nz

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Fri, 16 Aug 2013 03:10:23 GMT

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

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 310

 

v=0

o=root 981748528 981748528 IN IP4 118.90.114.xxx

s=Asterisk PBX 1.6.2.11

c=IN IP4 118.90.114.xxx

t=0 0

m=audio 14262 RTP/AVP 18 8 0 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

---

 

<--- SIP read from UDP:58.28.20.150:5060 --->

SIP/2.0 100 Trying

From: "asterisk" <sip:joe.bloggs@xport.co.nz>;tag=as283d252e

To: <sip:0210221xxxx@58.28.20.150>

Call-ID: 071a495c2d4c3116576b5f410d228e2c@xport.co.nz

CSeq: 102 INVITE

Via: SIP/2.0/UDP 118.90.114.xxx:5060;branch=z9hG4bK37daae17;rport=1024

Content-Length: 0

 

 

<------------->

--- (7 headers 0 lines) ---

 

<--- SIP read from UDP:58.28.20.150:5060 --->

SIP/2.0 604 Does not exist anywhere

From: "asterisk" <sip:joe.bloggs@xport.co.nz>;tag=as283d252e

To: <sip:0210221xxxx@58.28.20.150>;tag=3201400a-13c4-520d9820-7860173-63b5023c

Call-ID: 071a495c2d4c3116576b5f410d228e2c@xport.co.nz

CSeq: 102 INVITE

Via: SIP/2.0/UDP 118.90.114.xxx:5060;branch=z9hG4bK37daae17;rport=1024

Content-Length: 0

 

 

<------------->

--- (7 headers 0 lines) ---

Transmitting (NAT) to 58.28.20.150:5060:

ACK sip:0210221xxxx@58.28.20.150 SIP/2.0

Via: SIP/2.0/UDP 118.90.114.xxx:5060;branch=z9hG4bK37daae17;rport

Max-Forwards: 70

From: "asterisk" <sip:joe.bloggs@xport.co.nz>;tag=as283d252e

To: <sip:0210221xxxx@58.28.20.150>;tag=3201400a-13c4-520d9820-7860173-63b5023c

Contact: <sip:joe.bloggs@118.90.114.xxx>

Call-ID: 071a495c2d4c3116576b5f410d228e2c@xport.co.nz

CSeq: 102 ACK

User-Agent: Asterisk PBX

Content-Length: 0

 

 

---

Scheduling destruction of SIP dialog
'71bd3932-715e3741@192.168.x.xx' in 32000 ms (Method: INVITE)

 

<--- Reliably Transmitting (NAT) to 192.168.x.xx:5061
--->

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.x.xx:5061;branch=z9hG4bK-334ecd6d;received=192.168.x.xx

From: "6001" <sip:6001@192.168.x.yy>;tag=6e0f387ebc8d6065o1

To: "Jeff Mob" <sip:0210221xxxx@192.168.x.yy>;tag=as1eb2dc9d

Call-ID: 71bd3932-715e3741@192.168.x.xx

CSeq: 102 INVITE

Server: Asterisk PBX

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

Supported: replaces, timer

Content-Length: 0

 

 

<------------>

 

 

 

 

 


3594 posts

Uber Geek

Trusted
WorldxChange

  # 879615 16-Aug-2013 15:43
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are you licensed for g729 codecs ?




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications



8 posts

Wannabe Geek


  # 879628 16-Aug-2013 16:01
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Yes is an asterisk appliance and G729 is working with callcentric.

I tried u-law and a-law no difference.

Thank you

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