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Standard User deleted
(deleted) Wed 10-Jun-15 07:38:41
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Phone settings


[link to this post] 		 
Hi everybody,

I would like to use the hyperoptic phone connection but not the hyperoptic router. What are the settings for their voip connection? Is it a straight forward SIP connection which I could use with an Asterisk system?

Many thanks
Standard User dragon2611
(committed) Thu 11-Jun-15 10:21:44
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Re: Phone settings


[re: deleted] [link to this post] 		 
You will need to get into contact with hyperoptic support as we won't be able to tell you what your credentials are.

They used to give the settings if asked, not sure if they still do.

if you're using * then make sure you secure it properly as otherwise a very nasty bill could be the result.
ISP Representative Hyperoptic_CS
(isp) Fri 12-Jun-15 05:33:22
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Re: Phone settings


[re: deleted] [link to this post] 		 
Good morning Vassilis,

Yes it is possible to use a Hyperoptic number without our router. You will have to setup some settings and your personal Username and Password for your VoIP account.

It would be best either if you can contact us via email at [email protected] or call us on 0333 332 1111 to tell you all this since we will be giving you your login credentials, and that is something that has to be done privately since different people can use that as well.

Kind regards,

Nikola Beslin
Customer Support Agent

Customer Support
www.hyperoptic.com

Prefer to talk to the team? Call Customer Support on 0333 332 1111 or by email to [email protected]
The above post has been made by an ISP REPRESENTATIVE (although not necessarily the ISP being discussed in the post).


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Standard User deleted
(deleted) Fri 14-Aug-15 20:14:52
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Re: Phone settings


[re: Hyperoptic_CS] [link to this post] 		 
Hello Nikola,

I have a Snom 360 which works just fine with other VoIP providers.
I regularly make outgoing calls and receive incoming calls without troubles.

But with hyperoptic I cannot make outgoing calls.
Apparently, the problem relies on possibly some wrong RTP media type (codec) configuration.

Could you please recommend a sequence of codecs I should try on my end?

Below I've pasted the SIP trace taken from my phone.

Thanks

Sent to tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:404 (1370 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.51:2067;branch=z9hG4bK-p1zrwn8ya0fo;rport
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2067;transport=tcp;line=z1wry0hn>;reg-id=1
X-Serialnumber: 0004132394AD
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.7.3.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 534

v=0
o=root 801823659 801823659 IN IP4 154.58.120.99
s=call
c=IN IP4 154.58.120.99
t=0 0
m=audio 63880 RTP/AVP 8 0 18 18 3 99 108 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:1WMtDgai2eh9UKndqpMU/FnewF26WG/b4zqASEAe
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

Received from tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:485 (347 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.51:2067;received=154.58.120.99;branch=z9hG4bK-p1zrwn8ya0fo;rport=2067
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 1 INVITE
Server: Hyperoptic Proxy
Content-Length: 0

Received from tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:617 (509 bytes):

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TCP 192.168.1.51:2067;received=154.58.120.99;branch=z9hG4bK-p1zrwn8ya0fo;rport=2067
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.80c2
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 1 INVITE
Proxy-Authenticate: Digest realm="sip.hyperoptic.com", nonce="Vc4+zFXOPaBz4IAL7eeZ3H3Kg9FZEZCX"
Server: Hyperoptic Proxy
Content-Length: 0

Sent to tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:633 (471 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.51:2067;branch=z9hG4bK-p1zrwn8ya0fo;rport
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.80c2
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2067;transport=tcp;line=z1wry0hn>;reg-id=1
Content-Length: 0

Sent to tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:658 (1599 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.51:2067;branch=z9hG4bK-nc9rgdpeir9i;rport
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2067;transport=tcp;line=z1wry0hn>;reg-id=1
X-Serialnumber: 0004132394AD
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.7.3.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="HY97755",realm="sip.hyperoptic.com",nonce="Vc4+zFXOPaBz4IAL7eeZ3H3Kg9FZEZCX",uri="sip:[email protected];user=phone",response="35621410e803d017cf80c578a0a82073",algorithm=MD5
Content-Type: application/sdp
Content-Length: 534

v=0
o=root 801823659 801823659 IN IP4 154.58.120.99
s=call
c=IN IP4 154.58.120.99
t=0 0
m=audio 63880 RTP/AVP 8 0 18 18 3 99 108 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:1WMtDgai2eh9UKndqpMU/FnewF26WG/b4zqASEAe
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

Received from tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:723 (347 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.51:2067;received=154.58.120.99;branch=z9hG4bK-nc9rgdpeir9i;rport=2067
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 2 INVITE
Server: Hyperoptic Proxy
Content-Length: 0

Received from tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:823 (1089 bytes):

SIP/2.0 415 Temporarily Unavailable-Unsupported Media Type
Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=2thz38hpk2;did=e46.702;mpd=ii;rtpprx=yes;vsf=ZUNBYQUMWE5MWGEUcikOKTxsSEBnAllsf2cJR1RMCAIaFCFZ>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=2thz38hpk2;nat=yes;ngcplb=yes>
Record-Route: <sip:141.0.144.12;transport=tcp;r2=on;lr=on;ftag=2thz38hpk2;nat=yes;ngcplb=yes>
Via: SIP/2.0/TCP 192.168.1.51:2067;received=154.58.120.99;branch=z9hG4bK-nc9rgdpeir9i;rport=2067
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>;tag=4E619562-55CE3DA000078D07-A087C700
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 2 INVITE
Supported: 100rel
P-Asserted-Identity: <sip:[email protected]>
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Accept: application/sdp, application/dtmf-relay, application/isup, application/qsig, application/simple-message-summary, application/xml, audio/telephone-event, message/sipfrag
Content-Length: 0

Sent to tcp:141.0.144.12:5060 at 14/8/2015 20:12:31:839 (468 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.51:2067;branch=z9hG4bK-nc9rgdpeir9i;rport
From: "+44(20)8555-7030" <sip:[email protected]>;tag=2thz38hpk2
To: <sip:[email protected];user=phone>;tag=4E619562-55CE3DA000078D07-A087C700
Call-ID: 55ce3d9a99cc-geqze2yb0rs6
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2067;transport=tcp;line=z1wry0hn>;reg-id=1
Content-Length: 0

Received from tcp:141.0.144.12:5060 at 14/8/2015 20:12:32:673 (536 bytes):

OPTIONS sip:154.58.120.99:2067;transport=tcp SIP/2.0
Max-Forwards: 10
Record-Route: <sip:141.0.144.12;transport=tcp;r2=on;lr=on;ftag=1656f22;ngcplb=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=1656f22;ngcplb=yes>
Via: SIP/2.0/TCP 141.0.144.12;branch=z9hG4bK9787.64341b2abfcc07979c0cb9dcf0ea27a7.0
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
From: sip:[email protected];tag=1656f22
To: sip:154.58.120.99:2067;transport=tcp
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0

Sent to tcp:141.0.144.12:5060 at 14/8/2015 20:12:32:691 (561 bytes):

SIP/2.0 404 Not found
Via: SIP/2.0/TCP 141.0.144.12;branch=z9hG4bK9787.64341b2abfcc07979c0cb9dcf0ea27a7.0
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
From: <sip:[email protected]>;tag=1656f22
To: <sip:154.58.120.99:2067;transport=tcp>
Call-ID: [email protected]
CSeq: 1 OPTIONS
User-Agent: snom360/8.7.3.25
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0
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