Sagacious Himself — brevity in circumlocution: never blague — suffering genius

February 14, 2015

Manually configure Verizon advanced calling 1.0, VoLTE mmm SIP XCAP new switch

Filed under: Uncategorized — Sagacious Himself @ 2:20 am
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Invoke configuration menu (that’s on you). S5 N4 etc. That one Rom “upgrade” is not required.

Happy saint Valentine’s day.

Enable logs
Enable call info logging

In test mode DO NOT touch Manual Configuration switch

Service settings
.. IMS PDN
Check boxes one and four

PDN preferences
Check boxes one and three and four

180movie.com
2600:80d:1::10:161:*:*

If you cannot elucidate reasons why MDN and MIN might need differ reflect on the wisdom of proceeding.

//
Bearer

..MediainfoSettings are the exclusive province of sip veterans or practitioners of certain other dark arts. Don’t you touch them. If you haven’t developed your own craft for sip timers this isn’t for you.

Sip

..If you know why in particular to change the User Agent have at else NO touching.

(more…)

March 18, 2013

anveo feature request magic incantation

Filed under: bookmarkified — Sagacious Himself @ 9:36 am
Tags: , , ,

Genie, Genie,
Grant my wish.
May I have it,
Genie, Please?

http://www.dslreports.com/forum/r25087401-

courtesy Aaron@SipSorcery

to reduce call flow contact tedium I’m hoping to see anveo ldap support extra soonventually!

November 10, 2012

Filed under: Uncategorized — Sagacious Himself @ 7:20 am
Tags: , ,

SIP vs XMPP or SIP and XMPP?

http://p2p-sip.blogspot.com/2009/11/sip-vs-xmpp-or-sip-and-xmpp.html

now fold in

sRTP:OTR,
zRTP:OTR(authenticate identity)

sRTP: media encryption
OTR: conservation encryption (not stanza encryption)
zRTP: setup key exchange

 

since commenting there requires a google account I’ll never be commenting

 

March 17, 2009

goog-411 disingenuous about caller ANI – caller id blocking fruitless

Filed under: CIO,voip — Sagacious Himself @ 6:22 am
Tags: ,

google free 411 disingenuous about caller ANI

using star codes to block caller id will NOT prevent an ANI enabled line from fetching your information.

http://mobile.google.com/support/bin/answer.py?answer=76433&topic=14471

If you choose to connect to a business through the GOOG-411 service, your caller ID information will become visible to that business. We do not share your information with anyone except in the limited circumstances as outlined in our [alleged] privacy policy.

To avoid having any information associated with your phone number in the future, just block your caller ID before you call. With many phone services, you can do this by dialing *67 before the phone number. In most cases, you can also block your number through the menus on your mobile phone. For specific details on how to block your caller ID, contact your service provider. [be sure not to ask them about ‘Automatic Number Identification’]

“we won’t harvest your caller-id using ANI, honest, even if you block it, cuz we can’t see it then, but we’ll be able to pass it onto the next party”

google: powered by hippies with new money

more amusing: goog-411 sucks in comparison to microsoft’s offering: 800-2255-411 (800-CALL-411)

try it: http://gizmocall.com/18002255411 or with one of these SIP URI

sip:18002255411@tf.voipmich.com
sip: 18002255411@carriers.us
sip: 18002255411@sip.tollfreegateway.com
sip: 18002255411@tollfree.sip-happens.com
sip:18002255411@tollfreetollfree.com

other free toll free termincation providers

don’t like gizmo/skype?  FlaPhone.com is a pure flash (to you) SIP client

worship at the google altar? dial away otherwise use the SIP URI for goog to try it without being tracked and catalogged.

February 5, 2009

ViaTalk growth pains me

Filed under: CIO,voip — Sagacious Himself @ 3:48 am
Tags: , ,

painful using new ViaTalk

Oldschool ViaTalk is less vexing to use than VoicePulse behind NAT.  However the ‘new’ (secret) ViaTalk servers are painful to setup behind NAT — some NAT flavors more so than others.  Once their old servers evaporate so will much of their patronage.

Of their almost cutting edge feature set call record (*99) is the most amusing.  “Your call may be recorded for quality control” is delightful to rave back at the automated attended that puts you on notice whenever you call into most customer server queues.. so that the people you’re calling will similarly be unable to revoke consent.  Unfortunately it’s not very useful to have entire conversations archived on ViaTalk’s voicemail switch as it cannot be downloaded… easily.  As most mail providers will choke on attachments greater than 20 megs.. as they should because email != FTP.. email is obviously not an option.  There’s no sane reason not to offer WAV or FLAC downloads of voicemail, especially call recorded conversations, from the ViaTalk web control panel.

Of the dozen VSPs I have toyed with ViaTalk has the most appealing feature set and seemingly responsive support people.  They, however, are oft on par with godaddy: barely read the message before firing back an irrelevant reply.

ViaTalk does have a Bring Your Own Device (BYOD) plan which is great for those who like to tinker — which will be ESSENTIAL to create a reliable service.  Future Nine supports the BYOD model but VoicePulse does not.  Be sure to bring a device that allows user set dial plans of you’re in for a world of even more hurt. [The Obi110 is an excellent BYOD, but googlevoice is still evil]

Viatalk though is merely SIP and *NOT* IAX(2).

Both VoicePulse and ViaTalk have implementations of call filtering (never let that annoying person ring your line again) but ViaTalk makes redirecting calls to other numbers much easier.  VoicePulse only provides one number to forward to, which ViaTalk lets each rule forward to a number.

grab a free DDI (free DID) and make use of the second port

_

update: verizon’s walled garden gateway seems to actively sabotage some voip services with occasionally-disablable SIP ALG.  conf file mods not sufficient.. poking about in telnets [sic] might prove fruitful.

@ Verizon  It seems you enjoy provoking class action lawsuits.. instead why not simply cease abusing customers :gasp: and lower prices.  It’ll be far less traumatic and definitely well received by customers.

edit: overcome westwell sip alg for voip by connecting to the voip server on any port OTHER THAN 5060 (like 5062 or 5080)

[ Himself.wordpress.com ]

getting to like the new Future Nine

Filed under: CIO,DARPA,overview,voip — Sagacious Himself @ 3:35 am
Tags: , ,

like the new Future Nine

as pay as you go goes Future Nine is the first VSP to focus on satisfying the customer AND integrating customer feedback into the service offering

Sure their website looks like a highschool project from 1993 but their lead developer is very active in the forums which matter and is quick to respond to tickets

They use EVIL google checkout, and ethically impaired Paypal.  Also the minimum account funding is TEN dollars!! That’s a bit much for those who wish to purchase a DDI for $4

On the upside one can forward other (free) DDI to the issued SIP URI, and have dialtone upon instant activation to place toll free calls, and comes with voicemail

grab a free DDI (free DID) and receive incoming calls to help a developer community 😉

 

_

not a fan of VoicePulse… except SIP PBX

Filed under: CIO — Sagacious Himself @ 3:20 am
Tags: , , , ,

not a fan of VoicePulse

the only useful service offered by VoicePulse is SIP PBX

VoicePulse connect is evolving… voicepulse VSP is lingering in feature death despire a recent site polishing.  I do not believe they have asterisk engineers in their employ, but instead contract them as needed.

_

January 18, 2009

flaming history: asterisk DNS SRV records

Filed under: DARPA — Sagacious Himself @ 11:48 pm
Tags: , ,

.

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg38322.html

and then there was enum

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