[Spce-user] Intro

Skyler skchopperguy at gmail.com
Wed May 8 15:01:23 EDT 2013 

scratch that. I forgot that peering rule is not required for inbound. 
Looking again. brb.

On 5/8/2013 11:49 AM, Skyler wrote:
> go into "Viteleity Inbound" peering group. Under 'Peering Rules' you see
> a yellow highlighted error. Click the + to add a blank rule.
>
> Test again.
>
> Skyler
>
> On 5/8/2013 11:37 AM, Mike Hammett wrote:
>> No dice and here's my results.
>>
>>
>>
>> -----
>> Mike Hammett
>> Intelligent Computing Solutions
>> http://www.ics-il.com
>>
>> ----- Original Message -----
>> From: "Skyler" <skchopperguy at gmail.com>
>> To: spce-user at lists.sipwise.com
>> Sent: Wednesday, May 8, 2013 11:45:26 AM
>> Subject: Re: [Spce-user] Intro
>>
>> Mike,
>>
>>    No there isn't. Best thing is to:
>>
>> ngrep-sip b -d any -qt -W byline port 5060 > /tmp/vitelity-problem.txt
>>
>> Then email that to the list.
>>
>> BTW, looking at your congestion.txt and seeing the other suggestions to
>> re-write rule(s). Seems you are trying to strip the leading '1', but
>> from what I see, you need to add a '1'. Like so:
>>
>> ^([2-9][0-9]+)$   1\1   10-digits to e164
>>
>> Put the above into your rewrite rule set, then ensure the set is enabled
>> for the subscriber AND the Vitelity peer(s).
>>
>> Once confirmed, run a test call while running the ngrep above and send
>> to the list.
>>
>> --Skyler
>>
>> On 5/8/2013 9:02 AM, Mike Hammett wrote:
>>> That didn't pan out for me either.
>>>
>>> Is there some sort of debugging that has more verbosity than ngcp-sip?
>>>
>>>
>>>
>>> -----
>>> Mike Hammett
>>> Intelligent Computing Solutions
>>> http://www.ics-il.com
>>>
>>> ----- Original Message -----
>>> From: "Kevin Masse" <kmasse at questblue.com>
>>> To: "Mike Hammett" <sipwise at ics-il.net>, spce-user at lists.sipwise.com
>>> Sent: Tuesday, May 7, 2013 4:31:15 PM
>>> Subject: RE: [Spce-user] Intro
>>>
>>> Mike, try to verify this:
>>>
>>> Do you have a Peering Server for Vitelity
>>> You should have a Peering Server Vitel-Inbound with an IP that you
>>> are instructed to use from Vitelity
>>>
>>> Then a Peering Rule under that Peering Server
>>> In this area just put a 1 in the Callee Prefix nothing in Callee
>>> Pattern or Caller Pattern
>>>
>>> Then in REWRITE RULES
>>>
>>> Set the following in Outbound Rewrite Rules for Callee   (YES I did
>>> say Outbound Rewrite Rules, don’t think of it the way it may appear)
>>>
>>> Match Pattern: ^(1|)([1-9][0-9][0-9]+)$   Replacement Pattern:
>>> \2     Description:  Take off the 1
>>>
>>> Now inbounds calls should arrive and work with either your e164
>>> number assigned to the subscriber /OR/ your alias number assigned to
>>> that subscriber.
>>>
>>>
>>> Vitelity was quite clear with their error message sent to you with
>>> the call fail notification.
>>> That indicates that you are close.
>>>
>>> Now, there is one reason that most will agree with in the forum.
>>> This is offered for free and basically do your best to figure out.
>>>
>>> Sipwise offers paid services and a commercial product which will work
>>> out of the box.  I think if they truly offered all the answers in a
>>> manual it could be more difficult to offer their commercial product /
>>> pro product.
>>>
>>> I am with you and most others, a good detailed screen shot with
>>> examples it really missing.
>>>
>>> Let me know how you make out with the above.  Using vitelity is
>>> really straight forward once you get the above setting working.
>>>
>>> The examples I provided you are with us running it a SBC mode we do
>>> not connect any devices to sipwise and only route traffic to asterisk
>>> boxes in the field for our clients.  No SIP devices register to our
>>> configuration.  We use trusted sources and send traffic to the end
>>> user PBX.
>>>
>>> I hope this helps.
>>>
>>> Kevin
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: spce-user-bounces at lists.sipwise.com
>>> [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Mike Hammett
>>> Sent: Tuesday, May 07, 2013 4:44 PM
>>> To: spce-user at lists.sipwise.com
>>> Subject: Re: [Spce-user] Intro
>>>
>>> I did change it over to IP authentication. The inbound calls hit my
>>> SIPwise box, but it rejects them. I don't have any devices associated
>>> with it, but I do have VM setup. Ideas?
>>>
>>> BTW: Maybe a beginner\quick setup guide and then a more advanced
>>> manual or set of wiki articles for the more advanced topics? The
>>> basics should include what I have to do to connect to origination and
>>> termination peers, provision a client device and have incoming calls
>>> route to that client and their voicemail.
>>>
>>>
>>>
>>> -----
>>> Mike Hammett
>>> Intelligent Computing Solutions
>>> http://www.ics-il.com
>>>
>>> ----- Original Message -----
>>> From: "Mike Hammett" <sipwise at ics-il.net>
>>> To: spce-user at lists.sipwise.com
>>> Sent: Monday, May 6, 2013 7:01:18 AM
>>> Subject: Re: [Spce-user] Intro
>>>
>>> =====
>>> A call to your DID 815981XXXX  from 815739XXXX has failed at 5:48am
>>> on 05/06/2013 MDT. We received 'CONGESTION'  when attempting to route
>>> the call to your server or device. This number is configured to route
>>> to the peer 65.182.XXX.XXX.
>>>
>>> This error usually means your server or device does not recognize the
>>> number being dialed. If using asterisk, make sure you have the
>>> correct inbound context specified on your inbound trunk and that you
>>> have correctly added an inbound route/extension logic for this DID.
>>> =====
>>>
>>> So apparently whatever I did is wrong.
>>>
>>>
>>>
>>> -----
>>> Mike Hammett
>>> Intelligent Computing Solutions
>>> http://www.ics-il.com
>>>
>>> ----- Original Message -----
>>> From: "Mike Hammett" <sipwise at ics-il.net>
>>> To: spce-user at lists.sipwise.com
>>> Sent: Monday, May 6, 2013 6:23:41 AM
>>> Subject: Re: [Spce-user] Intro
>>>
>>> I did see where to change to IP based authentication and did so. Why
>>> is IP authentication preferred over user authentication? Well, other
>>> than it is more complicated to do this with SIPWise.
>>>
>>> I haven't tested it yet, but I made two peer groups, one for inbound
>>> and one for outbound. I didn't create any peering rules for the
>>> inbound group.
>>>
>>>
>>>
>>> -----
>>> Mike Hammett
>>> Intelligent Computing Solutions
>>> http://www.ics-il.com
>>>
>>> ----- Original Message -----
>>> From: "Kevin Masse" <kmasse at questblue.com>
>>> To: "Skyler" <skchopperguy at gmail.com>, spce-user at lists.sipwise.com
>>> Sent: Monday, May 6, 2013 6:08:15 AM
>>> Subject: Re: [Spce-user] Intro
>>>
>>> Good morning, I noticed something that needs to be addressed when
>>> using Vitelity.
>>>
>>>>>> Proxy:        sip28.vitelity.net (66.241.99.27)
>>>>>> Outbound Proxy:        outbound.vitelity.net
>>>
>>> Not all accounts with Vitelity will use the same inbound and outbound
>>> settings.
>>> Vitelity may send you traffic from any of the following
>>> 64.2.142.0/24
>>> 66.241.99.0/24
>>>
>>> If you are using settings provided by another Vitelity user it may
>>> not be the same as yours.  Make sure you know what your provisioned
>>> for with Vitelity and use those settings.
>>>
>>> Additionally it is not suggested to use registration with providers.
>>> If you can avoid it, use IP Authentication.  In the Vitelity portal
>>> go to the support link, then click on sub accounts. You will be able
>>> to setup your IP there for IP Authentication.
>>>
>>> Thanks,
>>> Kevin
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: spce-user-bounces at lists.sipwise.com
>>> [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Skyler
>>> Sent: Monday, May 06, 2013 4:14 AM
>>> To: spce-user at lists.sipwise.com
>>> Subject: Re: [Spce-user] Intro
>>>
>>> Woops, apparently I need sleep. sry.
>>>
>>> On 5/5/2013 9:53 PM, Skyler wrote:
>>>> Jeremie,
>>>>
>>>>     Do you also have user/pass on your vitality peer?
>>>>
>>>>     "I'm not sure how I'm supposed to put this into Sipwise."
>>>>
>>>>
>>>>
>>>> On 5/5/2013 6:10 PM, Jeremie Chism wrote:
>>>>> You enter the information into sip peers. I use vitality as one of my
>>>>> upstreams. I think we have 5 or so now. If you use the accept all
>>>>> rule for the peer it will allow any call to come in. Rewrite rules
>>>>> are used to normalize incoming calls to e164 format. A rewrite rule
>>>>> can be used for many things. Ex. If you want to use *98 for voicemail
>>>>> instead of the default 2000 a rewrite rule will do that. The users
>>>>> manual is pretty detailed for peers. The only thing you may need help
>>>>> for is the rewrite rules.
>>>>>
>>>>> Sent from my iPhone
>>>>>
>>>>> On May 5, 2013, at 7:35 PM, Mike Hammett <sipwise at ics-il.net> wrote:
>>>>>
>>>>>> Little of the peering section makes sense to me. One of my upstreams
>>>>>> has given me the following information:
>>>>>>
>>>>>> Proxy:        sip28.vitelity.net (66.241.99.27)
>>>>>> Outbound Proxy:        outbound.vitelity.net
>>>>>> Login:        XXXXXXXXXX
>>>>>> Password:        XXXXXXXXXX
>>>>>>
>>>>>> I'm not sure how I'm supposed to put this into Sipwise. I want to be
>>>>>> able to receive calls from this provider as well as send them calls.
>>>>>> As I add more providers, I want to be able to receive calls from
>>>>>> them and send my outbound to them.
>>>>>>
>>>>>> I see there was a lot of attention paid to the Rewrite Rule Set
>>>>>> section, of which I'm not sure why you even need that. Just have a
>>>>>> check box on each client (Force outbound caller-ID to DID). Done, or
>>>>>> not.
>>>>>>
>>>>>>
>>>>>>
>>>>>> -----
>>>>>> Mike Hammett
>>>>>> Intelligent Computing Solutions
>>>>>> http://www.ics-il.com
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Jon Bonilla" <jbonilla at sipwise.com>
>>>>>> To: spce-user at lists.sipwise.com
>>>>>> Sent: Thursday, May 2, 2013 5:10:06 AM
>>>>>> Subject: Re: [Spce-user] Intro
>>>>>>
>>>>>> El Sun, 28 Apr 2013 16:38:02 -0500 (CDT) sipwise at ics-il.net
>>>>>> escribió:
>>>>>>
>>>>>>> I'm just setting it up now. I've used Asterisk (pre-1.2 aka flat
>>>>>>> files, no gui help) and a few Asterisk based systems over the
>>>>>>> years, so I at least have the basics of VoIP down. The install was
>>>>>>> much more pleasant than PBX in a Flash. That asks a million
>>>>>>> unnecessary questions and takes forever.
>>>>>>>
>>>>>>> Thus far my commentary on the setup is thorough, but very confusing.
>>>>>>> Very
>>>>>>> little of the terminology used makes sense to me. I am forging
>>>>>>> ahead through the setup, but I'm not sure what I'm doing or how the
>>>>>>> pieces fit together.
>>>>>>> Maybe I'll understand better once it's running, but for now...  I'm
>>>>>>> lost.
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> Well, the installation process is quite unattended as you've seen.
>>>>>> The Handbook
>>>>>> gives you an overview of what you have but also a quick start guide.
>>>>>> Follow it
>>>>>> and try to understand the terms behind it. Don't try to think in
>>>>>> "pbx" terms because the spce is not a pbx or an asterisk system.
>>>>>>
>>>>>> Ask here when you get stucked. And you can always ask for a personal
>>>>>> training/webinar to our sales team. We're organizing trainings in
>>>>>> several countries too.
>>>>>>
>>>>>> cheers,
>>>>>>
>>>>>> Jon
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Spce-user mailing list
>>>>>> Spce-user at lists.sipwise.com
>>>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>>>
>>>>>> _______________________________________________
>>>>>> Spce-user mailing list
>>>>>> Spce-user at lists.sipwise.com
>>>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>>
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>>>
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>>
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