[Spce-user] Intro

Skyler skchopperguy at gmail.com
Wed May 8 19:46:54 EDT 2013 

Wait a second. Hold the phone.

The Subscriber Preferences.pdf you sent shows

a8159814493 at voip1.ics-il.net

Why is there an "a" in your subscriber number?

Delete that man.

Skyler

On 5/8/2013 4:25 PM, Skyler wrote:
> Wow. Alright. Need to see your proxy log then.
>
> Prepare for an inbound test call, just before you 'send' execute:
>
> tail -f /var/log/ngcp/kamailio-proxy.log >
> /tmp/vitelity-inbound-proxy-log.txt
>
> hangup the failed call then CTRL+C the terminal.
>
> Prepare for an inbound test call, just before you 'send' execute:
>
> tail -f /var/log/ngcp/kamailio-lb.log > /tmp/vitelity-inbound-lb-log.txt
>
> hangup the failed call then CTRL+C the terminal.
>
> Send vitelity-inbound-proxy-log.txt and vitelity-inbound-lb-log.txt to
> the list.
>
> Skyler
>
> On 5/8/2013 2:40 PM, Mike Hammett wrote:
>> No dice.
>>
>>
>>
>> -----
>> 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 2:22:24 PM
>> Subject: Re: [Spce-user] Intro
>>
>> ok, so its 404'ing before loading the rewrite.
>>
>> 1. Set 'Inbound Vitelity 3' rewrite rule under (each) Domain Preferences.
>> 2. Unset rewrite rule for the subscriber(s) and peer(s).
>>
>> This should apply the rewrite to all INVITE(s) _prior_ to the start of
>> processing the call. So the '1' will be added and proxy will find the
>> right subscriber.
>>
>> 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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>>>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>>>
>>>>
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>>>> http://lists.sipwise.com/listinfo/spce-user
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>>>> http://lists.sipwise.com/listinfo/spce-user
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>>>> http://lists.sipwise.com/listinfo/spce-user
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>>>>
>>>
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>>>
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>>
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>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>>
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