Re: [SR-Users] Evariste Systems to drop Kamailio market, become Acme Packet VAR

[David J]

Wow! Alex and the kamailio list are at a perplexing loss. We wish you luck
with your future endeavors. We hope that your exclusive Acme contract does
not become some sort April fools joke to get you out of open source.
On Mar 31, 2013 6:33 PM, "Alex Balashov"  wrote:

> For immediate release:
>
> ATLANTA, GA (1 April 2013)--Evariste Systems LLC, an Atlanta-based
> consultancy specialising in Kamailio-based VoIP infrastructure solutions
> for the ITSP and CLEC market, has announced that beginning in the second
> quarter of 2013, it will be abandoning its Kamailio-based technology
> portfolio to focus on its new role as a preferred VAR (Value Added
> Reseller) for Acme Packet (NASDAQ:APKT).
>
> "It is with a heavy heart that we abandon five years of Kamailio-oriented
> work and the Canonical SIP Routing Platform product derived from it,"
> said Alex Balashov, the principal of the company.
>
> "However, the reality is that investment in open-source VoIP technology
> is a dead end.  From a technological point of view, we have lagged very
> badly in meeting the needs of today's sophisticated VoIP market, and it's
> time to cut our losses.  Asterisk, Kamailio, FreeSWITCH--all this stuff
> just hasn't kept up with the pace of evolution of 3GPP, ETSI, and ITU
> standards.  We are tired of saying 'sorry, we don't support IMS or
> H.323' to our resultingly dwindling customer base.  Does anyone
> actually run an all-SIP network?"
>
> Starting in early April, Evariste will begin providing value-added
> consultancy related to the implementation of the Acme Packet Net-Net
> Session Director.  In Balashov's view, "the Net-Net SD is the only
> product capable of meeting the perimeter security, routing and peering
> needs of today's VoIP service delivery environment."
>
> Fred Posner, the director of Team Forrest, a Palner Group integration
> and consultancy operation based in the Jacksonville, Florida area,
> agreed:
>
> "SIP is a tiny piece of the telephony puzzle. The big boys of
> ClueCon [an interoperator revenue-sharing consortium] want DIAMETER-based
> interdomain peering policy control, H.323, MGCP, and IMS.  IMS is pretty
> much how VoIP architecture is done now.  We got out of the Asterisk
> business just in time, right before Mitel swallowed the PBX world.
> I'm glad to see Evariste is finally seeing the light, and I'm sure its
> shareholders are too."
>
> Posner also believes Evariste's lack of support for TDM interfaces
> accounted for dwindling market share.
>
> "Have you seen CSRP?  It's SIP in, SIP out.  Real inter-LATA haulers
> and application service providers use TDM and leave SIP for things
> like voicemail.  I can't plug my DS3s into a SIP proxy, so I just
> don't think there was any real demand for the sort of thing they
> were doing."
>
> Noting Oracle's US$2.5bn acquisition of Acme Packet in early February,
> as well as its more recently announced buyout of Tekelec, a Siris
> Capital Group portfolio company, Balashov remarked: "The obvious
> shift to an Oracle-centric telephony paradigm was a kind of validation,
> if you will, of our decision to unload our dead weight and sign on
> to the revolution in unified communications."
>
> Sean McCord, of CyCORE Systems, an Atlanta-based software consulting
> house and long-time Evariste creditor, agreed that there was a natural
> synergy between Evariste's shift to Acme Packet and Oracle's dominance
> of telephony infrastructure.
>
> "Oracle is a forward-thinking telecom pioneer," McCord said.
> "The telephone is Oracle, and Oracle is the telephone."
>
> Balashov also noted that a tightening regulatory environment and new
> consumer protection rules helped hasten the decision to embrace the
> more professionalised Acme Packet product portfolio.
>
> John Knight, Senior Engineer at Hendersonville, NC-based Ringfree
> Communications, one of Evariste's oldest channel partners, said:
> "As one of Evariste's long-time disties, we were jittery about exposure
> to CALEA and the QA requirements of large call centers.  We tried to
> make do, but at some point we just had to put the relationship on
> stop.  I'm all in favour of open, but there's just no open-source
> software out there that does call recording, and that's the bottom line
> for us.  In the end, we had to restructure some debt just to get
> bondholders to let us source a proprietary solution on tick."
>
> In a thematically related move, Evariste will be dropping its heavy
> use of the open-source PostgreSQL database manager for its rating and
> reporting tools.
>
> "The business case for standardising on Oracle's databases could not be
> clearer.  With Oracle Database 11g's support of warehousing and OLTP,
> the real mystery is why we didn't go there sooner," said Balashov.
>
> Carlos Alvarez, a director at Televolve, a growing Phoenix-area VoIP
> operator, recently spearheaded a move away from Evariste's PostgreSQL-
> based call detail record (CDR) storage solution to one running atop
> Microsoft SQL 

Re: [SR-Users] Shared Call Appearances module

[David J]

This looks really awesome. Thanks for sharing
On Nov 19, 2012 4:50 PM, "Andrew Mortensen"  wrote:

> I've been working on a Shared Call Appearances module for the past several
> months. It implements the Broadsoft SCA feature as laid out in Broadworks
> SIP Access Side Extensions Interface Specifications document. (Another
> implementation of the same feature was added to Freeswitch a few years ago,
> but we didn't want to use Freeswitch.)
>
> We've been testing and improving the module over the last month, and I
> think it's ready to share. The module is available at GitHub here:
>
> 
>
> To date, we've only tested the module with Polycom handsets.
>
> I'm happy to answer questions. I hope this will prove useful for others.
>
> Best,
> andrew
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Re: [SR-Users] Sync nonce between various servers

[David J]

Is the database shared? If so maybe when they authenticate add a secure
token to the header that the second proxy can use for auth?

Just a suggestion not sure if its the answer your looking for or perhaps I
didn't understand the scenario well enough.
On Nov 19, 2012 7:53 AM, "Andreas Granig"  wrote:

> Hi,
>
> There are lots of parameters controlling the creation of nonce values on
> a server, and I'm curious if there is a way to kind of "sync" them
> between servers.
>
> The use case would be to have a UA send for example its registration to
> Proxy1. Proxy1 would challenge it, UA will send the registration again,
> this time with credentials. Proxy1 would look up the user based on
> $au/$ar in the subscriber table, and if it's not found, will look up the
> responsible proxy from another table (with key being $au@$ar), forward
> it to Proxy2, which then would be able authenticate the user.
>
> The reason for this is that the auth credentials are unique across all
> servers and reliably identify a user, whereas for example From could be
> something else (e.g. in case of an IP-PBX sending a CLI in the
> From-userpart).
>
> Challenging the user on the second proxy again would theoretically be
> possible, but if the UA gets a 401 twice (once from Proxy1, once from
> Proxy2), it'll most likely pop up a password form for soft-clients, so I
> want to avoid that.
>
> Any ideas how to accomplish that?
>
> Andreas
>
>
>
>
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>
>
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Re: [SR-Users] Kamailio direct interconnectivity with PRI

[David J]

Well any gateway software that does protocol conversion is going to be more
overhead than pure sip.

Something has to do the protocol conversion.

Typically if you want to avoid using asterisk etc. The you can just buy a
true media gateway like audio codes which does exclusively that. Usually
the hardware solution has better performance
On Oct 11, 2012 9:18 AM, "SamyGo"  wrote:

> Hello Sir,
>
> With due respect, the only answer for your question  I have is that, I
> think that the capacity handling of currently popular media servers is less
> than that of Kamailio.
>
> I agree that all of current media servers are getting better and stronger
> but why should Kamailio get bottlenecked by Media-Servers in front of it.
>
> By the way I completely understand the current option of installing
> Asterisk/FreeSWITCH/Yate and tell them to send calls to Kamailio or
> anything else. Proudly enough I've done such setups happily before. I just
> wanted to reduce a hop between kamailio and the PRIs.
>
> Thanks,
> Sammy
>
>
>
>
> On Thu, Oct 11, 2012 at 6:04 PM, Olle E. Johansson  wrote:
>
>>
>> 11 okt 2012 kl. 13:57 skrev SamyGo :
>>
>> :) "Soon..." But Not Today.
>>
>> Not everyone can afford the Gateways. Thanks for the replies. I was
>> hoping maybe someone else be thinking of freeing the kamailio from
>> Asterisks or Freeswitchs when it comes to interconnecting with PSTN.
>>
>> Why? We have plenty of good media servers out there that can handle those
>> use cases. Few of them handle SIP correctly and include new SIP features
>> like presence in a proper way. Combining them is a good match.
>>
>> :-)
>>
>> /O
>>
>> Cheers
>>
>> Sammy
>>
>>
>> On Oct 11, 2012 4:09 PM, "Neill Wilkinson" <
>> neill.wilkin...@btinternet.com> wrote:
>>
>>> You might consider:
>>>
>>>
>>> http://sangoma.com/products/voip_gateways/netborder_software/netborder_express.html
>>>
>>> Then put Kamailio in front of that... Simple Gateway PRI -> SIP.
>>>
>>> Neill;o)
>>>
>>> Aeonvista Ltd
>>> Opening Up New Ideas
>>>
>>>
>>> On 11 October 2012 11:43, SamyGo  wrote:
>>>
>>>> Hi David,
>>>>
>>>> Thanks for this useful information.  What I don't like is that if I've
>>>> couple of Sangoma cards, each supporting 120 channels each,  is using any
>>>> of this B2BUAs in between the cards and Kamialio and get a limited capacity
>>>> application in front of Kamailio. !
>>>>
>>>> If there is any driver for kamailio where possibly use
>>>> media-proxy/rtpproxy for handling the PRI-Channels media and then
>>>> distribute my calls to media-servers i.e SMES/Asterisk/yate/FS/XYZ
>>>>
>>>> PRIs <===>> Driver+Kamailio <=> Asterisks/FreeSWITCHs
>>>>
>>>> Just want to know if technically any such driver program is doable or
>>>> not !
>>>>
>>>> Thanks,
>>>> Sammy
>>>>
>>>>
>>>> On Thu, Oct 11, 2012 at 3:31 PM, David J  wrote:
>>>>
>>>>> Asterisk yate or free switch.
>>>>>
>>>>> You need something as a gateway between PRI and sip. Kamailio does not
>>>>> handle this conversion
>>>>> On Oct 11, 2012 6:24 AM, "SamyGo"  wrote:
>>>>>
>>>>>>  Hello,
>>>>>>
>>>>>> I've a scenario in which I've to deploy a couple Sangoma PRI cards
>>>>>> with kamailio. What I wish is that I've some drivers for this purpose and
>>>>>> so I don't ned to install FreeSWITCH or Asterisk in between the PRIs and
>>>>>> Kamailio.
>>>>>>
>>>>>> Kindly give any feedback on what are the possibilities and options.
>>>>>>
>>>>>> Thanks,
>>>>>> Sammy
>>>>>>
>>>>>>
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>>>>>>
>>>>>>
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Re: [SR-Users] Kamailio direct interconnectivity with PRI

[David J]

Asterisk yate or free switch.

You need something as a gateway between PRI and sip. Kamailio does not
handle this conversion
On Oct 11, 2012 6:24 AM, "SamyGo"  wrote:

> Hello,
>
> I've a scenario in which I've to deploy a couple Sangoma PRI cards with
> kamailio. What I wish is that I've some drivers for this purpose and so I
> don't ned to install FreeSWITCH or Asterisk in between the PRIs and
> Kamailio.
>
> Kindly give any feedback on what are the possibilities and options.
>
> Thanks,
> Sammy
>
>
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>
>
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Re: [SR-Users] LUA Authentication.

[David J]

Thanks Fred.

I think this was it.

I think there were some updates to the app_lua module since.

Do you know if this is still relevant?
On Sep 3, 2012 6:24 PM, "Fred Posner"  wrote:

> Hi David,
>
> I believe this is the example you're looking for. It's on the Asipto KB
> site:
>
> http://kb.asipto.com/kamailio:usage:k32-lua-routing
>
>
> ---fred
> http://qxork.com
>
> On Sep 3, 2012, at 5:06 PM, David | StyleFlare wrote:
>
> > I think I saw once an example from miconda using LUA for Auth?
> >
> > Does anyone else remember seeing that?
> >
> > I wanted to do a custom Auth using LUA.
> >
> > Thanks in advance for any pointers.
> >
> > David.
>
>
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Re: [SR-Users] $200 bounty

[David J]

Sorry Daniel. I didn't see your message until I replied.

Understood.

On to 3.3...
On Jun 20, 2012 6:27 PM, "David J"  wrote:

> Dave.
>
> Understandably. But my point was missed if you think that anyone here is
> trying to monopolize on the list please do understand that they usually
> "contribute" to the project. Besides they are responsible for making
> kamailio what is today. All so you can benefit. To say that there rates are
> outrageous is an insult to the great software your going to use. You should
> look at the overall value that the software provides and calculate how much
> you saved by nit having to build it yourself for the last 10yrz yourself.
>
> Once you put it in to perspective its really a very good value to use a
> seasoned developer on the list.
>
> I am sure they can respect your budget but equally respect their time and
> contribution.
> On Jun 20, 2012 6:13 PM, "copycall"  wrote:
>
>> david,
>>
>> invariably, this happens on most open source projects and lists.
>>
>> a few of developers begin to think all the roads lead through them.
>>
>> and, they are out today to protect their turf
>>
>> so, they start with the barber-shop membership, pseudo-licensing, and a
>> separate list: the gatekeeping begins.
>>
>> i have contacted some of the "developers" on the list over the last 18-24
>> months about other projects.
>>
>> their pricing was outrageous for a small service provider and the
>> specific project, other alternatives had to be found.
>>
>> i concluded they were looking for the "home run" client, an att or
>> verizon.
>>
>> it's the same today, a small innocuous job elicits the mocking,
>> compulsive rule citations, and in your case, a slighting of all the
>> kamailio users who are not on the developer's list.
>>
>> maybe you are right, maybe no one can do this job but a developer on the
>> list. obviously, if this is the case, money will not be exchanged.
>> however, not one developer has offered their services at any price. i will
>> welcome these proposals.
>>
>> maybe some users do not want to be on the developers list, or they are
>> not ready, or have the time at the moment. but, maybe they want to do a
>> small job to see where it leads, or for fun, or they need a few extra bucks.
>>
>> i don't mind the possible risk in giving someone a chance.
>>
>> regarding the pricing you mentioned, my job as a buyer is to get the best
>> value for my money. but, thanks for your price list.
>>
>> since several people have already offered their services, i think you are
>> a bit misinformed about the competitive marketplace.
>>
>> if your goal is to grow the number of service providers using kamailio,
>> you have employed an odd strategy to implement it.
>>
>> dave
>>
>>
>>
>> On Wed, Jun 20, 2012 at 2:20 PM, David  wrote:
>>
>>>  To chime in here,
>>>
>>> On behalf of the professional developers on the list, it seems "a guy
>>> giving a try" may get you stated but someone with experience will get you
>>> there faster with a more predictable result.
>>>
>>> I would think the going rate today for an experienced kamailio developer
>>> is about $150/hr US. its possible someone in a developing country on odesk
>>> will try and charge less like $60 USD which is the equivalent of someone
>>> here charging $1500/hr, but they most probably dont have 8-10 years
>>> experience, and also a very high probability they don't contribute to the
>>> community. Its also more than likely they wont finish the project or get
>>> you much past whats available to you for FREE on the WIKI.
>>>
>>> You have to realize professionals charge professional rates, seeing that
>>> it was "Alex bashing", If you actually search the mailing list, Alex
>>> happens to answer many difficult questions for users free of charge. So if
>>> you think Alex is getting rich quick of this list, your obviously very
>>> fresh to the list.
>>>
>>> Your certainly welcome to post a more specific question on the list like
>>> how come my server only processes 1 cps...
>>>
>>> I am sure the community will be very eager to help.
>>>
>>> Good luck on your project, if you do happen to succeed please inform the
>>> list, we would be happy to welcome your star developer into the community.
>>>
>>> All the best.
>>>
>>>
>

Re: [SR-Users] $200 bounty

[David J]

Dave.

Understandably. But my point was missed if you think that anyone here is
trying to monopolize on the list please do understand that they usually
"contribute" to the project. Besides they are responsible for making
kamailio what is today. All so you can benefit. To say that there rates are
outrageous is an insult to the great software your going to use. You should
look at the overall value that the software provides and calculate how much
you saved by nit having to build it yourself for the last 10yrz yourself.

Once you put it in to perspective its really a very good value to use a
seasoned developer on the list.

I am sure they can respect your budget but equally respect their time and
contribution.
On Jun 20, 2012 6:13 PM, "copycall"  wrote:

> david,
>
> invariably, this happens on most open source projects and lists.
>
> a few of developers begin to think all the roads lead through them.
>
> and, they are out today to protect their turf
>
> so, they start with the barber-shop membership, pseudo-licensing, and a
> separate list: the gatekeeping begins.
>
> i have contacted some of the "developers" on the list over the last 18-24
> months about other projects.
>
> their pricing was outrageous for a small service provider and the specific
> project, other alternatives had to be found.
>
> i concluded they were looking for the "home run" client, an att or verizon.
>
> it's the same today, a small innocuous job elicits the mocking, compulsive
> rule citations, and in your case, a slighting of all the kamailio users who
> are not on the developer's list.
>
> maybe you are right, maybe no one can do this job but a developer on the
> list. obviously, if this is the case, money will not be exchanged.
> however, not one developer has offered their services at any price. i will
> welcome these proposals.
>
> maybe some users do not want to be on the developers list, or they are not
> ready, or have the time at the moment. but, maybe they want to do a small
> job to see where it leads, or for fun, or they need a few extra bucks.
>
> i don't mind the possible risk in giving someone a chance.
>
> regarding the pricing you mentioned, my job as a buyer is to get the best
> value for my money. but, thanks for your price list.
>
> since several people have already offered their services, i think you are
> a bit misinformed about the competitive marketplace.
>
> if your goal is to grow the number of service providers using kamailio,
> you have employed an odd strategy to implement it.
>
> dave
>
>
>
> On Wed, Jun 20, 2012 at 2:20 PM, David  wrote:
>
>>  To chime in here,
>>
>> On behalf of the professional developers on the list, it seems "a guy
>> giving a try" may get you stated but someone with experience will get you
>> there faster with a more predictable result.
>>
>> I would think the going rate today for an experienced kamailio developer
>> is about $150/hr US. its possible someone in a developing country on odesk
>> will try and charge less like $60 USD which is the equivalent of someone
>> here charging $1500/hr, but they most probably dont have 8-10 years
>> experience, and also a very high probability they don't contribute to the
>> community. Its also more than likely they wont finish the project or get
>> you much past whats available to you for FREE on the WIKI.
>>
>> You have to realize professionals charge professional rates, seeing that
>> it was "Alex bashing", If you actually search the mailing list, Alex
>> happens to answer many difficult questions for users free of charge. So if
>> you think Alex is getting rich quick of this list, your obviously very
>> fresh to the list.
>>
>> Your certainly welcome to post a more specific question on the list like
>> how come my server only processes 1 cps...
>>
>> I am sure the community will be very eager to help.
>>
>> Good luck on your project, if you do happen to succeed please inform the
>> list, we would be happy to welcome your star developer into the community.
>>
>> All the best.
>>
>>
>>
>>
>>
>> On 6/20/12 3:28 PM, copycall wrote:
>>
>> alex,
>>
>> if it offended you, then i apologize for telling you to "get a pair".
>>
>> well, i have received several offers to do what i am looking for, not all
>> at my asking price, which you think is too low.
>>
>> maybe some kamailio users don't consider themselves developers, but are
>> still interested in trying something new.
>>
>> i am willing to give someone an opportunity, including you.
>>
>> dave
>>
>>
>>
>>  On Wed, Jun 20, 2012 at 12:12 PM, Fred Posner wrote:
>>
>>> On Jun 20, 2012, at 2:40 PM, Alex Balashov wrote:
>>>
>>> > On 06/20/2012 02:38 PM, Fred Posner wrote:
>>> >
>>> >> I know a great bakery that can offer dessert kamailio pricing.
>>> >> Generally, $6/portion starting.
>>> >
>>> > What's the going rate for the Millenium Falcon cake these days?  :-)
>>> >
>>>
>>>  They start at $250 and increase by size. =)
>>>
>>> Death Stars are much cheaper. =)
>>>
>>> With best regards,
>>>
>>> Fred

Re: [SR-Users] kamailio with flowroute

[David J]

You can use IP auth its simple and works.
On Mar 8, 2012 4:19 PM, "romon.zaman"  wrote:

>   hello room,
>
> i was trying to add sip provider(like. flowroute,vitelity) with user-pass
> authentication in kamailio to accept inbound calls.
>
> any help?
>
> thanks
>
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Re: [SR-Users] Registration Limits

[David J]

Yes set max-contacts in usrloc module
On Mar 2, 2012 5:45 AM, "Reda Aouad"  wrote:

> Hi,
>
> Is there a way to ensure single-registration per user-agent for a user,
> which overwrites previous registration ?
> Or is there a way to limit the number of registrations per user, but
> overwriting the earliest registration for each new one ?
>
> Thanks,
> Reda
>
>
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Re: [SR-Users] [Kamailio-Business] Kamailio presentations at Cluecon 2011

[David J.]

Hmm...we will see next year; :)




Seems like we have turned ClueCon into KamailioCon. Good work, all of you!



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Re: [SR-Users] Chicago, ClueCon next week

[David J.]

Hello Daniel;

Will be there next week too; Going only for Thursday (Last Day);

Hope to have a chance to meet you.

Be in touch.

David.



On 8/2/11 5:15 PM, ifeanyi okoye wrote:

Hello Daniel,

My name is Ify and I'm a research voip engineer.I would be attending 
the conference. Please can I reach you via telephone.


Regards,
Ify

--- On *Tue, 8/2/11, Daniel-Constantin Mierla //* 
wrote:



From: Daniel-Constantin Mierla 
Subject: [SR-Users] Chicago, ClueCon next week
To: "kamailio users" ,
busin...@lists.kamailio.org
Date: Tuesday, August 2, 2011, 1:59 AM

Hello,

I'll be in Chicago few days next week for ClueCon conference. If
you are in the area, but not attending the ClueCon conference,
drop me a message in case you want to meet.

Cheers,
Daniel

-- Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda


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Re: [SR-Users] Chicago, ClueCon next week

[David J]

Will you also be there the last day as well?
On Aug 2, 2011 6:15 PM, "Daniel-Constantin Mierla" 
wrote:
> Hello Ify,
>
> On 8/2/11 11:15 PM, ifeanyi okoye wrote:
>> Hello Daniel,
>>
>> My name is Ify and I'm a research voip engineer.I would be attending
>> the conference. Please can I reach you via telephone.
>>
> the phone might not be the most convenient at that time, so the best is
> to ask the organizers to point you to me, unless you can spot me based
> on the pictures you can find on the web. I will have the presentation on
> the 2nd day, but I will be around on 1st day as well.
>
> Looking forward to meeting you,
> Daniel
>
>>
>> Regards,
>> Ify
>>
>> --- On *Tue, 8/2/11, Daniel-Constantin Mierla //*
>> wrote:
>>
>>
>> From: Daniel-Constantin Mierla 
>> Subject: [SR-Users] Chicago, ClueCon next week
>> To: "kamailio users" ,
>> busin...@lists.kamailio.org
>> Date: Tuesday, August 2, 2011, 1:59 AM
>>
>> Hello,
>>
>> I'll be in Chicago few days next week for ClueCon conference. If
>> you are in the area, but not attending the ClueCon conference,
>> drop me a message in case you want to meet.
>>
>> Cheers,
>> Daniel
>>
>> -- Daniel-Constantin Mierla -- http://www.asipto.com
>> Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
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>>
>>
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>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
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[SR-Users] Aliases Using the Default Script;

[David J.]

I am using the kamailio default script on 3.1.3.

I was wondering what happens when I added an Alias in dbaliases?

For example if I add 18005551...@mydomain.com alias to 1...@mydomain.com

when an invite comes in; it works perfect I got a 200 back. (1001 Device 
rings.)


If I add another alias to a remote server; ie.

18005551...@mydomain.com alias to 1800555...@someotherdomain.com

I get a 404 back.

I wonder what happens internally?

What should I modify to enable this case;

Thanks.

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Re: [SR-Users] xHTTP module.

[David J.]

Daniel;

Sorry for not being clear;

I understand the HTTP stuff;

I was asking if it is a simple as just calling the dialog bridge method 
within the HTTP event route.


for example;

event_route[xhttp:request] {

dlg_bridge("sip:m...@myproxy.com", "sip:y...@yourproxy.com",
"sip:myproxy.com:5080");

   ...
} 

Thanks for the clarification

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Re: [SR-Users] xHTTP module.

[David J.]

Thanks Daniel;

What creates the Initial Dialog and REFER method

Just calling

dlg_bridge("sip:m...@myproxy.com", "sip:y...@yourproxy.com",
   "sip:myproxy.com:5080");



When I get an HTTP event;









On 5/20/11 9:32 AM, Daniel-Constantin Mierla wrote:

Hello,

On 5/20/11 3:25 PM, David J. wrote:

I was wondering if I can make an HTTP request to Kamailio;
and then have kamailio do a lookup based on passed parameters
to connect callers.

I am trying to make a click-to-dial type application;

I was looking at the HTTP server inside kamailio;
It was interesting to me to try to use this as an alternative to 
php-asterisk;


(I know kamailio is not a full blown application server;)

If anyone could help me with this I would appreciate it;
you can get the caller and callee from HTTP GET parameters and do 
click-to-dial (implemented with REFER) using dlg_bridge() function:

http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2966104

Cheers,
Daniel




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[SR-Users] xHTTP module.

[David J.]

I was wondering if I can make an HTTP request to Kamailio;
and then have kamailio do a lookup based on passed parameters
to connect callers.

I am trying to make a click-to-dial type application;

I was looking at the HTTP server inside kamailio;
It was interesting to me to try to use this as an alternative to 
php-asterisk;


(I know kamailio is not a full blown application server;)

If anyone could help me with this I would appreciate it;

Thanks in advance.

David.

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Re: [SR-Users] Custom voicemail table

[David J]

Actually I might of made a mistake
I don't think the Voicemail table is used at all in that tutorial. Asterisk
only uses it. Kamailio does not use it.
On Apr 5, 2011 3:25 PM, "David J"  wrote:
> Of course. Look at the config it should be very easy. Just replace with
the
> name you want to use
> On Apr 5, 2011 3:19 PM, "Lucas Alvarez"  wrote:
>> Hi, is it possible to change the name of the table voicemessages for
>> voicemail profile in a kamailio-asterisk integration? I mean of the
> kamailio
>> side.
>> Thanks in advance.
>>
>> Lucas
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Re: [SR-Users] Custom voicemail table

[David J]

Of course. Look at the config it should be very easy. Just replace with the
name you want to use
 On Apr 5, 2011 3:19 PM, "Lucas Alvarez"  wrote:
> Hi, is it possible to change the name of the table voicemessages for
> voicemail profile in a kamailio-asterisk integration? I mean of the
kamailio
> side.
> Thanks in advance.
>
> Lucas
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Re: [SR-Users] The SIP protocol v2 - we're giving up.

[David J.]

For a second you really had us going.

Good Job
:)

On 4/1/11 4:54 AM, Olle E. Johansson wrote:

Friends,

After having spent many years working with the Asterisk SIP channel driver, 
Kamailio and the SIPv2 protocol, I have finally realized that this is a dead 
end. It's getting nowhere and it's way too complicated to set up, run and 
support in working code.

After realizing this, I started a new standardization project together with my 
friends in Canada, Simon and Marc, to develop a working solution based on the 
combination of IPv6 and SIP. We have gotten great feedback and now the IETF, 
the ITU and the IPv6 forum jointly launches the new standard, SIP-six.

> From the press release:

"”We realize that 99% of the SIP users use SIP for PSTN phone calls. The 
original SIP standards was written with other applications in mind, a vision that 
never came true.” said Bob Plug, area director in the IETF. ”That’s why we sat down 
and said to ourselves that this should be way more simple.”

The SIP-six standard totally removes the dependency of domains and URI syntax. 
There’s no point in using this, since everyone seems to think that IP 
addressing is more than enough. The new standard use part of the vast IPv6 
address space to incorporate the E.164 phone numbers as addresses. This is the 
reverse of the reverse phone number usage in the enum standard, which is no 
longer needed in SIP-six.

By using IPv6 mobile IP, phone users register their phones and get access to 
their phone number. Users that need security can easily integrate IPsec into 
their setup. Media and signalling uses the same addressing scheme and is mixed 
so that both SIP-six, RTP and RTCP only uses one port address - but in this 
case a port address with 32 bit subaddress identifying the media stream. This 
finally solves a lot of the firewall traversal issues that SIP v2.0 had. With 
the combination of mobile IP and use of public IPv6 addresses NAT traversal 
won’t be an issue.

The testbed for SIP-six has been running for a year at six choosen large SIP 
carriers, with the assistance of Edvina AB in Sweden and ViaGenius in Montreal, 
Canada. In an International effort, the testbed is today put in production and 
Roboid phones all over the world is automatically connected to this worldwide 
network."


You will be able to find out more about it here:
http://bit.ly/sipsix

SIP-six is implemented as a channel driver in Asterisk 2.0, as a replacement 
for SIP2.0 in Kamailio 4.0 and a channel module in FreeSwitch - all releases to 
be released later today. Softphones for testing will shortly be available from 
Blink and Zoiper.

Looking forward to continue this project with the rest of the 
Kamailio/SIP-router community!
Special thanks to Daniel for the reference implementation in Kamailio 4.0!

Have a nice weekend!

/Olle
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[SR-Users] Bad File Descriptor

[David J.]

 5(20390) ERROR:  [udp_server.c:586]: ERROR: udp_send: 
sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file 
descriptor(9)

 5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed


I see this error when I try to restart kamailio after crash; I see the 
cause of this problem is 'stale' entries in the location table;


If I delete this entries kamailio starts fine; any suggestions to 
prevent this from continuously happening.


Thanks.



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[SR-Users] Refer Using UAC.

[David J.]

I realize that kamailio is not a b2bua;
But because we are using Asterisk in the path;

To extend the Asterisk Realtime Tutorial;

I was wondering if I could do something like this...

Kind of like how we use UAC to send a register to Asterisk;
Could we do the same and modify the method to use REFER instead?

I know it is more complex; but I am not sure where to handle this case;

Thanks for any pointers.

if(is_method("REFER")){
$var(rip) = $sel(cfg_get.asterisk.bindip);
$uac_req(method)="REFER";
$uac_req(ruri)="sip:" + $var(rip) + ":" + 
$sel(cfg_get.asterisk.bindport);

$uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft;
$uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt;
$uac_req(hdrs)="Contact: + ":" + $sel(cfg_get.kamailio.bindport) 
+ ">\r\n";

if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + 
$sel(contact.expires) + "\r\n";

else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + 
$hdr(Expires) + "\r\n";



 uac_req_send();


}

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[SR-Users] Handling call transfer in the Asterisk Realtime setup.

[David J.]

I am trying to add support for call transfer in the Asterisk realtime 
tutorial on Asipto;


I am not sure what I would have to do to get this feature working;

Perhaps I have to handle "refer" messages; but I am not sure how I send 
that to Asterisk;


Any advice would be greatly appreciated.



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[SR-Users] 404 Not Found - If User Not Registered

[David J.]

if I do lookup()

What case does lookup return if entry exists but user not registered?

lookup("location");
switch ($retcode) {
case -1:
case -3:
sl_send_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Not Found");
exit;
};


should I just wrap this like this:

if(registered("location")){

lookup("location");
switch ($retcode) {
case -1:
case -3:
sl_send_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Not Found");
exit;

}
};

The purpose I ask this rather simple question is because I want to redirect to 
voicemail if user is not registered.
In the default script we have this case;

failure_route[FAIL_ONE] {
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
##if (t_check_status("486|408")) {
##  sethostport("192.168.2.100:5060");
##  append_branch();
##  # do not set the missed call flag again
##  t_relay();
##}
}

But it seems this case is never met;

Because this code


lookup("location");
switch ($retcode) {
case -1:
case -3:
sl_send_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Not Found");
exit;
};


Seems to send a 404 response to the UA rather than go to failure route?


Maybe I am wrong...

Any advice would be appreciated.





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[SR-Users] 404 failure route;

[David J.]

I am using the default script;

When I do a lookup for a user that is not registered; I get 404 back;

I added in my failure route to the list of codes 404; ie:

First on INVITE I have

t_on_failure("FAIL_ONE");


failure_route[FAIL_ONE] {
if (t_check_status("486|408|404")) {
sethostport("mydomain.com:5060");
  append_branch();

}
}

But for some reason I dont see it getting caught in the failure route 
and sending to the server specified;


Thanks.



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[SR-Users] Purple Error.

[David J.]

If anyone can direct me how to resolve.


I load presence.so,pua.so,purple.so

When I run kamailio I see.


Dec  9 14:36:05 localhost /usr/local/sbin/kamailio[1605]: ERROR: purple 
[purple.c:148]: can't import load_tm


Thanks.

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[SR-Users] Psuedo Variable.

[David J.]

Hello List;

I was following along the Kamailio Asterisk Realtime Integration; I 
modified the script to get it working on Kamailio v 3.1.


One area I had a question about was here...

We have the following block;

# Forward REGISTER to Asterisk
route[REGFWD] {
if(!is_method("REGISTER"))
{
return;
}
$var(rip) = $sel(cfg_get.asterisk.bindip);
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" + 
$sel(cfg_get.asterisk.bindport);

$uac_req(furi)="sip:" + $au + "@" + $var(rip);
$uac_req(turi)="sip:" + $au + "@" + $var(rip);
$uac_req(hdrs)="Contact: + ":" + $sel(cfg_get.kamailio.bindport) 
+ ">\r\n";

if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + 
$sel(contact.expires) + "\r\n";

else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + 
$hdr(Expires) + "\r\n";

uac_req_send();
}

If I change the line to read : $uac_req(hdrs)="Contact: "@" + $ad
from : $uac_req(hdrs)="Contact: $sel(cfg_get.kamailio.bindip)


I get the following error;

loading modules under 
/usr/local/kamailio-3.1-proxy/lib/kamailio/modules_k/:/usr/local/kamailio-3.1-proxy/lib/kamailio/modules/

$ad
 0(13715) ERROR:  [pvapi.c:445]: bad parameters
 0(13715) :  [cfg.y:3409]: parse error in config file 
/usr/local/kamailio-3.1-proxy/etc/kamailio/kamailio.cfg, line 717, 
column 48-47: unknown script pseudo variable

ERROR: bad config file (1 errors)


I would imagine that if the "$au" pseudo variable is available the "$ad" 
variable would be available as well.


Please help point out my mistake.

Thanks.

David.



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