Re: [OpenSIPS-Users] MWI on phones

[Mark Sayer]

I've never been able to get the sipsak hack to work satisfactorily and
have configured opensips to pass the subscribe to askerisk and the
resulting MWI back to the phone. Works fine.

Mark

On Tue, May 15, 2012 at 5:38 AM, Vallimamod ABDULLAH
 wrote:
> Hi,
>
> Have you tried the "externnotify" option in asterisk voicemail.conf and use a 
> custom script to generate the notify with sipsak 
> (http://www.voip-info.org/wiki/view/Asterisk+Realtime+MWI+Hacks) ?
>
> This hack does not handle the reboot case. You can eventually setup a cron 
> job that checks for new voicemails and sends the corresponding notify 
> regularly.
>
> Best Regards,
> - vma
> .
>
>
>
> On May 14, 2012, at 9:02 PM, Schneur Rosenberg wrote:
>
>> My phones are registered to opensips, the Voicemail is handled by
>> asterisk, Asterisk has a list of all sip devices, but it has no idea
>> if phones are connected or not, I would like to get the MWI from
>> asterisk sent over to the opensips, so opensips can send it to the
>> phone.
>>
>> Is there anyway to have Opensips notify Asterisk when a phone
>> connects, so asterisk is aware what phones are connected? would this
>> help for the MWI issue?
>>
>> I'm not sure if I'm right but it seems that some phones send a
>> subscribe for MWI, with these phones I was able to forward the
>> subscribe to asterisk and it works, but some sip devices just wait for
>> Asterisk to send them, this makes it more complicated.
>>
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Re: [OpenSIPS-Users] Blind Transfer to Asterisk

[Mark Sayer]

I assume you have the Asterisk "GOTO_ON_TRANSFER" parameter set?

Mark

On Wed, May 16, 2012 at 2:44 AM,   wrote:
> You need something in your default context that will figure out the correct
> context to send it too. I'm not sure what to do with REFER messages but my
> contexts are named after my customers Domains. So in the default context you
> could have something like
>
> exten => _NX,1,Set(dm=${SIP_HEADER(TO):15})
> exten => _NX,n,Set(dm=${CUT(dm,>,1)})
> exten => _NX,n,GotoIf(${dm} != ${CONTEXT}?${dm},${EXTEN},1)
>
>
> Once again this totally depends on your OpenSIPS and Asterisk setup. We
> would need more information.
>
>
> On , Schneur Rosenberg  wrote:
>> I use opensips to send calls via Asterisk, I share the sip usernames
>>
>> and passwords from opensips with Asterisk, and thats how Asterisk
>>
>> knows what context, caller id etc to use.
>>
>>
>>
>> everything works fine, the only issue is when doing a Blind Transfer
>>
>> OpenSIPS sends a REFER to asterisk, but for some reason when that
>>
>> happens, asterisk does not realize which user sent the REFER and it
>>
>> tries to call the number from the default context.
>>
>>
>>
>> Is there anyway I can create a new INVITE request and drop the REFER
>>
>> request, whenever a REFER comes in?
>>
>>
>>
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>>
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>>
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Re: [OpenSIPS-Users] Source IP address on Asterisk integration

[Mark Sayer]

I guess I'm wondering why you want to do that?

Mark

On Mon, May 21, 2012 at 5:49 PM, ksy  wrote:
> Ronald Cepres  writes:
>
>>
>>
>> Hi all,
>>
>> I'm trying to set up Opensips so that it simply relays the requests it
> receives to Asterisk on the same server, only using a different port. The 
> set-up
> is working but my problem is Asterisk uses the IP of OpenSIPS as peer contact
> even if the domain on the Contact header is from the actual sender of the
> request. The peer's setting on Asterisk is nat=yes, and I'm not allowed to
> change this value. Can I tweak something on Opensips so that Asterisk can see
> the real sender's IP address even with nat=yes on Asterisk?
>>
>> Thanks!
>>
>> Regards,
>> Ronald
>>
>
> Hi, Ronald! Have the same issue. I wonder if you have solved the problem.
>
>
>
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Re: [OpenSIPS-Users] CDRTool and B Party Billing

[Mark Sayer]

The Australian numbers to which you are referring require charges to
be made to both the caller and the callee. Calls made to 6113x numbers
are charged "the cost of a local call" so those should need no changes
other than to treat them as local calls. Calls received by way of a
6113x number (by the number owner) are charged a per minute call rate
similar to the FreeCall 6118x numbers. Treat them the same.

I don't believe changes are required to OpenSIPS.

Mark

On Sat, Nov 20, 2010 at 11:10 AM, Mike O'Connor  wrote:
> Hi Guys
>
> Does any one know how to setup CDRTool to handle B party billing. (ie
> the person called pays for the call) ?
>
> Australia has numbers a person/company can purchase which are almost
> free to the person calling but are charged to the numbers owner at a per
> minute rate. The numbers are in the format 6113XX and 6113.
>
> I assume I'll need to make changes to OpenSIPS, but I'm not totally sure
> what those changes should be.
>
> Thanks for any ideas
> Mike
>
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Re: [OpenSIPS-Users] OpenSIPS as regional/national office solution

[Mark Sayer]

Here is one suggestion:
- single OpenSIPS & Asterisk at central office
- use Asterisk as gateway to PSTN (for all offices)
- connect remote office PBXs to central office using using multi-port
FXS gateways
- 110ms is no problem
- single system admin point, single cpu, 200 or more concurrent calls
- no admin, low cost at remote offices

Mark

On Sun, Jan 16, 2011 at 8:21 AM, Sean Kelly  wrote:
> Hello all,
>  I am new to OpenSIPS but have a problem which I believe OpenSIPS can solve.
> Due to the power of OpenSIPS there are many ways to solve any particular
> issue it seems. I wondered if the expertise of the group would assist in
> pointing me in a relevant direction with regards to where to focus my
> education into a solution using OpenSIPS.
>  We currently have 10 very old pbx's, 1 at each office and these connect
> directly to the telco. As such, the calls between offices are charged
> long-distance. All offices have more than adequate Internet connectivity.
> What I would like to do is put * servers in each region where office phones
> will register with the central * and have those trunk to OpenSIPS which will
> direct internal calls between servers (free national calls), and also keep
> track of calls out to the PSTN to match/compare with telco billing. OpenSIPS
> will trunk to a CLEC providing inbound/outbound service. Some offices are
>>=40ms from head office, some are as high as 110ms.
> Goals: Single domain, regionally dispersed, free internal calls nationally,
> high-availability (region & national), single or load-balanced in/out from
> CLEC and bill tracking.
>
> Questions:
> 
> 1. The CLEC charges trunk setup fees, so is it feasible to have all in/out
> PSTN calls going through OpenSIPS nationally so as to only have 1 or 2
> trunks with provider? (as opposed to 10)
> 2. Should I have 2 OpenSIPS in this scenario with each regional *
> trunking/fail-over to each or have 2 OpenSIPS+Asterisk setup regionally with
> each region trunking to CLEC? .. or both?
> 3. Which modules / combination of modules would you recommend I focus on to
> achieve the above goals?
> 4. CID is important as well, internal & pstn. Does OpenSIPS simply
> pass-through CID or do I need to look into scripting a solution for this to
> work like a traditional telco? (do not see CID passing in lab).
> 5. For inbound, how would I direct calls to the appropriate regional *
> server?
> 6. How would you break this down to ease the start of the complete solution?
> ie: focus on national inbound/outbound or regional first?
>  Presently I have graduated from the OpenSIPS_Live_DVD_VM to having OpenSIPS
> installed/setup & working with phones in my lab on Debian 5. Have been
> following the OpenSIPS telephony book and also watching videos and reading
> articles on opensips.org which have all been excellent resources so far in
> getting me through the basics.
>
>   Any suggestions on which modules or part of the solution I should focus on
> next? General comments / suggestions?
>
>
> Thanks,
> Skyler
>
>
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Re: [OpenSIPS-Users] OpenSIPS as regional/national office solution

[Mark Sayer]

Skyler -

We are a South Pacific regional provider of hosted PBX services so I
may be prejudiced toward a like infrastructure. Some of our customers
are 3000kms from our servers but the ping times are still less than
50ms so I'm curious why yours are so long. That said, 200ms is sort of
the magic number you don't want to exceed. (Having said that, we do
get some pretty decent call quality connecting to some terminators who
are over 250ms away. 50+250 and its still OK.) Call quality is 99%
Internet connection. OpenSIPS + Asterisk works perfectly with every
call but if the Internet (which you can't control) plays up you get
flack for providing a bad service.

I'd recommend spending some time looking at your Internet connections.
Can you get them all from the same provider? (I don't even know what
sort of connections you are talking about. We actually get business
grade voice quality from ADSL over copper.) Can you locate your server
in a data center that has good connections to both your ISP and your
terminator? My dream has always been to have a large rack of equipment
in the back office but to make our service work I've had to locate in
a major data centre hundreds of kms away. Our office isn't nearly as
impressive as our service is but that's what the customers pay for.

I'd only put servers in the offices if there was some reason that
functionality was needed there. Even if you need a receptionist at
each office that can all be handled from a single Asterisk box.

Just more thoughts.
Mark

On Mon, Jan 17, 2011 at 6:11 PM, Skyler  wrote:
> Hi Mark,
>  Thanks for the reply. So if I understand correctly, I am thinking too big.
> K.I.S.S as some say.
> The existing PBX's are extremely old, so breakdowns & phones are a problem
> and we don't want to repair anymore. In the suggested scenario would you
> recommend replacing the existing hardware (as they breakdown) with IP phones
> and Asterisk at each office then or just ditch the Asterisk and have all the
> phones register to OpenSIPS directly at HQ? My concern is call quality with
> 110ms to HQ then 75ms to provider = 185ms from furthest office, is this
> still not an issue?
> Thanks,
> Skyler
>
> On Sat, Jan 15, 2011 at 4:55 PM, Mark Sayer  wrote:
>>
>> Here is one suggestion:
>> - single OpenSIPS & Asterisk at central office
>> - use Asterisk as gateway to PSTN (for all offices)
>> - connect remote office PBXs to central office using using multi-port
>> FXS gateways
>> - 110ms is no problem
>> - single system admin point, single cpu, 200 or more concurrent calls
>> - no admin, low cost at remote offices
>>
>> Mark
>>
>
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Re: [OpenSIPS-Users] OpenSIPS as regional/national office solution

[Mark Sayer]

We use NAThelper and find it to be a perfect solution for us.

Regarding Bogdan's comments ... I agree 100%. The centralised network
works for us and I recommend that others consider it but you have to
meet your own requirements.

Mark

On Tue, Jan 18, 2011 at 9:28 PM, Skyler  wrote:
> Mark -
>  Thanks for sharing your thoughts, they are definitely helping to put the
> pieces of this puzzle together. Today I spent most of the day mapping out
> each office via the net and found the common backbone interconnects. At
> these x-connects I found 2 data centers. All offices are 30-40ms from one or
> the other and both DC's are 15-20ms from each other. I couldn't figure out
> what the distance would be from the DC to the provider, though I know the
> provider is in a major DC and one Province over so it can't be more than
> 15-20ms across the backbone.
>  Both DC's offer dedicated servers, so we are going to look into putting one
> server at each DC and ditch the original regional/national plan for a more
> conservative and easy to manage plan. I'm confident now that there will be
> better overall quality going this way.
> Now its time to unscramble the mess that is my install notes and document a
> clean OpenSIPS+Asterisk install before moving further. After that I'm a bit
> lost though as I know that we need NAT but not sure which solution is best /
> easiest to work with (RTPproxy, NAThelper, MediaProxy). From what I've read
> up on each, Nathelper seems to be built into Osips whereas RTPproxy and
> MediaProxy require a possibly troublesome install vs loading module/adding
> code. Searching the mailing archives hasn't been enough for me to decide on
> a winner.
>  From what it sounds like, you have a lot of experience in the setup that
> I'm working on building. Out of curiosity, which method do you prefer for
> resolving far-end NAT issues?
>
> Skyler
>
>
> On Mon, Jan 17, 2011 at 1:26 AM, Mark Sayer  wrote:
>>
>> Skyler -
>>
>> We are a South Pacific regional provider of hosted PBX services so I
>> may be prejudiced toward a like infrastructure. Some of our customers
>> are 3000kms from our servers but the ping times are still less than
>> 50ms so I'm curious why yours are so long. That said, 200ms is sort of
>> the magic number you don't want to exceed. (Having said that, we do
>> get some pretty decent call quality connecting to some terminators who
>> are over 250ms away. 50+250 and its still OK.) Call quality is 99%
>> Internet connection. OpenSIPS + Asterisk works perfectly with every
>> call but if the Internet (which you can't control) plays up you get
>> flack for providing a bad service.
>>
>> I'd recommend spending some time looking at your Internet connections.
>> Can you get them all from the same provider? (I don't even know what
>> sort of connections you are talking about. We actually get business
>> grade voice quality from ADSL over copper.) Can you locate your server
>> in a data center that has good connections to both your ISP and your
>> terminator? My dream has always been to have a large rack of equipment
>> in the back office but to make our service work I've had to locate in
>> a major data centre hundreds of kms away. Our office isn't nearly as
>> impressive as our service is but that's what the customers pay for.
>>
>> I'd only put servers in the offices if there was some reason that
>> functionality was needed there. Even if you need a receptionist at
>> each office that can all be handled from a single Asterisk box.
>>
>> Just more thoughts.
>> Mark
>>
>> On Mon, Jan 17, 2011 at 6:11 PM, Skyler  wrote:
>> > Hi Mark,
>> >  Thanks for the reply. So if I understand correctly, I am thinking too
>> > big.
>> > K.I.S.S as some say.
>> > The existing PBX's are extremely old, so breakdowns & phones are a
>> > problem
>> > and we don't want to repair anymore. In the suggested scenario would you
>> > recommend replacing the existing hardware (as they breakdown) with IP
>> > phones
>> > and Asterisk at each office then or just ditch the Asterisk and have all
>> > the
>> > phones register to OpenSIPS directly at HQ? My concern is call quality
>> > with
>> > 110ms to HQ then 75ms to provider = 185ms from furthest office, is this
>> > still not an issue?
>> > Thanks,
>> > Skyler
>> >
>> > On Sat, Jan 15, 2011 at 4:55 PM, Mark Sayer 
>> > wrote:
>> >>
>> >> Here is one suggestion:
>> >> - single OpenS

Re: [OpenSIPS-Users] Call from Asterisk to Opensips

[Mark Sayer]

You have provided us with the error message from Asterisk but what
have you looked to see what OpenSIPS is doing? Is ext1001 currently
registered with OpenSIPS? There are a number of ways that Asterisk and
OpenSIPS might be configured to operate together. You will have to
give us more information on your setup.

Mark

On Fri, May 6, 2011 at 1:53 PM, Duong Manh Truong
 wrote:
> Hi all,
> I've created sip trunk on Asterisk and defined asterisk server ip on address
> table of opensips
> Then, from extension of Opensips , i can dial out to pstn through Asterisk
> Now, i want to route PSTN call to the extension
> but when Asterisk receive the call from PSTN and dial Opensips through the
> Sip Trunk
> i always got the message in the asterisk's console:
>  Called to-opensips/1001
>     -- SIP/to-opensips-0745 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
> (1001 is the extension of Opensips)
> Then the call hangs up.
> Anyone got this problem ? please help me the way to deal with!
> Thanks so much!
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Re: [OpenSIPS-Users] Using OpenSIPS as Proxy to Asterisk

[Mark Sayer]

A "standard" setup would change Asterisk to listen on 5061. Users
would register to OpenSIPS using 5060 and pass the call. via 5061 to
Asterisk for media handling.

Mark

On Wed, Jul 20, 2011 at 3:25 PM, Michael  wrote:
> Hello,
>
> We would like to use OpenSIPS as a Proxy to Asterisk.
>
> It's currently installed and running on the same server, listening to port
> 5260.
>
> What should we do to set SIP clients to connect to this port and be routed
> to the Asterisk that listens on port 5060? FYI, port 5060 is blocked for
> access from the Internet due to numerous daily access attempts.
>
> Thanks,
>
> Michael
>
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Re: [OpenSIPS-Users] openSIPS vs Kamailio vs SIP-Router

[Mark Sayer]

Ditto what Nick said. Both will likely work for you equally well. The
split of OpenSER into Kamailio and OpenSIPS was a personality issue
between the main developers that didn't really please the wider user
community.

(Not trying to start a flame war, but that's how it felt from here.)

Mark

On Mon, Aug 15, 2011 at 1:02 PM, David J.  wrote:
> Nick,
>
> All are used; SIP-Router is really just kamailio; But in terms of OpenSIP's
> vs Kamailio they are virtually the same software.
>
> As far as performance goes; I think they are about even. You will notice
> some differences for example OpenSIPs has a B2B module that you wont find in
> Kamailio; that is used for certain scenarios like topology hiding; if that
> is important to you then that maybe a reason to go with opensips.
>
> They both support common DB access like mysql, postgres.
>
> Each installation has their own criteria, but in terms of scale, both
> projects are used in carrier grade deployments. And in some cases both are
> used for various reasons in the same network.
>
> Hope that helps answer some of your questions.
>
>
>
> On 8/14/11 10:41 PM, Nick Khamis wrote:
>>
>> Hello Everyone,
>>
>> I can only imagine how many times this question has come up since post
>> 2008. Please forgive
>> my reoccurring of  the issue.
>>
>> We are looking to provide carrier grade sip services to our clients
>> world wide. What we need is a
>> lightweight, robust and scalable solution that will allow us to
>> terminate sip calls to our different carriers.
>> Performance, and high throughput are factors very important to my
>> employer. Features such as caller
>> authentication, database back-end, load balancing, and
>> interoperability with asterisk are things we are
>> interested in, as was offered using OpenSER.
>>
>> With three+ open source proxy servers available on the net puts us in
>> a situation where we have more
>> solutions to choose from, at the same time wish the features from one
>> were available in the other, and
>> vice versa.
>>
>> With this in mind, we will have to fall back to other factors such as
>> the most reliable, proven and active
>> projects. As mentioned, we would choose functional stability over
>> endless features that we will never use
>> and that add to the projects fingerprint...
>>
>> I understand that all three projects are forks from OpenSER, people
>> would naturally like to know what
>> differentiates one from the other.
>>
>> Thanks in Advance,
>>
>> Nick Khamis
>> Toronto Hydro Telecom
>>
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[OpenSIPS-Users] OpenSER+Asterisk+BLF

[Mark Sayer]

We are needing to modify the configure of a currently operating 
OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that 
are sent between Asterisk and a phone that supports BLF (like the 
Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 & 
Asterisk 1.4.17 in the same box. OpenSER performs all registration, 
authentication and NAT. Asterisk handles the media and the accounting.

In a pure Asterisk environment a "hint" would be setup in the 
Asterisk extensions.conf file and the phone (UA) would SUBSCRIBE to 
that HINT. Once Asterisk has registered that UA to the HINT then it 
sends SIP-NOTIFY messages to the UA as the status of the channel 
changes (available, ringing, busy).

Our current openser.cfg file makes no mention of either SUBSCRIBE or 
NOTIFY which is an obvious reason that my Asterisk installation never 
registers the UA to the HINT.

Is anyone interested in getting paid to fix this for us (we're too 
stupid to do it ourselves) or to offer another solution for 
controlling BLF in this setup.

Thanks,
Mark


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Re: [OpenSIPS-Users] OpenSER+Asterisk+BLF

[Mark Sayer]

All right I'm convinced and have started the migration from OpenSER 
1.2 to OpenSIPS 1.6. Can anyone point me to list of database changes 
between these two? I've gotten 1.6 to run right out of the can which 
is nice and the config file differences don't look too bad. VMA - I 
might get back to you with any problems as you've done this recently.

Mark

At 02:28 a.m. 09/03/2010, you wrote:
>Hi Mark,
>
>On Sunday7Mar, 2010, at 1:21 AM, Mark Sayer wrote:
>
> > We are needing to modify the configure of a currently operating
> > OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that
> > are sent between Asterisk and a phone that supports BLF (like the
> > Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 &
> > Asterisk 1.4.17 in the same box. OpenSER performs all registration,
> > authentication and NAT. Asterisk handles the media and the accounting.
>
>I am facing the exact same issue on my platform. The only solution I
>have found is to upgrade to opensips 1.6 and implement the presence on
>the proxy. AFAIK asterisk can't handle presence for devices not
>directly registered to it.
>
>BTW, the config migration from 1.2 to 1.6 was less complicated than I
>feared...
>
>Regards,
>- vma
>.
>
>
>
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Re: [OpenSIPS-Users] When do I need MediaProxy?

[Mark Sayer]

No, you probably don't need MediaProxy

Mark

At 12:13 p.m. 04/04/2010, you wrote:
>On Sat, Apr 3, 2010 at 3:05 PM, Brian Chamberlain  wrote:
> > Why are you splitting the registrations from Asterisk?
>
>I've had issues with Asterisk causing large groups of phones to go
>unreachable for a minute or so.
>This is with approximately 500-600 devices registered at one time on one box.
>Asterisk does not seem to be designed to be able to handle
>registrations for that many devices,
>especially when it is also under load from handling 50+ simultaneous calls.
>
>-- James
>
> >
> > On 3 Apr 2010, at 20:31, James Lamanna wrote:
> >
> >> Hi,
> >> I'm trying to use OpenSips as a registration server for Asterisk
> >> (assuming we can get presence working ok).
> >> Do I need to setup and use MediaProxy (or similar)? Or is the
> >> nathelper stuff good enough?
> >> I've made test calls from phones behind NAT to opensips to asterisk
> >> and I haven't experienced any
> >> one-way audio problems.
> >> Also, the phones are not allowed to reinvite, because I need to keep
> >> Asterisk in the media path.
> >>
> >> -- James
> >>
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Re: [OpenSIPS-Users] Hosting multiple Companies PBX on one OpenSIPS Server

[Mark Sayer]

The simple answer is yes, but not with OpenSIPS alone. You'll need
something like Asterisk or FreeSWITCH to handle the PBX functions.

Mark

On Thu, Aug 12, 2010 at 8:48 PM, Deon Vermeulen
 wrote:
> Good Day List
> Hope you well.
> I would like to find out if someone could assist me.
> I am in the process of setting up a SIP Server hosting a few companies PBX
> Functionality, but require some assistance/guidance.
> What I am looking for is:
> 1) Support for Multiple Domains - Each Company with their own domain and
> Extensions (Users).
> 2) Support for Multiple Servers - LDAP integration with each Companies AD
> Server. This should also be in a Secure way that each Company can only view,
> search within their own Domain. Domains should be transparent to each other.
> 3) Support for multiple PSTN/LCR connections with each company having their
> own auto attendant console, Dial-Plans, etc.
> Can this be done with OpenSIPS?
> Thanks
>
> Deon
>
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Re: [OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS

[Mark Sayer]

Those "virtual PBX" functions, like your present voicemail, cannot be
provided by OpenSIPS. They are Asterisk-style functions.

Mark

On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor  wrote:

> Hi Guys
>
> I've been using OpenSIPS now for about 9 month (after upgrading from
> OpenSER 1.2 used that for about 2 years) for my core SIP routing and
> billing.
>
> I'm now getting questions from customers about Virtual PBX functionality
> and I would like the opinion of the group about how well this could be
> done using OpenSIPS, Mediaproxy and maybe SEMS.
>
> My current core system has voicemail, call forwarding and T38 fax using
> sip forwards to asterisk, but as normal with Asterisk I do get
> occasional calls issues, mostly related to codec negotiation.
>
> I want to be able to have all the normal PBX functions like Auto
> attendant, Call forwarding on busy or absence, Call Park, Call pickup,
> Call transfer, Call waiting, Conference Call, Custom Greeting, Voice
> Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC
>
> So your comments requested.
>
> Thanks
> Mike
>
>
>
>
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Re: [OpenSIPS-Users] Paid Consultation Request

[Mark Sayer]

I'm afraid I have to echo Geoff's response to this. It's fascinating 
to see so many people telling him that this community isn't going to 
help him. It would seem that if you can't or are unwilling to help 
for whatever reason, then just press delete. His request, while 
perhaps a bit naive, was honest.

Mark


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Re: [OpenSIPS-Users] db mysql realtime asterisk

[Mark Sayer]

The simple answer would appear to be that your openser.subscriber 
table has not been created with a field named 'first_name'. Are you 
using mysql 5.x?

Mark

At 11:46 a.m. 12/04/2009, you wrote:
>Hi list , I am trying to integrate opensips with asterisk realtime,
>but when I want to create the asterisk charts in mysql it throws me an
>error:
>
>ERROR 1054 (42S22) at line 21: Unknown column 'first_name' in 'field list'
>
>this is the line 21
>
>CREATE VIEW vmusers AS
>
>
>realtime.sql
>
>use asterisk;
>
>CREATE TABLE `voicemessages` (
>   `id` int(11) NOT NULL auto_increment,
>   `msgnum` int(11) NOT NULL default '0',
>   `dir` varchar(80) default '',
>   `context` varchar(80) default '',
>   `macrocontext` varchar(80) default '',
>   `callerid` varchar(40) default '',
>   `origtime` varchar(40) default '',
>   `duration` varchar(20) default '',
>   `mailboxuser` varchar(80) default '',
>   `mailboxcontext` varchar(80) default '',
>   `recording` longblob,
>   PRIMARY KEY  (`id`),
>   KEY `dir` (`dir`)
>) ENGINE=InnoDB;
>
>CREATE VIEW vmusers AS
>SELECT id as uniqueid,
>   username as customer_id,
>   'netsoluciones' as context,
>   username as mailbox,
>   vmail_password as password,
>   CONCAT(first_name,' ',last_name) as fullname,
>   email_address as email,
>   NULL as pager,
>   datetime_created as stamp
>FROM opensips.subscriber;
>
>CREATE VIEW sipusers AS
>SELECT username as name,
>   username,
>   'friend' as type,
>   NULL as secret,
>   domain as host,
>   CONCAT(rpid, ' ','<',username,'>') as callerid,
>   'netsoluciones' as context,
>   username as mailbox,
>   'yes' as nat,
>   'yes' as qualify,
>   username as fromuser,
>   NULL as authuser,
>   domain as fromdomain,
>   NULL as insecure,
>   'no' as canreinvite,
>   NULL as disallow,
>   NULL as allow,
>   NULL as restrictcid,
>   domain as defaultip,
>   domain as ipaddr,
>   '5060' as port,
>NULL as regseconds
>FROM opensips.subscriber;
>
>
>
>I am following this tutorial this works me perfect with the version of
>openser 1.3.3
>
>http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3
>
>any idea?
>
>regards
>
>--
>rickygm
>
>http://gnuforever.homelinux.com
>
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Re: [OpenSIPS-Users] db mysql realtime asterisk

[Mark Sayer]

So, back to my original comment - The simple answer would appear to 
be that your openser.subscriber table has not been created with a 
field named 'first_name'.

Mark

At 04:03 a.m. 13/04/2009, you wrote:
>yes , I am using mysql-5.0.45-7.el5
>
>regardss
>
>2009/4/11 Mark Sayer :
> > The simple answer would appear to be that your openser.subscriber
> > table has not been created with a field named 'first_name'. Are you
> > using mysql 5.x?
> >
>
>--
>rickygm
>
>http://gnuforever.homelinux.com


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Re: [OpenSIPS-Users] Multi Tenant System

[Mark Sayer]

I suggest using the pieces as they work best. Let OpenSIPs handle the 
registration & NAT. Let Asterisk handle the media & connections to 
terminators or PSTN. The only issue is that Asterisk will only handle 
about 200 concurrent calls per box so a large installation might have 
a single OpenSIPs box and multiple Asterisk boxes. Relatively simple 
to setup and manage, stable, proven. Asterisk itself can be 
"partitioned" through careful construction of the extensions.conf 
file to do what you want.

Mark

At 07:38 p.m. 14/01/2009, you wrote:


>On 1/12/09, Brett Nemeroff 
><br...@nemeroff.com> wrote:
>Sure you can build a multi-tenant system with Opensips, asterisk, 
>and or freeswitch.
>
>
>You should understand that your question is kind of like walking 
>into a hardware store and asking, "Hey, can I build a house with this stuff?"
>
>
>Specifics on how to build these systems are all over the net. 
>However, there arn't a whole lot of good Multi-Tenant solutions out 
>there presumably because they are worth too much to give away.
>
>
>If you have a specific question, perhaps we can give you a more 
>specific answer? :)
>
>
>
>Hi Brett and all
>
>thanks for your suggestion
>
>Iam confused here some of the option how they going to work in real time
>
>lets take example
>
>1. User register with Opensips and calls Opensip users ( local)
>
>then they want to get another person in to conference, using PSTN or 
>SIP ( from asterisk)
>
>how this going to work ?
>
>2. Conference Bridge
>
>3. Multitenant
>
>X company having extention of 100 200 300
>Y company having extention of 100 200 300
>
>how do they call each other ?
>
>is there any mechanism ?
>
>I have manythings in mind to put here, let me do one at a time to 
>understand better.
>
>Ram
>
>
>
>
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Re: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity

[Mark Sayer]

We successfully run things a bit differently. Our in-bound from pstn 
goes to OpenSER (and may be translated to an Asterisk extension using 
dbalias) then to Asterisk. If the final destination is a UA then it 
goes back out thru OpenSER. Out-bound to pstn goes directly out from 
Asterisk. OpenSER handles all registration and NAT. Works well in 
production and config for both OpenSER and Asterisk is relatively 
simple and easy to maintain.

Mark

At 08:19 a.m. 21/01/2009, you wrote:
>Ok, just hacking around I succeeded with the following code snippet 
>to rewrite the domain part:
>
>avp_db_query("SELECT domain FROM uri WHERE uri_user='$rU'", 
>"$avp(i:678)");avp_pushto("$ru/domain", "$avp(i:678)");
>
>And the IP address of the SER server in the INVITE gets replaced by 
>the domain and everything works well. But I can't imagine that this 
>is what people do in a situation like mine...
>
>
>-Original Message-
>From: users-boun...@lists.opensips.org 
>[mailto:users-boun...@lists.opensips.org] On Behalf Of Robert Borz
>Sent: Tuesday, January 20, 2009 7:23 PM
>To: users@lists.opensips.org
>Subject: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity
>
>Hi,
>
>I'm currently setting up OpenSIPS/OpenSER with Asterisk as a PSTN 
>gateway. As starting point I'm using the configuration wizard from 
>sip:wise [1].
>
>After some modifications to the module configurations it works fine, 
>without any (obvious) failures.
>
>Both, SER and Asterisk has public IP addresses on their its 
>interfaces and SER is setup for multiple domains. Multi-Domain 
>support also works fine for calls handled within SER, and also does 
>PSTN termination when forwarding non-local calls to the right domain 
>(sip proxy) and non-local-pstn-calls to the Asterisk machine.
>
>The problem occurs by receiving calls from the PSTN, from the 
>Asterisk machine:
>
>- Incoming Call from the PSTN to 555123456 gets forwarded from the Asterisk
>   machine to the SER.
>
>- The INVITE is from sip:555123456@ to
>   sip:555123456@
>
>- And the uri table in the database backend looks like this:
>
>id | username  |   domain   |   uri_user   |   last_modified
>   +---++--+
> 1 | user1 | domain1.de | 555123456| 2008-12-26 15:32:18.761647
> 4 | user10001 | domain1.de | 555123457| 2008-12-29 20:47:11.740234
> 5 | user10002 | domain2.de | 555123458| 2008-12-30 10:46:36.455437
>
>Well, for the uri_db module the use_domain parameter is disabled 
>(set to 0), so the is_uri_local() function consideres the call as a 
>local call (that's right). But now trying to lookup the location 
>table the user is not seen as online. The user for 555123456 is 
>registered as 555123...@domain1.de.
>
>My uri_user's are unique and are equal to the PSTN numbers 
>associated with the users (including the area prefix). Users from 
>different domains hosted by the same SER gets PSTN calls from the 
>same Asterisk server.
>
>What's the usual way to lookup the domain part and rewrite the 
>INVITE so SER can relay the call to the correct domain and the user 
>isn't seen offline/not registered by the lookup("location") method?
>
>I can't imagine to be the first facing this problem. ;)
>
>Any help or suggestions are appreciated...
>
>
>
>Bye,
>Robert
>
>
>[1] http://www.sipwise.com/index.php/products?start=3
>
>
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Re: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity

[Mark Sayer]

It was one way to make sure that we could accurately account for call 
times and provide all calls with PBX capability. I'm sure it's not 
the only way but it works well for us.

Mark

At 09:16 a.m. 21/01/2009, you wrote:
>Hmm, sounds interesting. If I understood it correctly, it looks 
>something like this for calls from the PSTN:
>
>---inbound pstn-->[OpenSER]--->[Asterisk]
> |
>[UA]<--[OpenSER]<---+
>
>Why did you choose _not_ to handle such calls directly within SER 
>without involving the Asterisk machine, here?
>
>
>Now I set the "use_domain" parameter to 0 for all SER modules used, 
>and everything works fine. But I'm not sure if this is really what I want...
>
>
>Robert
>
>-Original Message-
>From: users-boun...@lists.opensips.org 
>[mailto:users-boun...@lists.opensips.org] On Behalf Of Mark Sayer
>Sent: Tuesday, January 20, 2009 9:05 PM
>To: users@lists.opensips.org
>Subject: Re: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity
>
>We successfully run things a bit differently. Our in-bound from pstn
>goes to OpenSER (and may be translated to an Asterisk extension using
>dbalias) then to Asterisk. If the final destination is a UA then it
>goes back out thru OpenSER. Out-bound to pstn goes directly out from
>Asterisk. OpenSER handles all registration and NAT. Works well in
>production and config for both OpenSER and Asterisk is relatively
>simple and easy to maintain.
>
>Mark
>
>At 08:19 a.m. 21/01/2009, you wrote:
> >Ok, just hacking around I succeeded with the following code snippet
> >to rewrite the domain part:
> >
> >avp_db_query("SELECT domain FROM uri WHERE uri_user='$rU'",
> >"$avp(i:678)");avp_pushto("$ru/domain", "$avp(i:678)");
> >
> >And the IP address of the SER server in the INVITE gets replaced by
> >the domain and everything works well. But I can't imagine that this
> >is what people do in a situation like mine...
> >
> >
> >-Original Message-
> >From: users-boun...@lists.opensips.org
> >[mailto:users-boun...@lists.opensips.org] On Behalf Of Robert Borz
> >Sent: Tuesday, January 20, 2009 7:23 PM
> >To: users@lists.opensips.org
> >Subject: [OpenSIPS-Users] Multi-Domain Setup with PSTN Connectivity
> >
> >Hi,
> >
> >I'm currently setting up OpenSIPS/OpenSER with Asterisk as a PSTN
> >gateway. As starting point I'm using the configuration wizard from
> >sip:wise [1].
> >
> >After some modifications to the module configurations it works fine,
> >without any (obvious) failures.
> >
> >Both, SER and Asterisk has public IP addresses on their its
> >interfaces and SER is setup for multiple domains. Multi-Domain
> >support also works fine for calls handled within SER, and also does
> >PSTN termination when forwarding non-local calls to the right domain
> >(sip proxy) and non-local-pstn-calls to the Asterisk machine.
> >
> >The problem occurs by receiving calls from the PSTN, from the
> >Asterisk machine:
> >
> >- Incoming Call from the PSTN to 555123456 gets forwarded from the Asterisk
> >   machine to the SER.
> >
> >- The INVITE is from sip:555123456@ to
> >   sip:555123456@
> >
> >- And the uri table in the database backend looks like this:
> >
> >id | username  |   domain   |   uri_user   |   last_modified
> >   +---++--+
> > 1 | user1 | domain1.de | 555123456| 2008-12-26 15:32:18.761647
> > 4 | user10001 | domain1.de | 555123457| 2008-12-29 20:47:11.740234
> > 5 | user10002 | domain2.de | 555123458| 2008-12-30 10:46:36.455437
> >
> >Well, for the uri_db module the use_domain parameter is disabled
> >(set to 0), so the is_uri_local() function consideres the call as a
> >local call (that's right). But now trying to lookup the location
> >table the user is not seen as online. The user for 555123456 is
> >registered as 555123...@domain1.de.
> >
> >My uri_user's are unique and are equal to the PSTN numbers
> >associated with the users (including the area prefix). Users from
> >different domains hosted by the same SER gets PSTN calls from the
> >same Asterisk server.
> >
> >What's the usual way to lookup the domain part and rewrite the
> >INVITE so SER can relay the call to the correct domain and the user
> >isn't seen offline/not registered by the lookup("location") method?
>