[Freeswitch-users] recommended settings for max-proceeding param
Hi Everybody, if it is not too much of a trouble can somebody point to a recommended value for max-proceeding in sofia.conf.xml ? If there is no recommended value what should be taken under consideration in order to determine one. I dug into the archives and discovered a thread called "Freeswitch freezes under increased call load" and there together with session per sec and max allowed sessions was recommended max-proceeding under sofia.conf.xml to be changed. I've just installed 1.0.3 version and checked the sofia.conf.xml file . I was not able to find a default setting for max-proceeding , when i added it i started getting cored umps. Regards Chav ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] js session.streamFile() interrupt slowness
Try this one
http://pastebin.freeswitch.org/7391
I just tested this on latest trunk and it stopped instantly.
On Sat, Feb 21, 2009 at 1:35 PM, Stephen Crosby wrote:
> Verry sorry for the list spam, this is the link to the corrected script:
> http://pastebin.freeswitch.org/7389
>
> --Stephen
>
> On Sat, Feb 21, 2009 at 11:33 AM, Stephen Crosby
> wrote:
> > There was a small error in that last script I sent, please test using
> > this version: http://pastebin.freeswitch.org/7388
> >
> > Thanks.
> >
> > On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby
> wrote:
> >> Sure, I've stripped down the script somewhat to something smaller that
> >> still produces this effect and you can see it at:
> >> http://pastebin.freeswitch.org/7388
> >>
> >> The file sound 'VR1' continues to play for a short time after I
> >> interrupt it with a DTMF event. It does interrupt, but it sounds a
> >> little awkward because of the delay. I was probably wrong in my
> >> estimate of the delay which seems to be about a full second, not two
> >> or three. I'm hoping I can adjust it somehow to feel more immediate.
> >> Any ideas?
> >>
> >> --Stephen
> >>
> >> On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins
> wrote:
> >>> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby
> wrote:
> I have a few scripts that use the javascript
> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function
> that returns false to interrupt the streaming file. There is a short
> delay between the time when I press a key and the time the file stops
> playing. Is there anything I can adjust that would affect that? It's
> only maybe 2-3 seconds, but it "feels" too long to me.
>
> --Stephen
> >>>
> >>> Could you pastebin your entire script plus the relevant dialplan
> >>> entry? Also, could you tell us which operating system and FS revision?
> >>> -MC
> >>>
> >>> ___
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users@lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >
>
> ___
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>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR
Rupa Schomaker (lists) wrote: >> do i have to upgrade to the latest trunk in order to pass channel >> variables to mod_lcr? >> Currently the version used is 1.0.trunk (12134M) >> Regards >> Chav >> >> > > You need at least 12204. > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Thank you very much. I'll try it and will keep you postged. Chav ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR
> do i have to upgrade to the latest trunk in order to pass channel > variables to mod_lcr? > Currently the version used is 1.0.trunk (12134M) > Regards > Chav > You need at least 12204. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR
Rupa Schomaker (lists) wrote: >> "default" is a reserved profile name -- I should probably prevent that >> from loading. >> > > correction, default is not reserved... > > >> Regarding passing the callerid to the custom sql, let me see what I can >> come up with... >> > > You can now specify channel variables in your custom sql. So, you > should be able to pass CID or a subset of CID to your custom sql. I've > updated the wiki with info on this. > > beware: channel vars only work when called in the context of a session. > Using a profile that uses a custom sql with channel variables from the > commandline will result in an error. > > -Rupa > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > do i have to upgrade to the latest trunk in order to pass channel variables to mod_lcr? Currently the version used is 1.0.trunk (12134M) Regards Chav ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] js session.streamFile() interrupt slowness
Verry sorry for the list spam, this is the link to the corrected script:
http://pastebin.freeswitch.org/7389
--Stephen
On Sat, Feb 21, 2009 at 11:33 AM, Stephen Crosby wrote:
> There was a small error in that last script I sent, please test using
> this version: http://pastebin.freeswitch.org/7388
>
> Thanks.
>
> On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby wrote:
>> Sure, I've stripped down the script somewhat to something smaller that
>> still produces this effect and you can see it at:
>> http://pastebin.freeswitch.org/7388
>>
>> The file sound 'VR1' continues to play for a short time after I
>> interrupt it with a DTMF event. It does interrupt, but it sounds a
>> little awkward because of the delay. I was probably wrong in my
>> estimate of the delay which seems to be about a full second, not two
>> or three. I'm hoping I can adjust it somehow to feel more immediate.
>> Any ideas?
>>
>> --Stephen
>>
>> On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote:
>>> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby
>>> wrote:
I have a few scripts that use the javascript
session.streamFile('somefile.wav', onDtmf); where onDtmf is a function
that returns false to interrupt the streaming file. There is a short
delay between the time when I press a key and the time the file stops
playing. Is there anything I can adjust that would affect that? It's
only maybe 2-3 seconds, but it "feels" too long to me.
--Stephen
>>>
>>> Could you pastebin your entire script plus the relevant dialplan
>>> entry? Also, could you tell us which operating system and FS revision?
>>> -MC
>>>
>>> ___
>>> Freeswitch-users mailing list
>>> Freeswitch-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
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Re: [Freeswitch-users] js session.streamFile() interrupt slowness
There was a small error in that last script I sent, please test using
this version: http://pastebin.freeswitch.org/7388
Thanks.
On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby wrote:
> Sure, I've stripped down the script somewhat to something smaller that
> still produces this effect and you can see it at:
> http://pastebin.freeswitch.org/7388
>
> The file sound 'VR1' continues to play for a short time after I
> interrupt it with a DTMF event. It does interrupt, but it sounds a
> little awkward because of the delay. I was probably wrong in my
> estimate of the delay which seems to be about a full second, not two
> or three. I'm hoping I can adjust it somehow to feel more immediate.
> Any ideas?
>
> --Stephen
>
> On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote:
>> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote:
>>> I have a few scripts that use the javascript
>>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function
>>> that returns false to interrupt the streaming file. There is a short
>>> delay between the time when I press a key and the time the file stops
>>> playing. Is there anything I can adjust that would affect that? It's
>>> only maybe 2-3 seconds, but it "feels" too long to me.
>>>
>>> --Stephen
>>
>> Could you pastebin your entire script plus the relevant dialplan
>> entry? Also, could you tell us which operating system and FS revision?
>> -MC
>>
>> ___
>> Freeswitch-users mailing list
>> Freeswitch-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
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Re: [Freeswitch-users] js session.streamFile() interrupt slowness
Sure, I've stripped down the script somewhat to something smaller that
still produces this effect and you can see it at:
http://pastebin.freeswitch.org/7388
The file sound 'VR1' continues to play for a short time after I
interrupt it with a DTMF event. It does interrupt, but it sounds a
little awkward because of the delay. I was probably wrong in my
estimate of the delay which seems to be about a full second, not two
or three. I'm hoping I can adjust it somehow to feel more immediate.
Any ideas?
--Stephen
On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote:
> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote:
>> I have a few scripts that use the javascript
>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function
>> that returns false to interrupt the streaming file. There is a short
>> delay between the time when I press a key and the time the file stops
>> playing. Is there anything I can adjust that would affect that? It's
>> only maybe 2-3 seconds, but it "feels" too long to me.
>>
>> --Stephen
>
> Could you pastebin your entire script plus the relevant dialplan
> entry? Also, could you tell us which operating system and FS revision?
> -MC
>
> ___
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>
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Re: [Freeswitch-users] Deployment information and use cases
We're still in the construction and design phase, but my company is building a multi-tenant Freeswitch based PBX for a Research Park in South Alabama. We expect to handle about 120 concurrent calls, and 6-700 registered UAs. The system will be based on commodity house-built SuperMicro servers, with mod_xml_curl handling all configuration. We will have PRIs for fax and 911, and SIP trunks to upstream ITSPs for most call volume. -anm On Tue, Feb 17, 2009 at 6:20 PM, Raul Fragoso wrote: > Hello FreeSWITCHERS, > > My company is currently creating a suite of applications which uses > FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > prospect to have our first customer installation - a governmental > department. That is a tender to have an IP-PBX installation to connect > their four office branches, each one with about 300 users - which I am > sure FreeSWITCH is able to handle. Since this is an official tender, > it's part of their protocol to ask about real sites using the product. > > Having said that, would you mind sharing some information about your > experience with FreeSWITCH deployments ? > > No need to give many details, but a short summary with company name (if > possible), when it was deployed, server equipment, number of users, > number of concurrent calls, what kind of functions and services are used > and overall capacity of the system. > > I would really appreciate if you can share that information. And if you > guys agree (and explicitly manifest your agreement), I can compile the > information in the FreeSWITCH wiki under a "Use Cases" page so it can > serve as a common reference as well. > > Kind regards, > > Raul Fragoso > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ESL
I have ported all the Perl samples to python, and they appear to be working fine. They are available in svn rev 12210 and > -anm On Thu, Feb 19, 2009 at 12:44 PM, Brian West wrote: > FreeSWITCHers, >Not sure anyone is paying attention or not but Anthony wrapped the > ESL library up so you can use it from Perl, Python, Lua, Ruby and > PHP. What I'm requesting from our community is to help flex it out.. > write examples and populate the Wiki page with information about it. > > http://wiki.freeswitch.org/wiki/Esl > > Collins and I are going to start filling in the page but I want > someone thats good with Ruby, Python, PHP to help in those areas.. > kick in some lua and perl if you like. > > It works with OES and IES... (Outbound Event Socket and Inbound Event > Socket) Not sure those names are official but we have been calling > them that ;) > > Thanks, > Brian West > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Suggestion for xml_curl performance
Is this a good application for the new ESL (Event Socket Library)interface? -anm On Sat, Feb 21, 2009 at 9:11 AM, Shannon wrote: > I'd recommend having a look at fastcgi as well. > > On 2/21/09, shehzad p wrote: >> >> Hi Brian, >> >> My setup is to use FS as basic calls routing. >> 1. Calls are coming to FS from more than one customer Gateways, and I need >> to authenticate them and check for enough balance based on database, >>[Caller Gateways] ===> [FreeSWITCH] ===> >> [Provider Gateways] >> 2. After knowing that Caller Gateways is valid, then based on dialed number >> it search in database for Provider Gateway and bridge the call there. >> 3. After call finish CDR is inserted back into database. >> >> My old setup was using Javascript which works fine in traffic of 10 to 20 >> calls, but then increase of traffic causes many problems. >> >> Now I eliminate use of any of the script (javascript or any other) for call >> routing, and route calls directly from dialplan, >> So I have setup test system using xml-curl to generate dynamic dialplan, >> I used below xml_curl PHP example as reference: >> http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example >> For CDR processing I used xml_cdr, with help of the example in FS source >> :scripts/contrib/trixter/xml-cdr. >> >> >> Waiting for any better suggestions, any comments... >> >> thanks >> msp. >> >> Brian West-3 wrote: >>> >>> it all depends on what you're doing.. can you elaborate? >>> >>> /b >>> >>> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >>> Recently I faced some performance bottleneck by using Javascript. >>> >>> >>> ___ >>> Freeswitch-users mailing list >>> Freeswitch-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > Shannon > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Suggestion for xml_curl performance
I'd recommend having a look at fastcgi as well. On 2/21/09, shehzad p wrote: > > Hi Brian, > > My setup is to use FS as basic calls routing. > 1. Calls are coming to FS from more than one customer Gateways, and I need > to authenticate them and check for enough balance based on database, >[Caller Gateways] ===> [FreeSWITCH] ===> > [Provider Gateways] > 2. After knowing that Caller Gateways is valid, then based on dialed number > it search in database for Provider Gateway and bridge the call there. > 3. After call finish CDR is inserted back into database. > > My old setup was using Javascript which works fine in traffic of 10 to 20 > calls, but then increase of traffic causes many problems. > > Now I eliminate use of any of the script (javascript or any other) for call > routing, and route calls directly from dialplan, > So I have setup test system using xml-curl to generate dynamic dialplan, > I used below xml_curl PHP example as reference: > http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example > For CDR processing I used xml_cdr, with help of the example in FS source > :scripts/contrib/trixter/xml-cdr. > > > Waiting for any better suggestions, any comments... > > thanks > msp. > > Brian West-3 wrote: >> >> it all depends on what you're doing.. can you elaborate? >> >> /b >> >> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >> >>> Recently I faced some performance bottleneck by using Javascript. >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
Hi Andrew,
Thanks for your help so far, I hope you can help me a bit further as I
don't get any reply from the FS erlang node, or so it seems.. Here is
what I've done:
- The erlang_event.conf.xml is unchanged:
- mod_erlang_event is not loaded in FS.
- First I start "epmd -d -d"
epmd: Sat Feb 21 13:12:56 2009: epmd running - daemon = 0
epmd: Sat Feb 21 13:12:56 2009: try to initiate listening port 4369
epmd: Sat Feb 21 13:12:56 2009: starting
epmd: Sat Feb 21 13:12:56 2009: entering the main select() loop
- After that I "load mod_erlang_event" in FS:
2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1324
mod_erlang_event_load() sections 16
2009-02-21 13:13:36 [CONSOLE] switch_loadable_module.c:858
switch_loadable_module_load_file() Successfully Loaded
[mod_erlang_event]
2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:240
switch_loadable_module_process() Adding Application 'erlang'
2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:260
switch_loadable_module_process() Adding API Function 'erlang'
2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1401
mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031
2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1426
mod_erlang_event_runtime() Connected and published erlang cnode at
freeswi...@erlyfs
- For which epmd gives the following output:
epmd: Sat Feb 21 13:13:36 2009: opening connection on file descriptor 4
epmd: Sat Feb 21 13:13:36 2009: got 25 bytes
* 00 17 78 1f 5f 68 00 00 05 00 01 00 0a 66 72 65
|..x._h...fre|
* 0010 65 73 77 69 74 63 68 00 00 |
eswitch..|
epmd: Sat Feb 21 13:13:36 2009: ** got ALIVE2_REQ
epmd: Sat Feb 21 13:13:36 2009: registering 'freeswitch:1', port 8031
epmd: Sat Feb 21 13:13:36 2009: type 104 proto 0 highvsn 5 lowvsn 1
epmd: Sat Feb 21 13:13:36 2009: got 4 bytes
* 79 00 00 01 |y...|
epmd: Sat Feb 21 13:13:36 2009: ** sent ALIVE2_RESP for "freeswitch"
- Then I start an erl shell on that same machine with "erl -sname ldr -
setcookie ClueCon". Output of epmd:
epmd: Sat Feb 21 13:16:24 2009: opening connection on file descriptor 5
epmd: Sat Feb 21 13:16:24 2009: got 18 bytes
* 00 10 78 8e 2c 4d 00 00 05 00 05 00 03 6c 64 72
|..x.,M...ldr|
* 0010 00 00 |..|
epmd: Sat Feb 21 13:16:24 2009: ** got ALIVE2_REQ
epmd: Sat Feb 21 13:16:24 2009: registering 'ldr:1', port 36396
epmd: Sat Feb 21 13:16:24 2009: type 77 proto 0 highvsn 5 lowvsn 5
epmd: Sat Feb 21 13:16:24 2009: got 4 bytes
* 79 00 00 01 |y...|
epmd: Sat Feb 21 13:16:24 2009: ** sent ALIVE2_RESP for "ldr"
As far as I understand the freeswi...@erlyfs node cannot be seen with
nodes() ? So does that mean that I also cannot net_adm:ping() it ?
Anyway, I tried sending some tuples as is shown on the wiki, but I get
no reply:
(l...@erlyfs)1> {foo, freeswi...@erlyfs} ! {api, status, ""}, receive X
-> X after 1000 -> timeout end.
timeout
(l...@erlyfs)2>
- Epmd gives some logs:
epmd: Sat Feb 21 13:19:09 2009: opening connection on file descriptor 6
epmd: Sat Feb 21 13:19:09 2009: got 13 bytes
* 00 0b 7a 66 72 65 65 73 77 69 74 63 68
|..zfreeswitch|
epmd: Sat Feb 21 13:19:09 2009: ** got PORT2_REQ
epmd: Sat Feb 21 13:19:09 2009: got 23 bytes
* 77 00 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 65 |
w.._h...free|
* 0010 73 77 69 74 63 68 00 |
switch.|
epmd: Sat Feb 21 13:19:09 2009: ** sent PORT2_RESP (ok) for "freeswitch"
epmd: Sat Feb 21 13:19:09 2009: closing connection on file descriptor 6
- And in tcpdump on lo, I see that epmd is contacted after which some
traffic was sent to FS:
13:19:09.535293 IP 172.31.0.13.34678 > 172.31.0.13.4369: S
2875169966:2875169966(0) win 32792
...
13:19:09.536834 IP 172.31.0.13.4369 > 172.31.0.13.34678: . ack 15 win
512
13:19:09.536923 IP 172.31.0.13.47054 > 172.31.0.13.8031: S
2868322908:2868322908(0) win 32792
13:19:09.536935 IP 172.31.0.13.8031 > 172.31.0.13.47054: R 0:0(0) ack
2868322909 win 0
Shouldn't FS then send a message back to the process in my erl shell ?
I tried logging all events in fs_cli, by entering "/event plain all",
but I see no events at all coming from erlang, just some heartbeats..
Also, I recompiled the module with EI_DEBUG defined as suggested on
the wiki. Still I don't see anything in the CLI when set to debug
logging.
Thanks again,
Leon
On Feb 20, 2009, at 8:08 PM, Andrew Thompson wrote:
> On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote:
>> Hi,
>>
>> I wanted to try out the mod_erlang_event module. I have Erlang R12B5
>> compiled and it's in the same location as the Makefile specifies (/
>> usr/
>> local/lib/erlang/...), but running make in the src/mod/
>> event_handlers/
>> mod_erla
Re: [Freeswitch-users] Deployment information and use cases
Ben, thank you for your story. I would very much like to add this to the wiki if you don't mind and everyone else agrees. What do you think guys? Use cases are _ALWAYS_ a good thing for new users. Mesquita On Feb 19, 2009, at 6:23 PM, Ben Holtsclaw wrote: Raul, I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what you describe. When I was trying to procure funds for a FreeSWITCH solution, I looked for the same information you're after, but came up with little. I'll briefly describe what we're trying to accomplish, and the tools I'm using to do it. This is probably more information than what you are looking for, but maybe it will also benefit someone else. We had several schools with aging or dying PBX's or KSU's. Each site had something different system, and was supported by a different VAR. Of course, the VAR's charged some outlandish fee to make onsite repair visits. Some number of Centrex lines supplied each school's dial tone. All in all, we had a very outdated and financially draining mess. Our district's long term goal had been to move to a more unified phone system. That made sense for many reasons, the chief of which was cost. We already had a strong fiber WAN in place. Why not use that for trunking and eliminate the monthly cost of the Centrex lines? That's the path we started down. Being a public entity, we had to be sure to explore all possible avenues. We looked at everything from traditional PBX's with IP add- on modules for trunking to a full blown Cisco CallManager solution. With third party proprietary systems, we were just never able to find the sweet spot between required feature set and cost. Would Cisco have been a workable solution? Absolutely. Could our small, rural, K12 public school district afford that? Not in a million years. I looked at several software packages -- some open source, some not -- but always came back to FreeSWITCH. The scalability and active development community were major factors for us. Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored disks set up with enough space to accommodate users' voicemail. Each server will average only about 60 voicemail boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We have always been a Novell shop, so SLES is naturally our Linux distribution of choice. We chose to go with server hardware at each site so that in the event of a WAN outage, we would still at least have intra- building and emergency communication (see below). Telephony Hardware. Each of our servers includes Sangoma hardware. We actually looked at doing IP trunking to a carrier from our network core, but decided to use telco provided PRI's instead. Presently, we have two PRI's that connect to a FreeSWITCH server at the center of our network via a Sangoma A102 dual port telephony card. All calls to and from the PSTN traverse this primary server. Servers at each remote site include one of Sangoma's A200 analog cards. Emergency calls to 911 route out over this analog card through one of at least two POTS lines that remain connected at each site. Not only does this provide some redundancy in the event of a WAN outage, but it ensures proper caller location is delivered to the 911 dispatcher. Granted, there are some other solutions for the latter, but this seemed to be the most cost effective solution for us. Telephone Desksets. We chose to go with Aastra for the telephones. The standard phone that we will place in each classroom and office is the 9143i. This is an attractive phone with an adequate feature set at a price we can afford. The person that is primarily responsible for answering the phone at each site will have an Aastra 57i and some number of 560M expansion modules. We have purchased roughly 300 Aastra desksets. Logical Layout. As new sites come online, their primary phone number is being moved from the Centrex to our PRI group. All inbound calls hit our primary server, and then FreeSWITCH bridges to the appropriate secondary server based on the DID it received. On the reverse, each servers dial plan is set up to route outbound calls (save 911) to the primary server where FreeSWITCH bridges with Openzap. Site to site calls, accomplished via four digit dialing, do not hit the primary server. Outbound calls to the PSTN deliver the site's DID as the calling number. In other words, if a user from site two calls my cell phone, I see site two's published telephone number on my caller ID. Our dial plans are set up so that receptionists at each site still answer all outside calls. If not answered, the call fails over to an IVR. Should we ever decide to do so, we are now perfectly positioned to have all inbound calls to the dis
