Re: [Freeswitch-users] MFC-R2 support for FreeSWITCH
Thanks a lot, moy, this is great. I'll check to see if there's somewhere I can test it. On Sun, Aug 23, 2009 at 2:40 AM, Diego Viola wrote: > Nice work, keep up the great work :). > > On Fri, Aug 21, 2009 at 6:29 PM, Moises Silva > wrote: > > So, I finally took some days to put up OpenR2 working with OpenZAP, which > > means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has > > support for. Including Mexico, Brazil, Argentina and others. The stack > has > > been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers > most > > countries that users may be interested in, support for new variants will > be > > added on-demand only (in any case users can always tweak the advanced > > configuration file to create their own variants as a last resort). > > I created a wiki page to illustrate the basic > > setup: http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 > > Now is time for testing. I just did minimal testing on my development > > environment, no serious testing, and I know that some stuff is not > working > > at this point (I had some issues with variable length DNIS and ANI) which > > should be fixed soon. > > If anyone around happens to have an R2 link and wants to test R2 support > in > > OpenZAP, I can give them a hand with the configuration and any issues you > > may find. You can find me on IRC at #freeswitch, #freeswitch-dev and > > #openzap as moy. > > > > -- > > Moises Silva > > Software Developer > > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 > > Canada > > t. 1 905 474 1990 x 128 | e. m...@sangoma.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 > > > > > > ___ > 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 > -- Arnaldo M Pereira ___ 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] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
On Tue, Apr 21, 2009 at 5:47 PM, Mathieu Rene wrote: > You can use ESL (Event Socket Library) to talk to freeswitch and do > the same thing as fastagi. > > Check out http://wiki.freeswitch.org/wiki/Esl (it builds a native > python module too!) There's also the eventsocket protocol for twisted, which allows one to create inbound and outbound socket apps within the same code: http://code.google.com/p/eventsocket > > > Mathieu > > On 21-Apr-09, at 3:13 PM, technologyinspired wrote: > > > Hello users, > > > > I am new to FreeSwitch. I have used Asterisk a lot but now want to > > use FreeSwitch. Presently I write complex Asterisk applications > > using Fast AGI in Python. How could I achieve the same from > > FreeSwitch? The main reason is that the server has heavy call load > > and I want to shift that load on two systems one the FreeSwitch and > > another the application server. > > > > Could you also give a pointer to such an example or documentation > > where it says how to run FreeSwitch applications in client server > > mode (Fast AGI mode in Asterisk). Is there any good documentation on > > mod_python + Django support in FreeSwitch? > > > > Thanks, > > > > Regards, > > Vin > > ___ > > 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 > -- Arnaldo M Pereira ___ 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] echo cancellation on PRI cards
Sharing my humble experience: in Brazil we usually need echo cancellation to have reliable DTMF detection _and_ voice quality over E1 lines (be it on MFC/R2 - r2d - or ISDN PRI lines), either for sip/tdm gateway devices or IVR applications. Usually there's no need for echo cancellation on links from some Telcos, in some specific places. But we need it in the majority of cases, even when my box is just a gateway between legacy pbxes. This represents just a subset of the available E1s in the world and it's just a practical experience, but it's a fact for me. If I don't have a card with echo cancellation, I don't offer reliability to my customer; I've done that in the past and didn't work out. I'm not theoretically discussing anything, just sharing what I've been through in the last 4 or 5 years. 2009/3/17 David Knell > Steve Underwood wrote: > > David Knell wrote: > > > Steve Underwood wrote: > > > [whopping big snip] > > > > The first bit of that's a tad patronising, isn't it, > > > > You are the one who started out being offensive. > > > > I'm sorry if you find disagreement offensive; you might not wish to > read beyond this > point if so. > > > and, in the case of the decade-old Aculab > cards which which I'm most familiar, is also untrue. > > > > I can't find too much about the old cards on the web now, but I found > http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html > which is pretty much a copy and paste from the old Dialogic web pages, > and you'll see it says "Cut through : Local echo cancellation permits > 100% detection with a >4.5 dB return loss line". The Aculabs did the > same thing for sure. They just couldn't work without cancellation. There > were some very early Dialogic cards, using DTMF receiver chips and OKI > ADPCM chips, and had no general purpose DSPs. They performed really > badly because of the lack of cancellation, and were quickly replaced > with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms > into a Motorola 56k DSP chips. > > > > The same document, under the bit which you've quoted, says: > "(E-1) Digital trunks use separate transmit and receive paths to network. > Performance dependent on far end handset's match to local analog loop." > - i.e. the card does no echo cancellation. > > > Your messages are starting to looked deranged. Why would they only apply > echo cancellation to T1s? Its a bizarre idea, and you must realise its > wrong. Are you so desperate to support a wrong answer you'll clutch at > straws? :-\ > > > More insults. Answer me this: if there were echo cancellation in use, why > would > DTMF detection performance depend on the far-end handset's match to the > loop? > > And the follow-up question (which you've already pretty much asked) - if > the > card doesn't echo cancel for E1s, why would it for T1s? > > As an aside, I'm not convinced that the document's not talking about return > loss > on the T1 line itself, the implication being that the T1 is being carried > on a single > pair, which makes the first sentence about E1s make a bit more sense. But > that's > just a guess. > > Aculab didn't even offer echo cancellation on Prosody for years and, > when they did, it > consumed prodigious amounts of DSP. Nonetheless, the DTMF detection > worked > perfectly well, even across 120 channels per 40MHz SHARC - there's > just no way > that those DSPs had enough horsepower to do echo cancellation across > that manychannels. > > > This page > http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html > seems to support what you say. It also implies DTMF detection sucks > unless you echo cancel. The statement "If the outgoing signal is a tone > of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" > is telling you to keep your outgoing signal in the same frequency range > as dial-tone where the dial-tone filter on the DTMF receiver will > obviate the need for an echo canceller. They are freely admitting > exactly what I have said. If you want a normal IVR with cut-through to > work you better turn that echo canceller on. > > My only experience with Aculab was fitting a box designed by other > people into a system. That one definitely echo cancelled, as it worked > as well as the Dialogic based boxes we developed ourselves. > > > That only holds true if your premise - that you need echo cancellation for > good > DTMF detection - is correct, which I don't believe it is. > > An Asterisk box with an el-cheapo quad E1 card in that I use for > TDM-SIP gatewaying > detects DTMF perfectly well with no echo cancellation. > > > You must have very low standards for "works well". > > > Nothing like a good old ad hominem attack. Beats reasoned argument any > day. > > You just don't need echo cancellation to achieve perfectly acceptable > DTMF detection. > > > Well, not if you expect people to wait for silence before entering DTMF, > but who would to
Re: [Freeswitch-users] Please end the torment
On Thu, Mar 5, 2009 at 2:37 PM, Brian West wrote: > Well you can always shoot an email to the mailing list... or wait for > someone that can answer it.. which reminds me we really need people to > volunteer to help out on the IRC channel so we have skilled people > able to answer questions 24/7, I personally cover 12-16 hours a day > most of the time. Even weekends! I was even up at 4am today and > answered a few. I'm usually on the channel starting 9am (GMT -3). Can't answer every question like you guys, but I can surely help. > > > /b > > On Mar 5, 2009, at 11:27 AM, Ben Holtsclaw wrote: > > > The problem with take-a-number is what if I'm not there when someone > > can answer it? > > > ___ > 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 > -- Arnaldo M Pereira ___ 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] Please end the torment
Besides #freeswitch-social and #openzap, there are other channels related to
the project. List:
http://wiki.freeswitch.org/wiki/Main_Page#Community_and_Support
On Thu, Mar 5, 2009 at 2:19 PM, Mathieu Rene wrote:
> The bot even takes people's question and keeps them in list so when someone
> that knows shows up he can answer it, thats the ~take-a-number option...
> On 5-Mar-09, at 12:16 PM, Brian West wrote:
>
> I have been trying to push all the social talk into #freeswitch-social to
> keep #freeswitch on topic.. sometimes after hours in the US it gets a bit
> off topic. I'm usually alive in the channel till around 11PM+ CST most
> days. I take questions and answer questions at all hours if I'm awake... I
> too am guilty of going off topic.
> /b
>
>
> On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote:
>
> I agree with Harry. I do not like the mailing list. Those that do like the
> mailing list always advocate Nabble. For those that advocate that solution,
> do you even realize that you can't post on Nabble unless you *are* subscribed
> to the mailing list? I am also not a fan of IRC. Before I came upon
> FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client
> at the same time I uninstalled my NNTP reader. Most of the time, I actually
> find it difficult to obtain support in the #freeswitch channel. Once you ask
> the question, if somebody doesn't happen to be there that knows the answer,
> then you're screwed. How many times have I asked a question only to wait 30
> seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the
> question again? I *have* found the conversation in #openzap to be much
> more focused. Thank goodness I'm using that module! In that channel, I never
> see conversations about cd burners, somebody's girlfriend in South America,
> or off color jokes about someone's sexual proclivity. And because I know
> I'll get flamed for saying that, just look at this:
>
> [23:10] <{tasker}> me, too, but i'm a different animal
> [23:10] <{tasker}> in NY and in Miami i went nutz
> [23:10] lol
> [23:10] * jefferai is now known as lollerai
> [23:10] yeah i love her
> [23:10] <{tasker}> latinas everywhere
> [23:11] * lanwifie (n=s...@99-196-39-200.cust.wildblue.net) has left
> #freeswitch
> [23:11] * jjg (n=...@76.21.4.40) Quit (Read error: 110 (Connection timed
> out) )
> [23:11] here its blond blue eyed girls
> [23:11] * lollerai is now known as lolferai
> [23:11] brazilians... hopefully she's hot. i've seen some pretty
> dodgy looking chicks from there
> [23:11] <{tasker}> diego: go back, buy her a ticket and tell her,
> "yeah, it's just for a few days"
> [23:11] * martyn-dev (n=marty...@190.26.4.61) has joined #freeswitch
> [23:11] <{tasker}> blonde / blue eyes are overrated
> [23:11] <{tasker}> give me a latina any day
> [23:11] best thing around here though
>
> If I'm going into #freeswitch at 11pm at night, it's probably because I
> really need some help with some problem I've run into after hours. Can you
> imagine me injecting a question about a SIP profile into that conversation??
>
> ALL that aside... I'm willing to use a carrier pigeon if that's the way the
> three primary developers wish to communicate. They have been instrumental in
> getting my project where it is today. You know the saying... beggars can't
> be choosers.
>
>
> Ben Holtsclaw
> Network Engineer
> Avery County Schools
> Ph: 828.733.3567 x2301
>
>
> ___
> 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
>
>
--
Arnaldo M Pereira
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Re: [Freeswitch-users] Web-based forum?
On Tue, Feb 24, 2009 at 4:19 PM, Michael Collins wrote: > > Maybe this question has been raised before, but if not: There's so > > much traffic in this mailing list that I was wondering if adding a > > web-based forum on the site was in the works? > > We are upgrading the freeswitch.org site soon to drupal 6.9. We are > considering turning on the forum feature there. No definitive decision > has been made but this request has come in several times. However, we > are trying to make it so that the devs don't have yet another place to > have to monitor for user questions, etc. so we will need to figure out > a way to make it easy to use for the experts... -1 for a forum. > > > -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 > -- Arnaldo M Pereira ___ 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] monitoring events in Python
You might use freepy, which can be found in freeswitch tree, on dir scripts/socket/freepy. fseventlistener.py is a good example to follow. You'll want to pass your custom event to sniff_custom_events(). On Sat, Feb 14, 2009 at 3:45 AM, Adam Wilt wrote: > I'm trying to use custom events for a conference call in a Python script. > I set-up the events in the conference.conf.xml file, and I send "bgapi event > plain CUSTOM conference::maintenance" to enable them. But I don't know how > to look for these events in my script. Does anybody have some example code, > or maybe just point me in the right direction? Thanks. > > > > > ___ > 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 > > -- Arnaldo M Pereira a...@arnaldopereira.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
Re: [Freeswitch-users] Auto dialing ...
On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey wrote: > OK ... Here goes another I'm doing this with AST ... but I want to move it > to FS. Searched via google site:lists.freeswitch.org auto dialer and > others ... nothing useful. > Today I have a platform for auto dialing with AST (centrally managed ... > about 10 machines) and we do this: > -- Remote machines query central DB for numbers to call based on certain > configs > -- Use AMI to generate the call > -- If call gets answered, extension info queried via rta (central db > again) > > The nice thing about all of this is it's relatively easy to manage (through > one central web interface we built) and it works ... the bad part is > reporting ... as anyone knows on this list that has used AST for auto > dialing in this way (via .call or AMI) every call looks like it fails > instead of showing a real cause code. > > So ... conceptually I'm trying to accomplish the same thing ... > > Today we use FS a lot for termination of VoIP traffic ... all done via > XML_CURL ... which is awesome! > > Would like to do something like: > -- originate request > -- on answer XML_CURL posts info > Auto dialing is one of the many areas where freeswitch is much superior than asterisk. You can accomplish what you need in some ways, one would be to listen for a CHANNEL_ANSWER event on event_socket interface and, then, take whatever needed action. > > > ___ > 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 > > -- Arnaldo M Pereira http://lustyscripps.wordpress.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
Re: [Freeswitch-users] GotoIf
On Sat, Oct 4, 2008 at 5:21 AM, henkoegema <[EMAIL PROTECTED]>wrote:
>
> I discovered some small errors in previous threads concerning this item.
> (transfer i.s.o bridge)
>
> To conclude:
>
> Asterisk:
> ---
> exten => s,n,GotoIf($[${CALLERID(num)}=32476478861]?default,1000,1)
>
>
> FS:
>
>
>
> data="sofia/internal/1000%$${domain}"/>
>
>
Remember that this analogy is wrong. Asterisk's dialplan runs a goto when
the expression is true, while FS' originates a call and bridge the two
parties.
>
>
>
> Henk
> --
> View this message in context:
> http://www.nabble.com/GotoIf-tp19793170p19810038.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> ___
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>
--
Arnaldo M Pereira
[EMAIL PROTECTED]
http://www.arnaldopereira.com
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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
Jon, why don't you let the number of calls increase and see how FreeSWITCH behave? If it starts dropping the calls, then you might want to review the process or get some more help, otherwise you're done. On Mon, Sep 29, 2008 at 6:06 PM, Anthony Minessale <[EMAIL PROTECTED]> wrote: > > > What are you using to generate the load? > What are you using to deuce the load values? > You do realize that asterisk does not send RTP when you call it with sipp > etc unless you have it configured to send some audio. Asterisk cannot > generate it's own audio in most cases. So that 100 calls may not even be > doing anything but sitting there blocking. > Also, if you are basing it on the load average, FS always has a high load > average because it distributes the load on the threads to get max usage of > the CPU thus the idle time and load average number can hover in the 100's > eaisly when the box is still completely responsive. > > > Perhaps you should just let us into your machine and call you on the phone > so we can get this over with for once and for all. We don't have time to > keep going back and forth like this. > > We are now 4/0 vs ppl saying there was a performance issue that boiled down > to us ssh'ing to the box and finding them doing something wrong or > misinterpreting the results so let's just get it over with. > > > > > > On Mon, Sep 29, 2008 at 3:47 PM, Jon Bruel <[EMAIL PROTECTED]> wrote: >> >> Thanks for the input. >> >> >> >> I have now upgraded the OS to i86_64 (FC9), and I configured without >> libcurl, as recommended by Anthony. The FS loaded without problems apart >> from some "database is locked" errors, which may be harmless? I ran the >> ulimit as recommended to unlimit all resources. I started the FS after the >> ulimit. The same performance test showed a slightly better result. Without >> trying to make this too scientific, the improvement by going from 32 bit OS >> to 64 bit OS has been 10-20%, at the most. Still the performance is around >> 30-40% of the performance of Asterisk. >> >> >> >> With 100 a-leg (no b-leg, as all calls are answered by the echo >> application) permanent channels the CPU idle is 10%, with 60% system-load >> and 30% CPU-load. So something is rotten in Denmark… /Jon >> >> ___ >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED] > pstn:213-799-1400 > > ___ > 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 > > -- Arnaldo M Pereira [EMAIL PROTECTED] http://www.arnaldopereira.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
Re: [Freeswitch-users] FreeSWITCH in latin america countries
On Sun, Jun 8, 2008 at 4:19 PM, Santiago Gimenez Ocano <[EMAIL PROTECTED]> wrote: > I'm native speaker and I know that my english isn't quite good. As far I can > see there are two issues here: > > 1. Having people that can support in spanish > > 2. Splitting the community > > Even thogh there is the slang problem, I think that the support given by > Brian, Anth, Mike, Norm, etc is excellent and I don't know if anyone how > speaks spanish can do as same as they do. No, their support is the best we could get, none spanish nor portuguese speaker could achieve that now. At least none known. > > Also, I think is better for the project to have one direction and having > only one support chat is important to achive this. > > On top of this, what would happend if somebody else wants to have a chat > room in his/her own language? > > It's possible to have people supporting in other languages in #freeswitch? I think we can go more objective in this thread: do you guys think it's a better idea to drop the spanish/portuguese channel and think about it when it's actually needed ? The feedback about the idea of having #freeswitch-la was bad... and I, of course, don't want to split anything here. :) > > Regards, > Santiago > > > >> Date: Sun, 8 Jun 2008 13:47:01 -0500 >> From: [EMAIL PROTECTED] >> To: freeswitch-users@lists.freeswitch.org >> Subject: Re: [Freeswitch-users] FreeSWITCH in latin america countries >> >> J. Javier Maestro wrote: >> > On Jun Sun 08 2008 12:11, John Skopis (Lists) wrote: >> >> example: >> >> skopii: que estas me hombres >> >> bot: [skopii] where are my homies >> >> bkw: skopii: right here >> >> >> >> then the bot would translate bkw's response and pm it to me >> > >> > Wow!! Stop right there please! :) >> > >> > Where are my homies? --> Dónde están mis colegas/amigos? (España) >> > >> > And because slang words are frequently used in IRC, things like 'homies' >> > is >> > most surely translated in a completely different way in every >> > spanish-speaking country within latin america... so I really can't see >> > the >> > point on automatic translation. It's horrible! There is not enough A.I. >> > in >> > the software to do it correctly, so it is very doomed right from the >> > begining. >> >> Clearly my spanish is horrible. What is easier to understand though: >> >> que esta me hombres >> or >> dónde está mi homies >> >> and that is a simple sentence. Things only get more interesting with >> more complex constructs. >> >> > >> > It is a nice idea, but an impossible one :-/ >> > >> > My 2 cents! >> > >> >> My point was to make a "best effort". We all know it wouldn't be >> perfect. It's certainly not impossible. >> >> If everyone is in the same channel, everyone benefits from the >> discussion. In addition it would help non-english speakers learn english >> and english speakers pick up some other languages (bits and pieces at >> least). >> >> Also, what happens when there is no one that knows the answer in the >> non-english channel? The non-english speaker comes into the main channel >> (out of his/her comfort zone) and has a difficult time because no one >> understands him/her. It's hard enough for some people just to ask a >> question, let alone in a different language. That is why I think we >> should make a best effort to make it easier for everyone. >> >> Often times a translated block of text is easier to understand than a >> non-native english speaker attempting english. Granted, my perspective >> is one sided as I do not know if that is true for other languages (I >> only speak english). >> >> Anyway, I am wasting my breath at this point. Everyone (not just us >> americans) will do as they please anyway. ;] >> >> >> ___ >> 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 > > > > Blog your life in 3D with Windows Live Writer. Get Windows Live-Free > ___ > 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 > > -- Arnaldo M Pereira [EMAIL PROTECTED] http://www.arnaldopereira.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
Re: [Freeswitch-users] FreeSWITCH in latin america countries
I've already registered #freeswitch-la, but would surely pass the control of it to bkw or any other; that simply doesn't matter from my perspective. Answering MC, we could use Portuguese and Spanish on #freeswitch-la. About translations: they are just too poor to be relied. Portuguese, for instance, has too many exceptions and isn't a much logical language. I agree that people could just learn english, but we all know that it doesn't work that way, we just don't learn a language as fast as we need to learn about some technologies and/or softwares. Also, I think mixing the languages on a single channel would create a mess. #freeswitch number of users is increasing over time and it'll eventually reach a limit that will make the usage of more than one language just undoable. Of course we could use other languages on it right now and wait until it's messy to change our strategy, but we can avoid the mess. Of course this is just my opnion, which is based on an assumption: the number of users will continually increase. :) On Sun, Jun 8, 2008 at 2:35 PM, Luis Jimenez <[EMAIL PROTECTED]> wrote: > Im spanish native talker, please dont use google to translate "que estas me > hombre", what is that supposed to mean? > > Jaja. > > > On Sun, Jun 8, 2008 at 1:24 PM, Brian West <[EMAIL PROTECTED]> wrote: >> >> The bot can already translate automatically in line in the channel >> when we tell it to do so. >> >> /b >> >> On Jun 8, 2008, at 12:11 PM, John Skopis (Lists) wrote: >> >> > example: >> > skopii: que estas me hombres >> > bot: [skopii] where are my homies >> > bkw: skopii: right here >> >> >> ___ >> 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 > > -- Arnaldo M Pereira [EMAIL PROTECTED] http://www.arnaldopereira.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
Re: [Freeswitch-users] FreeSWITCH in latin america countries
On Sun, Jun 8, 2008 at 10:02 AM, Nicolas Brenner <[EMAIL PROTECTED]> wrote: > Brian, > > Although I agree that it is not a good idea to split the community, > this wouldn't much split it as increase it. There's a lot of people > who don't understand english, but have the skills to use or learn > about FreeSwitch and even help in the development. Creating a latin > irc channel, could potentially bring thousands to the FreeSwitch > community, people who are now left out because of the language > barrier. Thanks Nicolas, that's exactly my point. Instead of splitting the community, it'll end up giving help to more people and make FreeSWITCH usage broader. There's no sense on having multiple channels if we talk english on them all, who does know english will be encouraged to go (or also go) to #freeswitch. Any comments, Brian ? > > On Sun, Jun 8, 2008 at 6:24 AM, Brian West <[EMAIL PROTECTED]> wrote: >> Arnaldo, >>I really do not want to split the community up. I highly recommend >> everyone stay in #freeswitch. >> >> /b >> >> On Jun 8, 2008, at 1:59 AM, Arnaldo de Moraes Pereira wrote: >> >>> Hello, >>> >>> For those of you from latin american countries, please join >>> #freeswitch-la. I have plans to knock down the barriers to adopt >>> FreeSWITCH in third-world countries, specially Brazil, so this channel >>> is one of the steps to achieve that. Brazilian portuguese and spanish >>> are also welcomed languages. >>> >>> One of the biggest barriers to use FreeSWITCH as a TDM/SIP gateway, is >>> to have MFC/R2 support, which is being written by Steve Underwood in a >>> generic manner. I'll be focusing on the endpoint for the existing >>> unicall implementation, which we hopefully will merge when Steve has >>> finished his unicall work. So, anyone will be able to use MFC/R2 with >>> a Sangoma and probably other cards. >>> >>> Take care. >>> >>> -- >>> Arnaldo M Pereira >>> [EMAIL PROTECTED] >>> http://www.arnaldopereira.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 >> >> Brian West >> sip:[EMAIL PROTECTED] >> >> >> >> >> ___ >> 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 >> > > > > -- > Nicolás Brenner > > ___ > 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 > -- Arnaldo M Pereira [EMAIL PROTECTED] http://www.arnaldopereira.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
[Freeswitch-users] FreeSWITCH in latin america countries
Hello, For those of you from latin american countries, please join #freeswitch-la. I have plans to knock down the barriers to adopt FreeSWITCH in third-world countries, specially Brazil, so this channel is one of the steps to achieve that. Brazilian portuguese and spanish are also welcomed languages. One of the biggest barriers to use FreeSWITCH as a TDM/SIP gateway, is to have MFC/R2 support, which is being written by Steve Underwood in a generic manner. I'll be focusing on the endpoint for the existing unicall implementation, which we hopefully will merge when Steve has finished his unicall work. So, anyone will be able to use MFC/R2 with a Sangoma and probably other cards. Take care. -- Arnaldo M Pereira [EMAIL PROTECTED] http://www.arnaldopereira.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
Re: [Freeswitch-users] Ast/FS Rosetta Stone
On Sun, Jun 1, 2008 at 12:04 AM, Michael S Collins
<[EMAIL PROTECTED]> wrote:
> Just to confirm: you can run both dialplans simultaneously, no? I'm
> thinking that it might be helpful to have a few sample extensions in each dp
> type that are functionally identical. I'll see if I can come up with some
> practical examples.
I could do some tests and write them on the wiki page. I've used
asterisk's dialplan in many ways, so hopefully I can document most
used cases. I can come up with some results within the next two days.
After that, if anyone has some issue related to that, I'll be willing
to fulfill it the freeswitch way.
> -MC
>
> Sent from my iPhone
> On May 31, 2008, at 8:55 AM, "Anthony Minessale"
> <[EMAIL PROTECTED]> wrote:
>
> Another good transition tool to play with may be mod_dialplan_asterisk which
> is in tree.
>
> Once loaded in modules.conf.xml it can be used in your sofia profile by
> adding "asterisk" as the dialplan parameter.
> You can also transfer calls to it by specifying it as the dialplan param in
> the transfer or execute_extension apps.
>
> it's not entirely like the real asterisk dialplan but it is at least a close
> familiarity.
> You cannot include contexts in each other in ours because we do all our
> dialplan logic before the call.
>
> It's structured in the same way so you can create context default
>
> [default]
> ; and you can use comments the same way
> ; here is a demo that is close to the extensions.conf in tree that installs
> by default
>
> --
> [default]
>
> ; Things you're used to
> exten => music,n,Dial(SIP/[EMAIL PROTECTED]|120)
>
> ; similar pattern matching and caller-id match.
> exten => _1X,n,set(cool=${EXTEN})
> exten => _1X,n,set(myvar=true)
> exten => _1X,n,Goto(default|music)
> exten => 2137991400/1000,n,Goto(default|music)
>
> ; we also embellished things a bit and added some of our own goodies
> ; Some new magic you can do if you start the exten string with a ~ it
> implies PCRE regex
> ; *NOTE* the ,n, is there for familiarity purposes we do not parse it anyway
> nor will we honor line numbers.
>
> exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1)
> exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route})
>
> ; instead of exten, put anyting about the call you would rather match on.
> ; either the names of a field in caller_profile or a string of variables to
> expand.
> caller_id_number => 2137991400,n,Goto(default|music)
> ${sip_from_user} => bill,n,Goto(default|music)
>
>
> Anyway you will find the XML dialplan more flexible but this may serve as a
> stepping stone for beginners.
>
>
>
>
> On Fri, May 30, 2008 at 7:07 PM, Michael Collins <[EMAIL PROTECTED]>
> wrote:
>>
>> FYI,
>>
>>
>>
>> I create a new wiki page:
>>
>> http://wiki.freeswitch.org/wiki/Rosetta_stone
>>
>>
>>
>> I thought maybe those who know about Asterisk and FreeSWITCH both could
>> start dropping stuff in there. I think that people coming from Asterisk
>> might have an easier time if they had some sort of frame of reference. Note
>> that I put a disclaimer at the top – there isn't really a one-to-one
>> Ast-to-FS translation for most of the stuff, but if you said to an
>> experienced Asterisk user that "conf/dialplan/default.xml" is analogous to
>> "/etc/asterisk/extensions.conf" then he/she would at least have a starting
>> point. I'm hoping to add more stuff like the fact that there aren't really
>> "priorities" in FS but that you can use scripting langs, ivr menus, etc. to
>> mimic a lot of that kind of functionality.
>>
>>
>>
>> Anyway, your thoughts and suggestions are welcomed. I will do my best to
>> add stuff as I am able…
>>
>>
>>
>> -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
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:[EMAIL PROTECTED]
> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[EMAIL PROTECTED]
> iax:[EMAIL PROTECTED]/888
>
> ___
> 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
>
>
--
Arnaldo M Pereira
[EMAIL PROTECTED]
http://www.arnaldopereira.com
_
Re: [Freeswitch-users] parallel call forking back to dialplan
Wiki updated: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall On Thu, May 29, 2008 at 11:43 AM, Anthony Minessale <[EMAIL PROTECTED]> wrote: > the originate string can contain a list of , and | sep urls > > , means (&) forked simo dial > | means one at a time > > both at once: > > sofia/default/[EMAIL PROTECTED],sofia/default/[EMAIL PROTECTED] > > one at a time > sofia/default/[EMAIL PROTECTED]|sofia/default/[EMAIL PROTECTED] > > > you can put , sep list within a | sep list as well > > sofia/default/[EMAIL PROTECTED]|sofia/default/[EMAIL > PROTECTED],sofia/default/[EMAIL PROTECTED] > > > > > On Thu, May 29, 2008 at 9:01 AM, kokoska rokoska <[EMAIL PROTECTED]> > wrote: >> >> Hi all, >> >> I'd like to know if there is a way how to fork call to several >> destinations in the dialplan. >> Something like DisAsterisk(TM) "Dial(Local/x&Local/y)" or OpenSER >> "append_branch" and "branch_route"... >> >> Or - if it is not possible - could someone point to any workaround how >> to achieve this? >> >> Thanks a lot! >> >> Best regards, >> >> kokoska.rokoska >> >> >> ___ >> 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED] > pstn:213-799-1400 > ___ > 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] FS and Asterisk connectivity
On Mon, Apr 14, 2008 at 4:49 PM, Brian Snipes <[EMAIL PROTECTED]> wrote: > I wish to connect FS to * for interoperability testing and can't seem to > get my configs correct. Has anyone done this already and if so can you > post your configs? My configs are like yours, except for three things: 1. my asterisk is configured on outbound profile, instead of default's 2. FS registers to my asterisk 3. My rev: 8081 Are you sure the sampling rate are ok for both legs ? Besides that and comfortable noise generation (turned off, as: ), I can't think of anything else. My 8081 rev is working nicely with asterisk, maybe I should update and see what happens. > > I can call from FS to the asterisk side via a dialplan entry where > ^9(.*)$ passes to a gateway I've setup: > > > > > > > > sip_profiles/default/asterisk.xml : > > > > > > > > > > > The problem I get with this is no audio either way. I am on rev 8099. > > TIA, > Brian > > > ___ > 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] [Freeswitch-dev] Ideas for lecture
I'm also sending this message to freeswitch-users' list. On Thu, Apr 10, 2008 at 8:35 PM, Anthony Minessale <[EMAIL PROTECTED]> wrote: > sure let us know what you need. Despite FreeSWITCH didn't reach 1.0 yet, would be nice to list some cases of success. I've heard "We've never heard of FreeSWITCH, is anyone using it on production ? Who ?" sometimes, as the main argument against the software adoption. So, does anyone wants to share their deployment experiences ? Maybe it's a good idea to elaborate on http://wiki.freeswitch.org/wiki/Enterprise_deployment, or just create another wiki page for something like that... > > Maybe you can do the same presentation at ClueCon this year > http://www.cluecon.com Most likely I won't be able to be there. :( > > > > > On Thu, Apr 10, 2008 at 6:06 PM, Arnaldo de Moraes Pereira > <[EMAIL PROTECTED]> wrote: > > > > > > > > > Hello, > > > > I'll give a lecture, focusing on FreeSWITCH, as well as comparing it > > with other open source pbxes/switches/proxies. The talk will have a > > brief overall of the available software choices for the most common > > cases, but the main goal is to spread the word and show what this > > project is capable of. > > > > That'll happen in "Faculdade Impacta de Tecnologia" (Impacta college > > of Tecnology), which exports technology and teaching methodologies to > > some countries, etc. The contents of the talk aren't defined yet, I > > just told them the idea and it was very well welcomed. I would like to > > show the developers' perspective, as well as some real use cases, > > that's why I ask you guys to say whatever you think it's worth about > > those topics. > > > > There will be live examples of applications, I'd also be grateful > > receiving ideas for that. > > > > Thanks, > > Arnaldo > > > > ___ > > Freeswitch-dev mailing list > > [EMAIL PROTECTED] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED] > pstn:213-799-1400 > ___ > Freeswitch-dev mailing list > [EMAIL PROTECTED] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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] Best way to start with Freeswitch
On Mon, Mar 31, 2008 at 6:19 AM, kokoska rokoska <[EMAIL PROTECTED]> wrote: > > Brian West napsal(a): > > > Wasn't this what the whole Zend optimizer was for? So it would cache > > and pre-compiled scripts? > > > Thanks Brian for your answer! > > When I started with Asterisk in the past, I wrote all my AGI scripts in > PHP. And than I saw big latency in call setup and lack of performance. > Thus I try Zend optimizer, eAccelerator and APC too. You are right, all > of them help much, but I still wasn't satisfied. > After rewritting all of my code to ANSI C, everything change rapidly. > The same HW could setup nearly twice as much calls per second than with > PHP and bytecode cache... Even if you have to write C code for performance, it's easy to write an application module to handle all needed tasks. FreeSWITCH's architecture allows one to worry about accomplish the tasks when writing code, instead of worrying about workarounds to get your code to work the way you want, without breaking other things... and without hacks. Just wanted to say that, because I ended up in the same situation, having to write asterisk apps. I had a hard time writing them and was _very_ impressed when started writing FreeSWITCH's. All weird problems were just gone, the things just work and the API is consistent. I've just started using mod_python and my guess is that it'll fit your needs. When you're done testing, please share the results, so we can try to understand where are the bottlenecks, if any, and improve it. > > > > > Similar to how fastcgi does ... Either one is ok. > I have never heard about fastcgi. I will look at it. Thank you! > > > > You're over thinking... > > May be you are right :-) > I prepare test setup and let we see what happens :-) > > > > You're not going to put it all on ONE single machine but a > > web cluster so you can spread it out. > > > > Yes, I can. But one of the most important reasons for "switch to > Freeswitch" is better performance (call setup rate and concurrent calls) > on existing HW I believe in :-) > > > Thanks again for useful answers! I setup something in lab and - I'm > affraid - will ask for other unclear things... > > > > Best regards, > > kokoska.rokoska > > > ___ > 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] Segmentation fault on mod_python, with trunk r7982
On Sun, Mar 30, 2008 at 12:21 AM, Brian West <[EMAIL PROTECTED]> wrote: > Can you provide a full backtrace of the core on the bug please. Attached to the bug. If there's something else needed, please let me know. > > /b > > > > On Mar 29, 2008, at 10:19 PM, Arnaldo de Moraes Pereira wrote: > > > Hello, > > > > I've found a segfault running python scripts through mod_python and > > filled a bug on jira: http://jira.freeswitch.org/browse/MODLANG-54 > > > > Take care, > > Arnaldo M Pereira > > > > ___ > > 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 > ___ 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] Segmentation fault on mod_python, with trunk r7982
Hello, I've found a segfault running python scripts through mod_python and filled a bug on jira: http://jira.freeswitch.org/browse/MODLANG-54 Take care, Arnaldo M Pereira ___ 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
