December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Headers and use of "" in the header
by rod 23 Dec '08

23 Dec '08

Dear All, I've been playing with the freeswitch options for one month now, and I've been able to use it with kamailio working as a registrar. What I'd like to do is to add a diversion header using the following action in the dialplan: <action application="export" data="sip_h_Diversion=<123456789(a)10.10.10.254>;reason="unconditional""/> Please note, that I'd like to put the word "unconditional" between quotes, this is to comply with the SIP gateway to which I'm sending trafic. But I've been unable to set an escape character to use theses quotes, cause as you may understand, without escape character FS will consider this instruction instead: <action application="export" data="sip_h_Diversion=<123456789(a)10.10.10.254>;reason=" cause the quote after reason= is considered as a closing quote for data=". Is there a way to achieve this. Thanks. rod

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Memory question
by Dennis 23 Dec '08

23 Dec '08

after doing some testing with fs, i can see in the console, when entering "top", that fs uses 9.9% of the memory. when i do some more calls, the used memory will raise - the memory will not beeing released, till i do a restart of fs. is this a normal behavior or do i have some problems? thanks dennis

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Extra loud prompts when transcoded from L16@8000 to G.722
by Andy Spitzer 23 Dec '08

23 Dec '08

Woof! I've noticed that the percieved volume of prompts recorded at L16@8000 is much louder (to the point of distortion) when played back via G.722 on Polycom phones, vs when played back via G.711. The same prompts are also slightly louder when played back on SNOM phones via G.722 vs G.711, but not nearly as obnoxiously loud. I'm not sure if FS can do anything about this, it may be a Polycom issue, but is there a way to apply gain (negative in this case!) during the transcoding process so the percieved volume of the prompts is the same no matter which codec is selected? --Woof!

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making outbound call with IPv6
by Jason White 22 Dec '08

22 Dec '08

I'm trying to make an outbound call to an IPv6 host using the pre-supplied internal-ipv6 profile. <action application="bridge" data="sofia/internal-ipv6/nnnn(a)ipv6-host.domain"/> I can ping the host with ping6, and it has a DNS AAAA record. (There is no A record as the host is a friend's box behind a NAT). My own box is also behind a NAT, but it has IPv6 connectivity via a tunnel broker. The error in the freeswitch.log file is: [ERR] sofia_reg.c:1344 sofia_reg_handle_sip_r_challenge() No Matching gateway found It's quite possible I could be doing something wrong, or there might be a bug somewhere, or the provided internal-ipv6 profile might need some adjusting... Suggestions welcome.

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mod_java.so load issue
by jonathan augenstine 22 Dec '08

22 Dec '08

I am installing Freeswitch on Fedora. I was building/installing the mod_java.so module and I encountered the following load issue: 2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **/usr/local/freeswitch/mod/mod_java.so: invalid ELF header** Is this a build issue? I am assuming maybe there is a g++ option that is set incorrectly but my searches on Google and looking at gcc docs have not provided a solution. Jonathan

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If Bypass Media works why won't Proxy Media work?
by mszlazak＠aol.com 22 Dec '08

22 Dec '08

I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode. I'm trying to pass audio "internally" on the same machine between endpoints and have be advised that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode. Any comments??

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Re: [Freeswitch-users] Phone lines ring busy after awhile
by Adam Wilt 22 Dec '08

22 Dec '08

Sorry, I'm using freeswitch-1.0.latest.tar.gz on Fedora. > > Message: 5 > Date: Mon, 22 Dec 2008 12:16:08 -0800 > From: "Michael Collins" <msc(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Phone lines ring busy after awhile > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <87f2f3b90812221216i42cbaeedx52cee1863e68667a(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > what operating system and which revision of FreeSWITCH? > -MC > > On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree(a)gmail.com> wrote: > > > Hello. I have an installation of FreeSwitch runnnig, and I wrote a > Python > > script to answer an inbound call and play an IVR. It works fine for > awhile, > > but eventually it stops answering the phone line and the phone rings > busy. > > When I look in the FreeSwitch logs I don't see anything unusual. But I > don't > > think it's a problem with the VOIP phone service because when I restart > > FreeSwitch the problem goes away. Any ideas? > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)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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group call with BLF and pickup
by Seweryn Niemiec 22 Dec '08

22 Dec '08

Hi, I'm currently evaluating FreeSWITCH to see if I can migrate to it from Asterisk (which gives me now all functions I need, but has some problems with predictability). I need only one atypical functionality: 1. we have one main extension, which clients call. lets name it 5555 2. when someone calls 5555, then immediately rings only one phone (1001) and 15 seconds later 3 more phones (1002-1004). phone 1001 should be dialled once per call to 5555 3. all phones (1000-1010) have BLF monitoring 5555, so anyone can pick it up when someone calls it ad 3. it is acceptable that all phones monitor 1001 instead of 5555 to pick up connection to 5555 After reading and googling about possible implementation in FS for few hours I couldn't find anything useful. Could you tell if it is possible to implement in FS without artificial limbs (i dunno if it's best English word for what I mean :) like I had to do in Asterisk? greetings, Seweryn

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Phone lines ring busy after awhile
by Adam Wilt 22 Dec '08

22 Dec '08

Hello. I have an installation of FreeSwitch runnnig, and I wrote a Python script to answer an inbound call and play an IVR. It works fine for awhile, but eventually it stops answering the phone line and the phone rings busy. When I look in the FreeSwitch logs I don't see anything unusual. But I don't think it's a problem with the VOIP phone service because when I restart FreeSwitch the problem goes away. Any ideas?

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Mod Fax: Error, problems and questions...
by Dennis 22 Dec '08

22 Dec '08

hi, after we managed to setup fs with mod_fax and our socket outbound script, we have some questioons about an error and problems, when sending a fax: 1.) there is one error, we get always - no matter, if the fax was sent successfully or not. in the pastebin under http://pastebin.freeswitch.org/6338 you can see the error in the last line. this is the full log of a fax in fs console loglevel debug. 2.) fax works quite good. we couls send long faxes over a normal fax machine without any problem. but for fast testing we are using a softfax (fritz fax). here we have some more problems. mostly a fax with one page will pass through, but more pages will mostly fail. because we are new to the fax thing, we do not really know, what the messages tell us about failed faxes. here are the top 3 messages (unordered and always one at a time - nerver at once): variable_fax_result_text => Received a DCN while waiting for a DIS fax_result_text => The HDLC carrier did not stop in a timely manner fax_result_text => Unexpected message received could someone please tell us, where the problem might be? thanks dennis

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