- FreeSWITCH-users - FreeSWITCH Mailing Lists

"choppy" voice
by Jonas Gauffin 21 Dec '08

21 Dec '08

Hello I got problems with choppy voice. I just happens1 time of 10 or something like that. Incorrect call: http://pastebin.freeswitch.org/6479 working call: http://pastebin.freeswitch.org/6481 Any idea what is wrong? Thanks, Jonas

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Predictive Dialing
by Gopalakrishnan A.N 21 Dec '08

21 Dec '08

Hi, I would like to have predictive dialing. In asterisk we used manager api and for outbound we use originate. The originate command will dial a number where asterisk answer the call and then we predict the answering machine with the silence file. Inspite of that human voice is detected and transferred to agents. Like this how can we go ahead with Freeswitch? Any help would be appreciated. -- Thank you with regards, Gopal,

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sessions not ending up
by Gopalakrishnan A.N 21 Dec '08

21 Dec '08

Hi, I have configured the freeswitch, we are dialing through event socket, if i dial a call per day say around 200 to 300 calls, at the end of the day the sessions are not ending up in the freeswitch, i can able to see in the console till all the calls were hanged up, I am using .NET crm. Any suggestions would help us. -- Thank you with regards, Gopal,

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error creating IPv6 profile when corresponding IPv4 port in use
by Jason White 21 Dec '08

21 Dec '08

Here's the scenario (FreeSWITCH revision 10725). I have Asterisk listening on port 5060 under IPv4. This version of Asterisk does not support IPv6, and netstat -6 -a | grep sip suggests that, as expected, it isn't listening on port 5060 under IPv6. If I set up FreeSWITCH profiles to listen on port 5070, for example, under both IPv4 and IPv6 then it works fine. However, if I create an IPv4 profile to listen on port 5070, and an IPv6 profile to listen on port 5060, Sofia fails with a "port already in use" error, even though nothing should be using port 5060 under IPv6. >From the logs: tport_create(): 0x1b4d0e0 tport_bind_server(0x1b4d0e0) to */[2001:44b8:61::3b]:5060/sip tport_bind_server(0x1b4d0e0): calling tport_listen for udp tport_alloc_primary(0x1b4d0e0): new primary tport 0x1b4f010 tport_listen(0x1b4d0e0): bind(pf=10 udp/[2001:44b8:61::3b]:5060): Address already in use tport_destroy(0x1b4d0e0) nta: bind([2001:44b8:61::3b]:5060;transport=*): Address already in use nua: initializing SIP stack failed Obviously, I can easily use a different port, and hence this issue doesn't affect me. However, if it's a bug, I think it should still go into the bug list. I realize the problem might not be in Sofia or FreeSWITCH. Is Asterisk really taking over the IPv6 port as well? Environment: Debian Sid, kernel 2.6.26, x86_64 architecture. Note: once I learn enough about FreeSWITCH to configure it properly and set it up to meet my personal telephony needs, the Asterisk installation will be, shall we say, redundant.

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SQL Error
by fidibus83 19 Dec '08

19 Dec '08

Hello, Im a newbie in FS. I get an error from the freeswitch cli: 2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL ERR [database disk image is malformed] I dont know what to do to remove this error! Can you help me? Regards

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[Java] Catching dingaling messages
by kriko 19 Dec '08

19 Dec '08

Hello! I was wondering if it would be possible to catch messages from dingaling. I saw it can print out messages into console when a user types in a message, but it doesn't understand it. I would like to catch that and do something (like initiate a call). I know you have to call you program via a dialplan, so I don't know how would it be really possible to invoke your java app when other events occur. Is it possible? Cheers, -- kriko

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Redirecting a call from one FS to another FS?
by Dennis 19 Dec '08

19 Dec '08

i would like to know, what the best way is, to redirect an incoming call from one fs (fs1) to another fs (fs2). we use 3 freeswitch servers and the carrier passes calls to the three fs servers randomly. if on fs server is not offline, the carrier sends the call to the next fs. this is generally good, but for conferencing it not so good. i am using socket outbound and need to do this for conferencing. let's say, we have a conference going on on fs1. another person wants to enter this conference, but the call is passed to fs2. on fs2 we see, that the caller wants to enter the conference going on on fs1. now we have to redirect the call from fs2 to fs1. is this done with "redirect" and some according settings/params or are there other ways to do this? we would like to do this without our carrier doing something, to be a little more independant. thanks dennis

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Ending a bridged call with a touchtone
by Marc Orenberg 19 Dec '08

19 Dec '08

Thanks for the response Brian. I don't understand what bind_meta does, or how it can help me. Is it something I can use from my Python script? I searched for a description of it, but I was unable to find one. Could you please point me towards some documentation, or maybe quickly explain it? Thanks. >Date: Tue, 16 Dec 2008 11:30:46 -0600 >From: Brian West <brian(a)freeswitch.org> >Subject: Re: [Freeswitch-users] Ending a bridged call with a touchtone >To: freeswitch-users(a)lists.freeswitch.org >Message-ID: <F9E12702-5CA0-45A8-B408-3F2162F94A8E(a)freeswitch.org> >Content-Type: text/plain; charset="us-ascii" > >Try bind_meta, examples are in the default dialplan. > >/b > >On Dec 16, 2008, at 10:24 PM, Marc Orenberg wrote: > >> Hello. I'm trying to allow the A-leg of a bridged call to be able >> to press a touchtone to end the call. >> In my Python script, I set-up a DTMF callback function using >> setInputCallback, but it doesn't seem to have any effect during >> bridged calls. Is there another way to do this? Thanks.

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Java example
by kriko 19 Dec '08

19 Dec '08

I made a simple java example, following this guide http://wiki.freeswitch.org/wiki/Java so when someone calls it should print something in console. I've also modified dialplan/public.xml, what I want is to intercept calls from jabber and process them: http://pastebin.com/m35de11d9 But there is nothing printed into console, path to jar is ok, also class name is ok. Java module is being loaded on startup. What could be wrong? -- kriko

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Re: [Freeswitch-users] Core Dump
by Michael S Collins 19 Dec '08

19 Dec '08

Check out this page: wiki.freeswitch.org/wiki/Debugging_Freeswitch -MC Sent from my iPhone On Dec 18, 2008, at 6:38 AM, "peder(a)networkoblivion.com" <peder(a)networkoblivion.com > wrote: > What is the process for capturing and submitting a core dump? > > I am messing around with the Cisco 79x1 phones and tcp and multiple > reg. > I have a 7961 using tcp and a 7960 using udp both reg'd with the same > number and both showing up as registered. If I call out from the > phone > using tcp, it works. If I call out from the phone using udp, I get a > core dump. If I call in, it calls both phones and I am assuming the > call to the phone using udp causes a core dump as well. These are the > only two phones on the system and I am running version 10851 from > yesterday. If I only have the udp phone registered and the tcp > phone is > off, it works fine. It is only when I have a mix of a udp reg and a > tcp > reg on the same number that I appear to get a core dump. > > > Peder > > _______________________________________________ > 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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