December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Error loading mod_managed in CentOS
by srinivasula reddy 04 Dec '10

04 Dec '10

All, We are presently working on Windows Based FreeSwitch server but doing piloting on using CentOS platform for better performance. We are presently using modified Managed module (mod_managed.dll) using C# and .NET framework 3.5. We want to use the same dll in linux too. We found the possibility using Mono 2.8 but when we place the mono built dll in the /mod folder, we are getting error while loading saying that mod_managed.so is not found. i understand that we are trying to load dll. could someone please help us on how to link the dll so that FreeSwitch will load the module without errors? We also found that mod_mono_managed is also available. I built the mod_mono_managed.dll in mono windows and tried to load the module which also failed. Any help provided is highly appreciated. Thanks Srinivas

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confusing on skypopen module
by ha do 03 Dec '10

03 Dec '10

hi all i try to use the skypopen and i follow on wiki page there are some places that are very hard to understand 1/ let's edit the startskype script remember to add the removing of all the installed snd-* modules cp freeswitch/src/mod/endpoints/mod_skypopen/configs/startskype.sh ./ vi startskype.sh start the X servers and the Skype clients sh ./startskype.sh 2/ on "How to prepare the configuration directory of Skype clients on Linux", see http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_pr… cp /mnt/root/configskypenew.tgz ./ tar xzf configskypenew.tgz chown root.root .Skype configskypenew.tgz is not a published file - you must generate it yourself for your own configuration using the instructions at the above link. This is because it contains your authentication details. 3/ Copy and install the Skype clients configuration directory you previously prepared (see http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_pr… for how to prepare it): cd /root mount /dev/hda5 /mnt cp /mnt/root/skypeconfig2.tgz ./ tar xzf skypeconfig2.tgz chown -R root.root .Skype 4/ Install the skypopen configuration files, edit and execute the script that starts the Skype client instances: cd /usr/src cd freeswitch/src/mod/endpoints/mod_skypopen/ cd configs/ cp skypopen.conf.xml /usr/local/freeswitch/conf/autoload_configs/ ==> copy the config to autoload vi /usr/local/freeswitch/conf/autoload_configs/skypopen.conf.xml ==> what parts should i edit in the skypopen.conf.xml cp startskype.sh 2startskype.sh => where do i put the 2startskype.sh vi 2startskype.sh => what information should i input sh ./2startskype.sh when i start the sh ./startskype.sh i got the error skype@skype-desktop:/$ sh ./startskype.sh ERROR: Module snd_* does not exist in /proc/modules ERROR: Module snd_hda_intel is in use ERROR: Module snd_dummy is in use Fatal server error: Server is already active for display 101 If this server is no longer running, remove /tmp/.X101-lock and start again. Password: what is the password i should input here where i put the real password of skype account and the real account ID of skype account please help Thank you Ha

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Need help about lua
by dome＠tel.co.th 03 Dec '10

03 Dec '10

Dear All, After upgrade from 1.0.6 to git current version. i got problem about lua and i report to jira (http://jira.freeswitch.org/browse/FS-2893) i remove all lua devel package and recompile by make current (I'm sure mod_lua use FS liblua.a) but i still got problem i realy need to use luamemcahe , luasocket in my lua script Someone help me please BG Dome C.

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Fwd: : Skypopen error
by Jian Ren 03 Dec '10

03 Dec '10

Try sending again. ---------- Forwarded message ---------- From: Jian Ren <renjian(a)gmail.com> Date: Fri, Dec 3, 2010 at 11:13 AM Subject: Re: [Freeswitch-users] : Skypopen error To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Hi Giovanni, Unfortunately, my wife still got aborted calls with your fix, and mine(another phone). I could see the warning line you added: 2010-12-02 21:12:45.470780 [DEBUG] switch_rtp.c:2925 RTP RECV DTMF C:400 2010-12-02 21:12:45.470780 [DEBUG] mod_skypopen.c:792 [|] [DEBUG_SKYPE 792 ][interface00 ][UP,INPROGRS] interface00 CHANNEL SEND_DTMF 2010-12-02 21:12:45.470780 [DEBUG] mod_skypopen.c:793 [|] [DEBUG_SKYPE 793 ][interface00 ][UP,INPROGRS] DTMF: C 2010-12-02 21:12:45.470780 [DEBUG] skypopen_protocol.c:1122 [|] [DEBUG_SKYPE 1122 ][interface00 ][UP,INPROGRS] DIGIT received: C 2010-12-02 21:12:45.470780 [WARNING] skypopen_protocol.c:1127 [|] [WARNINGA 1127 ][interface00 ][UP,INPROGRS] Received DTMF DIGIT "C", but not relayed to Skype client because Skype client accepts only 0-9*# 2010-12-02 21:12:45.687811 [DEBUG] skypopen_protocol.c:173 [|] [DEBUG_SKYPE 173 ][interface00 ][UP,INPROGRS] READING: |||CALL 46932 DURATION 12||| But the call was still ended within 1 min. Then there were other errors like: 2010-12-02 21:14:08.708226 [DEBUG] skypopen_protocol.c:173 [|] [DEBUG_SKYPE 173 ][interface00 ][DOWN,IDLE] READING: |||ERROR 559 CALL: Action failed||| 2010-12-02 21:14:08.708226 [DEBUG] skypopen_protocol.c:228 [|] [DEBUG_SKYPE 228 ][interface00 ][DOWN,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed||| 2010-12-02 21:14:08.708226 [DEBUG] switch_core_session.c:1993 sofia/internal/1000(a)192.168.1.104 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Attached is the log file. I have turned on the debug level again to give your more details. The root cause might not be just the phone, it always happened after the server has been running for a while (couple hours to 2 days), once server rebooted, everything works fine. Should I create another issue or open the original one? Thanks! Jian On Thu, Dec 2, 2010 at 10:31 AM, Giovanni Maruzzelli <gmaruzz(a)celliax.org>wrote: > On Thu, Dec 2, 2010 at 3:34 PM, Jian Ren <renjian(a)gmail.com> wrote: > > Cool, I did a git clone last night after 7pm, which should contain the > fix > > already, right? > > Don't know, you can check like this: > > # cd /usr/src/freeswitch/src/mod/endpoint/mod_skypopen > # git log > > And you'll see the last commits that you got. > > -giovanni > > > Maybe that's why I haven't seen the problem so far. > > Besides, I disconnected her 2.4G cordless phone and replaced with a 900Hz > > one, in case if it's caused by the Wifi interference. > > > > Thanks! > > Jian > > > > On Thu, Dec 2, 2010 at 9:22 AM, Giovanni Maruzzelli <gmaruzz(a)celliax.org > > > > wrote: > >> > >> On Thu, Dec 2, 2010 at 3:15 PM, Rupa Schomaker <rupa(a)rupa.com> wrote: > >> > Her phone might have a *really* crappy DTMF detector. They are more > >> > likely > >> > to fail miserably with female voices. > >> > >> Thanks Rupa! > >> > >> Btw, should be fixed in mod_skypopen code since yesterday. You'll now > >> get a warning, but no more an error, and the call should be > >> unaffected. > >> > >> -giovanni > >> > >> > > >> > On Wed, Dec 1, 2010 at 11:18 AM, Jian Ren <renjian(a)gmail.com> wrote: > >> >> > >> >> Hmm, so far the only "customer" is my wife, who keeps complaining to > me > >> >> everyday. It explains why never happened to me. I will ask her > tonight > >> >> how > >> >> she sent A or D while calling. Or maybe it's the problem of the > phone. > >> >> She > >> >> is using a dual mode phone. Before, it's connected to PC with USB for > >> >> skype > >> >> calls on Windows. Now I stopped running any version of skype on > windows > >> >> and > >> >> the USB was disconnected so she is using the phone as a normal one. > >> >> Thanks! > >> >> Jian > >> >> > >> >> On Wed, Dec 1, 2010 at 11:56 AM, Giovanni Maruzzelli > >> >> <gmaruzz(a)gmail.com> > >> >> wrote: > >> >>> > >> >>> Hi Jian, > >> >>> > >> >>> I hope in English "bizarre" does not sound bad, in Italian would be > >> >>> like "original and out of standard in a funny way" :). > >> >>> > >> >>> >From the log you attached to the Jira, the incoming SIP calls that > >> >>> are > >> >>> then bridged to skypopen are sending both the "A" dtmf and the "D" > >> >>> dtmf (can't remember if any other). > >> >>> > >> >>> You can peruse the log looking for "error 21", and you'll see is > >> >>> anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs > that > >> >>> mod_skypopen duly passes to skype. > >> >>> > >> >>> Problem is: the Skype client does not accept or relay dtmf "A-D", > and > >> >>> spit out an error. > >> >>> > >> >>> Out of curiosity you may want to check why your customers are using > >> >>> dtrmf A-D, but is not an absolut need. > >> >>> > >> >>> Anyway, I'll fix this in mod_skypopen code asap, so that if another > >> >>> channel (SIP in your case) try to send A-D to skype, that dtmf will > be > >> >>> ignored and a warning line will be emitted to console and to > logfile. > >> >>> And no more errors or aborted calls. > >> >>> > >> >>> -giovanni > >> >>> > >> >>> > >> >>> > >> >>> On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian(a)gmail.com> wrote: > >> >>> > Also this happened during a call, which got hanged up because of > the > >> >>> > error. > >> >>> > So the phone sends out A-D during a call? > >> >>> > Jian > >> >>> > On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian(a)gmail.com> > wrote: > >> >>> >> > >> >>> >> Interesting. I don't have these keys on my phone. Here is the > >> >>> >> dialplan > >> >>> >> string I am using in the ATA(SPA1001): > >> >>> >> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.) > >> >>> >> Shouldn't it only take numbers? > >> >>> >> Could this be caused by any dialplan XML files in freeswitch? > >> >>> >> Thanks! > >> >>> >> Jian > >> >>> >> > >> >>> >> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists > >> >>> >> <freeswitch-list(a)puzzled.xs4all.nl> wrote: > >> >>> >>> > >> >>> >>> On 12/01/2010 04:03 PM, Jian Ren wrote: > >> >>> >>> > Hi Giovanni, > >> >>> >>> > Could you explain more to a bizarre person like me :-) > >> >>> >>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone > sent > >> >>> >>> > to > >> >>> >>> > my > >> >>> >>> > skype through chat? If it's abnormal, I think I should fix the > >> >>> >>> > root > >> >>> >>> > cause. > >> >>> >>> > Thank you very much for looking at the issue, with 2M logs > :-). > >> >>> >>> > >> >>> >>> DTMF A-D seems to be used only on military phones: > >> >>> >>> > >> >>> >>> > >> >>> >>> > http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html > >> >>> >>> > >> >>> >>> Regards, > >> >>> >>> Patrick > >> >>> >>> > >> >>> >>> _______________________________________________ > >> >>> >>> 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 > >> >>> >> > >> >>> > > >> >>> > > >> >>> > _______________________________________________ > >> >>> > 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 > >> >>> > > >> >>> > > >> >>> > >> >>> > >> >>> > >> >>> -- > >> >>> Sincerely, > >> >>> > >> >>> Giovanni Maruzzelli > >> >>> Cell : +39-347-2665618 > >> >>> > >> >>> _______________________________________________ > >> >>> 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 > >> >> > >> >> > >> >> _______________________________________________ > >> >> 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 > >> >> > >> > > >> > > >> > > >> > -- > >> > -Rupa > >> > > >> > _______________________________________________ > >> > 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 > >> > > >> > > >> > >> > >> > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> 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 > > > > > > _______________________________________________ > > 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 > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > 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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Nibblebill
by Francis Trevor 03 Dec '10

03 Dec '10

Is it possible to bill two separate nibblebill accounts from the same dialplan entry? For instance, can I bill a customer and a vendor account I have in Nibblebill during the same call with different rates? This would be useful in a wholesale scenario. Further, if that isnt possible, can a call be posted to two separate account codes, so as to create an entry in two separate CDR files. Ultimately what I am wanting to accomplish is to make accounting entries in a customer and a vendor account for the same call. I can rate them later, but it would be useful to not have to query the master CDR file for TP's. Am I going about this in a rather complicated way and/or is my logic not lucid enough? Thoughts? TGF

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Enquiry - Comparing Asterisk and FreeSwitch
by Rodrigo Lang 03 Dec '10

03 Dec '10

Good day list, My questions may seem a bit silly, but I always worked with Asterisk and I'm seeing the potential of FreeSwitch. Two questions that I have are the following: 1 - FreeSwitch has an interface similar to the Asterisk AMI? Where an external software can connect and monitor everything that happens in real time and have some control over the FS? 2 - Can you put your settings in FS in database like the Asterisk Realtime? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/>

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how set caler info in deflect command
by Kozak Vladimir 03 Dec '10

03 Dec '10

Hi all. Operation deflect have next structure "uuid_deflect <uuid> <sip URL>" or "<action application="deflect" data="sip:someone@somewhere.com" />" How I can set specified caller info in this operation? It need set caller name and caller phone. I try set sip_invite_from_params, sip_callee_id_name, sip_callee_id_number, sip_invite_req_uri, origination_caller_id_name, origination_caller_id_number variables before use deflect command. It dosn't work. What I need to do?

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Export a channel variable within Dial String
by Bernhard Suttner 03 Dec '10

03 Dec '10

Hi, I call multiple destination with bridge like that: [var1=value]sofia/default/user1,[var1=another_value]loopback/user2 Is it somehow possible, that var1=another_value will be exported but var1=value not? What I want is, that I could specify a variable to be exported _within_ the dialstring? Thanks for every hint. Best regards, Bernhard

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TEST
by Brian West 02 Dec '10

02 Dec '10

Ignore please. /b

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is mod_python supported on windows?
by jun yang 02 Dec '10

02 Dec '10

just don't find the instructions on how to build it. wiki is about on linux only.

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