- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Sendmail & Email template
by sharad 28 Dec '11

28 Dec '11

Hello Do we have any API to make to shoot an email using some predefined template. Regards Sharad

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Building Freeswitch on OSX Lion - ./configure openssl error
by James 28 Dec '11

28 Dec '11

I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) and running into an error configuring Freeswitch on OSX Lion: ./configure: line 12283: syntax error near unexpected token `openssl,' ./configure: line 12283: ` PKG_CHECK_MODULES(openssl, openssl,' configure: error: ./configure.gnu failed for libs/iksemel ./bootstrap.sh runs fine; ./configure eventually spits out the error above. Any tips for me? I am mainly trying to set it up for local development purposes, not production. Thanks.

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require in LUA
by Juan Antonio Ibañez Santorum 28 Dec '11

28 Dec '11

Hello! I have one doubt about requier in Lua. I have one module coded in a fille called logging.lua. Why must I create '/usr/local/share/lua/5.1/logging/lua.lua' to be included in my main lua script instead creating '/usr/local/share/lua/5.1/logging.lua'? Regards

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prevention of duplicate calls.
by Rodney 27 Dec '11

27 Dec '11

is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up and continuing the first call.

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mod shout win32
by Rodney 27 Dec '11

27 Dec '11

sean, or anyone. what are the prerequisites and how do you actually build mod shout on windows machines? is there a document somewhere you can help me with? all i see on the wiki is linux. thanks. rodney ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, December 27, 2011 7:20 AM Subject: FreeSWITCH-users Digest, Vol 66, Issue 173 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: mod_shout on win32 (Sean Devoy) 2. does streamFile callback provide file position? (peter(a)uringme.com) 3. sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) 4. Building Freeswitch on OSX Lion - ./configure openssl error (James) 5. Re: Call log - multiple entries CDR?? Billing? (Sherif Omran) 6. Re: Caller ID on Analog Lines (guru singh) Thanks. FYI, I did build from source after getting all the pre-reqs. Sean -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Sunday, December 25, 2011 10:20 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_shout on win32 that module is not built in the default package. you will have to build from source to get that module as it stands. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-on-win32-tp7126980p7 127192.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 I'm looking at the examples for mod_spidermonkey where streamFile can be interrupted by a DTMF. There's dtmfcallback.js (http://wiki.freeswitch.org/wiki/Examples_dtmfcallback.js) and the Session_streamFile doc itself (http://wiki.freeswitch.org/wiki/Session_streamFile) They both use the third argument to the callback to get digits.digit or .duration, which I'm assuming is the length that the DTMF was pressed. But, do the third or fourth arguments also include the position within the file? I'm looking for a way to save the location in a voice prompt where a person pressed the DTMF -- either in seconds or samples or bytes from the beginning of the file. Also, for my own knowledge, is there a doc somewhere that explains all the possibile members of the third and fourth arguments (usually data and arg) in the callback from streamFile? Thank you. Hi people! I got my A200 board running with 1 FXS module on Linux along with mod_freetdm, but I am facing problems getting it to run on Windows. From Sangoma I followed the instructions to set up the board on Windows7 winpipe module, which works so far. How do I get freeswitch with mod_freetdm to run on Windows that I can make use of the board (pbx) on a win machine? for any ideas, I would thank you. Tamer I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) and running into an error configuring Freeswitch on OSX Lion: ./configure: line 12283: syntax error near unexpected token `openssl,' ./configure: line 12283: ` PKG_CHECK_MODULES(openssl, openssl,' configure: error: ./configure.gnu failed for libs/iksemel ./bootstrap.sh runs fine; ./configure eventually spits out the error above. Any tips for me? I am mainly trying to set it up for local development purposes, not production. Thanks. out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details Call logs are duplicated for 1 call? How can i prevent this? --- On Tue, 12/27/11, curriegrad2004 <curriegrad2004(a)gmail.com> wrote: >From: curriegrad2004 <curriegrad2004(a)gmail.com> >Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? >To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> >Date: Tuesday, December 27, 2011, 6:33 AM > > >xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for > > >On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: > >Hi, >> >>I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? >> >>thanks in advance >> >>regards, >>Sherif Omran >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>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 >> >> > >-----Inline Attachment Follows----- > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > Hi, I am the person whose thread you mentioned. I've also had somebody contact me off the list regarding this. What I did in my case was jut get some sort of dtmf converter. It costs Rs 200 and is Made in China. You plug in your FXO line in it and take another wire from it to your sangoma card. Caller ID is detected then. The only labeling it has on it is NetSonic. Somebody who contacted me tried to find it online but wasn't able to I think. I was referred to a shop here by the guy who I buy my Sangoma cards from. Muhammad Shahzad is right about polarity reversal, I confirmed this with the telco (Airtel). I do have hardware to test this or give access to somebody who wants to test a fix. But I just moved to a new place and dont have anything set up right now. It'll take a week before I have servers set up. Regards guru On Sun, Dec 25, 2011 at 7:42 AM, Ryan V <ryan(a)kaevee.com> wrote: > Caller ID on PRI line works fine. I have problem only with calls coming in > on FXO. > > Thanks, > > Venkatesh K > > > On Sun, Dec 25, 2011 at 1:33 AM, Muhammad Shahzad > <shaheryarkh(a)googlemail.com> wrote: >> >> I think India uses polarity reversal to indicate start of caller id on PRI >> line, which i guess is not fully supported by FreeTDM yet. However, you can >> try the following FreeTDM options, though i am not sure if they will work. >> >> <param name="tonegroup" value="in" /> >> <param name="enable-callerid" value="true" /> >> <param name="answer-polarity-reverse" value="true" /> >> >> You may try to set last parameter to false and test again. Do let us know >> if it works or not. >> >> I am not much familiar with FreeTDM code, nor got the equipment to test, >> so for those who have, please see below link regarding same problem in >> Asterisk and how they solved it, >> >> https://issues.asterisk.org/print_bug_page.php?bug_id=6683 >> >> Hope this helps. >> >> Thank you. >> >> >> On Sat, Dec 24, 2011 at 6:57 PM, Ryan V <ryan(a)kaevee.com> wrote: >>> >>> Hi, >>> >>> Thanks to all the support we got in mailing lists and our telco, we put >>> our freeswitch PBX online today. >>> >>> We are using Sangoma A101 and A200 Cards. We are successfully receiving >>> the calls on our PRI and Analogue lines. But, we don't get caller-id on our >>> analogue Lines. In India, Caller id comes in as DTMF tones and I could see >>> the caller ID being detected correctly in debug. >>> >>> I have seen this issue raised earlier on the list and Moises Silva from >>> Sangoma offered to fix it if given access to the server. Here is the link >>> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 >>> >>> I wanted to know whether anyone has come up with a solution. I am open to >>> give access to our server if Moises Silva or anyone else in Sangoma would >>> like to have a look at it. >>> >>> Thanks, >>> >>> Ryan >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk(a)hotmail.com >> Email: shaheryarkh(a)googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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

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Caller ID on Analog Lines
by Ryan V 27 Dec '11

27 Dec '11

Hi, Thanks to all the support we got in mailing lists and our telco, we put our freeswitch PBX online today. We are using Sangoma A101 and A200 Cards. We are successfully receiving the calls on our PRI and Analogue lines. But, we don't get caller-id on our analogue Lines. In India, Caller id comes in as DTMF tones and I could see the caller ID being detected correctly in debug. I have seen this issue raised earlier on the list and Moises Silva from Sangoma offered to fix it if given access to the server. Here is the link http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 I wanted to know whether anyone has come up with a solution. I am open to give access to our server if Moises Silva or anyone else in Sangoma would like to have a look at it. Thanks, Ryan

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Failover with Postgresql
by B.Tietz＠pinguin.ag 27 Dec '11

27 Dec '11

Hi, I try to make sofia recover with two servers. Setup is slightly done like described in Wiki. But if I try to make a sofia recover the call is not recovered. Here is the Error Message from CLI: 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr data, call not recovered Both server can reach the pgsql-server via odbc. Data is written in sip_recovery. I think the coding fort he database is wrong. Can anyone tell me what would be the best encoding? UTF8 and ANSII is bad! regards, Benjamin

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Re: [Freeswitch-users] Gateway Caller ID Help required
by Zohair Raza 27 Dec '11

27 Dec '11

Hi, See these links http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_10:_Route_to_a_gateway… http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_18:_Add_international_… http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_20:_Fix_invalid_caller… <extension name="asterlink.com"> <condition field="caller_id_number" expression="^1000$"/> <condition field="destination_number" expression="^(\d{10})$"> <action application="set" data="effective_caller_id_number=8001231234"/> <action application="set" data="effective_caller_id_name=800 Number"/> <action application="bridge" data="sofia/gateway/asterlink.com/1208$1"/> </condition> </extension> Your debug message doesn't show caller id, so what I've understood is, you need to make a check for 10 digit caller id, so that could be, and leave caller id as it is <condition field="caller_id_number" expression="^(\d{10})$"/> similarly you can make other checks as per your requirements. Regards, Zohair Raza On Tue, Dec 27, 2011 at 5:59 AM, Sherif Omran <sherifomran2000(a)yahoo.com>wrote: > Hi Zohair, > > I made it differently, each gateway forwards to an extension and in the > dial plan i control the incoming caller id. I think, this may not be > correct because of using 2 numbers instead of only 1 external number that > is connected to the same extension. > If you have an idea, i would appreciate. > > Regarding the german gateway, I can not add 0049 because i am calling it > from my swiss number and the number is transmitted correctly, however what > i see is 10 digits only. see the following output: > > 2011-12-27 01:53:32.275392 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_1/+41793940965(a)bluesip.net) Running State Change > CS_REPORTING > 2011-12-27 01:53:32.275392 [DEBUG] switch_core_state_machine.c:662 > (sofia/sipinterface_1/+41793940965(a)bluesip.net) State REPORTING > 2011-12-27 01:53:32.275392 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_1/sip:1000@88.64.50.28:61000) Running State Change > CS_HANGUP > > kind regards, > Sherif > > --- On *Tue, 12/27/11, Sherif Omran <sherifomran2000(a)yahoo.com>* wrote: > > > From: Sherif Omran <sherifomran2000(a)yahoo.com> > > Subject: Re: [Freeswitch-users] Gateway Caller ID Help required > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Date: Tuesday, December 27, 2011, 3:14 AM > > > Hi Zohair, > > I found it > > <action application="set" > data="effective_caller_id_number=001${caller_id_number}"/> > > but i have 2 gateways, this works with the usa gateway. However, I need to > put a condition in the dialplan > > if it comes from german gateway, add 0049 > if it comes from usa gateway, add 001 > > I don't know the variable name of the gateway > > best regards and many thanks > > Sherif > > > --- On *Mon, 12/26/11, Zohair Raza <engineerzuhairraza(a)gmail.com>* wrote: > > > From: Zohair Raza <engineerzuhairraza(a)gmail.com> > Subject: Re: [Freeswitch-users] Gateway Caller ID Help required > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Date: Monday, December 26, 2011, 3:02 PM > > Hi, > use effective_caller_id variable in dialplan to append + > > Regards, > Zohair Raza > > > > > On Mon, Dec 26, 2011 at 4:58 PM, Sherif Omran <sherifomran2000(a)yahoo.com>wrote: > > Hi > > I have setup an IPkall gateway, so that any call to my number in USA be > connected to my extension. However, the caller ID that appear is 10 digits > and i want it to show +1 stating for usa. > > In the mean time, I have another german gateway, the number is transmitted > fully including the country code, but it appears on my phone in the form of > last 10 digits only. > > any help is appreciated > > thank you in advance > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <http://mc/compose?to=consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org<http://mc/compose?to=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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Fwd: Configure openVOX card FXO card.
by Farooq Hussain 27 Dec '11

27 Dec '11

Hello, I am new to freeswtich. Please let me know how to configure FXO openvox 4ports card configure on freeswtich. -- Thanks Farooq Hussain

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FreeTDM & libisdn question
by Stuart Mills 27 Dec '11

27 Dec '11

Hi, I'm new to freeswitch as I'm trying to migrate a system from Asterisk. The Asterisk system has a digium primary rate card in it, so I am trying to get this working, everything else is great it is literally just the card. The model of this card is a 2nd generation Wildcard TE410P and supposedly freetdm does support this when compiled with libisdn, the problem I am having relates to libisdn not compiling properly when I run "make && make install" Has anyone experienced this issue before, or even perhaps tried to do this with success? Regards, Stuart

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