- FreeSWITCH-users - FreeSWITCH Mailing Lists

Building FS with mod_java support
by Victor Medina 08 Dec '14

08 Dec '14

Hi all! I´ve been trying to build Freeswitch 1.4 with Java support. It compiles just fine, installation fails with: making install mod_java make[4]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make install-am make[5]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Entering directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[6]: Nothing to be done for `install-exec-am'. cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts cp: cannot stat ‘freeswitch.jar’: No such file or directory make[6]: *** [install-data-local] Error 1 make[6]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[5]: *** [install-am] Error 2 make[5]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[4]: *** [install] Error 2 make[4]: Leaving directory `/home/administrador/freeswitch/src/mod/languages/mod_java' make[3]: *** [mod_java-install] Error 1 make[3]: Leaving directory `/home/administrador/freeswitch/src/mod' make[2]: *** [install-recursive] Error 1 make[2]: Leaving directory `/home/administrador/freeswitch/src' Can someone help? Sin mas a que hacer referencia, Victor Medina

2 2

Re: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI
by Notify Me 08 Dec '14

08 Dec '14

The delay is very short, they send a100 Trying and then a 480 temporarily Unavailable back, as I can see in tcpdump. After they send the fail and its acknowledged, they end it and send an SMS back. This is a full dump 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none], proto UDP (17), length 1146) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118 INVITE sip:sina.nowahala@sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 5 From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE Contact: <sip:mod_sofia@my.public.ip.address:5080> User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Support: update_display,send_info Remote-Party-ID: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address s=FreeSWITCH c=IN IP4 my.public.ip.address t=0 0 m=audio 22290 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none], proto UDP (17), length 384) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356 SIP/2.0 100 Trying Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber> Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Content-Length: 0 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none], proto UDP (17), length 1015) othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 4 From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 INVITE User-Agent: FreeSWITCH-mod_sofia/1.2.14 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 X-FS-Display-Name: sina.nowahala X-FS-Display-Number: sip:sina.nowahala@sip.othersipgw.com Remote-Party-ID: "sina.nowahala" <sip:sina.nowahala@sip.othersipgw.com>;party=calling;privacy=off;screen=no 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none], proto UDP (17), length 412) 10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384 ACK sip:sina.nowahala@sip.othersipgw.com;+234diallednumber SIP/2.0 Via: SIP/2.0/UDP my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q Max-Forwards: 5 From: "234diallednumber" <sip:+234diallednumber@my.public.ip.address>;tag=04tDF6pXQU4Ng To: <sip:sina.nowahala@sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09 CSeq: 68640179 ACK Content-Length: 0 So you recommend a transfer on fail wav to be played? I have been tearing my hair out trying to play sounds before the call is transferred to them, so that the dialler hears something. On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus <avi(a)avimarcus.net> wrote: > How long is the delay? I assume they return a normal_clearing or fail 4XX? > If the delay is short and they have a "real" status return, then.. for > success, you can just queue a "success.wav" to play after. > And for fails, you can use > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer to > an extension that plays an error message. > > This SMS sending sounds like a normal API call.. which if it was, could be > done with curl or inside a lua/js script. > > -Avi > > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me <notify.sina(a)gmail.com> wrote: >> >> Hi Avi! >> >> Thanks for the reply, I did figure it out, thanks! This is what I did, >> >> <include> >> <extension name="firstsipgw-inbound"> >> <condition field="destination_number" expression="^0123450(\d{2})$"/> >> <condition field="caller_id_number" expression="^0(\d{10})$" >> require-nested="true"> >> <action application="set" data="effective_caller_id_number=+234$1"/> >> <action application="set" >> data="caller_id_number=+234${effective_caller_id_number}"/> >> <action application="set" data="transfer_ringback=$${us-ring}"/> >> <action application="set" >> data="transfer_ringback=file_string://$${hold_music}"/> >> <action application="set" >> data="transfer_ringback=local_stream://connecting"/> >> <action application="bridge" >> >> data="sofia/external/sip:user.name@sip.othersipgw.com;${effective_caller_id_number}"/> >> </condition> >> </extension> >> </include> >> >> I can see from the logs that dials the othersipgw.com URI, and I can >> see tcpdumps that corresponds to the traffic being sent, and the >> other side gets it. The other side immediately drops the call and does >> some processing that returns an SMS to the dialler. >> >> I dont know if I can trouble you for more help.. there is no ringing >> when the transaction happens, and the dialler is unsure what is >> happening, if the call actually connected. As you might have guessed >> from the above I am trying to make it ring for a few seconds before >> the call drops. Is this possible? I would like for the dialler to be >> able to hear it ring a few times before the call is cut. >> >> Any help gratefully accepted. >> >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus <avi(a)avimarcus.net> wrote: >> > Hi - did you figure this out yet? >> > >> > One comment: >> > <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/> >> > -- I >> > don't know if $1 is available anymore. You might want to just set that >> > as >> > part of the actual number to route, e.g. add a 1212 prefix and match it >> > again or set it as a channel variable. You can see in the logs if the $1 >> > is >> > resolving correctly. >> > >> > -Avi >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.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 > >

2 1

outgoing speech path is broken sometimes in 1.4.14
by Den Pavlovsky 08 Dec '14

08 Dec '14

Hi I installed FS 1.4.14 from git Now I test outgoing calls with Radius authorization. The problem that I encountered is that sometimes there is no RTP from Freeswitch. I see it in tcpdump dump. Iptables -no restrictions. I also tested when there is no mod_rad_auth loaded (I commented it out in autoload_configs/modules.conf.xml). - the same result And when I put the call on hold - then when I return - the call path is normal. Any help on how to debug this issue is appreciated. Does anyone encountered the same behaviour? The version FreeSWITCH Version 1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19 22:11:13Z 64bit) P.S. I am new to FS. I came from Asterisk world.

1 1

Re: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform
by DanB 08 Dec '14

08 Dec '14

Hi Deon, If you are looking for a billing engine (which I find it more suitable than complete billing system for hosted environments, same approach as FreeSWITCH +/or Kazoo - it does not enforce any business rules), take a look on CGRateS. It is opensource, very fast (12k+ requests per second), real-time, scalable (fire up more instances as you need them), feature-rich (multi-tenancy, supports advanced concepts as multiple calls out of same account, multiple balances inside one account, derived charging - eg: reseller chaining, grouped rating - charge using various billing increments within same call), API driven (full set of APIs available in both JSON and GOB) and production ready (runs already at MVNOs, ISPs, wholesale suppliers sites live). Happy to help you more! DanB

2 1

outgoing speech path is broken sometimes in 1.4.14
by Dmitry 08 Dec '14

08 Dec '14

Hi I installed FS 1.4.14 from git Now I test outgoing calls with Radius authorization. The problem that I encountered is that sometimes there is no RTP from Freeswitch. I see it in tcpdump dump. Iptables -no restrictions. I also tested when there is no mod_rad_auth loaded (I commented it out in autoload_configs/modules.conf.xml). - the same result And when I put the call on hold - then when I return - the call path is normal. Does anyone encountered the same behaviour? The version FreeSWITCH Version 1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19 22:11:13Z 64bit) P.S. I am new to FS. I came from Asterisk world.

1 0

Re- LUA audio playback on Windows 64bit not working? (FOUND & FIXED)
by Andrew Keil 08 Dec '14

08 Dec '14

To all, Found the issue below: Removed the inline="true" from the xml dialplan and everything works as expected. <action inline="true" application="lua" data="ASKTestLua.lua $2"/> Now: <action application="lua" data="ASKTestLua.lua $2"/> Not sure why I put it there in the first place. Sorry about that. Andrew From: Andrew Keil Sent: Monday, 8 December 2014 1:18 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Re- LUA audio playback on Windows 64bit not working? To FreeSWITCH users, I am just starting to use Lua inside FreeSWITCH and I am having difficulty with playing back audio to the caller. It seems the Media is not ready or available even though I have answered the caller using session:answer(). It also seems that session:ready() always returns false after session:answer(). Very strange. Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro (installed from freeswitch.msi) - default installation except for conf\vars.xml (default_password changed) conf\dialplan\default\00_ASKTestLua.xml: <include>  <extension name="ivr_demo"> <condition field="destination_number" expression="^(9101)$">    <action inline="true" application="lua" data="ASKTestLua.lua $2"/> </condition> </extension> </include> scripts\ASKTestLua.lua: -- Test Lua application -- answer the call session:answer(); session:sleep(1000); prompt = "ivr/ivr-welcome_to_freeswitch.wav"; freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt)); session:execute("playback",prompt); --session:streamFile(prompt); freeswitch.consoleLog("INFO", "Prompt played\n") -- hangup session:hangup(); Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] switch_core_session.c:2725 Application playback Requires media! pre_answering channel sofia/internal/1000(a)192.168.15.74<mailto:sofia/internal/1000@192.168.15.74> EXECUTE sofia/internal/1000(a)192.168.15.74<mailto:sofia/internal/1000@192.168.15.74> playback(ivr/ivr-welcome_to_freeswitch.wav) vs. conf\dialplan\default\00_ASKTestLua.xml: <include>  <extension name="ivr_demo"> <condition field="destination_number" expression="^(9101)$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>  </condition> </extension> </include> Which works!! However this is not using Lua. I have tried using "session:streamFile(prompt);" instead of "session:execute("playback",prompt);" and this also does not work. I would appreciate any insight into why this is happening. Kind Regards, Andrew Keil

2 2

Re- LUA audio playback on Windows 64bit not working?
by Andrew Keil 08 Dec '14

08 Dec '14

To FreeSWITCH users, I am just starting to use Lua inside FreeSWITCH and I am having difficulty with playing back audio to the caller. It seems the Media is not ready or available even though I have answered the caller using session:answer(). It also seems that session:ready() always returns false after session:answer(). Very strange. Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro (installed from freeswitch.msi) - default installation except for conf\vars.xml (default_password changed) conf\dialplan\default\00_ASKTestLua.xml: <include>  <extension name="ivr_demo"> <condition field="destination_number" expression="^(9101)$">    <action inline="true" application="lua" data="ASKTestLua.lua $2"/> </condition> </extension> </include> scripts\ASKTestLua.lua: -- Test Lua application -- answer the call session:answer(); session:sleep(1000); prompt = "ivr/ivr-welcome_to_freeswitch.wav"; freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt)); session:execute("playback",prompt); --session:streamFile(prompt); freeswitch.consoleLog("INFO", "Prompt played\n") -- hangup session:hangup(); Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] switch_core_session.c:2725 Application playback Requires media! pre_answering channel sofia/internal/1000(a)192.168.15.74 EXECUTE sofia/internal/1000(a)192.168.15.74 playback(ivr/ivr-welcome_to_freeswitch.wav) vs. conf\dialplan\default\00_ASKTestLua.xml: <include>  <extension name="ivr_demo"> <condition field="destination_number" expression="^(9101)$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>  </condition> </extension> </include> Which works!! However this is not using Lua. I have tried using "session:streamFile(prompt);" instead of "session:execute("playback",prompt);" and this also does not work. I would appreciate any insight into why this is happening. Kind Regards, Andrew Keil

1 0

remove country code from destination_number
by Paul Klenk 07 Dec '14

07 Dec '14

Hi, I'd like to have the country code "+49" removed from destination_number and changed to "0" I tried: <extension name="international" continue="true"> <condition field="destination_number" expression="^\+49(.*)"> <action inline="true" application="set" data="destination_number=0$1"/> </condition> </extension> but after correctly executing this extension-statement it reverts to the original version of destination_number

4 3

Recommendation for Billing for Hosted PBX Platform
by Deon Vermeulen 07 Dec '14

07 Dec '14

Hi, I've been looking at the Kazoo as a very viable option as a HostedPBX platform, but don't find any information with regards to Billing. I've also been tracking the thread about recommended GUI where 2600Hz are in the process for updated the code in git for Blue.box. This also looks like a possibility for a HostedPBX platform, but then again I have no idea about the Billing software to use for this. FusionPBX is a top contender if no the number 1 for a HostedPBX platform but again information for Billing is lacking. I would like to know what the feel out there is for Billing on a HostedPBX Platform, i.e. Cloud. If would be great if there is an opensource Billing system. I've purchased the jBilling Telecom guide, but no where can I find information about integrating with FreeSWITCH nor implementing it for a HostedPBX platform in the Cloud. Kind Regards

5 4

how to select 2nd row of sql query result in lua
by Shabbir abbasi 06 Dec '14

06 Dec '14

if i execute a query in mysql select dialprefix,rateinitial from cc_ratecard where (dialprefix='001' or dialprefix='00123') ORDER BY LENGTH(dialprefix) DESC; result row1 | 001 | 0.01000 | row22 | 00123 | 0.02000 | and this is lua code local rates = {} local count = ""; assert(dbh:query(query, function(qrow) for key, val in pairs(qrow) do rates[key] = val freeswitch.consoleLog("INFO"," rates :"..key.. "..="..val.." \n") end count = qrow.count; end)) questios is 1 how to select 2nd row from result ? 2 how to count total rows ? 3 how to print all 2 rows in console ? any suggestios wellcome

2 1