- FreeSWITCH-users - FreeSWITCH Mailing Lists

SIMPLE IM in UTF-16
by Alex Chan 04 Nov '13

04 Nov '13

Hi folks, I'm using FreeSWITCH as a SIP IM B2BUA. When the content-type was in UTF-16BE, the message body will be dammed at where the zero byte 0x00 first appears. Any ideas? Thanks in advance. Alex

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bind_digit_action does not work when the call is sent to a script through socket (ESL)
by Ali Pey 04 Nov '13

04 Nov '13

Hello, bind_digit_action does not work when the call is sent to a script through socket. I have tried setting it before or inside the ESL perl script. It works before the call is sent to ESL, but it stops working after. Would this be a bug? Any guidance or help would be greatly appreciated. We have been trying to get it to work for days with no success. Thanks, Ali Pey

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Inbound Call to Registered Users
by Brandon Lee 04 Nov '13

04 Nov '13

Hello, We have 2 FreeSWITCH servers that share the same db at different locations and when a user registers at one location, the other location is unable to dial the registered user. The hangup clause is USER_NOT_REGISTERED. Also in the logs I see Cannot locate registered user 9897sx3243(a)74.23.24.23. It seems that the registered user lookup filters by freeswitch server ip as the domain is the same on both servers. I noticed if I register at both locations, and send the call to the server I am not registered to, it works just fine. Would anyone know how I can enable freeswitch to perform the lookup even if the user is registered at another location?

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slow ODBC
by Juan Antonio Ibañez Santorum 03 Nov '13

03 Nov '13

Hello, I have one custom module accessing to a mysql DB using FS ODBC functions. Sometimes I see those accesses get slow but I don't know why. CPU is not very intensive. How many ODBC connections make FS? May be neccesary to get more? Any idea to try to get if the problem stays in FS, ODBC or MySQL server? Regards

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Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle
by Johannes Stads 03 Nov '13

03 Nov '13

Sergey I tend to agree with the statement in your first post, it's a shame that Asterisk works better in your setup. But i think there the problem is not in Freeswitch. The problem is in gsmopen module, in its features and support. I made a question about gsmopen back in september, and did not receive any answer. Gsmopen is in a beta state, we should not require any stable operation of FS with gsmopen module in a production environment. I have a long experience with Asterisk, and decided to give a try to Freeswitch to see its superior performance. I started using a linux virtual machine. Then tried it also with BeagleBone Black which is just a bit more powerful beast than PI. All compiled well, including the latest version of FS. Building of a mod_gsmopen was a bit of a pain, but finally worked too. What linux flavor were you using with FS on PI? In my setup I used bare minimum config, and again can confirm that FS is a superior platform with plenty of advanced features, though its concept is more difficult for understanding initially. Although I was able to send SMS and make calls out of the box, gsmopen worked highly unstable and sound quality was not good, to say the least. To prove my concept, today i took a fresh SD card to put Debian Wheezy on it and installed Asterisk with chan_dongle. The platform used was BBB. What can i say, the experience was different. To compare, FS build time on BBB/Debian/SD card was such: ./configure (about 1h) make (about 2h) Asterisk was up and running in about 30 min. chan_dongle was built smoothly without a need to tweak libraries. In my FS setup gsmopen did not show a claimed 'auto discovery' feature and some exercise with UDEV rules was required to make sure that modems are loaded in right order. In chan_dongle it is nicely done out of the box. Overall, chan_dongle is much more advanced piece of software, I want to use this opportunity to thank its author. If your goal is calling to mobiles, you have to consider chan_dongle against gsmopen, and not FS against *. Cheers, Jo -------------------------------------------- On Wed, 10/30/13, Sergey Zhuravlov <4orbit(a)gmail.com> wrote: Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, October 30, 2013, 4:16 AM I am glad to tell you that: 1) mod_gsmopen and e1550 work on Raspberry Pi (incoming, outgoing SMS in both directions). 2) and the stable version FS and the latest FS compile on Raspberry Pi, but it's a long time. 3) The latest version does not run on my raspberries. Need advice on optimizing the FS for raspberry Pi, as low-productivity device. Can any startup options? Maybe someone knows what modules you can disable resource-intensive. Something else? Does somebody have a positive experience with Raspberry Pi, mod_gsmopen GSM dongle ??? I use Raspberry Pi B and e1550. It's a shame that with an asterisk all works fine -- RasPBX. http://www.raspberry-asterisk.org/ But I prefer the FreeSWITCH ;-) On the basis of experience, I thought that PS will work best for this modest hardware faster and require fewer resources. But it turns out it is not. Asterisk with a web server, MySQL database, and other, less load than FS. In this connection, other parameters may be used starting FS to save resources? And the main question! Silence an incoming call on 5000 (IVR) as well as an outgoing in gsm network. Those are not sound. -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com -----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

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RSS menus odd behavior
by Jack 02 Nov '13

02 Nov '13

I have been experimenting with the RSS module and I am having problems with the menus. It wants to speak the break time along with the menu items. Is there a way to suppress this behavior? On windows 7 . Thanks, Jack EXECUTE sofia/external/Anonymous(a)anonymous.invalid rss() 2013-11-02 14:29:29.909247 [DEBUG] mod_rss.c:278 Raw Codec Activated 2013-11-02 14:29:30.875302 [DEBUG] switch_ivr_play_say.c:2212 Speaking text: ,<break time="500ms"/>Main Menu. <break time="600ms"/> Select o ne of the following news sources, or press 0 to exit. ,<break time="600ms"/>1: England Cricket. <break time="600ms"/><break time="2000ms"/> 2013-11-02 14:29:49.329357 [DEBUG] switch_ivr_play_say.c:2412 done speaking text 2013-11-02 14:29:50.269411 [DEBUG] switch_ivr_play_say.c:2212 Speaking text: ,<break time="500ms"/>Main Menu. <break time="600ms"/> Select o ne of the following news sources, or press 0 to exit. ,<break time="600ms"/>1: England Cricket. <break time="600ms"/><break time="2000ms"/> 2013-11-02 14:29:50.509425 [DEBUG] switch_rtp.c:5126 RTP RECV DTMF 1:760 2013-11-02 14:29:50.509425 [DEBUG] switch_channel.c:471 RECV DTMF 1:760 2013-11-02 14:29:50.509425 [DEBUG] switch_ivr_play_say.c:2412 done speaking text 2013-11-02 14:29:51.749496 [DEBUG] switch_ivr_play_say.c:2212 Speaking text: ,<break time="500ms"/>England news from ESPN Cricinfo.com. Visi t Cricinfo.com for up-to-the-minute cricket news, breaking cricket news, live cricket commentary, ball-by-ball commentary, cricket video, cr icket audio and features.. . local time: 02:29 PM, Press 0 for options, 5 to change voice, or pound to return to the main menu. 2013-11-02 14:30:15.728867 [DEBUG] switch_ivr_play_say.c:2412 done speaking text 2013-11-02 14:30:15.888877 [DEBUG] switch_ivr_play_say.c:2212 Speaking text: Story 1. Everything 'fantastic' for Prior 2013-11-02 14:30:16.868933 [DEBUG] switch_rtp.c:5126 RTP RECV DTMF 0:760 2013-11-02 14:30:16.868933 [DEBUG] switch_channel.c:471 RECV DTMF 0:760 2013-11-02 14:30:16.868933 [DEBUG] switch_ivr_play_say.c:2412 done speaking text 2013-11-02 14:30:17.928993 [DEBUG] switch_ivr_play_say.c:2212 Speaking text: Press star to pause or resume speech. To go to the next item, p ress six. To go back, press 4. Press two to go faster, eight to slow down, or 7 to resume normal speed. To change voices, press five. To res tore the original voice press 9. To hear these options again, press zero or press pound to return to the main menu. 2013-11-02 14:30:25.189408 [DEBUG] switch_rtp.c:5126 RTP RECV DTMF 6:760 2013-11-02 14:30:25.189408 [DEBUG] switch_channel.c:471 RECV DTMF 6:760 2013-11-02 14:30:25.189408 [DEBUG] switch_ivr_play_say.c:2412 done speaking text 2013-11-02 14:30:25.349418 [DEBUG] switch_ivr_play_say.c:2212 Speaking text: Story 2. England left with much to ponder 2013-11-02 14:30:29.048629 [DEBUG] switch_ivr_play_say.c:2412 done speaking text 2013-11-02 14:30:29.668665 [DEBUG] switch_ivr_play_say.c:2212 Speaking text: England travel to Hobart with questions about the composition o f their side for the first Test still largely unanswered after completing a none-too-convincing draw in their opening tour match against a W estern Australia XI

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And the Weddings Back on!
by Ken Rice 01 Nov '13

01 Nov '13

Its my pleasure to announce Brian West and Greg Dunn will resume their wedding on Monday! After a good delay due to a health issue, Greg is now able to travel! So you guys send the congrats to Brian and Greg! The happy couple has posted a list of items they can use @ http://www.amazon.com/gp/registry/wedding/11UO8CGGBUU59/ref=cm_sw_em_r_wr_Mo 8Csb1EXEX9W_wb Again Congrats Brian & Greg! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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Scalability and Bridge
by Valter Cardoso 01 Nov '13

01 Nov '13

I'm thinking in using multiple FreeSwitch instances to support high traffic load, but I don't know how to deal with bridges in this scenario. If I had two incoming calls, each one created in a different FS instance, it's not possible to create a bridge with these calls, right? How should I deal with bridge using multiple FS instances? Thanks, Valter

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RPS vulnerability, Yealink responds.
by Ken Rice 01 Nov '13

01 Nov '13

Hey Guys, As you know theres been a big buzz about Remote Provisioning Services and vulnerabilities that are possible. We have been in touch with Yealink, and they have released the following statement. ----- Snip ------- Recently we received reports stating there were security vulnerabilities in Yealink phones¹ RPS feature. We have investigated in these reports and found out that this problem was already solved in our V71 version of firmware. Correction and Precaution Solutions for RPS Security Vulnerability Issues Yealink V71 version firmware could completely guard the Yealink IP Phones from the security risks. How to operate: 1) To upgrade the Yealink IP Phones to V71 firmware; 2) To encrypt the configuration files with Yealink cfg encryption tool. The Yealink cfg encryption tool could be downloaded per below link, http://www.yealink.com/Upload/T2X/T2X-V71/Yealink%20Configuration%20Encrypti on%20Tool.zip Yealink take our customers¹ security very seriously and are constantly working to improve the security features of our products. Yealink therefore also highly recommend our customers regularly update firmware to the latest version to benefit from our new features and facilities. ----- Snip ------ So if you are using Yealink and RPS encrypt those configs and upgrade to atleast the V71 firmware. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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ACK disappeared from SIP session, Help
by "Гришин В.С." 01 Nov '13

01 Nov '13

the scheme is PBX---E1--Cisco 2811-----VoIP SIP-----FreeSWITCH-----VoIP SIP---asterisk(1.8.20)---IP Phone FreeSWITCH IP is 191.21.19.12 Asterisk IP is 162.48.13.26 Call comes to FreeSWITCH from a Cisco 2811. SIP session between freeswitch and asterisk(1.8.20) |Time | 191.21.19.12 | | | | 162.48.13.26 | |21.048754| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.048758| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.065794| 100 Trying| |SIP Status | |(5065) <------------------ (5060) | |21.122406| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |21.143580| RTP (g711A) |RTP Num packets:50 Duration:0.922s SSRC:0xB5320A6 | |(21866) <------------------ (12772) | |22.282684| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |22.296762| 181 Call is being forwarded |SIP Status | |(5065) <------------------ (5060) | |27.558268| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |35.039883| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.307222| RTP (g711A) |RTP Num packets:1490 Duration:29.773s SSRC:0x530CAB37 | |(21866) <------------------ (12772) | |35.476614| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.532995| RTP (g711A) |RTP Num packets:2966 Duration:29.680s SSRC:0x308FC816 | |(21866) ------------------> (12772) | |36.436846| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |38.282686| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |42.025399| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |45.637280| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |49.343115| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |53.119940| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |56.907800| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |60.791453| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |64.606730| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |65.219392| BYE | |SIP Request | |(5065) <------------------ (5060) | |65.234354| 200 OK | |SIP Status | |(5065) ------------------> (5060) | |65.234359| 200 OK | |SIP Status | |(5065) ------------------> (5060) | It is visible that there is no package with ACK confirmation from a FreeSWITCH to Asterisk. Asterisk sends several times SIP OK on a freeswitch, and without having waited confirmations (ACK) after 23 seconds sends SIP BY on a freeswitch and on IP Phone. After that call stops. Before disconnection subscribers hear each other of 29 (+-) seconds In tcpdump between Cisco and FreeSWITCH it is visible that the SIP ACK packet from Cisco arrives on a freeswitch. In what place the SIP ACK in a FreeSWITCH disappeared? What to me to analyse in SIP session? Maybe someone solved a similar problem? please, Help! Vladislav Grishin

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