April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch on PowerPC Ubuntu errors... possible?
by Jeff Bernhardt 25 Apr '13

25 Apr '13

I’m trying to build the latest Freeswitch from git on an old Mac PPC running Ubuntu 12.10 PPC. Having the following error(s) on make, but it’s kind of a strange setup, so I don’t know if it’s maybe just not possible or if I just need to tweak something or if maybe gcc is messed up. I get several errors like this: error: type of bit-field 'x' is a GCC extension [-Werror=edantic] ...where 'x' is various things in different errors. Is this is a quick fix or will it take some hacking? Was just messing around with this old Mac, so if it doesn't work, no big deal. Thanks. Jeff

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ESL in c : How to pass MyCallBack function a parameter ?
by julien terrasson 25 Apr '13

25 Apr '13

Hi again, While trying to build my own esl outbound server (server_esl.c) I found that a callback function was called from the main block whenever a socket is opened. Prototype of MyCallBack is : static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock, struct sockaddr_in *addr) I would like to extend this prototype to communicate this callback function a Queue structure that is defined in the main thread. Let's take an example : int main(void) { Queue *Q = createQueue(MaxNumberOfElements); if(pthread_create(&thread1, NULL, processQueue, Q)) { esl_listen_threaded("localhost", 8084, mycallback, 100000); } I would like to be able to pass myCallBack the Q structure so that i can queue up all the request incoming from freeswitch. But my knowledge of C is a bit dusty. So it would be nice if i can get in touch with esl_listen_threaded coder so i can define the smartest way to extend the code to transmit a C struct to the callBack Function.. Can you help me to contact him ? Regards, Julien

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Mult-tenant FS with ReSIProcate?
by Jeff Bernhardt 24 Apr '13

24 Apr '13

We have several small clients we run Asterisk deployments for. We have a FreePBX server on VMware ESXi onsite for each of them and they have trunks using a local SIP provider. Very simple setups. We're thinking of running a muli-tenant Freeswitch at our office/datacenter and ditching all their FreePBX servers (although they've served us well). I haven't setup multi-tenant stuff in Freeswitch, but I have some idea how it works... however, what confuses me beyond belief is this: we also want to do this whole thing with federated VOiP working as well via ReSIProcate (Repro). Getting Repro up and running by itself isn't much work, and getting Freeswitch up and running by itself isn't much work, but getting them both up with multi-tenancy working together is over my head, and I wonder if it's even possible. The opentelcoms.org and rtquickstart.org guides have provided lots of good info, but they don't talk about multi-tenancy or go in depth for newbie people like me about how the routing for this whole thing would work even for single tenancy. Does anyone know of any clear description/tutorials of how this could all work that doesn't assume decades of working in telephony? Just want to know 1) if it's possible and 2) generally what needs to routed to where, etc (this page has a general explanation, but it left me with some questions http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch) Also, we've chosen Repro over Kamilio because supposedly Repro is easier to setup for the uninitiated, but is this so? And what about OpenSIPS? I see 2600hz uses it for their SIP proxy. Is there any reason this would be better than Repro for our purposes? Thank you. Jeff

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Fwd: How to change Music-on-hold while digit_bind_action is used ? (Michael Collins)
by julien terrasson 24 Apr '13

24 Apr '13

Hi Michael, Thanks for your answer (quoted hereafter).. "To answer your initial question: yes, what you are trying to accomplish is definitely possible. However if you want the caller ("A" in your scenario) to press only a single key then you can't use bind_meta_app (BMA). BMA requires two key presses: * and one other DTMF. That's why your scenario requires ** to trigger." ... but i'm already using bind_digit_action :-) I probably gave too much information and made the problem description unclear. Sorry about that & let me try to explain again : Every time the star key is pressed, it will correctly execute the lua script, no problem here.. But When executing that lua script : The first time, the standard freeswitch moh is heard (on the b_leg) when playAndGetDigits is called (on the a_leg). The 2nd time, my custom freeswitch moh is heard (on the b_leg) when playAndGetDigits is called (on the a_leg). Here are the instruction which are causing me headaches : */* Change the moh on the A leg */ *api:executeString("uuid_setvar ".. b_leg_uuid .. " hold_music /usr/local/freeswitch/sounds/fr/ca/june/voxService/*fr_wait_**email.wav*");* /* Play and request digit (B leg) */* digit = session:playAndGetDigits(1, 1, 1, 5000, '#', '/usr/local/freeswitch/sounds/ fr/ca/june/voxService/fr_ask_email.wav', '', '\\d+'); I was expecting fr_wait_email.wav to be played every time, but it look like "uuid_setvar <b_leg_uuid> hold_music <custom_wav>" is not taken into account on-the-fly, but only after bind_digit_action is finished. Is that a normal behaviour ? If yes, what would be the correct way of changing the moh when bind_digit_action is executed ? Regards, Julien

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FreeSWITCH Community Conference Call Today
by Michael Collins 24 Apr '13

24 Apr '13

Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_24 The folks from JeraSoft will be doing a presentation on their routing/billing solution that works with FreeSWITCH. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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gsmopen help
by Seven Du 24 Apr '13

24 Apr '13

Hi, I just tried on Raspberry Pi with a Huawei E1550 and I got 3devices - USB0,1,2, however, when making a call the USB device name seems changed to USB3,4. Seems whenever making a call the device changes even no mod_gsmopen loaded. I tested I can send text message without problem. questions - 1) Is this because the audio is locked? How do I check if it's locked or not? 2) If not, why the device name keeps changing? it there a way to avoid that? 3) I got 3 devices when plug the USB, is that normal? how do I tell which one is control and which one is audio? It seems the last one is control. And the third one maybe 3G? Thanks, below is some log from /var/log/message Apr 16 20:41:44 raspberrypi kernel: [40750.001897] usb 1-1.2: USB disconnect, device number 9 Apr 16 20:41:44 raspberrypi kernel: [40750.002605] option1 ttyUSB0: GSM modem (1-port) converter now disconnected from ttyUSB0 Apr 16 20:41:44 raspberrypi kernel: [40750.002686] option 1-1.2:1.0: device disconnected Apr 16 20:41:44 raspberrypi kernel: [40750.002963] qmi_wwan 1-1.2:1.1: wwan0: unregister 'qmi_wwan' usb-bcm2708_usb-1.2, WWAN/QMI device Apr 16 20:41:44 raspberrypi kernel: [40750.010340] option1 ttyUSB4: GSM modem (1-port) converter now disconnected from ttyUSB4 Apr 16 20:41:44 raspberrypi kernel: [40750.010425] option 1-1.2:1.2: device disconnected Apr 16 20:41:44 raspberrypi kernel: [40750.018266] option1 ttyUSB5: GSM modem (1-port) converter now disconnected from ttyUSB5 Apr 16 20:41:44 raspberrypi kernel: [40750.018360] option 1-1.2:1.3: device disconnected Apr 16 20:41:44 raspberrypi kernel: [40750.269178] usb 1-1.2: new high-speed USB device number 10 using dwc_otg Apr 16 20:41:44 raspberrypi kernel: [40750.382002] usb 1-1.2: New USB device found, idVendor=12d1, idProduct=1003 Apr 16 20:41:44 raspberrypi kernel: [40750.382036] usb 1-1.2: New USB device strings: Mfr=2, Product=1, SerialNumber=0 Apr 16 20:41:44 raspberrypi kernel: [40750.382052] usb 1-1.2: Product: HUAWEI Mobile Apr 16 20:41:44 raspberrypi kernel: [40750.382066] usb 1-1.2: Manufacturer: HUAWEI Technology Apr 16 20:41:44 raspberrypi kernel: [40750.512562] usb 1-1.2: USB disconnect, device number 10 Apr 16 20:41:49 raspberrypi kernel: [40755.099352] usb 1-1.2: new high-speed USB device number 11 using dwc_otg Apr 16 20:41:49 raspberrypi kernel: [40755.210452] usb 1-1.2: New USB device found, idVendor=12d1, idProduct=140c Apr 16 20:41:49 raspberrypi kernel: [40755.210486] usb 1-1.2: New USB device strings: Mfr=2, Product=1, SerialNumber=0 Apr 16 20:41:49 raspberrypi kernel: [40755.210502] usb 1-1.2: Product: HUAWEI Mobile Apr 16 20:41:49 raspberrypi kernel: [40755.210516] usb 1-1.2: Manufacturer: HUAWEI Technology Apr 16 20:41:49 raspberrypi kernel: [40755.218266] scsi25 : usb-storage 1-1.2:1.0 Apr 16 20:41:49 raspberrypi kernel: [40755.224063] scsi26 : usb-storage 1-1.2:1.1 Apr 16 20:41:49 raspberrypi kernel: [40755.227981] usb 1-1.2: USB disconnect, device number 11 Apr 16 20:41:56 raspberrypi kernel: [40762.009397] usb 1-1.2: new high-speed USB device number 12 using dwc_otg Apr 16 20:41:56 raspberrypi kernel: [40762.120656] usb 1-1.2: New USB device found, idVendor=12d1, idProduct=140c Apr 16 20:41:56 raspberrypi kernel: [40762.120688] usb 1-1.2: New USB device strings: Mfr=2, Product=1, SerialNumber=0 Apr 16 20:41:56 raspberrypi kernel: [40762.120705] usb 1-1.2: Product: HUAWEI Mobile Apr 16 20:41:56 raspberrypi kernel: [40762.120720] usb 1-1.2: Manufacturer: HUAWEI Technology Apr 16 20:41:56 raspberrypi kernel: [40762.131735] option 1-1.2:1.0: GSM modem (1-port) converter detected Apr 16 20:41:56 raspberrypi kernel: [40762.132274] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB0 Apr 16 20:41:56 raspberrypi kernel: [40762.138276] qmi_wwan 1-1.2:1.1: cdc-wdm0: USB WDM device Apr 16 20:41:56 raspberrypi kernel: [40762.142683] qmi_wwan 1-1.2:1.1: wwan0: register 'qmi_wwan' at usb-bcm2708_usb-1.2, WWAN/QMI device, ea:1a:89:2e:60:ec Apr 16 20:41:56 raspberrypi kernel: [40762.148396] option 1-1.2:1.2: GSM modem (1-port) converter detected Apr 16 20:41:56 raspberrypi kernel: [40762.148977] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB3 Apr 16 20:41:56 raspberrypi kernel: [40762.153480] option 1-1.2:1.3: GSM modem (1-port) converter detected Apr 16 20:41:56 raspberrypi kernel: [40762.154368] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB4 Apr 16 20:41:56 raspberrypi kernel: [40762.156892] scsi31 : usb-storage 1-1.2:1.4 Apr 16 20:41:56 raspberrypi kernel: [40762.161995] scsi32 : usb-storage 1-1.2:1.5 Apr 16 20:41:57 raspberrypi kernel: [40763.153006] scsi 31:0:0:0: CD-ROM HUAWEI Mass Storage 2.31 PQ: 0 ANSI: 2 Apr 16 20:41:57 raspberrypi kernel: [40763.164198] scsi 32:0:0:0: Direct-Access HUAWEI MMC Storage 2.31 PQ: 0 ANSI: 2 Apr 16 20:41:57 raspberrypi kernel: [40763.166151] sr0: scsi-1 drive Apr 16 20:41:57 raspberrypi kernel: [40763.175965] sd 32:0:0:0: [sda] Attached SCSI removable disk -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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How to tigger an external command when registering
by Luis Daniel Lucio Quiroz 24 Apr '13

24 Apr '13

For a reason, I need to tigger an external command when user register. Is that possible? LD

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IP infomation in SDP
by Jimmy Chang 24 Apr '13

24 Apr '13

Hi, In the default configuration, the internal profile listen port 5060 and external profile listen port 5080. And follow the setting in http://wiki.freeswitch.org/wiki/NAT_Traversal vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=59.120.xx.xx"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> internal.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> When user make a SIP call from public network using external profile(5080), he encountered no audio. Here is the SDP of 200 OK from FS to user. /v=0// //o=FreeSWITCH 1366781127 1366781128 IN IP4 10.0.110.33// //s=FreeSWITCH// //c=IN IP4 10.0.110.33// //t=0 0// //m=audio 10568 RTP/AVP 18 101// //a=rtpmap:18 G729/8000// //a=fmtp:18 annexb=no// //a=rtpmap:101 telephone-event/8000// //a=fmtp:101 0-16// //a=silenceSupp:off - - - -// //a=ptime:20// //m=video 10530 RTP/AVP 116// //a=rtpmap:116 H264/90000// //a=fmtp:116 profile-level-id=428014/ When user make a SIP call from public network using internal profile(5060), everything OK. Here is the SDP of 200 OK from FS to user. /v=0// //o=FreeSWITCH 1366780920 1366780921 IN IP4 //59.120.xx.xx// //s=FreeSWITCH// //c=IN IP4 //59.120.xx.xx// //t=0 0// //m=audio 10892 RTP/AVP 18 101// //a=rtpmap:18 G729/8000// //a=fmtp:18 annexb=no// //a=rtpmap:101 telephone-event/8000// //a=fmtp:101 0-16// //a=silenceSupp:off - - - -// //a=ptime:20// //m=video 10178 RTP/AVP 116// //a=rtpmap:116 H264/90000// //a=fmtp:116 profile-level-id=428014// / I can't figure out which parameters in sofia profile or in dialplan make port 5060 set the public IP in SDP and work fine. Any advice? Thanks in advance. Jimmy

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Re: [Freeswitch-users] Multi-tenat PBX using xml_curl
by Chris B. Ware 24 Apr '13

24 Apr '13

Anthony, any domain has the same dial-string: <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> and when an extension calls another extension I use: <action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/> Date: Mon, 22 Apr 2013 16:58:13 -0500 From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] Multi-tenat PBX using xml_curl To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Message-ID: <CAKbxfG9YOB9pPPyBok-srfYQ8a1KRxbE2S7TaKESYwxiG_uFng(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" What is the dial string you are using to call them? On Mon, Apr 22, 2013 at 4:51 PM, Chris B. Ware <chrisbware(a)yahoo.it> wrote: > Hi guys, > > I've looked for a help on this issue on irc and mailing-list archive > without success. > I need to configure a multitenant PBX using xml_curl. I've followed wiki > to do it. > > I've two domains: > > DomainA.voip.com > DomainB.voip.com. > > I can register extensions to those domains but when I call one extensions > from another I receive > an "INVALID PROFILE" error. > > The only way I've found to fix this issue is to add those domains as > aliases under internal profile. > > I thought they were automatically added: what should I set to generate > their aliases automatically? > > Please, help. Thanks. > > _________________________________________________________________________ > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org

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sip to h323 video call via freeswitch
by Navnath Sonavne 24 Apr '13

24 Apr '13

Anthony, any domain has the same dial-string: <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> and when an extension calls another extension I use: <action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/> Hi Anthony, You said any domain has same dial-string so i searched it in various files but i did not find it. Can you tell me in which file i can find it? In reply you said when an extension calls another extension I use: <action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/> I used this action in my default.xml but no success. FYI I am able to convert sip protocol to h323 protocol for Audio call. Here is my extension from default.xml to convert sip to h323 : <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.103"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension> I am using this for video call also but no success. I am attaching vars.xml , default.xml , internal.xml , h323.conf.xml , opal.conf.xml and sofia.conf.xml. Please look in these files and tell me if am I going wrong and what should be done to make sip to h323 video call? -Navnath

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