July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

create dummy channel to record conference
by Alexandre Mouillé 07 Oct '14

07 Oct '14

Hi, I hope I'm not posting this in the wrong mailing list. I'm working on a custom conference that uses the conference module. I need a record function that keeps recording (or at list appends the same wav file) when no one is in a given conference. The solution that was used in the Asterisk program I'm porting was to create a dummy channel that goes in the conference and records what it hears in a file. - Can I create a dummy channel with an "originate /loopback" that would go into a conf and record what it hears (I'm not sure what loopback really is) - Otherwise, can I use originate from the FS box to call itself, one side connects to the conference and the other records. (I'm not sure how to do that, and how I could transmit variables (to tell which file to record to) to the called program) Thanks

5 8

Goip GSM Gateway works great with FreeSwitch!
by Dissident 12 Sep '14

12 Sep '14

http://freeswitch-users.2379917.n2.nabble.com/file/n6691087/single-channel-… Hi, I've configured this GSM gateway with Freeswitch and it works great. I'm posting this so that if somebody tries to configure this Gateway won't have to spend as much time as a I did :) This is the configuration: *Gateway* *freeswitch/conf/sip_profiles/external/goip.xml* *Tips to check that everything is OK - Remember to remove the PIN number from your SIM card. Double check your SIM card with a Phone. It should not ask you for a PIN. Disable it! Period :) - *sofia status profile internal* (the Goip extension user must appear, I chose ext. 1000 for it...) - * sofia status* (to check that the Goip gateway is properly configured. - On the Goip status page the Phone Status and your GSM Status must be LOGIN - On the Goip status page GSM operator GSM signal must show something. If it doesn't it's because it has not logged properly with your GSM Sim card. Double check it with a Phone. - If the Goip is logged properly with FS you can use your favorite SIP client to log directly into GOIP that is to say. if your GOIP IP is 192.168.2.2 you can log into 1000(a)192.168.2.2 *(again this is not your FS IP is the GOIP IP)* and make an outbound call to an ordinary cell phone. This is useful in case you have trouble with the outbound call. * Things to find out :?? - First I used X-Lite, Windows XP and miniSip server 20 clients to test this gateway. It's kind of strange that with miniSip every time I call to the extension I chose for Goip a tone appears an then I can dial a cell phone number. FS says that this extension is not available, it shows up with the sofia status profile internal though. *I've used with Linux Ubuntu: - FreeSWITCH Version 1.0.head (git-decfdbb 2011-08-11 14-15-26 -0500) - Jitsy Java Sip Client - 3 Android Phones with CSipSimple, Sipdroid and 3CXPhone Sip Client - 2 linksys spa941 - 1 Goip GSM GateWay http://goip.com.ua/download/goip-series-manual.pdf http://goip.com.ua/download/goip-series-manual.pdf http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… I hope you find this info useful, if you have any comments or requests or corrections... please let me know... best regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 5

Skype is likely to stop supporting desktop API by Dec. 2013
by Henry Huang 12 Sep '14

12 Sep '14

Here is the link: http://www.engadget.com/2013/11/02/skype-killing-old-api/?ncid=rss_truncated I don't know if this mean that we'll no longer be able to interconnect with Skype anymore. And I just want to share the news with this community. Henry

8 8

Params to FS Native.Api() changed. How to get handle of CoreSession?
by Srini K 10 Sep '14

10 Sep '14

Hi, In older version of FreeSWITCH I have been using FreeSWITCH.Native.Api(). For eg, string uuid = new FreeSWITCH.Native.Api().ExecuteString("create_uuid"); activeCalls = new FreeSWITCH.Native.Api().ExecuteString("show calls") I updated FS to latest version. Now Native.Api expects CoreSession as param. FreeSWITCH.Native.Api(FreeSWITCH.Native.CoreSession) How to get handle of CoreSession. Iam using mod_managed. Thanks in advance. Srini

4 8

Outbound calling using originate
by Malay Thakershi 04 Sep '14

04 Sep '14

Hello, I use originate to make outbound calls. originate {origination_caller_id_name='CH calling',origination_caller_id_number=1XXXXXXXXXX,origination_uuid=XXXXXXXX,absolute_codec_string=PCMU,ignore_early_media=true,originate_timeout=45,}sofia/gateway/chsbc01/1XXXXXXXXXX 1056 XML default I would like people to hear a couple of rings before they hear the first voice prompt. Would increasing originate_timeout value allow users longer to answer (thus delaying going to VM)? Thanks. In public.xml dial plan: <extension name="provider-inbound"> <condition field="destination_number" expression="^XXXXXXXXXX|XXXXXXXXXX$"> <action application="pre_answer" /> <action application="sleep" data="1000" /> <action application="transfer" data="1056 XML default" /> </condition> </extension> In default.xml dial plan: <extension name="CHPhone"> <condition field="destination_number" expression="^1056$">  <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 1980 Factory Records, Inc."/> <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/> <action application="set" data="RECORD_ARTIST=CH"/> <action application="set" data="RECORD_COMMENT=CH call record"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="recording_follow_transfer=true"/> <action application="sleep" data="2000" /> <action application="managed" data="clsAsmtApp" /> </condition> </extension>

4 4

mod_curl setting content type header
by Jeff Stark 04 Sep '14

04 Sep '14

Hi, Is there a way to set the Content-Type header when using mod_curl in a Lua script? Using curl on the command line I can do the following: curl -H "Content-Type: application/json" -H "Accept: application/json" -X POST -d '{"field1":"val1","field2":"val2"}' http://mydomain.com/ And in a Lua script I can do the following: local my_json = "{\"field1\":\"val1\",\"field2\":\"val2\"}" api = freeswitch.API() response_data = api:execute("curl", "http://mydomain.com/ post" .. my_json) But I don't see a way in the freeswitch API to set the Content-Type header in the above script. Appreciate any help or insight that can be offered. Thanks. Jeff

3 4

Inband DTMF Woes
by dotnetdub 02 Sep '14

02 Sep '14

Hi List, We are using a carrier who insists on sending DTMF INBAND. I was using the start_dtmf application in my dialplan but this was generating a lot of talk off. I've switched to spandsp_start_dtmf application and this seems to work without any issue.! My next issue is that we have a customer who is using asterisk and an IVR that ends up in the voicemail application. Freeswitch seems to be stripping the 'BEEP' out of the RTP that the voicemail application sends. It does it a bit randomly - and I initally thought it was the spandsp start dtmf application. I've been playing with the thresholds and twists etc but it doesn't improve. I've captured the RTP and the leg from the asterisk PBX has the voicemail 'BEEP' but the leg to the carrier doesn't. I've done /event plain all when the server isn't busy to see if I can spot something happening when the BEEP is in the RTP but nothing comes up. It doesn't seem to be the span dsp dtmf stuff causing this. I'm at a bit of a loss. Anyone any ideas? Thanks, Brian

4 3

Not able to send SIP chat
by Juan Antonio Ibañez Santorum 28 Aug '14

28 Aug '14

Hello, I am doing some test to send SIP MESSAGE using chat api without success. SIP domain is 10.3.1.231 Destination reg info is: Call-ID: E950F7906ACA4DC4B2CB5ADEC4C4882D0x0a030191 User: cabinacr1(a)10.3.1.231 Contact: "user" <sip:cabinacr1@10.3.1.145> Agent: SJphone/1.65.377a (SJ Labs) Status: Registered(UDP)(unknown) EXP(2013-08-22 10:23:00) EXPSECS(294) Host: fscentos6 IP: 10.3.1.145 Port: 5060 Auth-User: cabinacr1 Auth-Realm: 10.3.1.231 MWI-Account: cabinacr1(a)10.3.1.231 Command 'chat sip|cabinacr2@10.3.1.231|cabinacr1@10.3.1.231|Hola' returns 'Error! Message Not Sent' Command 'chat sip|sip:cabinacr2@10.3.1.231|sip:cabinacr1@10.3.1.231|Hola' returns 'sofia_presence.c:4309 Not sending message to ourselves!' What is the correct way to send the message? Regards

2 1

multiple sip gateway configuration
by Louie Liu 13 Aug '14

13 Aug '14

Hi, There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions: 1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that's case, what do I need to change in the sip profile external.xml file? 2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? 3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good. External.xml <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> External/sip_provider_A.xml <gateway name="Sip_Provider_A"> <param name="username" value="xxx"/> <param name="password" value="xxx"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> External/sip_provider_B.xml <gateway name="Sip_Provider_B"> <param name="username" value="aaa"/> <param name="password" value="bbb"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> Cheers, Louie

9 14

PD: SIP -> XMPP calls
by Muszalski Michał - Korpo TP 12 Aug '14

12 Aug '14

> Hello, > > We are trying to make some kind of a gateway between SIP and XMPP domain. > We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is > registered as a component in OF). Calls from XMPP to SIP are working fine. > The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing > tone, a the callee (XMPP) has a 'connecting...' message after answer the > call. > Do you have any experience in making calls between SIP and XMPP users? Is > it possible to establish the video calls also? > > Best Regards, > > Michał Muszalski > Senior Specialist > > Orange Labs Poland > Obrzeżna 7, 02-691 Warsaw >

4 4