October 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Manual generation of CDRs
by Dan Christian Bogos 13 Oct '14

13 Oct '14

Neven, It is important to not misunderstand the sip looping I am proposing with loopback endpoint of FreeSWITCH. In terms of consequences, I guess it depends on how you do it but if you do not proxy media through that profile and maybe use even another sip profile for the looping (goes in separate thread) I think it is safe to say that very small processing penalties occurs extra. In terms of how you do it, you need to see it as a normal call generation, something like this: sofia/internal/1000@mydomain.com;fs_path=sip:127.0.0.1 DanB On 13.10.2014 16:19, freeswitch-users-request(a)lists.freeswitch.org wrote: > @Dan: > Please can you tell me is it looping of call making disturbance to it or > there are some consequences to it? > So does this mean that I can do something similar to: sofia/internal/ > 1000@mydomain.com;fs_path=sip:1000@mydomain.com > Sorry for this stupid question, but is this how to make loopback on call?

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Re: [Freeswitch-users] Manual generation of CDRs
by Dan Christian Bogos 13 Oct '14

13 Oct '14

Neven, It is important to not misunderstand the sip looping I am proposing with loopback endpoint of FreeSWITCH. In terms of consequences, I guess it depends on how you do it but if you do not proxy media through that profile and maybe use even another sip profile for the looping (goes in separate thread) I think it is safe to say that very small processing penalties occurs extra. In terms of how you do it, you need to see it as a normal call generation, something like this: sofia/internal/1000@mydomain.com;fs_path=sip:127.0.0.1 DanB On 13.10.2014 16:19, freeswitch-users-request(a)lists.freeswitch.org wrote: > @Dan: > Please can you tell me is it looping of call making disturbance to it or > there are some consequences to it? > So does this mean that I can do something similar to: sofia/internal/ > 1000@mydomain.com;fs_path=sip:1000@mydomain.com > Sorry for this stupid question, but is this how to make loopback on call?

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First setup, Cannot hear anything
by Delifisek Tux 13 Oct '14

13 Oct '14

Hello, I'm trying to build Freeswitch 1.4 from source on debian 7.6 After setup, I try to do some test. However I cannot hear anything. I barrely modify default setup, (just change default ip, password etc) I try to do some test, everhing ok on log, fs_cli. no Errors lots of progress debugs. However I cannot hear anything from sip phone Like calling 9664 does not produce any sound. Even I try ther wav files no sound. What can be wrong... My best regards.

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Help on header condition
by Chris B. Ware 13 Oct '14

13 Oct '14

Hi, on mod_sofia.c there's this line of code: header = (tech_pvt->cid_type == CID_TYPE_RPID && !switch_stristr("aastra", ua)) ? "Remote-Party-ID" : "P-Asserted-Identity"; If you have an Aastra phone, you get header = "P-Asserted-Identity". Since I've snom phones and I'd like to get header = "P-Asserted-Identity", which var should I set to do it? I've tried setting sip_cid_type=pid before bridge, inside bridge and exporting it also. No luck. I suspect my compiler (gcc version 4.7.2) execute AND before ==! :( Any help?

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update freeswitch time+date
by Shlomi Agiv 13 Oct '14

13 Oct '14

In our system, when the freeswitch goes up, the time and date is 1970, and only later it's updated. I noticed that the freeswitch doesn't update the time accordingly. Is there a way to tell the freeswitch to get the time from the linux, or alternatively, a command to tell it to sync the time? Thanks, Shlomi Agiv

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Incoming call ignored by SIP client
by Gavin Hamill 13 Oct '14

13 Oct '14

Hi all, I'm running FS 1.4.9 on a CentOS 6.5 machine with a single NIC. The machine is behind NAT, and I have a working VPN so that internal calls work. The local IP of the machine is 192.168.196.101, and the remote VPN subnet (no NAT - clear routing) is 192.168.41.0/24. I am using port 5080 for internal calls (auth-calls=true) and port 5060 for incoming calls from a SIP provider. The machine has an internal IP and port-forwarding handles public IP -> private IP. When I dial from a mobile phone to a number on the external SIP provider, I do get activity on the console: 2014-10-10 10:52:20.655471 [NOTICE] switch_channel.c:1055 New Channel sofia/external/07951357XXX(a)87.238.73.181 [78cc73d0-e9d1-4d7d-9f22-f7adac8ba0c0] 2014-10-10 10:52:21.259045 [NOTICE] switch_ivr.c:1844 Transfer sofia/external/07951357xxx(a)87.238.73.181 to XML[106@default] 2014-10-10 10:52:21.259045 [WARNING] mod_dptools.c:1628 incoming call from [07951357XXX] to [106] 2014-10-10 10:52:21.259045 [NOTICE] mod_sofia.c:2237 Pre-Answer sofia/external/07951357XXX(a)87.238.73.181! 2014-10-10 10:52:21.259045 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:106@192.168.41.198:5060 [a7aaa63b-fd85-4b1f-8855-180a218a388e] At this point there is a pause and a timeout after 30 seconds. There is a phone registered to extension '106' listening at that IP address on port 5060: freeswitch@internal> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 1572302163(a)192.168.41.198 User: 106(a)freeswitch.bashton.eu Contact: "" <sip:106@192.168.41.198:5060> Agent: qutecom/rev-g-trunk Status: Registered(UDP)(unknown) EXP(2014-10-10 10:32:04) EXPSECS(2796) Host: freeswitch IP: 192.168.41.198 Port: 5060 Auth-User: 106 Auth-Realm: freeswitch.bashton.eu MWI-Account: 106(a)freeswitch.bashton.eu Total items returned: 1 ================================================================================================= freeswitch@internal> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.196.101 SIP-IP 192.168.196.101 URL sip:mod_sofia@192.168.196.101:5060 BIND-URL sip:mod_sofia@192.168.196.101:5060;transport=udp,tcp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN PCMU,PCMA CODECS OUT TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 2 FAILED-CALLS-IN 1 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts Dialplan XML Context default Challenge Realm auto_to RTP-IP 192.168.196.101 Ext-RTP-IP 192.168.196.101 SIP-IP 192.168.196.101 Ext-SIP-IP 192.168.196.101 URL sip:mod_sofia@192.168.196.101:5080 BIND-URL sip:mod_sofia@192.168.196.101:5080 ;maddr=192.168.196.101;transport=udp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN PCMU,PCMA CODECS OUT PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 REGISTRATIONS 1 freeswitch@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@192.168.196.101:5060 RUNNING (0) external::magrathea gateway sip:bashton@sipgw.magrathea.net REGED internal profile sip:mod_sofia@192.168.196.101:5080 RUNNING (0) ================================================================================================= 2 profiles 0 aliases The SIP client is definitely listening: gdh@gdh-e7440:~$ netstat -anp | grep 5060 tcp 0 0 192.168.41.198:5060 0.0.0.0:* LISTEN 17566/qutecom udp 0 0 192.168.41.198:5060 0.0.0.0:* 17566/qutecom And I can see that when the call comes in, there is traffic received by the SIP client (tcpdump + wireshark): INVITE sip:106@192.168.41.198:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.196.101:5080;rport;branch=z9hG4bKX71j6K414rQNa Max-Forwards: 67 From: "None " <sip:07951357XXX@freeswitch.bashton.eu>;tag=rBeB1Nv1Bampm To: <sip:106@192.168.41.198:5060> Call-ID: f850b9a6-cb05-1232-7693-02b891c191f6 CSeq: 66131306 INVITE Contact: <sip:mod_sofia@192.168.196.101:5080> User-Agent: FreeSWITCH-mod_sofia/1.4.9~64bit 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: 253 X-CALLINFO: cdr=5437AC53=94F6B14; X-FS-Support: update_display,send_info Remote-Party-ID: "None " <sip:07951357XXX@freeswitch.bashton.eu >;party=calling;screen=yes;privacy=off The incoming call is handled in the public context with: <action application="transfer" data="106 XML default"/> and in extension 106 in the default context, there's this: <action application="bridge" data="user/$1"/> I have tried both qutecom and linphone with the same behaviour. Any ideas? Cheers, Gavin.

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Change some variables.
by Mimiko 13 Oct '14

13 Oct '14

Hello. How can I change this variable: Caller-Username: Caller-Caller-ID-Name: Caller-Caller-ID-Number: Caller-Orig-Caller-ID-Name: Caller-Orig-Caller-ID-Number: Caller-ANI: Other-Leg-Username: Other-Leg-Caller-ID-Name: Other-Leg-Caller-ID-Number: Other-Leg-Orig-Caller-ID-Name: Other-Leg-Orig-Caller-ID-Number: Other-Leg-ANI: in bleg? This contain some sensitive data which I don't want to send to external gateways of voip providers. -- Mimiko desu.

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Hook flash/ flash support in FreeTDM
by Kalyani Kulkarni 13 Oct '14

13 Oct '14

Hello, I have a Sangoma A200 with 2 FXS and 2 FXO ports plugged into FreeSwitch and configured through wanpipe and FreeTDM. Incoming and outgoing calls work fine controlled through ESL. FS version is 1.4.8 wanpipe driver: 7.0.8 libsng_isdn: 7.29.0 for x86_64 machine OS: centos 6.4 I'm testing call hold/swap functionality. I'm using send_dtmf api to send 'F' followed by the number to simulate flash and dial a number on FXO line but the FS log shows following error: 2014-09-10 11:13:29.425811 [DEBUG] ftdm_io.c:3983 [s2c1][1:1] Generating DTMF [F1xxxxxx6146] 2014-09-10 11:13:29.425811 [ERR] ftmod_wanpipe.c:909 [s2c1][1:1] Wanpipe failed to execute command 18: Success 2014-09-10 11:13:29.425811 [ERR] ftdm_io.c:3576 I/O backend does not support command 18! command 18 corresponds to FTDM_COMMAND_FLASH but looks like it is not supported. On FS mailing list I saw one post where it was mentioned that this support is present but not clear whether it works end to end ( http://lists.freeswitch.org/pipermail/freeswitch-users/2010-September/06264…) What could be wrong? I've set analog-options in freetdm.conf.xml. Has anyone experienced this? Any suggestions? Thanks, Kalyani

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Manual generation of CDRs
by Neven Vrenko 12 Oct '14

12 Oct '14

Hi guys, I'm new to Freeswitch and I have one question. It's regarding CDRs, what I would like to do is: 1. Customer calls number 2. IVR responds and prompts for PIN 3. PIN is provided by customer 4. NEW CDR IS GENERATED FROM DIAL PLAN 5. Call proceeds Is it possible to somehow generate new CDR "manually" from DialPlan? If not what is your proposal for solving this? Thank you in advance, Neven

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Change language of voice conference announcements to different language
by flow27＠meta-matrix.de 12 Oct '14

12 Oct '14

Hello, I try to make the pre-recorded voice conference announcements available in English as well as in German. From an old howto I found a few steps. I downloaded the German sound file and extracted it into freeswitch/sounds/de/at/ute/... then I made this adjustment to freeswitch/conf/lang/de/de.xml: <include> <language name="de" say-module="de" sound-prefix="$${sounds_dir}/de/at/ute" tts-engine="cepstral" tts-voice="david"> <phrases> <macros> <X-PRE-PROCESS cmd="include" data="demo/demo.xml"/>  <X-PRE-PROCESS cmd="include" data="vm/tts.xml"/> </macros> </phrases> </language> </include> and I made sure the de.xml is included in freeswitch.xml: <section name="languages" description="Language Management"> <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/> I was hoping that by setting the default_language parameter: <action application="set" data="default_language=de"/> before calling the conference application I could exchange the English annoucements with the German ones but this does not work. I read in the docs that the default language parameter is mainly for the say-module but which parameter then selects the language of the pre-recorded announcements? Is the German say-module even required when I dont want to use the say command at all but just want to playback the German soundfiles? In the logfile I see that the language is changed: 83d9e7fc-5212-11e4-a335-2d65638c2a61 Dialplan: sofia/external/04124124124(a)myprovider.de Action set(default_language=de) But when the conference starts it is using the English sound prefix: 2014-10-12 15:20:34.503660 [INFO] mod_conference.c:10305 using channel sound prefix: /opt/freeswitch/sounds/en/us/callie How can I change the voice conference annoucement language without simply replacing the audio files because I want to have the English announcements also available for English-only participants. Thanks for any help

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