July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

TLS setup failed (after changing key type)
by mehroz 22 Aug '13

22 Aug '13

Hi all, My SIP over TLS is working perfect as long as i used gentls_cert script. I have changed script to create ECC keys and changed that script replacing with : openssl ecparam -name secp160r2 -out CA_CURVE.pem openssl req -out "${CONFDIR}/CA/cacert.pem" -new -x509 -keyout "${CONFDIR}/CA/cakey.pem" -newkey ec:CA_CURVE.pem -config "${TMPFILE}.cfg" -nodes -days ${DAYS} -sha1 >/dev/null openssl req -new -out "${TMPFILE}.req" -newkey ec:CA_CURVE.pem -keyout "${TMPFILE}.key" -config "${TMPFILE}.cfg" -nodes -sha1 And since then, i see "TLS setup failed"... any one? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/TLS-setup-failed-after-changi… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 2

How to compile and install the prerequisites gsmlib & libctb for freeswitch to use gsm modem
by Ashish Mishra 22 Aug '13

22 Aug '13

I have seen the prerequisites of making freeswitch use gsm modem to make call but cannot actually figure out how to compile and install gsmlib & libctb...kindly help

2 1

Codec order flipped in proxy-media mode re-INVITE
by Jeff Pyle 16 Aug '13

16 Aug '13

Hello, FreeSWITCH Version 1.3.17+git~20130318T211211Z~2dc3b47db1. The config is a basic SBC to handle NAT traversal between public and private interfaces. Proxy-media mode is configured. Inbound call from carrier hits us with G711u and G729 in the SDP, in that order. Inside PBX chooses G711u and call is established. 30 minutes later the carrier does a session refresh with a re-INVITE with an identical SDP as the initial INVITE. When FreeSWITCH the re-INVITE leaves FreeSWITCH towards the inside PBX, the attached SDP has G711u and G729 are flipped such that G729 is now preferred. The PBX accepts the G729 request and the session switches to G729. There are no SDP smarts in the dialplan. Codecs are not handled at all by design. Is this something that could be caused by a misconfiguration? Or, is my next step a jira? - Jeff

2 3

Bypass_media_after_bridge doesn't works
by Carlo Dimaggio 08 Aug '13

08 Aug '13

Hi all, I have an installation with several phones (yealink) behind NAT and freeswitch withaggressive-nat-detection=true in internal SIP Profile. In dialplan I set bypass_media_after_bridgein order to eliminate on-way audio problems (and use less bandwidth), but I'm encountering a strange issue. When I call two extensions, sometimes freeswitch bypassthe media between them (so the SDP sent by freeswitch have the private IP) and sometimes freeswitch send the SDP with the NAT IP(in this case there is no audio). What could be the problem? Is this a bug or my misconfiguration? Best regards, Carlo Dimaggio

2 2

How to attche custom "content-type" in Freeswitch SIP message
by xiaofengcanyuexp＠163.com 05 Aug '13

05 Aug '13

Dear support, I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example. It firstly addresses "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can write private "content-type" like "application/sdp". Now I can see the application/sdp is encapsulated via variable "switch_r_sdp". Is there anyway to encapsulated other customized "content-type"? Appreciate to get your reply. --------------------------------------------------------------------------------------------------------------- Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup". INVITE sip: 87896677(a)dance.com;user=phone;SIP/2.0 From: "Caller"<sip:678923001@skynetwork.com> To: <sip:87896677@dance.com>;user=phone Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP(a)skynetwork.com Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR MIME-Version: 1.0 Content-Length: 433 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/sdp User-Agent: ENSR2.5.46.6-IS2-RMRG36-RG20-CPO487 Content-Length: 142 v=0 o=- 1706944438 1706944438 IN IP4 192.168.1.105 s=ENSResip t=0 0 m=audio 6793 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/isup; version=ansi;base=ansi00 Content-Disposition: signal; handling=optional 01 00 60 01 0a 03 05 0b 02 c0 90 06 03 10 78 98 66 77 0a 07 83 13 76 98 32 00 0f 00 --QRLVLNKeKxWDHAuwlEkR-- Thanks ------------------- 2013-05-25

3 2

Issue with export_on_answer with Sangoma A108/A108
by DP Siddharth 05 Aug '13

05 Aug '13

Hi All, I am trying to bridge a call & want to play a sound file when legb answers. here is dialplan: <extension name="incoming"> <condition field="destination_number" expression="^4948667$"> <action application="answer"/> <action application="bridge" data="{execute_on_answer=playback /srv/sounds/oriental-insurance-deepawali-demo.wav}freetdm/1/a/9873244228"/> <action application="hangup" /> </condition> </extension> Here I am using Sangoma card & libsng_isdn library (I have tested with latest stable FreeSWITCH, wanpipe driver & libsng_isdn). FreeSWITCH: FreeSWITCH Version 1.0.head (git-9bf2726 2012-03-14 23-26-09 -0500) I have updated with v1.2.stable but same observation WANPIPE Release: 3.5.28 libsng_isdn.so.9.9.9, I have now updated it with libsng_isdn.so.7.27.2 I am getting following error: 2013-07-22 12:44:53.725215 [WARNING] ftmod_wanpipe.c:1033 [s1c2][1:2] Failed to read 4096 bytes from sangoma device: No buffer space available (-65) 2013-07-22 12:44:53.725215 [WARNING] ftdm_io.c:4059 [s1c2][1:2] raw I/O read filed 2013-07-22 12:44:53.725215 [WARNING] mod_freetdm.c:776 Failed to read from channel FreeTDM/1:2/09873244228 device 1:2! 2013-07-22 12:44:53.805207 [WARNING] ftmod_wanpipe.c:1033 [s1c2][1:2] Failed to read 4096 bytes from sangoma device: No buffer space available (-65) 2013-07-22 12:44:53.805207 [WARNING] ftdm_io.c:4059 [s1c2][1:2] raw I/O read filed I also raised this issue with sangoma ( http://support.sangoma.com/index.php?/Default/Tickets/Ticket/View/9246) but got no conclusive response in last 6 months. Kindly help in resolving same. thanks & regards, Siddharth

2 2

FreeSWITCH Utilization in Government and Municipalities
by McKnight, Jeffrey 01 Aug '13

01 Aug '13

I am implementing FreeSWITCH in my county in the very near future. I have an approved preliminary plan to roll out a couple of hundred VOIP phones, but need case studies of FreeSWITCH utilized in government or municipal applications in order to proceed on a larger scale. Fire departments, law enforcement, emergency operations, and 911 applications would also be helpful. It appears that some feel that government is so different than commercial applications, that it warrants further investigation. If you are involved in or are aware of specific FreeSWITCH applications in these areas, I would appreciate any details that you could provide. I will be attending ClueCon next week in Chicago and will continue to solicit contacts while there. Hope to see you there. I appreciate your time and consideration. Regards, Jeff Jeff McKnight, PMP(r) IT Telecommunications Manager Brevard County BoCC 321 633-2135 Jeffrey.McKnight(a)brevardcounty.us <mailto:Jeffrey.McKnight@brevardcounty.us> <javascript:ClickThumbnail(37)> ----------------------------------------- Under Florida Law, email addresses are Public Records. If you do not want your e-mail address released in response to public record requests, do not send electronic mail to this entity. Instead, contact this office by phone or in writing.

5 4

Sending NOTIFY using sendevent / ESL
by Martyn Davies 01 Aug '13

01 Aug '13

I've been trying to generate a NOTIFY using sendevent, following some of the examples in http://wiki.freeswitch.org/wiki/Mod_event_socket and also various stuff found in Jira and historic message trails. I can't get it to do anything. If I switch on events for notify, I can see that the sendevent command generates some kind of message (and the fields seem to have appropriate URIs and so on), but at the SIP level nothing at all happens (I've got wireshark at the client end). I've switched on all events on fs_cli, and sofia trace but see nothing to suggest that a notify is sent. I'd appreciate any suggestions that could make a message flow. Failing that, what logs can I switch on to give a clue about where/why the message is discarded before making it to SIP? I was using FS 1.5.1, just changed to 1.5.3 today (no difference). Client software is brand-new Mac Blink client bought last week. Regards, Martyn

3 5

Just curious, is there any standard for FS API command output?
by Rafal Gwizdala 01 Aug '13

01 Aug '13

Guys, I'm communicating with FS through an inbound event socket and almost every command has some different standard for output data structure. Some responses are CSV, some (like 'sofia status profile internal reg') use a 'vertical name:value format , i've seen also different formats I don't remember now. What's worse, the event returned after executing a command (SOCKET_DATA api/response) doesn't indicate if the command succeeded or not. Some commands are nice enough to return +OK for success but not all of them. Am I missing something here? Are there more programmer-friendly methods of getting command results from FS? Thanks RG

3 4

What ports are really necessary?
by Jim Lynch 31 Jul '13

31 Jul '13

I'm running a simple switch. No phones are connected to the system from outside of my local network. The only wan connections I have are a couple of voip providers. None of my phones on the lan call outside of the lan. All they talk to is the switch. I see a bunch of ports listed at http://wiki.freeswitch.org/wiki/Firewall but don't know if I need them all. I suspect I don't. I want to close up the firewall as much as I can. Thanks, Jim.

6 10