Re: [Freeswitch-users] Server Configuration for 50 concurrent sessions
Amplify Query... not enough data to make a logical compilation of requested data. /b On Dec 30, 2009, at 12:17 AM, Sharad wrote: > Hi > > I just want to know what should be the approx configuration of the server > for 50 concurrent call sessions having 3000-4000 users. > > Regards ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
You should join IRC and join in JM and really start the official softphone project. /b On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > I had wrote a Air based GUI, is it make sense? > > http://wiki.freeswitch.org/wiki/FsAir ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
Sounds like a plan to me... who wants to take the lead on the project... we'll host it.. setup SVN, provide jira access, fisheye and wiki space... /b On Dec 29, 2009, at 6:44 PM, João Mesquita wrote: > Why don't we evolve FSGui to be a softphone? I could use a couple of > experienced programmers to help out with it since I pretty much suck at it... > FSGui is extensible using plugins so I think that a softphone would be > nothing more then just another plugin. > > Math? Would you like to join in there and put a bit of work with me? > > João Mesquita > FreeSWITCH™ Solutions > t: +1 (646) 4959927 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Cisco 501's
Anyone have access to these phones? Two of them if possible and provisioning information? Thanks, Brian ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
Guessing the biggest issue is I want to create a softphone project using FreeSWITCH as the core of the project... is this something people would be interested in joining? /b On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > I am using iSoftPhone, works great with FreeSWITCH. > > Ivan > > Den 29. des. 2009 kl. 22.14 skrev Brian West: > >> Does it only do IAX? If so we'll need someone to re-write an IAX2 stack >> since the libiax2 from Digium is no longer updated to keep pace with >> Asterisk and is now incompatible. Which is the main reason we are thinking >> about dropping IAX support unless someone writes a license compatible lib or >> updates and takes over mod_iax aka owns it as their own. >> >> /b >> >> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: >> >>> There is one softphone for OSX that doesn't suck. It's called >>> JackenIAX. > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. /b On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: > There is one softphone for OSX that doesn't suck. It's called > JackenIAX. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bypass Media True Disables MOH
But it doesn't go back to bypass after Maybe you can post a bounty for that functionality. /b On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote: > > When I uncomment the following tag, internally held calls no longer hear > MOH. > > > > Is there a way to have the above uncommented and still provide MOH to held > calls? > > Best Regards, > Jerry > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
I would love to have a FreeSWITCH based softphone for all three platforms... I just feel a project like that would be kick ass. Must work on 32bit and 64bit of Windows, Mac and Linux ... and not suck like most softphones do. /b On Dec 29, 2009, at 2:08 PM, EdPimentl wrote: > Add me to the app list I use MAC mostly ... > > Also can you list the new (better, gentler) list of commands to install > FreeSwitch on a MAC OSX ... ? > > Thanks in advance, > -E ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
Ivan, I have been trying to gather up everyone to start a FreeSWITCH based softphone project for Mac, Linux and Windows... you think we could collaborate with you to accomplish this? I think if we do this right we can have a really nice phone with lots of options. Thanks, /b On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my > intel Macs with great success. > I am also developing a GUI application using Cocoa. I started that a year > ago, but haven't looked at it for a while, but this Christmas I have started > working on it again. > > Ivan > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Hold is broken in trunk 16055
Its null because the device on the other side didn't send one. We pass it as
is... fix the broken device or don't use proxy media.
/b
On Dec 29, 2009, at 9:37 AM, Lei Tang wrote:
> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
> code in sofia.c send the 200ok response
> sofia.c
> function sofia_handle_sip_i_state
>.
> switch(ss_state)
>
> case nua_callstate_received:
> .
> else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) &&
> !r_sdp) {
> nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END());
> sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE);
> goto done;
> }
>
> The cause is r_sdp is null, but I don't known why tl_gets don't return remote
> sdp tag, it's quite strange.
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Re: [Freeswitch-users] Hold is broken in trunk 16055
the 200ok is not from FS.. its from the end point... so its not us thats not putting the SDP into the 200ok but the device you're talking to because in proxy media they are passed as is. /b On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: > Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip > agent I'm using is x-lite and wxCommunicator. > I will test if trunk 16055 work when I set proxy media mode to false tomorrow. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Hold is broken in trunk 16055
Also can you join #freeswitch-dev, include full siptrace+debug log and put it on pastebin. What phone are you using? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > Hi, I think hold function in trunk 16055 is broken, I have also tried some > old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any > sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both > included. > > sip trace for trunk 16055 > re-invite request sent to fs when client hold the line > INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ==sip trace for fs 1.0.4 > =re-invite request sent to FS when client want to hold the all > INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90
Re: [Freeswitch-users] Hold is broken in trunk 16055
Hold is working fine I just tested it... I would need to see the whole dialog to see what is wrong... I tested with Polycom, Snom and Aastra. Are you doing proxy media or anything like that? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > Hi, I think hold function in trunk 16055 is broken, I have also tried some > old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any > sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both > included. > > sip trace for trunk 16055 > re-invite request sent to fs when client hold the line > INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ==sip trace for fs 1.0.4 > =re-invite request sent to FS when client want to hold the all > INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 7
Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4
I'm still not done with this I think we found a bug in the lib... Viktor fixed it today and I'm going to retry after I get done testing G729 more today! ;) /b On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote: > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn trunk, > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone > client, however I am not seeing the enrollment option popup in zfone 0.92 > build 218 on windows in front of an x-lite client. > > Any suggestions on what I should look at to troubleshoot this? > > I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile, but > until then ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Local call uses public context?
acl.conf.xml sofia profile: and Then here is an example of a user: Now save that.. restart freeswitch and you now let that user in from 1.2.3.4/32 and set the user_context to default. /b ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] twitter.com/freeswitch (its not ours)
Dear FreeSWITCHers, Someone has registered the freeswitch name and is squatting on twitter with it. They haven't used it in over a year and I would like to have this for our project as its clearly confusing. If you own this account please contact me off list. Thanks, Brian ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SNOM shared lines with TLS problems?
Shared will require some testing with TLS. I need traces, console logs and you to do some foot work to see if you can provide more details. /b On Dec 24, 2009, at 8:35 AM, Yehavi Bourvine wrote: > Hello, > > Is there anyone who is using SNOM with TLS encryption and shared lines and > it works? > > We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM > phones. The TLS is defined by adding transport=tls to the registrar field > (proxy is left blank). We noticed the following behaviour: > > With non-shared line UDP and TLS both work ok. > With shared lines UDP works ok. > with shared line TLS works as long as only one phone is registered. > After the second TLS shared line registers we get busy for this extension. > From the SNOM trace there is no incoming call attempt at all from FreeSwitch. > Anyone has this setup working and can share some tips? > > Thanks, __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after Remote SDP:
If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail. /b On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: > This is all I see and then registration fails. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Local call uses public context?
You're letting the phones register via the ACL so they never actually do a directory lookup.. domains acl is built from the cidr= attribute on the users in the directory. You have bigger problems if you can't register properly with digest authentication. What does your directory entry look like? /b On Dec 24, 2009, at 2:31 PM, Lars Zeb wrote: > Thanks for the reply, Michael. > > I tried the digest authentication using the cidr and copying the > conf/sip_profiles/internal.xml from the distribution, where > > As a result, one endpoint could not register and another was unauthorized. > > http://pastebin.freeswitch.org/11634 > > Then I went changed the context in internal.xml from public to default and > name="apply-register-acl" value="192.168.10.0/24"/> > > And the phones registered OK. So my confusion persists. > > Lars ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
"all" is no longer needed. /b On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote: > make all install sounds-install moh-install. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Local call uses public context?
2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by acl "192.168.10.0/24[]". Access Granted. Because the context is set on the profile as public... and you really didn't auth the user so user_context was never set. /b On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: > I am trying to setup a second FS box from scratch using v16048. > > What can cause a local call (81002, or 9996) to use context public? It’s a > standard vanilla install. > > http://pastebin.freeswitch.org/11629 > > Thanks, Lars > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] forcing ptime settings
You might also have to set the codec negotiation to scrooge /b On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote: > You can disable auto-adjust in the sip profile., but that might just make it > worse, no warranty: > > > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mr...@avgs.ca > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] forcing ptime settings
That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff. /b On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: > I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm > having some trouble playing .wav files into the media stream using FreeSWITCH. > > The audio either comes out really slow, or really fast. So a 60 second .wav > file is either finished playing in 90 seconds (really slow) or finishes > playing in 20 seconds (really fast). I believe this is caused by different > ptime values that are being setup in the session. In the FreeSWITCH console I > often received this error > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant > to say was 20 > > I tried forcing the codec and ptime using absolute_codec_string='p...@30i' > and it seemed to fix the really slow playback problem. > > but now I'm getting a > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant > to say was 10 > > error and in some sessions the audio is playing back too fast (at 3x the > speed). > > Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the > ptime values? Are there any other work arounds? > > > --matt ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Multitenant dialplans
Yes DNS is required for this to work properly. /b On Dec 23, 2009, at 9:43 AM, John wrote: > Still having this issue. Do separate domains need to be real fully > qualified domains, or can they just be added as in Company1, 2, 3, etc? > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RTP/RTCP media whilst recording
What does pretty much mean to you? Can you give me an exact rev? /b On Dec 23, 2009, at 8:26 AM, TTNC - Technical wrote: > Oh, I'm running pretty much the latest svn truck. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Choosing a Codec.
VMD will force a transcode anyway too. /b On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote: > My setup is as follows: > > FreeSWITCH -> SIP Trunk -> PSTN. > > From freeswitch, I'm making outbound calls using event socket via the > "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is > default settings. Using "playback" application, I'm playing a mu-law audio. > I'm also starting the "vmd" application, so that I can replay the message on > beep. > > Thanks for your suggestion on native format. I'll try it. > > Thanks, > Vinuth. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Choosing a Codec.
If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: > The audio quality is a lot different when it plays on the landline. And the > quality degrades a bit when the message played is lengthy >30s. So I thought > it would be better if I have the file in mu-law and play it as is.. > > Thanks, > Vinuth. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Choosing a Codec.
Why? You don't have to avoid it... why bother? /b On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: > My basic intent is to avoid on-the-fly transcoding, while having a high > quality audio playing on PSTN. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] "a1-has" param in gateway setting
I'm not too sure you can put an a1-hash on outbound auth. /b On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote: > Hi, > > Does any body know or has test the "a1-hash" parameter with gateway > setting? I am not sure if it is even allowed. I have the following > gateway setting but when the freeswitch starts up it simply ignores this > provider without any error message or attempt to register in the log > file. Thank you for your help in advance. > > > > > > > > > > > > > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Multitenant dialplans
The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote: > I have Freeswitch setup and working as a single tenant > system mostly using the default configuration. Trying to > convert to a multitenant environment, I have used both the > Multi-tenant and Multiple Companies wiki's. I get the phone > to register, can call out using the external profile to a > ITSP, can call music on hold; however I can not call other > users in the company. > It appears that when logged in with single company and > default context it sucessfully calls other internal phones > with bridge to > "sofia/internal/sip:exters...@public-ip:translated-port"; > however when I log into "Company1" with the phones, it tries > "sofia/internal/dialed-extens...@company1" ... I also get > "User not Registered". The dialplans are the same either > way. > > Any ideas? > > Thanks > John > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] WARNING On Inbound Call Question
You know that warning is meaningless. Search the archives we have talked about this to no end it seems. And I'm sure Moy fixed this. /b On Dec 21, 2009, at 6:24 PM, Jerry Richards wrote: > Okay, I upgraded to 1.0.5pre9 and tried this test again and I do not see the > WARNING in the Freeswitch log. However, it still behaves the same way. That > is, the internal callee rings for about 12 seconds, then stops ringing, and > the PSTN caller just hears ringback for about 60 seconds and is not given the > opportunity to leave voice mail. In contrast, an internal-to-internal call > will go to voice mail after 30 seconds. > > I put a new 11595 log into the pastebin. Is there some Sangoma Wanpipe > driver (or Freeswitch) setting that would correct this? > > Best Regards, > Jerry > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skypiax: Skype account frozen
So says the man with his Skype username in his sig! :P /b On Dec 21, 2009, at 12:37 PM, Itamar Reis Peixoto wrote: > the best answer is don't use skype. > > > > > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: ita...@ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after Remote SDP:
Can you get me siptraces please. /b On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote: > Thanks Brian and Gad, > > I have stun set and if I do a 'sofia status profile internal', I see > the external IP address of the 3102 ATA, so I assume that stun is > working correctly on the SPA3102. > > These are the options that I have set (according to the 3102 manual). > > • Handle VIA received: yes > • Handle VIA rport: yes > • Insert VIA received: yes > • Insert VIA rport: yes > • Substitute VIA Addr: yes > • Send Resp To Src Port: yes > • STUN Enable: Choose yes. > • STUN Server: stun.freeswitch.org > > I assume that is all is needed? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports?
The funny part is... it won't matter. Their are times when people post questions or issues and its well into debugging the issue before we realize "oh, you're on windows?". For the most part the windows installer is one of the most popular files on our website. /b On Dec 19, 2009, at 10:18 PM, Jason White wrote: > Gabriel Gunderson wrote: > >> Funny that you assume his desktop is running Windows (maybe it is). I >> would have guessed that the average person on this list doesn't run >> Windows on the desktop. But, what do I know? > > Some of us on the list have never run Windows on anything. > > It's Debian on my desktop, by the way, with FreeSWITCH acting as a soft-phone > via a USB head set, and also handling my Snom 320 SIP phone. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to debug TLS handshake errors?
You have to watch it with TLS. Make sure your distro didn't mess up your SSL libs due to the recent vulnerability found. I havn't tested with my polycom in a few weeks but it was working on my Polycom after I uploaded the ca cert and marked it as trusted/used on the phone. /b On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: > I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a > Polycom-501 which does not have an internal certificate, thus only one-way > certificate validation is needed. I've downloaded the root certificate to he > Polyciom, and Freeswitch gives me the following error: > > Peer did not provide X.509 Certificate > I understand that it tries to do mutual authentication which is not possible > in this case. How can I tell FreeSwitch to ignore the client's certificate? > > BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. > > Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after Remote SDP:
On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: > DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a > grain of salt. Welcome to the community. > I have a similar setup (and problem) - the wiki documentation refers to it as > "double nat". Like you, my FS and client are behind different NATs and I can > register my remote endpoint and make calls (in my case, to the the FS demo > ivr at 5000). > > Since your external endpoint (spa3102) is registering, you've likely setup > your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). Your endpoint need only insert rport and FreeSWITCH will do the right thing. > 1) Setup stun on your remote endpoint (spa3102 in your case) > 2) Add value="NDLB-connectile-dysfunction"/> to the directory xml file that > describes your spa3102 endpoint The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar. > Option 1 worked for me right away (eyebeam in my case) and, as expected, the > remote sdp had the correct (remote) IP address, since the endpoint is using > stun to correctly identify its IP address to FS. However, option 2 has not > made a difference (for me). Is it just me or is it strange that SIP works > without stun, but RTP doesn't? > > I guess I've been spoiled by the way Asterisk handles NAT and was hopeful > that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have > to tell users to setup stun on their clients. Maybe a FS user with some > experience with this type of NAT setup and these settings can help. I'd be > interested in knowing how to correctly setup remote NATted endpoints without > stun - or, at least, hear from someone that this setting works for them > without stun. > > Anyway, hope this helps you with your SPA3102. Bottom line is enable rport and use stun on the SPA and it'll just work. /b ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after Remote SDP:
You'll need to fix your device to know its IP and it should stop doing that. /b On Dec 20, 2009, at 5:58 AM, Mark Campbell-Smith wrote: > Hi! > > I'm sure this is a NAT issue, but I'm not sure what options to use. > > I have a Linksys SPA3102, NAT'd on the internet (remotely) and > connected to my FS on the otherside of the world, which is also > natted. A PAP2T is connected on the same subnet as the FS. The 3102 > registers successfully and a call can be set up from the PAP2 to the > 3102. > > However, after FS receives the Remote SDP the audio stops (ring tone > stops in my case) > > 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel > sofia/internal/sip:2...@192.168.1.3:56885 entering state > [completing][200] > 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: > v=0 > o=- 18490612 18490612 IN IP4 192.168.1.3 > s=- > c=IN IP4 192.168.1.3 > t=0 0 > m=audio 16432 RTP/AVP 2 100 101 > a=rtpmap:2 G726-32/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > I notice that the ip address in the o and c fields indicate a local IP > address. Should this IP address be an external IP address of the 3102 > instead? > > Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
Sounds like a plan. We will pursue it through the consult...@freeswith.org route. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, December 18, 2009 3:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consult...@freeswitch.org and provide us with access your test environment and let us investigate the possibility of making improvements. On Fri, Dec 18, 2009 at 2:16 PM, Brian wrote: Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of "having robots call the conference in a way that probably does not match reality". In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 A
Re: [Freeswitch-users] mod_conference scalability
Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of "having robots call the conference in a way that probably does not match reality". In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. My scenario is not a hypothetical one of having robots call the conference in a way that probably does not match reality. In fact, this will very much reflect the reality of the application Im building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. Im trying to find a real solution to a real problem. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, December 18, 2009 11:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.org <mailto:sip%3a...@conference.freeswitch.org> This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: "FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS." Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > I did a test with the trunk version for the one conference case, and > it is the same results as for 1.0.4. The audio failed at around 300 > listeners. Oddly though, it consumed less %CPU (240% instead of 300%), > and yet the audio still failed at the same number of listeners. > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.miness...@gmail.com] > Sent: Thursday, December 17, 2009 3:49 PM > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > We didn't post it anywhere but we just get overwhelmed with them and > many of them are unfounded and take up a lot of time to track down. > That does not mean you have not found a real problem but the first > step is trying trunk. > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > wrote: > > I didnt realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didnt get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.miness...@gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > > > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a fo
Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable. /b On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote: > Should I open a JIRA for this? > > Best regards > Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
I've got FS running on a 64 bit OS, and here is more info on the test procedure. I've got one server (primary) that hosts the speaker call (this is meant to be a primary conference with a few speakers, but my test simplifies this to just one speaker). I've got a second server (secondary) that hosts the conference that all the listeners go into, and I have two other servers that I use automate the listener calls. The goal is to have several secondary servers to scale the listener side of things, but for this initial test I've only got one secondary server. The primary server dials into the secondary conference server so that the listeners can hear the speaker conference on the primary server. The automated listener servers start dialing into the listener conference at a combined rate of 5 calls per second (i.e. 2.5 calls per second each). The play an audio loop that represents noise on their end, which since they are listeners, should be ignored anyway. As I ramp up the automated listener calls, I manually call into the conference from either my SIP phone, or from a land line using a DID that I have directed to the conference. All calls are using SIP with uLaw 8000hz codec. Also, I've set up the profile for the listener conference to disable many of the events: I do have caller controls for the listener, since in my production I will need to generate and handle events for listener DTMF. To compare FreeSWITCH vs Asterisk, I just swap out the secondary conference server and everything else stays the same. Brian. From: Brian West [mailto:br...@freeswitch.org] Sent: Thursday, December 17, 2009 5:20 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability What exactly are you doing I know it goes better than that.. are you using 64bit? / b On Dec 17, 2009, at 3:41 PM, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sip message logging and analysis
So is wireshark UI and its free! :P /b On Dec 17, 2009, at 6:33 PM, Chris Fowler wrote: > I’m using VQManager (there is a 30 day trial) and it’s useful for seeing who > does what / when per call; it’s very easy to install… > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Handling REFER...
Also when can we expect little KK's running around? :P Congrats on the marriage /b On Dec 17, 2009, at 6:27 PM, Michael Collins wrote: > I love it when users go all Chuck Norris and Rambo in answering their > questions AND documenting the info! Thanks KK. > > -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
What exactly are you doing I know it goes better than that.. are you using 64bit? / b On Dec 17, 2009, at 3:41 PM, Brian wrote: > I did a test with the trunk version for the one conference case, and it is > the same results as for 1.0.4. The audio failed at around 300 listeners. > Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio > still failed at the same number of listeners. > > Brian. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
Hi Dave, That was one of the questions I had in my original post, was there an alternative way to implement a single speaker, many listener case? There was a suggestion proposed to use local streams instead of the conference. I'm not familiar with it, and I'm in the process of reading the wiki and source code to see what can be done with that. Thanks, Brian. -Original Message- From: David Knell [mailto:d...@3c.co.uk] Sent: Thursday, December 17, 2009 4:07 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker. As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple audio path switching - maybe someone more knowledgeable than me will correct me if I'm wrong? I presented some stuff at ClueCon which would address this kind of simple application and ought to scale well beyond what you've seen with FS or Asterisk. It's still pretty basic [I'd do more with it if I wasn't so busy joshing with the other Brian on Facebook], and has never been deployed in anger but, if you're interested, drop me a note off-list. --Dave > I didn’t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn’t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.miness...@gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production. > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn’t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I’ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test – more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I’m doing wrong, but I don’t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:m...@jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > >
Re: [Freeswitch-users] mod_conference scalability
I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: I didn't realize there was a policy about load testing questions. What forum should I have used for this? I didn't get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production. On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:m...@jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total
Re: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl
In your case don't store them in the domain put them in the gateways tags on the profile directly. /b On Dec 17, 2009, at 2:46 PM, Paulo Vicentini wrote: > Hi, > FS was sending (while loading modules) such request: [purpose] => gateways > But I was not aware of that...so that I am replying FS with my Gateways > now... > > But now I am wondering...suppose I have 1000 domains and two different > gateways per domain (2K Gateways) > Should I reply FS request with such huge XML on startup? > > > Thanks for your backings > > Paulo ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
I didn't realize there was a policy about load testing questions. What forum should I have used for this? I didn't get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production. On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:m...@jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNS
Re: [Freeswitch-users] mod_conference scalability
Yes, while it is true that does make a profound difference but if he has many listeners and not very many talkers... just tapping into the conference and streaming that audio out would scale well. /b On Dec 17, 2009, at 1:50 PM, Steve Underwood wrote: > I don't think you have mentioned which codecs are involved. This can > have a profound effect. > > Steve ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
If you're going to have that many listeners then it would be best to use something like shoutcast to broadcast the stream out to a local stream on various different boxes... then tie the callers into a stream... when they have questions uuid_transfer them into the conf.. then back to the stream when done. This would scale to very large numbers because you could split it out into 100's of boxes if needed. /b On Dec 17, 2009, at 1:29 PM, Brian wrote: > Hi Mike, > > I didn’t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? If I > want to put this into a production environment, I would need a stable > version, which as far as I know is the 1.0.4 version. > > However, I did test on Asterisk 1.4 using app_conference, and doing the same > scenario was able to get 1 speaker and 600 listeners on a single conference > with no audio issues. The CPU at that point was just over 300%, same as where > the single conference scenario failed on FreeSWITCH with 300 listeners. I > was able to push it to over 700 listeners before I reached 400% CPU usage (I > guess maxing out my quad-core processors), and asterisk finally crashed. But > up until that point, there were no audio problems. > > I’ve read a lot about how FreeSWITCH is supposed to be more scalable than > Asterisk, but unless there is something wrong with my FreeSWITCH setup, > Asterisk was clearly the winner in this test – more than doubling FreeSWITCH > capacity in this case. Again, maybe there is something on the FreeSWITCH side > that I’m doing wrong, but I don’t see what it could be. > > Brian. > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:m...@jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] detecting rtp packet for zombie channels
Please try on SVN trunk. I might toss a PRE10 sooner. /b On Dec 17, 2009, at 1:05 PM, Juan Backson wrote: > Hi, > > I am using 1.0.4 (exported) with proxy media, and I have enable-timer = true > and minimum-session-expires=120. > > Is this the correct way of setting the sip session timers? > > thanks, > jb ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9
This would have nothing to do with receiving a 502 on sip. /b On Dec 17, 2009, at 12:08 PM, Jerry Richards wrote: > I found the issue with this. I did an svn checkout from the trunk, and then > I did a local svn export to another local folder. For some reason, the svn > export did not include the libs/openzap folder (which was not the case when > I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap > folder? > > Best Regards, > Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl
I'm going to guess you removed these lines from your profile:
parse=true causes the profile to parse the domain looking for gateways and
register them..
/b
On Dec 17, 2009, at 11:18 AM, Paulo Vicentini wrote:
> Hi,
> I am trying to define Gateways (for inbound and outbound calls via SIP
> provider) within Directory (under "internal" sample profile) using XML CURL
> But I am getting this warning:
> 2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not
> found.
>
> And
>
> sofia status gateway MyGW
> API CALL [sofia(status gateway MyGW)] output:
> Invalid Gateway!
>
>
> This is my configuration (overlook language details )
>
> ""+
> ""+
> ""+
> ""+
> ""+
>" "+
> ""+
> ""+
> ""+
> ""+
> ""+
> ""+
> ""+
> ""+
> " value=\"{presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}\"/>"+
> ""+
>
> ""+
>""+
>
> ""+
> ""+
> ""+
> ""+
> ""+
> ""+
> ""+
> ""+
> "";
>
> User id "test" is able to register and call other internal users
>
> In my sip_profiles/internal.xml I have:
>
>
>
>
>
>
>
> Can you help me with this issue?
>
> Thank you
> Paulo
>
>
> Keep your friends updated— even when you’re not signed in.
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Re: [Freeswitch-users] Mirroring wiki with wget for offline browsing?
I would rather you not do that with wget you beat the hell out of the wiki resources... how often do you do this? I would try doing a printable version. /b On Dec 17, 2009, at 10:56 AM, Fred-145 wrote: > > Hello > > I'm no wget expert, and figured I should ask here first: I'd like to > download the whole wiki using wget for off-line reading. > > Using the following didn't work: > > wget -m -np http://wiki.freeswitch.org/wiki/Main_Page > > If I move the wiki/ directory to the root directory of my web server, and > try to open http://localhost/wiki/Main_Page, FireFox tries to download the > page with this dialog box: > > "You have chosen to open > Main_Page > which is a: application/octet-stream" > > I assume wget can do this, but I don't know enough. Has someone succeeded in > downloading the whole wiki with wget and could give the right switches to > use? > > Thank you. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Voicemail->Email
What SVN rev. exactly?
/b
On Dec 17, 2009, at 10:13 AM, Oliver Schönbeck wrote:
> Hello,
>
> we are running freeswitch 1.0.trunk and are currently trying to get the
> mod_voicemail to send the received messages to the user by using exim4 on a
> debian machine.
>
> So far we followed the instructions in the wiki article (
> http://wiki.freeswitch.org/wiki/Mod_voicemail ).
>
> I added some lines to the bash script to enable some kind of logging:
> #! /bin/bash
> typeset LOG="/tmp/${0##*/}.out"
> mv $LOG ${LOG}.old >/dev/null 2>&1
> [[ -t 1 ]] && echo "Writing to logfile '$LOG'."
> exec > $LOG 2>&1
> exim4 -t -v >> $LOG
>
> If I run the script from the command line everything is working as expected.
> If the script gets called by freeswitch I get the following result in my
> logfile:
> /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault
> (core dumped) exim4 -t -v >> $LOG
>
> Has anybody seen similar effects before?
>
> Any advice whats going wrong is heavily appreciated.
>
> Thanks
>Oliver
>
>
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Re: [Freeswitch-users] How to debug TLS handshake errors?
Also what device are you using? I haven't tested with many so far... Polycom, Snom and a few others do TLS (see interop page on wiki) others do it wrong. /b On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: > You could try ssldump: > > http://www.rtfm.com/ssldump/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to set the Session Name on a SDP?
Why are you needing to change it? /b On Dec 17, 2009, at 5:21 AM, Oscav wrote: > > I just found that this is related to the username of the profile. It needs to > be set as parameter. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] detecting rtp packet for zombie channels
We need more info... svn rev, gcore, back trace and what not... please see the reporting bugs link on the wiki. http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Dec 16, 2009, at 11:53 PM, Juan Backson wrote: > Hi > > I have rtp-timeout-sec set to 300 s but I am still getting calls with > duration of 1 day long. > > Is there any other ways to check for zombie channels? > > jb ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote
Works on my CentOS 5.4 box just fine... /b On Dec 17, 2009, at 7:34 AM, Neil Patel wrote: > Hi Mike, > > This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can > setup ssh access for you to check things out. > > In case this wasn't apparent I am trying to install FS from trunk. > > Thanks, > Neil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ACLs through proxy
it needs to be an ACL from acl.conf or a ip/cidr /b On Dec 17, 2009, at 5:41 AM, Bill W wrote: > Okay, I added: to my sofia > profile and restarted sofia, and still no joy. > > I'm on FreeSWITCH Version 1.0.trunk (15764) > I've got in > the directory, but I'm still being rejected by the acl: > > 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 > Rejected by user acl 190.218.103.12/32 > > Here's what I believe is the appropriate snippet of the debug output: > http://pastebin.freeswitch.org/11531 > > Thoughts? > Thanks, > Bill ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9
Need siptrace with this type "sofia profile siptrace on" replace with
your profile.
/b
On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote:
> I upgraded to the latest 1.0.5pre9 and now if I try to call from an internal
> phone to an external number on my Sangoma PRI, I get a "502 Bad Gateway"
> reply. Below is the console loglevel 7 output. It says the destination is
> out-of-order. I'm not sure what this means. Any help is appreciated.
>
> 2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for proxy
> 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy [0]
> 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by
> acl "domains". Falling back to Digest auth.
> 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for proxy
> 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy [0]
> 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by
> acl "domains". Falling back to Digest auth.
> 2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/5...@192.168.72.141:5060
> [e58e763f-7688-4600-aa70-481bbc359f58]
> 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel
> sofia/internal/5...@192.168.72.141:5060 entering state [received][100]
> 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP:
> v=0
> o=TC 1100638826 1100638826 IN IP4 192.168.72.32
> s=session
> c=IN IP4 192.168.72.32
> t=0 0
> m=audio 1760 RTP/AVP 0 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000/1
> a=ptime:20
> a=ptime:20
>
> 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923
> (sofia/internal/5...@192.168.72.141:5060) State Change CS_NEW -> CS_INIT
> 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/5...@192.168.72.141:5060 [BREAK]
> 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/5...@192.168.72.141:5060) Running State Change CS_INIT
> 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/5...@192.168.72.141:5060) State INIT
> 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83
> sofia/internal/5...@192.168.72.141:5060 SOFIA INIT
> 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111
> (sofia/internal/5...@192.168.72.141:5060) State Change CS_INIT -> CS_ROUTING
> 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/internal/5...@192.168.72.141:5060 [BREAK]
> 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/5...@192.168.72.141:5060) State INIT going to sleep
> 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/5...@192.168.72.141:5060) Running State Change CS_ROUTING
> 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/5...@192.168.72.141:5060) State ROUTING
> 2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132
> sofia/internal/5...@192.168.72.141:5060 SOFIA ROUTING
> 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78
> sofia/internal/5...@192.168.72.141:5060 Standard ROUTING
> 2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing
> Anonymous->93491028 in context default
> Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing [default->unloop]
> continue=false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
> [default->tod_example] continue=true
> Dialplan: day of week[4] =~ 2-6 (PASS)
> Dialplan: hour[14] =~ 9-18 (PASS)
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match (PASS)
> [tod_example] break=on-false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Action set(open=true)
> Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
> [default->holiday_example] continue=true
> Dialplan: month[12] =~ 1 (FAIL)
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match (FAIL)
> [holiday_example] break=on-false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
> [default->Mediant1000] continue=false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL) [Mediant1000]
> destination_number(93491028) =~ /^8(\d+)$/ break=on-false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
> [default->SangomaPRI] continue=false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (PASS) [SangomaPRI]
> destination_number(93491028) =~ /^9(\d+)$/ break=on-false
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Action
> set(effective_caller_id_number=425740${caller_id_number})
> Dialplan: sofia/internal/5...@192.168.72.141:5060 Action
> bridge(openzap/smg_prid/a/3491...@g1)
> 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/
[Freeswitch-users] mod_conference scalability
Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] build errors :(
What SVN rev? /b On Dec 16, 2009, at 12:04 PM, RR wrote: > Hello All, > > I know you will probably ask me to check out a fresh copy from svn trunk and > all, but I assure you I have done that yet I keep getting these errors on > make: > > creating freeswitch > cc1: warnings being treated as errors > libs/esl/fs_cli.c: In function âcompleteâ: > libs/esl/fs_cli.c:440: warning: format â%ldâ expects type âlong intâ, but > argument 4 has type âintâ > libs/esl/fs_cli.c:440: warning: format â%ldâ expects type âlong intâ, but > argument 4 has type âintâ > make[2]: *** [fs_cli-fs_cli.o] Error 1 > > Any ideas? > > Thanks > RR ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Click-to-call and click-to-dial
see scripts/perl/call.cgi /b On Dec 16, 2009, at 9:59 AM, John Platts wrote: > > How can I perform click-to-call or click-to-dial in FreeSWITCH? > > Do you have any recommendations on programs capable of click-to-call or > click-to-dial from Microsoft Outlook or Microsoft Excel? > > _ > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. > http://clk.atdmt.com/GBL/go/171222985/direct/01/ > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] detecting rtp packet for zombie channels
Why not just set rtp-timeout-sec on the sofia profile and it'll do that for you. Unless something else is going on. /b On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: > Hi, > > I am having problem with around 1 % of the channels always get > zombilized. > > What I want to do is to have a background thread that regularly > check all the channels that have been in existance for like > 1 hr, > and then check to see if there is any RTP coming in and going out. > If there is no RTP, then I just hangup that channel. Does anyone > know if there is anyway to do that in a freeswitch module? Which > API can I use to accomplish this purpose? Alternatively, is there > anyway to configure freeswitch so that it will hangup the calls > where there is no media in and out for so many seconds? > > Thanks, > jb > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ACLs through proxy
use "apply-proxy-acl" on the sofia profile. /b On Dec 15, 2009, at 10:58 PM, Bill W wrote: > > However, having the proxy in the path effectively negates using IP > based > ACLS. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] PlayAndGetDigits multiple WAV files
Compile it yourself is the best bet to get the very latests and greatest code. /b On Dec 15, 2009, at 6:26 PM, Malay Thakershi wrote: Is it possible to only get updated files from the latest trunk? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] REDIRECT 503 not working
You have to be careful things like eyebeam will send the invite back to FS1 that did the redirect as if it were the proxy with the request URI as the URI you did in the 302 please post a sip trace of the entire exchange on pastebin. /b On Dec 15, 2009, at 4:00 PM, Ahmed Naji wrote: > People, > > I have a very simple call scenario where calls are hitting > FreeSWITCH, and I need to send a 302 REDIRECT to get them to go > elsewhere without answering them. > > It's for a phased migration requirement so that traffic can continue > to flow to the current site, but gets redirected to a new site. The > old site will eventually be decommissioned. > > Here is what I have in my conf/dialplan/public/test.xml: > > > > > data="sip:715...@aaa.bbb.ccc.ddd > "/> > > > > > FreeSWITCH is sending back the 302 back to the test end-point > (eyeBeam 1.5.20 build 54436), but the call is not reaching the > specified in the data portion of the redirect application. I know > it's sending it because of logs FreeSWITCH end and the info being > displayed on eyeBeam's client interface stating Call being > forwarded ...etc. ...etc. > > Has anyone had any similar experiences with a similar setup ? > > Oh, and one more thing, I have disabled firewalling on both the > proxy where eyeBeam is registered and the destination where I'm > sending the call. I have also verified that my new destination (also > a FreeSWITCH box) is accepting registrations, inviites and able to > route calls initiated by eyeBeam when directly registered on it. > > Has anyone had similar experiences ? better still, has anyone > successfully setup FreeSWITCH to be an SBC and can give me feedback ? > > Regards, > > Ahmed. > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS, CCIE > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4
if you don't have ZRTP compiled in as per the wiki it won't work... their are a few changes coming to this code soon. /b On Dec 14, 2009, at 8:01 PM, Harondel J. Sibble wrote: > Hmm, I emailed the zfoneproject folks about an hour ago asking about a > release date for zfone3 and was surprised about a half hour later > with a call > from PRZ himself. > > Here's what I got from the call > > 1) the currently released version of zfone already has support for > secure pbx > enrollment > > 2) the tivi softphone client which I am using on windows mobile and > symbian > smartphones does not yet have secure pbx enrollment support > > I contacted Tivi support and the 2.0.7 client with support for > secure pbx > enrollment is due out close to the end of the year, depending on > various > factors yada, yada, yada. > > Am I correct in assuming that connecting via a softphone (eikga) on > a windows > machine also running the latest official zfone client, and calling > 9787 > should give me more than just a message saying the following? > > 1) call is secure > 2) welcome to the zrtp enrollment agent > 3) thank you for calling (about 1-2 seconds after item 2) > > then I get a few beeps, I see the zhone client saying security is > interrupted > and the call drops. > > This is the same class of behaviour I get with the Tivi clients. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] monday build
Also Pre9 is up now. /b On Dec 14, 2009, at 1:25 PM, Jeff Lenk wrote: > > Please post back to the list if you have problems with the windows > build! > Everything is working as far as I know. > If you have an existing build you should delete the following > directories > and let the scripts download it again. > > libs\pocketsphinx-0.5.99 <- delete > libs\sphinxbase-0.4.99 <- delete ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] monday build
I do pre releases and it'll be up shortly had to fix a couple of bugs. I don't do binary releases for windows you'll have to do that yourself or wait. /b On Dec 14, 2009, at 12:45 PM, Kendall Stauffer wrote: HI I tried to build the svn last Friday and it didn’t make the sphinx dll, so I thought I would wait for the Monday build (web site says update every Monday). Is there going to be a windows build today, and or is the sphinx dll build problem fixed if I build it myself? Thanks!! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound
%23 is # so the question is should we URL decode that before routing? I thought we did... what version are you using now? /b On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: > This line is basically saying that you have a call coming from > 4165551212 and it's looking for a destination number of > %23904161234. The key here is that it is coming in the public > context so you'll need to handle the routing in conf/dialplan/ > public.xml > > What should this call be doing once it comes in to FS? > > -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Does FS support STUN by default?
You don't have to do that usually... /b On Dec 11, 2009, at 5:38 PM, Fred-145 wrote: > I'll see if I can find a utility that checks that the ports are open > after > FS is up and running. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Please test www.bkw.org/sofia_autonat_static_ip.diff /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > Thanks Mathieu, but I am on SVN r15912 now. > > Chris ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix this. /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > Thanks Mathieu, but I am on SVN r15912 now. > > Chris ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] The Building Freeswitch blog
well mod_alas.c is for the N800 Please open a jira. /b On Dec 11, 2009, at 12:19 PM, Julian Lyndon-Smith wrote: Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Does FS support STUN by default?
FreeSWITCH on windows will already poke holes in the windows firewall using upnp. Just start FS and it works. Your outer nat is a larger issue... /b On Dec 11, 2009, at 12:09 PM, Fred-145 wrote: > > One last question: Does someone know of a utility for Windows that > can check > that a NAT router supports either UPnP or NAT-PMP? I guess it's no > big deal > to write a small diagnostic by connecting to free firewall checkers > to see > if the relevant ports are open, but if it's already available... > > Thank you. > -- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] gtalk dingaling G723
Can't use G723. /b On Dec 11, 2009, at 5:02 AM, zendel fernandez wrote: > > hi! > > Pls shed some light to the below dingaling/gtalk issue. > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] windows pre compiled asr
Can you confirm its fixed now? /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: It hasn't been included as of late since I'm getting an unresolved link error during the build. I'll need someone experienced in pocketsphinx to assist with this issue: 13>ngram_search.obj : error LNK2001: unresolved external symbol _ngram_model_flush 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: 1 unresolved externals regards, Carlos On Thu, Dec 10, 2009 at 6:30 PM, Kendall Stauffer wrote: I downloaded yesterdays latest pre compiled and seems to works great, but I get invalid Asr module when trying to run pizza app. It seemed to come pre configured with pocketsphynx, anything I should know before I spend a boat load of time on it? Rest seems real good,. thatks!!! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] windows pre compiled asr
Thats being fixed today! ;) /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: > It hasn't been included as of late since I'm getting an unresolved > link error during the build. I'll need someone experienced in > pocketsphinx to assist with this issue: > > 13>ngram_search.obj : error LNK2001: unresolved external symbol > _ngram_model_flush > 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: > 1 unresolved externals > > regards, > > Carlos > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Still cant find it
Download MSVC and compile it yourself is usually the best bet. /b On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote: Ok, So I have looked around a lot now, think I have read everything carefully, and don’t see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don’t see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! ___ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] embedded freeswitch compatable hardware
On Thu, Dec 10, 2009 at 09:20:39PM -0500, Michael Jerris wrote: > As a note, we are pretty aggressive about making sure all this stuff works > right out of svn without any patches so it should be easy to port freeswitch > to most platforms now. Thats good to hear. I am guessing this means I should use a recent version. I see there is an Ubuntu archive, wondering if that will work with Voyage Linux. If not, I should be able to build from the source. Anyway I sent an email to Yawarra to ask them if the net5501 computer <http://www.yawarra.com.au/product.php?productCode=HW-NT55> is compatible with the TDM400 cards. There is something about a kit for the dual rack mount computer for the TDM400, which would be good if I had a rack, and somewhere to put a rack. So presumably this means it should work for the non-rack mount system too. -- Brian May ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] embedded freeswitch compatable hardware
On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: > Lack of OpenZAP support might be an issue, I assume that would be > required to connect to an onboard analogue port... I assume I could just > install Debian or another distribution instead though. This is another distribution I found: http://linux.voyage.hk/ It comes with Asterisk out of the box, although I suspect it wouldn't be too hard to get Freeswitch working instead. -- Brian May ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Rocks!!!!!!!!!
Use the BKW method... three to four word sentences to describe what to do... its very poetic! Or is that haiku? /b On Dec 10, 2009, at 11:53 AM, Michael Collins wrote: > > I was just thinking of some way to learn FS gradually and > effectively. The > frequent problem with wiki's, is that the quality of articles is > uneven and > they don't have a good layout. But then, writing documentation is > hard and > time-consuming :-/ > > Amen, brothah! :) > -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [vars.xml] default_password=1234?
please look in conf/directory/default/*.xml /b On Dec 10, 2009, at 7:40 AM, Fred-145 wrote: > > Hello > > I'm going through the various XML files, and noticed this first line > in > vars.xml. > > > > What is this password used for? > > Thank you ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
I have confirmed it works with Polycom, Snom and a few others polycom is the hardest to set due to having to put the ca cert into the phone... but other than that its good. /b On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: > An intermediate report: > > Audiocodes: TLS works only on outgoing requests, incoming ones are > ignored. I am waiting for Audiocodes' help in order to debug it. > SRTP: worked when no TLS is active. When TLS is active the call is > disconnected when the remote party answers. Still debugging it. > > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions > how to enable TLS from the WEB interface. > > Regards, __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] embedded freeswitch compatable hardware
Jason White wrote: > Have a look at http://www.yawarra.com.au/ > Ok, found the net5501: http://www.yawarra.com.au/hw-net5501.php And here it is assembled for you: http://www.yawarra.com.au/product.php?productCode=HW-NT55 I am not quite sure on one aspect, for extensions to work the TDM400P card requires a IDE style power connector that provides 12V, 5V, etc. Presumably this would be possible somehow with the net5501, because those voltages would be required for a HDD which seems to be supported. Anyone know what are the "Pigtail" and "DIN rail clips" options? -- Brian May ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] embedded freeswitch compatable hardware
Kristian Kielhofner wrote: > The Soekris net5501 and standard case will (I believe) take a full > height card. Then again you could use any board and get an external > SIP gateway (ATA). We don't currently support OpenZAP with FS in > AstLinux but I'd love to add support for it eventually. > Ok, I found this: <http://www.soekris.com/net5501.htm>. It looks like room for a full height card. 4 network adaptors for a Freeswitch box. Hmmm. Suspect I would only find use for one ;-) Lack of OpenZAP support might be an issue, I assume that would be required to connect to an onboard analogue port... I assume I could just install Debian or another distribution instead though. Does this require a hard disk drive to boot Linux? I am guessing that compact flash could be used instead. Alternatively, if I used an external ATA, what is a good one to use? I think Jason has already made a suggestion, if so I have forgotten. I guess I get nervous going down this approach because it will add to the latency, but then again it won't use so much CPU power either, and the Digium cards send a lot of time-critical interrupts. > I'm currently working with the FS devs on getting some issues in > trunk resolved to get cross compiling working again. Until then you > can find ISOs with FreeSWITCH and AstLInux here if you'd like to check > it out: I am curious, how do you install ISOs onto a box like the net5501? I don't see any provision for CD-ROM drives. -- Brian May ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] embedded freeswitch compatable hardware
Hello, I asked this question on my local linux user group mailing list, and got the recommendation to ask here. Anyway, at the moment I am running Asterisk on an IP04 embedded system. http://www.rowetel.com/ucasterisk/ip04.html It works well most of the time, however there are some bugs that do, under circumstances lead to less then desirable behaviour (such as on some occasions which I don't fully understand sometimes the remote system fails to generate any audio packets when there is no audio - almost like silence suppression was supported by the remote system - and asterisk fails to generate any audio packets in return; on another slower computer running the same SIP software and on the same network everything works fine; as far as I can tell the software - twinkle - doesn't even support silence suppression). I suspect at least some - if not all - of the issues I have encountered may be resolved with Freeswitch, however I don't really want to replace my small, energy efficient, embedded system, with a large, power hungry computer system. Overkill. An added complication is I need at least 1 analogue port to connect to the Australian based telephone line (2 ports exchange ports and 1 extension port would be ideal but not essiential). Unfortunately, I have been told that the IP04 hardware isn't compatable with the requirements of Freeswitch. Such as not having a MMU. So there doesn't appear to be much effort porting Freeswitch to IP04 as a result. I do have a spare TDM400p card, although as it is full height, suspect this isn't going to help. Are there any other good alternatives? Thanks. -- Brian May ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FreeSWITCH 1.0.4 Bug Reports...
Dear FreeSWITCHers, As of Friday Dec. 11th we will NOT accept any more bug reports on 1.0.4. You need to be on a 1.0.5pre or SVN trunk. 1.0.4 is over 6 months old and I really suspect your issues in 1.0.4 are already fixed. We will release a new pre every monday morning till 1.0.5 is released please keep up to date if possible. We are working hard to get 1.0.5 out and be as stable as possible and its more stable than 1.0.4... their might be some edge or corner cases that aren't accounted for so we need you to please download SVN trunk in your test labs and try it out... report issues and help us make the best FreeSWITCH release possible. Thank you, Brian West ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Rocks!!!!!!!!!
That is what is nice about our community I'm more than willing to answer the questions if you document them... as are many others in the core team...we just have a lot to do and I think the best repayment is documentation! ;) /b On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote: On Thu, Dec 10, 2009 at 11:07 AM, Brian West wrote: Visit the friday meetings and we can help if you document it. ;) I would be willing to lend a hand with the documentation but I know so little (a complete freeswitch noob). For example I was trying to figure out how to tell if an extension was set up "show dialplan in asterisk". I could not find this anywhere. If I find out I would be happy to add it to the rosetta stone. I am currently working on getting outbound socket working. Once I get it going I would be happy to add it to the relevant section of the wiki (in this case ruby). ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Rocks!!!!!!!!!
Visit the friday meetings and we can help if you document it. ;) /b On Dec 9, 2009, at 3:56 PM, Tim Uckun wrote: > I found the rosetta stone useful though woefully lacking in volume. > > I guess that's true overall with the project. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] controlling calls handled within a fifo using event_socket
"fifo list" issue this API and get the fifo XML and get the caller's uuid out of the list. /b On Dec 9, 2009, at 10:50 AM, Luke Graybill wrote: > The short version of my question is this: how do I programmatically > determine which channel uuid the consumer channel in a fifo is > connected to? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones
Best option for you is to use 96 in the sofia profile you're using to talk to these broken devices. /b On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote: > Dear list, > > Some Nec phones sends DTMF RFC2833 with payload 101 during the call, > but have negotiated a different one on SDP. > When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 > we notice this phone sends the following INVITE packet and RTP > packets: http://pastebin.freeswitch.org/11433 > Whole wireshark capture file is on > http://gregianin.org/teste_voice_rfc2833.pcap > > Is there any parameter to tweak FS in such a way to force understand > 101 packets as DTMF? > Thank you in advance! > > Fernando Testa > PS: On pcap you have the following IPs: > FS at 10.91.10.210 > Nec Pbx 10.91.10.22 > phone 10.91.10.85 > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
Well the fun part is you can't link them. :P /b On Dec 8, 2009, at 10:38 AM, wrote: > That would require a dual-core processor. One core would be 32 bit and > the other core would be 64 bit. ;-) > > -- > Russell Mosemann ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
The fun part comes when you try to link that 32bit .a file into a 64bit so file. :P /b On Dec 8, 2009, at 9:49 AM, Mathieu Rene wrote: They provide you with a 32 bit library, with the header files to link with it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
I have resubmitted our request for the source. /b On Dec 8, 2009, at 9:58 AM, Michael Jerris wrote: > We have as of yet been unable to obtain source and we have been in > very close contact with skype all the way up to the lead technical > and business people on this project. We would of course welcome > access to the source but we have as of yet not been able to get a copy > > Mike ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
And you didn't open a Jira about this? These are the kinds of issues that you should report so we can fix them... sitting on them and NOT reporting them only delays the 1.0.5 release. /b On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote: Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
