Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail
try these drivers: ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.7.smg_pri.4.tgz Mike On Dec 29, 2009, at 6:17 PM, Jerry Richards wrote: > I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the > bug is still present. Would libpri possibly help? I'm currently using the > native wanpipe PRI stack and default openzap configs in Freeswitch. > > Best Regards, > Jerry > > > From: Anthony Minessale [mailto:anthony.miness...@gmail.com] > Sent: Monday, December 28, 2009 3:31 PM > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed > toVoice Mail > > you have to update the sangoma driver and probably FreeSWITCH for good > measure. > Its a known bug in the sangoma driver that has been fixed it the latest > release. > > > > On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards > wrote: > Hello All, > > I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. > > I am still having the problem where a PSTN-to-Internal call via a Sangoma > A101D card stops ringing the internal phone after about 10 seconds. It > should be ringing for 30 seconds and then go to Voice Mail (as an > Internal-to-Internal call does). > > Best Regards, > Jerry > > > -Original Message- > From: Jerry Richards [mailto:jerry.richa...@teotech.com] > Sent: Tuesday, December 22, 2009 8:02 AM > To: 'freeswitch-users@lists.freeswitch.org' > Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail > > > I have a Freeswitch PBX server with an installed Sangoma A101D card > connected to a PRI. Most everything works okay, however when I get an > inbound call from the PSTN, if the call is not answered within about 12 > seconds, the call ends (so it doesn't go to voice mail). If I make a call > from one internal phone to another, then it will go to voice mail after 30 > seconds. How can I get the external call to route to voice mail after 30 > seconds? > > I put a new 11595 log into the pastebin. Do you know any Freeswitch setting > that might cause this? > > If this issue has been addressed before, what string should I use to search > for it, because I can't find it. > > Thanks, > 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:+19193869900 > ___ > 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] Hold is broken in trunk 16055
This means there was no sdp sent. Did you confirm this with siptrace?
On Dec 29, 2009, at 10:37 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.
2009/12/29 Brian West
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.
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lei.tl...@gmail.com
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Re: [Freeswitch-users] Hold is broken in trunk 16055
There is no need for you to show us traces. The fact that you are using proxy media is enough to know that the issue is with your device. If you look at the full sip trace you will see the same. Mike On Dec 29, 2009, at 10:10 AM, Lei Tang wrote: The phone I'm using is x-lite and wxCommunicator, both are sip phone software. I have not used pastebin, Is it a bug trace tool like bugzilla? Can you tell me how to register a pastbin account? 2009/12/29 Brian West 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: s8fc27f844
Re: [Freeswitch-users] sound rpms
the build system already has targets for all of this and there are tarballs you can manually download and extract as well that are located in http://files.freeswitch.org/. if you NEED packages, you will have to wait until that work is complete or figure out what the error is. Mike On Dec 28, 2009, at 2:54 PM, Joseph L. Casale wrote: >> This is a total work in progress that has not even merged into tree. So it >> is not "known" >> to work or not work anywhere. Patches to correct issues are welcome. > > > Mike, > I took another look at this and don't really know enough about rpm building > to diagnose this. Frankly, the format of the latest spec is so wildly > different > from anything I have ever touched I am at a loss:) > > Is there a simple manual way for me to properly get the sounds for MOH etc > installed? > is it acceptable to simply run the buildsounds-callie.sh script with the > sounds_location > pointed to my /opt/freeswitch/sounds directory? > > Thanks! > jlc > ___ 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] What's problem in SVN ?
The issues you ran into are probably sorted out now. Give it a try and if its still not working, post the build errors. Mike On Dec 28, 2009, at 8:15 AM, Dome Charoenyost wrote: > Oh...sory i forgot chismas and new year. > if someone come to thailand please let's me know :) > > BG > Dome C. > > > 2009/12/28 Jason White : >> Dome Charoenyost wrote: >>> What's problem in SVN ? Not thing update after 23/12/2009 (16055) >> >> Surely the FreeSWITCH developers are entitled to spend time with their >> families/friends after a highly productive year of work. Note that there are >> holidays in many countries at this time of year. >> >> I would like to wish everyone involved in the FreeSWITCH project a pleasant >> and refreshing holiday, and much success in 2010. >> ___ 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 doesn't update rtp port when sip UPDATE message changed the rtp port
There is no such thing as freeswitch 1.5. Have you tried latest svn trunk to see if this behavior is the same? Mike On Dec 23, 2009, at 7:49 AM, Lei Tang wrote: Hi all, I'm using FS 1.5, doesn't somebody known something about this problem? My scenario is : A(FreeSwitch) B --INVITE ---> <100 Tring <180 Ring with sdp m=audio 55066 RTP/AVP 0 120 c=IN IP4 10.36.143.76response for UPDATE message < 200 OK response for INVITE message, with sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 ACK -> The problem is, B changed the rtp port in UPDATE message and "200 OK" response message, but FS didn't do update , so it still send and receive data from port 55066. Is this a bug in FS? Does someone known something about this problem? Any advice is appreciated! -- Lei.Tang lei.tl...@gmail.com ___ 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] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch
Of course there is a way. Depending on the interface your looking at either a freeswitch endpoiny module or an openzap module. Mike On Dec 23, 2009, at 4:54 AM, Kristoff Bonne wrote: Hi Rupa, None. That's exactly the point. Everything has to be done over the usb "HID" interface. I've been reading about HID yesterday. HID is a usb interface that can be used for a large number of things, ranging from keyboard and game-controllers up to "water-cooling and PC-chassis" and point-of- sale or coin changer devices. It also has a telephony-interface: see page 69 to 72 of this document: http://www.usb.org/developers/devclass_docs/HID1_11.pdf This include call-control, on-hook/off-hook detection, DTMF-related things, etc. Now, the question is this: Is there a way to "plug" this all into freeswitch? Cheerio! Kr. Bonne. Rupa Schomaker schreef: Interesting. It would have to do more than just dialtone/dtmf though. Need call control, caller id, etc. What do they ship with it as far as drivers go? On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne wrote: Hi all, This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" device for just 15 euro. This is a device which has on one side a USB-connector and on the other side 2 RJ-11 connectors (one FXO and one FSX). Internally, the device seams to contain a tigerjet 560C chipset. (see here: http://www.tjnet.com/chips/tiger560C.htm) What is interesting on this device is that is uses standard USB device-classes that are by default supported by most operating- systems: usb-sound and usb-hid. When I connect it to my server (mac mini 3G running debian), the system automatically recognises these two classes [168391.922479] usbcore: registered new interface driver hiddev [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on usb-0001:10:1b.1-1 [168391.939548] usbcore: registered new interface driver usbhid [168391.943984] usbhid: v2.6:USB HID core driver [168392.154596] usbcore: registered new interface driver snd-usb- audio And -behold- when I connect a handset in one of the port, I even get a dialtone and I can sent out DTMF-dialtone which are somehow partly (But I have no idea what program actually generates this dialtone !!!) Now, the question: Any idea if / how this can incorperated into freeswitch? Is there a way to use this device to connect a phone to freeswitch without having to go throu a SIP-client first. Cheerio! Kr. Bonne. -- jabber/gtalk: krist...@krbonne.net ___ 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 -- jabber/gtalk: krist...@krbonne.net ___ 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] Variables for install directories
Sounds right to me, just assign it to me if it lets you Mike On Dec 23, 2009, at 12:03 AM, "Joseph L. Casale" wrote: >> For the path in the dialplan I don't think we have any right now but >> file a bug on jira and I can try to add them. As for something in >> the >> script itself that is a bit more work but if anyone has a patch to >> inject some vars into scripts like that it would be a nice addition. >> >> Mike > > Ok, signed up for an account, where does the dialplan part go, FSCORE? > Thanks for the help! > jlc > ___ > 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] Choosing a Codec.
That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Tha being said that does not sound like what you are experiencing Mike On Dec 22, 2009, at 10:29 PM, David Knell wrote: > On the other hand, a u-law WAV turned into L16 and then back to u- > law to > be sent down the line shouldn't suffer any alteration at all - if it > does, the there's something wrong with the translation. > > The quality dropping over time is almost certainly down to something > else. Vinuth -can you get a recording to compare with the original? > > --Dave > > >> 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 >> > > > ___ > 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] Codecs and things
We expect the g729 sometime very soon, weeks not months away. Mike On Dec 22, 2009, at 7:45 PM, Rupa Schomaker wrote: > On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji > wrote: >> Hello people, >> >> Can someone please clear the following ambiguities with codecs: >> >> Are we definitively able to run pass-through codecs (e.g. G.729) in >> Proxy >> Media mode, or does FS need to be running in bypass-media ? the >> Wiki is not >> clear in this regard > > Yes, you can use proxy media, bypass media, or even regular mode if > you don't transcode (special for g729). Proxy media is really a > special hack that should only be used for T38 passthrough. If you are > using it for other purposes, think about it some more > >> When an A-leg has negotiated a pass-through media codec, can the B- >> leg be >> transcoded into a non-pass-through codec, and vice-versa ? think A- >> leg >> incoming with a G.729 codec, and target for B-leg needs to be setup >> with a >> GSM-codec, say > > That would require transcoding - which can't be done if the codec is > pass-through. > >> Where in the developer's set of documentation are codecs >> discussed ? I would >> like to start porting some code of mine for G.729a/b/ab form a ti DSP >> platform to FS. FS lacking full G.729 support is proving quite a >> hindrance, >> and there is no clear direction from the dev community as to when >> the same >> will be available. Incidentally, any news on this effort ? where >> are we with >> code, and what's an ETA for a Beta ? > > I'd say look at the broadvoice or other simple self-contained codecs > are done. Currently the only supported g729 solution is to use a > digium board with mod_dahdi_codec. > > I don't have any info on a software based g729 solution. > >> On the same lines as (3) above, there is a codec dev template in >> the source >> tree. Again, where can I find documentation relating to this ? the >> template >> has hardly any docs at all. >> >> Best regards and warm wishes for a Merry Christmas and a great New >> Year to >> one and all. >> >> Ahmed. >> >> >> -- >> Ahmed A. Ibrahim-Naji Al-Alousi >> Ph.D., MIEE, MBCS >> >> ___ >> 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 >> >> > > > > -- > -Rupa > > ___ > 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] Freeswitch not seeing Register requests
If your seeing the trafic in ngrep bit not in sip trace in Sofia when enabled, your firewall is blocking the traffic Mike On Dec 22, 2009, at 5:20 PM, Michael Collins wrote: On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote: Yes, the internal profile exists. Name Type Data State === === === === === === === === === == internal profile sip:mod_so...@192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_so...@[:: 1]:5060 RUNNING (0) external profile sip:mod_so...@192.168.10.25:5080 RUNNING (0) example.com gateway sip:joeu...@example.com NOREG 192.168.10.25 alias internal ALIASED === === === === === === === === === == 3 profiles 1 alias I would do a sanity check at this point: put this box and one phone on a completely separate network with nothing else and see what happens. -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 ___ 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] Force endpoint to use rfc2833 for dtmf
Not sure if we have an option to disable info. Even without this, dtmf should go across the bridge fine. Please open up a bug on jira about this Mike On Dec 22, 2009, at 6:40 AM, Peter P GMX wrote: > Hello, > > in a bigger installation with some thousand endpoints in the field we > see, that the endpoint equipment is always using INFO messages > (standard > setting is auto, so the endpoint decides which method to use). I > have 2 > questions to that scenario: > > 1. Is there a way that Freeswitch forces/restricts the endpoint to > use rfc2833 or not to send to allow INFO in the invite message? > 2. Currently INFO messages do not get forwarded from the caller > through freeswitch to called endpoint. How can we enable that FS > is fowarding the INFO messages? > > 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 ___ 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] Variables for install directories
For the path in the dialplan I don't think we have any right now but file a bug on jira and I can try to add them. As for something in the script itself that is a bit more work but if anyone has a patch to inject some vars into scripts like that it would be a nice addition. Mike On Dec 21, 2009, at 7:03 PM, "Joseph L. Casale" wrote: > Searching through the wiki for any indication as to what if any > variables exist > > for the install location in that I can leverage in a script. > > > > Can anyone point me along, I can’t seem to find anything. I want to > place a shell > > script in /opt/freeswitch/scripts that needs a reference to a conf > file that a binary > > it runs is calling. > > > > So now I have in two places hardcoded paths that I was hoping to > avoid, in the dialplan > > and in the shell script. When either of these is run, does there > exist something like > > > > shell_script.sh"/> > > > > and the same for use inside the shell script? > > > > Thanks! > jlc > > > > ___ > 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
If this is using prid it also requires the latest drivers from sangoma. I am pretty sure these are just in dev snapshots not release drivers yet. Something 3.5.8.6 or later iirc. Mike On Dec 21, 2009, at 7:52 PM, Brian West wrote: 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 ___ 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] sound rpms
This is a total work in progress that has not even merged into tree. So it is not "known" to work or not work anywhere. Patches to correct issues are welcome. Mike On Dec 21, 2009, at 3:49 PM, Joseph L. Casale wrote: > >Working on it, moving the repos around to do this right... > > > >http://jira.freeswitch.org/browse/FSBUILD-218 > > > >Mike > > Thanks, Is this known to not work with non root builds? It errored out after > creating some > messy hierarchies with the actual variable calls, instead of their values? > ___ 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] sound rpms
Working on it, moving the repos around to do this right... http://jira.freeswitch.org/browse/FSBUILD-218 Mike On Dec 21, 2009, at 2:56 PM, Joseph L. Casale wrote: > So the spec from trunk says “Soundfiles are moving into a separate spec” > but I can’t find this spec anywhere in svn? > > Anyone know where it is? > > Thanks! > jlc > > ___ > 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] RTP problems in recent revisions?
The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but after upgrading to revision 16003 I get > the > following. > > 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). > > 2. A PCMU call to a SIP provider is fine for the first 20 to 30 > seconds, then > the audio breaks up completely. > > I have ZRTP compiled in, if that makes any difference. > > Obviously there's a regression somewhere. Let me know if I can > provide further > help. > > > ___ > 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] SIP Re-invite
I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: > Mike, > > My latest traces that I captured were done within the FS box: > http://pastebin.freeswitch.org/11541 > > Thank you, > Dorn B. > From: Michael Jerris > To: freeswitch-users@lists.freeswitch.org > Sent: Thu, December 17, 2009 8:03:46 AM > Subject: Re: [Freeswitch-users] SIP Re-invite > > are you doing this trace from the freeswitch box itself? > > Mike > > On Dec 17, 2009, at 10:48 AM, DJB wrote: > >> Anthony, >> >> I have pasted the invite sip trace here: >> http://pastebin.freeswitch.org/11536 >> Please advise if you need further info. >> >> 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 ___ 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 is your dialplan on the secondary box? On Dec 18, 2009, at 9:08 AM, Brian wrote: I’ve got FS running on a 64 bit OS, and here is more info on the tes t procedure. I’ve got one server (primary) that hosts the speaker call (this is m eant to be a primary conference with a few speakers, but my test sim plifies this to just one speaker). I’ve got a second server (seconda ry) that hosts the conference that all the listeners go into, and I have two other servers that I use automate the listener calls. The g oal is to have several secondary servers to scale the listener side of things, but for this initial test I’ve only got one secondary ser ver. 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 th e 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 ___ 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
We are always doing enhancements and yes there are some real scalability enhancements in trunk compared to 1.0.4, I am just not sure if they effect conference significantly or not. I would guess that trunk is actually more stable than 1.0.4 at the moment. Give it a try and find out. Mike On Dec 17, 2009, at 2:29 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 > 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] Building on Windows
On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote: >> Hi, >> >> I'm probably going to regret this - I'm not sure that I'll be able to do >> this without a lot of pain (nothing to do with FS - more my lack of ability >> with Visual Studio), but.., I want to try building FreeSwitch from >> source rather than using the pre-built binaries. I have a couple of initial >> questions that, hopefully, someone can answer please ? >> >> 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the >> horizon for me. >> Having downloaded the SVN, I see there is a VS 2005 Solution, but it is >> marked as "Unsupported", although the Wiki says that you only need VC++2005. >> What does "unsupported" mean in this context ? I guess that support for >> VS2005 is being dropped, but is the VS2005 Solution still being maintained, >> and if so, for how long? I'd hate to get into the build thing and then find >> that I was stalled when VS2005 support was dropped altogether ? > > Install VS 2008 if at all possible (express edition is free). 2005 > support isn't maintained much if at all, so a lot of newer modules stand > a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. >> >> 2. The whole SVN thing is new to me but I've worked out that I need an SVN >> Client on Windows to work with the source. Can anyone recommend the best >> (free) SVN Client for Windows to use with FreeSwitch. I have installed >> TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work >> on my first build but it's not command line based so some of the tips given >> in the Wiki like "make current" and "make sounds" may be more awkward to >> achieve. Is anyone else using Tortoise and/or can give some tips on which >> SVN client to use ? >> > Tortoise SVN is fine and is probably the de-facto client for windows. > make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] >> 3. I built 15979 last night (with VS2005) and got some warnings, with data >> type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. >> >> 4. There was one fatal error in the build of mod_opal (missing file) >> (Some examples of the warnings and the error are shown below :-) >> > Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. >> 5. How do I specify which options (e.g., mod_flite, to be included iin the >> build. >> > You can enable the different sub projects somehow in the UI, I always > forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer >> 6. How do I build the sounds etc. ? >> > > The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. 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] Polycom 501 conferencing with FreeSwitch
I have not seen anyone mention it. Mike On Dec 17, 2009, at 11:07 AM, Yehavi Bourvine wrote: > I'll rephrase my question: Has anyone done that, or should I dig into it? > After all, Polycom is quite common... > > Thanks, __Yehavi: > > 2009/12/17 Michael Jerris > Its software, anything is possible with enough time and effort. > > Mike > > On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > > > After some discussions with Polycom support it seems that their > > conferencing support is based on draft-ietf-sipping-cc-conferencing-03 > > (which is not the latest and is not compatible with the latest one). > > > > Any idea whether it is possible to program Freeswitch to support this draft? > > ___ 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 Re-invite
are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > 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] mod_shout.so: undefined symbol: ogg_sync_wrote
if you contact me offlist, or better, join #freeswitch on irc.freenode.net and ping me (MikeJ) Mike On Dec 17, 2009, at 8:34 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 > > On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris wrote: > strange, can someone file a bug on this on jira.freeswitch.org and contact me > off list with ssh info so I can troubleshoot this on your box. > > Thanks > Mike > > On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > >> I'm also experiencing this problem, and I have verified I have libogg, >> libvorbis, and their dev packages installed. >> >> I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not >> listed in the dependency lib list. Is this related? >> >> -Neil >> >> On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: >> looks like ogg devel packages are installed but ogg lib is not? >> >> >> On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: >> >> > FreeSWITCH seems to be unable to read MP3 files, citing that it's an >> > unknown format. Looking through the log, I found this during startup: >> > >> > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error >> > Loading module /usr/local/freeswitch/mod/mod_shout.so >> > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: >> > ogg_sync_wrote** >> > >> > There don't seem to be any compile-time errors, yet I can't seem to >> > eliminate this issue. Any help would be appreciated. >> >> >> ___ >> 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 > > > ___ > 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] SIP Re-invite
if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: > We have a customer that we are sending calls to off the FS and here is the > issue: > > > > Call is initially setup fine and they send a first re-invite with media > 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first > re-invite fine > > > > They then send a second re-invite with their media IP to cut through media > and the FS sends a 200 OK to this fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because according to them it is > strictly for the purpose of “keep alive” per RFC 4028. This third re-invite > has the exact same media IP and UDP pot information as the second re-invite > does. The problem is FS does not respond to this third re-invite AT ALL. It > doesn’t send a 100 trying a 200 OK nothing so this causes the call to be > dropped as the other end does not recieve a response from FS. > > > > One more thing, we did not see the third re-invite in sofia siptrace, but we > do see it in ethereal, which is kind of odds. > > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > ___ 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] Polycom 501 conferencing with FreeSwitch
Its software, anything is possible with enough time and effort. Mike On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > After some discussions with Polycom support it seems that their conferencing > support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the > latest and is not compatible with the latest one). > > Any idea whether it is possible to program Freeswitch to support this draft? > >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] 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] xml_rpc.conf
On Dec 16, 2009, at 9:01 AM, Nameer Kazzaz wrote: > Hi all, >Can I set xml_rpc server to run on a specific interface I can set > the port but not the ip address to bind to. I have a linux server with > more then one interface. I don't want to use iptables to block it. No, but you always have the option of using iptables to block it. 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] mod_shout.so: undefined symbol: ogg_sync_wrote
strange, can someone file a bug on this on jira.freeswitch.org and contact me off list with ssh info so I can troubleshoot this on your box. Thanks Mike On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > I'm also experiencing this problem, and I have verified I have libogg, > libvorbis, and their dev packages installed. > > I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed > in the dependency lib list. Is this related? > > -Neil > > On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: > looks like ogg devel packages are installed but ogg lib is not? > > > On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: > > > FreeSWITCH seems to be unable to read MP3 files, citing that it's an > > unknown format. Looking through the log, I found this during startup: > > > > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error > > Loading module /usr/local/freeswitch/mod/mod_shout.so > > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > > ogg_sync_wrote** > > > > There don't seem to be any compile-time errors, yet I can't seem to > > eliminate this issue. Any help would be appreciated. > > > ___ > 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] SIP Error Message 480
Yep, there is your issue.. I missed it when you pasted the extension, its a typo in your condition. Dialplan: sofia/internal/12482578...@127.0.0.1:5080 Regex (FAIL) [VoipMs] destination_number(19059183027) =~ /expression=/ break=on-false Notice what it is comparing there .. and notice the typo in your condition. Mike On Dec 15, 2009, at 9:27 PM, bcxml wrote: > > > Here is the link to the debug log > > http://pastebin.freeswitch.org/11521 > > > Brian > > > mercutioviz wrote: >> >> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: >> >>> Try turning on debug logs, but from this it looks like its not matching >>> any >>> extensions. >>> >>> Agreed. "console loglevel debug" at the fs cli and then make a test call, >> capture output, drop into pastebin.freeswitch.org, and post the URL in >> this >> thread. >> -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 >> >> > > -- > View this message in context: > http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805343.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > ___ > 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] SIP Error Message 480
Try turning on debug logs, but from this it looks like its not matching any extensions. Mike On Dec 15, 2009, at 3:11 PM, bcxml wrote: > > I have Freeswitch and Microsoft Speech Server 2007 on the same box > > When Speech Server initiates a call, I get a sip error message 480 > > Here is the internal profile trace... > > 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel > sofia/inter > nal/12482578...@127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] > 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing > 12482578002- >> 19059183027 in context public > 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup > sofia > /internal/12482578...@127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] > send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: ___ 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
You can do that with phrase macros. Mike On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: > Hello, I create one WAV file that has: > > Question + Option 1 + Option 2 + Option 3 + … > > I noticed towards end of the file Cepstral Allison starts chopping and > speeding up. > > So my question text that gets converted to WAV file using swift EXE looks > like: > > Which is the biggest mammal on land? > Select one of the following choices. strength='weak'/>Or press star to skip the question > 1 Parrot > 2 Elephant > 3 T-Rex > 4 Blue Whale > > > And my csharp code looks like: > pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, > 5000, "*#", > > @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV ", > > @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", > "^\\d", ""); > > > What happens is, the voice just starts chopping and speeding up between > options. Even though I am not able to say that it only does that towards the > end, I think so. > > I thought, if I break each file into individual WAV instead of 1 big WAV, it > may help? > > Is there a way to play multiple (separate) WAV files in PlayAndGetDigits > function? > ___ 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] One-way Video
try just 1 video codec in freeswitch codec prefs and make sure you are using trunk, we fixed quite a few video issues recently. Mike On Dec 15, 2009, at 12:54 PM, Jerry Richards wrote: > I am trying to bring up a video call, but not having much luck. We are only > getting one-way video (i.e. the caller sees far-end video, but the callee > does not). I added the H263/H264 tags to the pre-process > "global_codec_prefs" and "outbound_codec_prefs" tags in vars.xml. > > Anyone have hints on making two-way video to work? ___ 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] What are the solutions for G729 support ?
We have not published costs yet, but expect it to be inline with other similar offerings. I expect the module will initially be available for linux and we will add other platforms as demand shows a need for it and I can get build servers up that will be used to produce the binaries. Windows will likely be one of the early alternatives but we have not yet tested the code on windows. Mike On Dec 14, 2009, at 5:01 PM, Oscav wrote: > > Hi Anthony, > > What kind of software?? Is there any related licensing cost? Will it be also > available for windows ?? ___ 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] Language settings for demo IVR
The issue is the demo ivr does not use phrase macros. The line in ru.xml is
for the phrase macros. We should probably change this in the future.
Mike
On Dec 15, 2009, at 5:58 AM, Dmitry Bely wrote:
> On Tue, Dec 15, 2009 at 12:15 AM, Michael Collins wrote:
>>
>>
>> On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely wrote:
>>>
>>> I'm playing with demo IVR from FreeSwitch distribution and have a
>>> problem with language settings. I would like to use Russian as a
>>> default language for voice messages so I set in vars.xml
>>>
>>>
>>>
>>> and installed Russian sound files. It works almost correctly: all
>>> phrases are played in Russian, but not explicitly specified .wav
>>> files; say for
>>>
>>>>>invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>
>>> I have
>>>
>>> 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File
>>>
>>> [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav]
>>> [System error : No such file or directory.]
>>>
>>> How to fix this and make it use the correct language?
>>>
>> What about this in vars.xml?
>>
>> > data="sound_prefix=$${base_dir}/sounds/en/us/callie"/>
>
> Yes, that does the job. Thank you! But it looks a bit inconsistent.
> Path to sound files is also set in $${base_dir}/conf/lang/ru/ru.xml.
> Why duplicate the settings? And another problem is that you cannot
> easily switch the language for your voice menu.
>
> - Dmitry Bely
>
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Re: [Freeswitch-users] Freeswitch and Gtalk
That should work fine. Mike On Dec 12, 2009, at 2:08 AM, Surajee Ratnayake wrote: > Hello.. > just want to get the following clarified from the friends in the same domain, > > since freeswitch is allowing multiple gtalk user registrations with gtalk > servers, assume we route gtalk voice calls coming to these gtalk users are > routed to sip extensions or to PSTN/PLMN? will google block some thing like > that or is it already happening? ___ 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 started on IVR Library
A good example of how to use this code would be in mod_rss or mod_voicemail in tree. I would say look at the doxygen at http://docs.freeswitch.org/group__switch__ivr.html but it appears that page is completely broken. I will try to take a look and figure out why this weekend, in the meantime, you can look at the doxygen comments inline in switch_ivr*.h. Mike On Dec 12, 2009, at 2:42 AM, Thangappan.M wrote: > Dear all , > > I've seen the IVR library functions which are implemented in C > language. Can any one please suggest how can I use that library or give idea > to do the IVR programs in C through this library. > ___ 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
Probably the best list is: http://wiki.freeswitch.org/wiki/FreeSwitch_Dependencies Due to the fact that we allow you to change modules after configure there is no great way to have it error out when you don't have the right deps other than to just have the compile errors when you try to build. Its probably time for a tool like make menuconfig but we do not have that as of yet. Mike On Dec 11, 2009, at 1:47 PM, Julian Lyndon-Smith wrote: > Thanks Mike. I understand why you don't want all to be built. However, > there are things that I would like - such as mod_java. However, that > fails to compile, I presume because of some missing dependency or > requirement. Is there any tool to tell me what is needed in order to > build a module ? > > Julian > > 2009/12/11 Michael Jerris : >> It just so happens I was looking at this same bug last night and having >> troubles chasing down a solution, if anyone comes up with anything good >> please let me know. The basics of this is that automake continues on to >> other subdirs if build in one subdir fails. >> >> Mike >> >> p..s. a note on the blog, I generally do not recommend just building >> everything, for example, mod_alsa is a module written specifically for the >> n800 due to mod_portaudio not working there. This module is barely touched >> and I would not use it unless you have a good reason to. >> >> >> On Dec 11, 2009, at 1:19 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 >> >> >> ___ >> 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] The Building Freeswitch blog
It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. The basics of this is that automake continues on to other subdirs if build in one subdir fails. Mike p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. This module is barely touched and I would not use it unless you have a good reason to. On Dec 11, 2009, at 1:19 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 ___ 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
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code On Dec 11, 2009, at 11:25 AM, Kendall Stauffer wrote: > Yes, I can do that , I don’t see where I download the source, Sorry to bug > 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: > Thanks Frank for sharing your experience. This is the behavior change just > starting within three days, maybe because of some code changes in mod_sofia > which I should change the settings accordingly > I noticed that the acl automatically having 192.168.0.0 set as "deny", that's > why I tried to changed the settings regarding nat acl and localnet acl. > > Chris > > > > On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: > On Fri, Dec 11, Chris Chen wrote: > > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN > > trunk 15905. > > Is this a change in behavior or is this the first time you've run freeswitch? > If this is your first time welcome aboard! Also if this is your first time > you've probably have some IPs aliased on your interface and you still have > stun enabled. This was the behavior I saw the first time I ran it on a box > with aliases on an interface. The stun server tells freeswitch after some > time that the IP is different then the one you've assigned. This is just one > possibility. If this isn't the case then we will need to see sip traces on > all of your profiles. > > --FC > > ___ > 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] embedded freeswitch compatable hardware
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. Mike On Dec 10, 2009, at 8:57 PM, Brian May wrote: > 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 ___ 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?
we also support natpmp and static ip setting. Mike On Dec 10, 2009, at 12:21 PM, Fred-145 wrote: > > Thanks for the clarification. So it's either UPnP or STUN/port-mapping. ___ 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
I think I fixed the spandsp cross compile issues tonight, but I suspect there is a good chance that I broke other builds in the process. I also did a bunch of work to make the OS X Snow Leopard build cleaner today. Testing would be much appreciated on both. Mike On Dec 9, 2009, at 10:47 PM, Kristian Kielhofner wrote: > Brian, > > I have been making efforts to fully support FreeSWITCH in AstLinux. > Our primary targets are low powered x86 boards like the Soekris and > Alix. x86, powerful enough, cheap enough (as low as $100), and about > 12 watts. Not bad. > > 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. > > 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: > > http://mirror.astlinux.org/freeswitch/daily/ > > Let me know what you think. > > On Wed, Dec 9, 2009 at 7:55 PM, Brian May > wrote: >> 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 >> > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > ___ > 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] no hang-up on B leg
src/switch_ivr_bridge.c This could just as well be a glare condition when the call is in process of tearing down. Mike On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote: > No doubt, but that’s a little difficult as this only happens occasionally and > I have 200 calls going on at the time. It’s needle in the haystack stuff. > > Here’s what I know. > > I have an external process listening for DTMF events. If I detect ‘*’ I do a > kill uuid on the B leg. On a number of occasions I get an error saying the B > leg doesn’t exist, so I now do a double kill on the associated leg which I > get from the event. I do not get a ‘doesn’t exist’ message for the A leg, > which leads me to believe that process of tearing down both bridged legs is > flawed. > > The kluge clears the B leg hang issue, so the pressure’s off for me, but when > I get a few nano seconds, I’ll look at the code to see if there’s anything > obvious. > > Can anyone give me a hint on what module handles bridged calls? (sorry, being > lazy and suffering from a lack of sleep) > > Regards, > > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael > Jerris > Sent: 08 December 2009 16:16 > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > We will really need debug logs and sip traces to be able to figure out what > exactly is going on here. > > Mike > > On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: > > > Sorry no, apart from the fact that I was seeing the hangup. > > > I’m wondering if this a bandwidth congestion issue. Is there anyway on a > bridged call I could trap on dtmf like look for ‘*’ and force a hangup? I > don’t seem to able to see this tone on the B leg though. > > Regards, > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: 07 December 2009 19:12 > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > > > On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton > wrote: > Hi all, > > I’ll slowly pulling my hair out on this one. I had FS successfully hanging > up both legs on a bridge, now today, with nothing changed, I’m not seeing a > hangup of the b leg at all. > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup > just fine. Before when I had an issue with the B leg not closing the bridge, > I was at least getting a hangup event, now it’s not being fired. Does anyone > have an idea what might be causing this? > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when > things seemed to be working so that you can compare? > -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 ___ 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] CLIP on FXS channels with mod_openzap
I recall implementing that back when we released openzap, it should be in there unless someone chopped it out for some reason. Look for "zap_channel_send_fsk_data" Mike On Dec 9, 2009, at 6:01 AM, François Legal wrote: > I'm still working on this issue, and decided to take a look at the openzap > code. > > First, I figured out that the parameter name for callerid is enable_callerid > rather than enable-callerid. > > I also figured out that this parameter defaults to TRUE (which is coherent > with the observed behaviour on my FXO span) > > > By further checking the code, I figured out that presenting the callerid on > an FXS port might not be implemented yet. I could see the code for retrieving > the callerid from FXO but nothing to send it. > > > Is my asumption (feature not implemented) correct ? > > > François > > ___ 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] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Our plan for 1.0.5 is that we will also have rpm and deb packages for many distros on our own repo. Stay tuned. This has been another major reason for the delay in 1.0.5. Mike On Dec 8, 2009, at 1:26 PM, Joseph L. Casale wrote: >> For those interested, here's how to compile and install Dahdi (which doesn't >> need Asterisk at all, unlike some docs on the Net seem to imply due to >> references to /etc/asterisk/*.conf): > > I understand that Some Debian based distro's have Dahdi in their repo's > making it > simple, but not many know that Digium runs its own repo for rpm based distros: > > http://packages.asterisk.org/ > > Can't get easier than that... > > jlc > > > ___ > 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] Force presence status manually
The best way to solve this is probably to share the db for presence and registration between those boxes. If you take a look at the default configs the settings should be commented there. Mike On Dec 8, 2009, at 11:02 AM, Peter P GMX wrote: > Hello, > > is there a way to manually force a presence status update? > In our scenario we have a Freeswitch cluster. As phones sometimes > register on one and one time on another machine via the load balancer, > we cannot dial via user/exten. Instead we dial each phone by it's > register string via xml-curl. That way -when a phone is called - other > phones who subscribed to this phone, do not receive a message to update > their presence status. > Is there a way to force the pesence status of a phone manually in the > dialplan? > We may then set the status before bridging and then reset it with a > hangup hook. > ___ 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
I changed the name of key to ikey in trunk. Mike > 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? > > > Jon Brüel > ___ 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] compilation error of skypiax_protocol.c
If you can off list provide me with remote login information to this box I can troubleshot the issue. Mike On Dec 8, 2009, at 4:09 AM, Jingwei Yang wrote: > Hi João, thanks for the reply. But I don't quite get you.. Could you please > elaborate a little bit? I tried installing libtiff and upgrading FS to the > latest revision, but still the same error. > > Here's how I normally update FreeSwitch: make clean && svn up && > ./bootstrap.sh && ./configure && make install > > If any step missing, please kindly let me know. In addition, my OS is CentOS > 5.3 and my gcc is version 4.1.2. > > Regards, > -Jingwei > > > 2009/12/8 João Mesquita > Maybe, just maybe isse that make target to reconf libtiff? > > Regards, > > JM > > > On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang wrote: > I installed libjpeg-7 following this website: > http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the > previous error is replaced by a new one: > > gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. > -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math > -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes > -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF > .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o > at_interpreter.c: In function ‘command_search’: > at_interpreter.c:5299: error: ‘COMMAND_TRIE_LEN’ undeclared (first use in > this function) > at_interpreter.c:5299: error: (Each undeclared identifier is reported only > once > at_interpreter.c:5299: error: for each function it appears in.) > at_interpreter.c:5308: error: ‘command_trie’ undeclared (first use in > this function) > at_interpreter.c: In function ‘at_interpreter’: > at_interpreter.c:5424: error: ‘at_commands’ undeclared (first use in this > function) > make[8]: *** [at_interpreter.lo] Error 1 > > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > However, I'm still able to start freeswitch and mod_skypiax and make skype > calls with no problem. > > Regards, > -Jingwei > > > > On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: > No, I didn't change or update the system libs. I just wanted to double check > whether my system has this libjpeg library. ./configure was definitely > executed before the source codes were rebuilt. > > Regards, > -Jingwei > > > On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: > Hi, > > That one is on your side. If you changed/updated system libs it might be > worth doing another ./configure > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mr...@avgs.ca > > > > > On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: > >> Hi Mathieu, thanks for the promptly reply. The error has been fixed. >> However, I encounter another one. >> >> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o >> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >> -lc >> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: >> cannot open shared object file: No such file or directory >> make[8]: *** [at_interpreter_dictionary.h] Error 127 >> make[7]: *** [all] Error 2 >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_voipcodecs-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> Do you have idea about this one? >> >> Thanks! >> >> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >> Consider it fixed. >> Committed revision 15765. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mr...@avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >> >>> Hi Guys, >>> >>> I got a compilation error of skypiax_protocol.c with the latest version >>> r15764. >>> >>> Compiling skypiax_protocol.c... >>> cc1: warnings being treated as errors >>> skypiax_protocol.c: In function ‘X11_errors_handler’: >>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c: In function ‘skypiax_send_message’: >>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c: In function ‘skypiax_do_skypeapi_thread_func’: >>> skypiax_protocol.c:1726: wa
Re: [Freeswitch-users] no hangup on B leg
We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: > Sorry no, apart from the fact that I was seeing the hangup. > > > I’m wondering if this a bandwidth congestion issue. Is there anyway on a > bridged call I could trap on dtmf like look for ‘*’ and force a hangup? I > don’t seem to able to see this tone on the B leg though. > > Regards, > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: 07 December 2009 19:12 > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > > > On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton > wrote: > Hi all, > > I’ll slowly pulling my hair out on this one. I had FS successfully hanging > up both legs on a bridge, now today, with nothing changed, I’m not seeing a > hangup of the b leg at all. > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup > just fine. Before when I had an issue with the B leg not closing the bridge, > I was at least getting a hangup event, now it’s not being fired. Does anyone > have an idea what might be causing this? > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when > things seemed to be working so that you can compare? > -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] esl for Mac OS X 10.4
Please re-test this with svn trunk of freeswitch and if it is still the case open up a bug on jira.freeswitch.org in the build system catagory assigned to me and attach the config.log and config.status from the libs/esl dir to the bug. Mike On Dec 7, 2009, at 1:34 PM, Kendall Stauffer wrote: > Any direction on where to start would be appreciated. I am trying to get > freepbx working with this, and everything works (I think) except esl > > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West > Sent: Monday, December 07, 2009 1:10 PM > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 > > The build system for libesl and everything below that won't work 100% on the > mac just yet. You have to make some changes to how its linked and you'll > have to compile php yourself to get everything in there properly. The perl > one however is much easier to fix. > > -SOLINK=-shared -Xlinker -x > +SOLINK=-dynamiclib -Xlinker -x > > > Thats all you usually fix for the mac. > > > /b > > > > On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: > > > I have downloaded and compiled freeswitch, and it runs fine, can compile > everything without error including spandsp, but can’t get esl to compile. My > version is earlier than the snow leopard that is mentioned in the general > install docs, and I have tried it with and without the compiler flags in the > freewswtch installation -> MAC os X. > I have also googled this, and don’t see what I am doing wrong. Anybody > there that can help? > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" > CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable > -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" > CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C php > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient > -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > _main > __convert_to_string > __efree > __emalloc > __estrndup > __zend_get_parameters_array_ex > __zend_list_find > __zval_copy_ctor > _compiler_globals > _convert_to_long > _zend_error > _zend_get_constant > _zend_hash_find > _zend_register_list_destructors_ex > _zend_register_long_constant > _zend_register_resource > _zend_rsrc_list_get_rsrc_type > _zend_wrong_param_count > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make: *** [phpmod] Error 2 > ___ > 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] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
If this issue continues after another update and re bootstrap/configure, please open up a bug on jira.freeswitch.org under build system, assign to me, and attach the config.log and config.status file from the root of your freeswitch src dir. Mike On Dec 7, 2009, at 2:39 PM, Anthony Minessale wrote: > try rerunning the ./bootstrap.sh > > > On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards > wrote: > When I got the latest trunk the make gets an error. Should I perhaps disable > the mod_amr? > > making all mod_amr > make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop > > The method I used to get the latest trunk follows: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch > > Best Regards, > Jerry > > From: Jerry Richards [mailto:jerry.richa...@teotech.com] > Sent: Monday, December 07, 2009 7:44 AM > To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' > Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE > When Gateway Sends RTP > > I am changing the 3pcc setting because one of my gateways sends INVITEs > without SDP. I will try to update to the latest trunk today and capture > traces as Anthony described. If I can't do it today, it might be at the end > of the week. > > Best Regards, > Jerry > > > From: Michael Jerris [mailto:m...@jerris.com] > Sent: Saturday, December 05, 2009 7:30 PM > To: Jerry Richards > Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE > When Gateway Sends RTP > > Jerry- > > Any update on this? > > Mike > > On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: > >> Why are you changing the 3pcc setting, is this an invite with no sdp? >> you need to take a trace from FS. >> >> 1) update to latest trunk first so line number match up. >> 2) issue these commands >> >> sofia profile internal siptrace on >> console loglevel debug >> >> save the output and put it on pastebin http://pastebin.freeswitch.org >> >> >> >> >> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards >> wrote: >> >> I have Mediant 1000 gateway, and for some reason, when I make an outbound >> call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A >> Wireshark trace shows that FS is replying to the gateway's inbound RTP >> packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP >> packets to the same port that FS specified in the outbound INVITE. It >> appears in the log that FS is discarding the 200 OK from the gateway. >> >> I disabled the Firewall and SELinux on the Freeswitch machine. I tried >> changing to "true" and also "proxy", but it has no effect. >> >> Anyone know what could be the issue? I posted the Freeswitch log in the >> pastebin. >> >> 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 >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_miness...@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:8...@conference.freeswitch.org >> iax:gu...@conference.freeswitch.org/888 >> googletalk:conf+...@conference.freeswitch.org >> pstn:213-799-1400 >> ___ >> 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com
Re: [Freeswitch-users] continue_on_fail
You definitely need to use the settings in combination for what you are trying to do. Can you explain a bit more what you want to do in what conditions and maybe we can suggest how to accomplish this. NORMAL_CLEARING is not a failure, so it can continue on after the bridge unless you specify otherwise. Mike On Dec 7, 2009, at 1:31 PM, Peter P GMX wrote: > I have a Problem with continue_on_fail. > > > I have setup a hunt group > > data="sofia/external/2...@10.11.12.243,sofia/external/2...@10.11.12.234,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245"/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. ___ 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
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 On Dec 8, 2009, at 9:39 AM, Kevin Green wrote: Their site (https://developer.skype.com/silk) specifies that they will provide the source, which as you say may not be 64-Bit compatible but could likely be tweaked to work. I think you just need to be specific in that you want a source copy not a binary copy of the codec. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris wrote: That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: It seems you can get a copy of either the binaries or the source by doing the following: Review & execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. Return executed agreement to silksupp...@skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN Skype will email you the SILK binary once we receive the executed agreement. Check out documentation, FAQ, and discussion forum (URL TBD) Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > wrote: Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West wrote: > We can ONLY hope someone will do this and BSD/MIT the library and NOT > GPL it... if they GPL it then we'll have to have someone write it all > over again... love the Open Source oil and water. > > /b > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > >>> it I suspect. >> >> Given that they released the codec specification, perhaps someone is >> writing >> an independent C implementation? (Not that I'm much interested, >> but...) > > > ___ > 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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 ___ 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
That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: It seems you can get a copy of either the binaries or the source by doing the following: Review & execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. Return executed agreement to silksupp...@skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN Skype will email you the SILK binary once we receive the executed agreement. Check out documentation, FAQ, and discussion forum (URL TBD) Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > wrote: Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West wrote: > We can ONLY hope someone will do this and BSD/MIT the library and NOT > GPL it... if they GPL it then we'll have to have someone write it all > over again... love the Open Source oil and water. > > /b > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > >>> it I suspect. >> >> Given that they released the codec specification, perhaps someone is >> writing >> an independent C implementation? (Not that I'm much interested, >> but...) > > > ___ > 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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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] Choppy sound with PCMU
Also I have seen some people reporting that the new tickless timers in newer kernels work better. You may want to try those. Mike On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote: > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable > monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to > sleep 1000 microseconds that is what we need it to sleep for or at least as > close as possible, and the main reason that thread is never sleeping is > because you can't actually count on it to run every 1ms but you mostly can. > Hence the whole philosophy on only making 1 thread run hot all the time to > ensure that the rest don't have to repeat the same algorithm. We focus on > high end performance this was the point of your experimentation because we > will need to use a compile time defines and other logic to make it more > efficient on your platform, a platform which we are not using. I am curious > what would happen if you install Kristian's astlinux on one of your devices, > i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do these > systematically each alone and all together with/without -hp or disable > monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer thread > which will make all the switch_cond_next share a 1ms conditional instead of > doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others work > better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it > should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > > Here is what I found... > > I tried high-priority scheduling as per your suggestion, reniced the program > explicitly, rewrote timer thread to sleep on cond. variable and activate > only when there are timers and only when the timer actually had to be > clicked, turned off SQL thread and removed polling from sofia profile > thread. > > That pretty much eliminated all idle 1ms sleepers that were there except for > three in sofia itself (su_epoll_port). And when I was about to be happy, I > found that two outgoing calls through my VOIP providers when bridged > together showed terrible distortions. I undid all my changes, tried 1.0.4, > trunk (noticed btw that when I bridge two calls via loopback in JS in the > trunk I must keep JS running, or the calls get terminated - NOT the same as > in 1.0.4 where exitting JS left calls running), got pretty much the same sad > results. At the same time calls bridged by freeswitch between LAN and any of > the VOIP providers behaved just fine. And calls bridged by Asterisk any way > were fine too. So that pretty much looked like the end of the freeswitch > trials for me. > > But then I timed your code, mine and found that all those 1ms sleeps that > your timer thread was doing (and all those pollers were doing as well) were > actually 4ms sleeps because you know what unless kernel is configured with > HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms > (HZ=100). Mine was 250. > > This actually meant that the original timer thread was firing once, sleeping > for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 > times back-to-back, etc. It was still firing 20ms timers on time, but 30ms > ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever > relied on runtime.reference or switch_micro_time_now() were kind of screwed > because both were running jumpy. Plus whoever assumed that apr_sleep(1000) > or cond_yield() was sleeping for 1ms were also in for a surprise. It felt > satisfying to find that, however it didn't explain why the same distortions > were observed with rewritten timer thread and disabled RTP timers. > > Anyway, I sighed (pretty much like you) and recompiled the kernel with > HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, > you need to hook up serial console and see what the heck went wrong. > > That eliminated distortions, ha! But made freeswitch more CPU hungry. Now > the remaining 1ms threads sitting in sofia epoll were really polling for > 1ms, not 4, and freeswitch was consistently sitting in the first line of the > top chart showing 3% CPU utilization when idle. > > Don't know whether it's because of t
Re: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario
Please report bugs to jira.freeswitch.org. Mike On Dec 6, 2009, at 11:45 PM, Seven Du wrote: > Hi, > > I know there's some chang on att_xfer, and after upgrade(re-bootstrap) > to trunk code, no sound after att_xfer. > > Then I rebuild FS 15807 with a fresh checkout, but still using the old > conf/ settings, sound is ok, but there are other problems: > > A call B, and B att_xfer C > > 1) origination_cancel_key not working. no even no DTMF log in FS when > I press # or any other key, I tried with Zoiper and Snom(on the B leg) > 2) when C answers, B immediately hangup, so B has no chance talk to C > > Could this be a problem? I pasted logs: > > http://pastebin.freeswitch.org/11417 ___ 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] Bad sound quality while eavesdropping
This bug has been now closed out in jira due to no response for requested information. If you wish to resolve this issue please follow up on your bugs when information is requested. Mike On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote: > > Nope, I wanted to make sure that this is indeed a bug. I opened an issue in > JIRA before regarding some other matter and it turned out to be my mistake, > so I decided to try mailing list first this time. > MA > > > > Brian West wrote: >> >> Did you open a jira and attach all the info? >> >> /b >> >> On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote: >> >>> Yes, I confirmed that with Wireshark (filter "rtp and ip.src == >>> ). RTP packets are sent every 20ms. >>> >>> MAniserowicz ___ 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] HA questions.
so your registering to the provider to get the calls? If so, this gets tricky, the provider likely does not support multiple registrations, even if they did they probably send the call to both registered endpoints. With this big unknown its not very easy to suggest a good solution. If I were looking to set this up without needing proxies I would want to use srv records and naptr records and a provider that would balance using these including failiover. Mike On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote: > On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote: >> The easiest place to do this is at the point you send the calls to >> FreeSWITCH. How are the calls coming in? >> > > From an as of now unkown SIP trunk provider (we are still in > negotiations with a couple of companies). > > ___ > 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] Choppy sound with PCMU
for hundreds of >>>> milliseconds, not for one. >>>> >>>> And there is even infrastructure present to do blocking pops: i.e. why >>>> couldn't sqldb thread do queue_pop() instead of queue_trypop() >>> intermixed >>>> with 1ms sleeps? This looping is such a waste... >>>> >>>> >>>> eaf wrote: >>>>> >>>>> As I see it, switch_cond_next() currently is just a do_sleep(1000). >>> Yes, >>>>> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >>>>> overrides that. >>>>> >>>>> Yeah, there is a global timestamp... It's easy to workaround that for >>> RTP >>>>> who calls switch_micro_time_now()... But if somebody accesses >>>>> runtime.timestamp directly, it's gonna be tough to grep for that. If >>> only >>>>> this was C++... >>>>> >>>>> I'll play around. Never liked polling too much. Never could've guessed >>>>> that polling could be so useful for scalability ;) My naive >>>>> implementation would've pulled timestamp via system calls and would've >>>>> done sleeping by passing exact interval to select() instead of syncing >>>>> with a pacing thread. Which would be dead-quiet at idle time, but, of >>>>> course, would stop scaling at some point due to excessive number of >>>>> system calls. >>>>> >>>>> Thanks. >>>>> >>>>> >>>>> Michael Jerris wrote: >>>>>> >>>>>> In short. No, you can not for many reasons. The milisecond tic is >>>>>> used throughout the code even when there is not any calls up. You >>> can >>>>>> grep for switch_cond_next if you would like to see where but it is >>>>>> required to keep our global timestamp and for pacing the scheduler >>>>>> among other services that run all the time. >>>>>> >>>>>> Mike >>>>>> >>>>>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >>>>>> >>>>>>> >>>>>>> Can I reduce resolution of that timer thread 10 times? I mean, I >>>>>>> glanced >>>>>>> through the code, and see that among others (are there others?) RTP >>>>>>> and IVR >>>>>>> set up their timers that are subsequently managed by this thread. >>>>>>> RTP timers >>>>>>> should be eliminated by that setting you've suggested. IVR timers >>>>>>> are set at >>>>>>> 20ms... So, if the thread is set to wake up every 10ms instead of >>>>>>> 1ms it >>>>>>> should be able to wake up those IVR timers just fine. Right? >>>>>>> >>>>>>> That's a cool design to have one dedicated thread that maintains >>>>>>> accurate >>>>>>> timing and then broadcasts via condition variables to hundreds of >>>>>>> other >>>>>>> threads events that they can register for. I'm sure it's one of the >>>>>>> reasons >>>>>>> why FS scales so much better than Asterisk. But for poor low-end >>>>>>> setups that >>>>>>> sit in the closet, eat only 6W of power and hardly ever run more >>>>>>> than two >>>>>>> calls at the same time, can I hack it somehow to be more UNIX- >>>>>>> friendly? I.e. >>>>>>> make it stuck in select() or recv() when there is nothing to do, >>> call >>>>>>> clock_gettime() right from the thread that wants and when it wants >>>>>>> to know >>>>>>> current time? >>>>>>> >>>>>>> Say, what if that thread is made to suspend on a condition variable >>>>>>> in case >>>>>>> if there are no timers registered in TIMER_MATRIX? Then, if some >>> other >>>>>>> thread comes up and adds its timer into the matrix, it could wake up >>>>>>> the >>>>>>> timer thread and enjoy accurate timing as needed, on demand? And in- >>>>>>> between >>>>>>> the calls, when there is no RTP or IVR, it will all go silent? I >
Re: [Freeswitch-users] Call transfer got broken for me
what revision were you at prior to upgrade or can you narrow the range of versions that broke this any more (or even better the exact version that broke this). Please post this bug to http://jira.freeswitch.org. Mike On Dec 3, 2009, at 10:30 AM, Milena wrote: > Hello, > > It was all ok until yesterday when i updated to svn 15761(last update before > that was about 4 days ago), Now I have this issue: > > someone from the pstn (555) calls through my FXO gw (10.1.1.90) to ext 200 > 200 picks up, then 200 transfers the call to 205 > call gets lost (it used to transfer normal until the moment I updated) > > Today I updated to 15771 and the issue is still there. > Can anyone help me figure out what is going on? > > Call log: http://pastebin.freeswitch.org/11374 > > 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 ___ 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] IAX? Issues connecting road warriors with SIP?
with the right clients, it nearly always works well. with a client that does not support stun or at least rfc 3581 the results are much more sketchy and require more hacks on the server side, but with enough effort can almost always be made to work. Mike On Dec 3, 2009, at 7:17 AM, Fred-145 wrote: > > Hello > > In a thread back in March, I read that support for IAX in FreeSwitch is a > bit of kludge and since there's not much demand for it, chances are it won't > improve in the foreseeable future. > > So I'd like some feedback from users who routinely connect to a FreeSwitch > server from various venues, ie. wifi hotspots at McD, Ethernet LAN in > hotels, etc. (in my case, the FreeSwitch server is located in a private > network behind a NAT router with SIP/RTP ports statically mapped.) > > Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) > ports fail being opened dynamically to work properly, or does SIP today > really work well over NAT firewalls? > > Thank you. > -- > View this message in context: > http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26625105.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > ___ > 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] Gateway issue with no audio
You may want to try this again with latest svn trunk. We have done quite a lot of work to make nat support much better sense 1.0.4 Mike p.s. I can't comment about version 1.4 due to broken flux capacitor. On Dec 3, 2009, at 4:36 AM, Henry Huang wrote: > My freeswitch is using public IP. I setup a gateway registering to voipstunt, > and put it under internal profile. I tried to make call, and I got no RTP > back from the provider... Tried treating NAT issue by changing IP address, > internal IP, external IP. But no use, still getting no audio. > > Finally, I gave up play around with the internal profile and put the gateway > settings under external profile. And magically, it worked. I am getting audio > now. But it leads me to wonders, what's the core difference between external > profile and internal profile. Even if I set the external SIP IP and exteranl > RTP IP to the public IP in internal profile, I am still getting no audio. Can > anyone clear the concept for me here? > > by the way, I am using freeswitch 1.4 stable version. > > > > -- > Henry Huang > > ___ > 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] can't register Inphonex
You can turn up the full freeswitch debug or enable the siptrace on the sip profile to get more information about this. This looks like a nat related issue getting no response from the provider. A sip trace is probably the best tool to figure this one out. sofia profile internal siptrace on Mike On Dec 2, 2009, at 10:35 PM, John Lalande wrote: > I am new to FS having ditched Asterisk a few weeks ago. I have iptel > registered but I cannot get Inphonex to work. I am using the settings > fromhttp://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no > avail. > > The error displayed in the console is "2009-12-02 21:32:55.243917 [ERR] > sofia_reg.c:1442 inphonex Registration Failed with status Request Timeout > [408]." > > Is there some way to debug this? sofia status displays: > > Name Type Data > State > = > external profile sip:mod_so...@192.168.125.15:5080 > RUNNING (0) > example.com gatewaysip:joeu...@example.com > NOREG > inphonex gateway sip:5285...@sip.inphonex.com > FAILED (retry: 28s) > iptel gateway sip:jlala...@sip.iptel.org > REGED > internal profile sip:mod_so...@192.168.125.15:5060 > RUNNING (0) > internal-ipv6 profile sip:mod_so...@[::1]:5060 > RUNNING (0) >192.168.125.15 alias internal > ALIASED > = > > > ___ > 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] HA questions.
The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? Mike On Dec 2, 2009, at 7:49 PM, Tim Uckun wrote: > I have read some of the archived emails about HA, loadbalancing, > failover etc and I am still a bit confused about how I could set up > some sort of resiliency with freeswitch. > > My situation is much less complex than the scenarios people were > talking about and I hoping the solution is similarly much less > complex. > > I have two machines. Both will run freeswitch and also an IVR > application with local databases. I will take care of the database, > application and configuration synchronization between the two > machines. Ideally the calls would be load balanced between the > machines and if any application falls down then the calls should go to > the other machine. Same if I take a machine down for whatever reason. > > If a machine goes down I am willing to "lose" those people who were > making a call at the time. I do have a flag in the application which > will stop answering the calls while processing the existing calls for > a graceful shutdown and hopefully the load balancer would shuttle the > calls to the other machine while this is happening. > > At this stage everything is done via SIP. > > My questions are... > > Do I have to have a sip proxy? If the answer is yes it seems like I > have to set up two sip proxies so I don't have another single point of > failure. Can I load the sip proxies on the same machine? Do I need two > more machines? > > If I take load balancing out of the picture would it be possible to do > a simple linux HA or a windows built in ip failover solution? Would a > simple IP failover work over UDP or would I have to use IAX and tcp/ip > ? > > Is it better to go the virtualization route? > > Sorry if these are dumb questions. I am just trying to get my head > wrapped around this. I don't need five nines (although that would be > awesome), I just want a reasonable degree of assurance that my app can > keep taking calls in case something weird happens. ___ 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] Eavesdrop error?
The behavior is probably expected, the unhelpful error is probably undesirable but it would make a mess of the dial-plan to clean that up. Mike On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote: > Is this reasonable given it was the only call in FreeSwitch at the time? How > can this situation be corrected in the future? > > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: Wednesday, December 02, 2009 3:35 PM > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] Eavesdrop error? > > it probably just means the uuid was not retrieved from the db when you > called the eavesdrop exten which does the lookup on the uuid for the hash > key based on what ext you hit to retrieve the most recent uuid that called > that ext. > > > On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: > Sorry, svn 15753 > > -Original Message- > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lars Zeb > Sent: Wednesday, December 02, 2009 2:08 PM > To: freeswitch-users@lists.freeswitch.org > Subject: [Freeswitch-users] Eavesdrop error? > > I tried to use eavesdrop today and it did not work. The error message in the > log is: > > [ERR] mod_dptools.c:334 Usage: [all | ] > > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this > incorrect? > > http://pastebin.freeswitch.org/11363 > > 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:213-799-1400 > > > ___ > 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] Best way to run originate calls through dial plan
http://wiki.freeswitch.org/wiki/Mod_commands#originate
Usage: originate |&()
[] [] [] [] []
You can do this via shelling out to fs_cli like your example below or using esl
directly from php:
http://wiki.freeswitch.org/wiki/Esl
Mike
On Dec 2, 2009, at 1:23 PM, eaf wrote:
>
> I need a way to start a call from the PHP script to the originating number,
> tell the party on that number to hold on, start another call to destination
> number, and bridge everything together. On both legs I need to pass custom
> caller ID. I can of course open direct connections to VOIP gateways right
> from PHP, but I want to reuse existing routing rules in the dial plan, hence
> I want to know what's the best way of making originate go through a specific
> context of the dial plan.
>
> As for the number of calls per second, it's going to be only occasionally
> used.
>
>
> mercutioviz wrote:
>>
>> On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote:
>>
>>>
>>> What would be the best way of making originate() run call through a dial
>>> plan
>>> (compared to directly going to a specified VOIP gateway). Would it be
>>> loopbacks, i.e. smth like this?
>>>
>>> /opt/freeswitch/bin/fs_cli -x "originate
>>>
>>> {ignore_early_media=true,origination_caller_id_number=xx}loopback/yy/default/XML
>>> '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js
>>> zz)'"
>>>
>>> The idea of this is that originate() sets up the first call, then
>>> webcall.js
>>> plays back a WAV, and bridges the first call with the second one (also
>>> set
>>> up via loopback).
>>>
>>>
>> Could you describe the problem that you're trying to solve? That would
>> make
>> it easier to know if what you've come up with is the best solution. How
>> many
>> calls per second were you wanting to generate with this setup?
>> -MC
>>
>> ___
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>>
>>
>
> --
> View this message in context:
> http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp26610094p26613841.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> ___
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Re: [Freeswitch-users] How to run a JS script periodically
You could also use the scheduler to run the jsrun command inside FreeSWITCH. Mike On Dec 3, 2009, at 8:31 AM, Rob Forman wrote: > What about cron? > > Create a cron entry like: > */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()" > > But if you're just dumping global variables, you could easily retrieve them > directly from fs_cli without running an app and process the output however > you'd like: > > /usr/local/freeswitch/bin/fs_cli -x "global_getvar" > > > On Thu, Dec 3, 2009 at 6:21 AM, Oscav wrote: > > Hi, > > Someone knows how to run periodically a JS script ?? The purpose is to write > to a db some global informations (Global Variables) about FS like every 5 > minutes. ___ 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] Choppy sound with PCMU
First off, maybe this conversation is better suited to the dev list, and second off, the current setup of where we do timers, where we poll, polling frequency and architecture is the result of 4+ years of ongoing testing and optimization. We have tried all different methods throughout. Sometimes what we found to be most efficient is not what we thought at first would be, but testing showed otherwise. We have always optimized the general case as to if there are many calls, and no suggestion would be implemented that hurts this case. That being said, if you could really come up with a way for this to be more efficient in any case, without sacrificing performance int he other cases, you are able to prove this with extensive test results, and you are able to prove that it does not impact for example call quality in any of the hundreds of edge cases that have led us to the point we are now, then we may be interested in taking such a patch. Mike On Dec 2, 2009, at 11:58 PM, eaf wrote: > > As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, it > could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" overrides > that. > > Yeah, there is a global timestamp... It's easy to workaround that for RTP > who calls switch_micro_time_now()... But if somebody accesses > runtime.timestamp directly, it's gonna be tough to grep for that. If only > this was C++... > > I'll play around. Never liked polling too much. Never could've guessed that > polling could be so useful for scalability ;) My naive implementation > would've pulled timestamp via system calls and would've done sleeping by > passing exact interval to select() instead of syncing with a pacing thread. > Which would be dead-quiet at idle time, but, of course, would stop scaling > at some point due to excessive number of system calls. > > Thanks. > > > Michael Jerris wrote: >> >> In short. No, you can not for many reasons. The milisecond tic is >> used throughout the code even when there is not any calls up. You can >> grep for switch_cond_next if you would like to see where but it is >> required to keep our global timestamp and for pacing the scheduler >> among other services that run all the time. >> >> Mike >> >> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >>> >>> Can I reduce resolution of that timer thread 10 times? I mean, I >>> glanced >>> through the code, and see that among others (are there others?) RTP >>> and IVR >>> set up their timers that are subsequently managed by this thread. >>> RTP timers >>> should be eliminated by that setting you've suggested. IVR timers >>> are set at >>> 20ms... So, if the thread is set to wake up every 10ms instead of >>> 1ms it >>> should be able to wake up those IVR timers just fine. Right? >>> >>> That's a cool design to have one dedicated thread that maintains >>> accurate >>> timing and then broadcasts via condition variables to hundreds of >>> other >>> threads events that they can register for. I'm sure it's one of the >>> reasons >>> why FS scales so much better than Asterisk. But for poor low-end >>> setups that >>> sit in the closet, eat only 6W of power and hardly ever run more >>> than two >>> calls at the same time, can I hack it somehow to be more UNIX- >>> friendly? I.e. >>> make it stuck in select() or recv() when there is nothing to do, call >>> clock_gettime() right from the thread that wants and when it wants >>> to know >>> current time? >>> >>> Say, what if that thread is made to suspend on a condition variable >>> in case >>> if there are no timers registered in TIMER_MATRIX? Then, if some other >>> thread comes up and adds its timer into the matrix, it could wake up >>> the >>> timer thread and enjoy accurate timing as needed, on demand? And in- >>> between >>> the calls, when there is no RTP or IVR, it will all go silent? I mean, >>> sitting on a wait queue in the kernel is way better than go back and >>> forth >>> incrementing counters that nobody even needs at the moment? >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> idle is a 4 letter word to a realtime application. >>>> >>>> The core keeps a single high-priority thread to keep 1ms timing and >>>> expands >>>> that broadcasting >>>> to hundreds or thousand of threads who need acc
Re: [Freeswitch-users] Choppy sound with PCMU
In short. No, you can not for many reasons. The milisecond tic is used throughout the code even when there is not any calls up. You can grep for switch_cond_next if you would like to see where but it is required to keep our global timestamp and for pacing the scheduler among other services that run all the time. Mike On Dec 2, 2009, at 7:31 PM, eaf wrote: > > Can I reduce resolution of that timer thread 10 times? I mean, I > glanced > through the code, and see that among others (are there others?) RTP > and IVR > set up their timers that are subsequently managed by this thread. > RTP timers > should be eliminated by that setting you've suggested. IVR timers > are set at > 20ms... So, if the thread is set to wake up every 10ms instead of > 1ms it > should be able to wake up those IVR timers just fine. Right? > > That's a cool design to have one dedicated thread that maintains > accurate > timing and then broadcasts via condition variables to hundreds of > other > threads events that they can register for. I'm sure it's one of the > reasons > why FS scales so much better than Asterisk. But for poor low-end > setups that > sit in the closet, eat only 6W of power and hardly ever run more > than two > calls at the same time, can I hack it somehow to be more UNIX- > friendly? I.e. > make it stuck in select() or recv() when there is nothing to do, call > clock_gettime() right from the thread that wants and when it wants > to know > current time? > > Say, what if that thread is made to suspend on a condition variable > in case > if there are no timers registered in TIMER_MATRIX? Then, if some other > thread comes up and adds its timer into the matrix, it could wake up > the > timer thread and enjoy accurate timing as needed, on demand? And in- > between > the calls, when there is no RTP or IVR, it will all go silent? I mean, > sitting on a wait queue in the kernel is way better than go back and > forth > incrementing counters that nobody even needs at the moment? > > > Anthony Minessale-2 wrote: >> >> idle is a 4 letter word to a realtime application. >> >> The core keeps a single high-priority thread to keep 1ms timing and >> expands >> that broadcasting >> to hundreds or thousand of threads who need accurate timing. >> >> Your choppy audio is caused by linksys lying about the packet len >> that >> it's >> using and we set our timer >> to the wrong speed. >> >> > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > ___ > 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] Choppy sound with PCMU
This is keeping track of a place in the music on hold so your hold music does not start back up at the same place every time. If you don't want to do this it is a module that you don't need to load and you can get your moh from any soundfile at your choice in configuration. Mike On Dec 2, 2009, at 10:35 PM, eaf wrote: > > Oh, looks like the timers are also used for streaming local data in > read_stream_thread(). Due to this there is always one timer active > with 20ms > interval. > > But wait a sec, why is freeswitch periodically trying to stream > /opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav > somewhere? > Every minute or so? Did I misconfigure it? > > > eaf wrote: >> >> Say, what if that thread is made to suspend on a condition variable >> in >> case if there are no timers registered in TIMER_MATRIX? Then, if some >> other thread comes up and adds its timer into the matrix, it could >> wake up >> the timer thread and enjoy accurate timing as needed, on demand? And >> in-between the calls, when there is no RTP or IVR, it will all go >> silent? >> I mean, sitting on a wait queue in the kernel is way better than go >> back >> and forth incrementing counters that nobody even needs at the moment? >> >> >> Anthony Minessale-2 wrote: >>> >>> idle is a 4 letter word to a realtime application. >>> >>> The core keeps a single high-priority thread to keep 1ms timing and >>> expands >>> that broadcasting >>> to hundreds or thousand of threads who need accurate timing. >>> >>> Your choppy audio is caused by linksys lying about the packet len >>> that >>> it's >>> using and we set our timer >>> to the wrong speed. >>> >>> >> >> > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26620518.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > ___ > 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] Problem with compiling revision 15739
I think I just fixed this a few minutes ago, it is running test builds on the build servers now to verify. On Dec 1, 2009, at 2:19 PM, John Platts wrote: > > I attempted to do a make current with revision 15739, but some of the Sofia > source files will not compile with revision 15739. Those source files were > not changed between revisions 15738 and 15739. I am using GCC 4.1.2 to > compile FreeSWITCH. I used the following to get revision 15738, which was the > previous revision, built: > make update-clean > svn update -r 15738 > make all install > > This does the same stuff as make current, except that revision 15738 is > checked out of the SVN repository. > > _ > Windows 7: Unclutter your desktop. Learn more. > http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid=1&media=aero-shake-7second&listid=1&stop=1&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 > ___ > 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] freeswitch changes c= to FS's IP with bypass_media=true
The only way this would happen would be if this is set to proxy media not bypass. Are you setting both? Mike On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: In the following trace,102 is FS IP, 104 is calling party and 13 is called party. with bypass_media, FS still changesc=IN IP4 192.168.1.102 Any idea why? freeswi...@localhost.localdomain> recv 951 bytes from udp/ [192.168.1.104]:5060 at 22:56:33.782715: --- - INVITE sip:90964...@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104: 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport From: ;tag=786224322 To: Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104 CSeq: 37 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite Content-Length: 397 v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv --- - send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: --- - SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104: 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 From: ;tag=786224322 To: Call-ID: 003c8e1b-f8dc-de11-a853-001a80565...@192.168.1.104 CSeq: 37 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Content-Length: 0 --- - 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel sofia/internal/phonerl...@192.168.1.102 [d4233c9a- ee3b-40d4-910d-3b1579f9a273] 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/ internal/phonerl...@192.168.1.102 entering state [received][100] 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 EXECUTE sofia/internal/phonerl...@192.168.1.102 info() 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/phonerl...@192.168.1.102] Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/phonerl...@192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259708193783162] Caller-Channel-Created-Time: [1259708193783162] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [phonerl...@192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [786224322] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964...@192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964...@192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [phonerl...@192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/phonerl...@192.168.1.102] variable_sip_call_id: [003C8E1B-F8DC-DE11- a853-001a80565...@192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_bypass_media: [true] variable_proxy_media: [true] variable_sip_via_rport: [5060] variable_m
Re: [Freeswitch-users] Passing incoming remote-party-id from called to caller
What is the jira bug number on this voicemail email issue? I don't recall seeing it. Mike On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine wrote: > > Are you on SVN trunk? As far as I recall the callee_id_number/name > stuff isnt in 1.0.4. > > No, because the SVN has problems with Emailing the voicemail... > > We use 1.0.4 and set sip_callee_id_number/name which works. I would > like to not set it and get it from the other side... > > 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 ___ 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] errors installing wanpipe drivers
make openzap is the correct way to build when using with openzap/freeswitch. If you are having issues with this you should check with sangoma support as to why that build of the drivers is not supporting it properly and what version you should be using. Mike On Nov 30, 2009, at 5:41 AM, François Legal wrote: > I did manage to build these drivers, but maybe you're not doing it the right > way. Sangoma document state that the drivers should be built by using their > ./Setup script that does all that is required. > > I did use ./Setup install which builds the kernel modules, the wanrouter > utilities and install all the required stuff. > > Then you can go back to freeswitch and build the mod_openzap/libopenzap. > > > François > > > On Mon, 30 Nov 2009 15:19:43 +0530, Neil Patel wrote: > > Hi All, > > I am currently installing a Sangoma A102 card to work with FS using wanpipe > drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related > modules to compile: > > > cd wanpipe-3.5.6.5/ > > make openzap > ... > make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' > make -C api/libstelephony clean > make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make[1]: *** No rule to make target `clean'. Stop. > make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' > make: *** [all_lib] Error 2 > > The libstelephony directory has no Makefile in it. Why is it missing? Is > there a version of wanpipe drivers that will work? I have been unsuccessful > with 3.4.4 and 3.5.6 in similar fashion. > > 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 ___ 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] custom call counter
It depends on the timing of when your increment and decrement are vs when the sql calls to push the events into the tables that are used for show calls are. Also, the sql calls are batched and queued causing a little delay (less than a second). If your doing a lot of short lived calls there is sure to be timing discrepancy. I am sure there is even more discrepancy if you look at the output of status which shows the current number of sessions (those are individual call legs) as that information is a little more real time. Mike On Nov 28, 2009, at 12:48 AM, Juan Backson wrote: > Hi, > > Instead of using "show calls count" to obtain the current call count stat, I > am writing some C code to increment a counter during on_answer_hook and > decrement the counter during on_hangup_hook. > > It looks like my counter result is very closed to "show calls count" when the > traffic is low, like 50 -60. But when traffic is high, like 1000 calls, my > counter is showing 30% less. > > When all calls are finished, my counter becomes 0 again, and that proves that > it does not increment/decrement more than it should. > > Is this normal? Does anyone have any idea why there is such discrepancy? > > 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] No NOTIFY MWI when registering via proxy.
Does the alias you added match the one that you saw in the event? The alias is 100% for sure the fix for this issue, please check again. Mike On Nov 26, 2009, at 6:55 PM, Peter P GMX wrote: > I tried now with phones directly attached to the freeswitch (without an > OpenSIPS in between). I also added the alias. But the behaviour is as > before: > No notify message from freeswitch, neither after register nor after a > voicemail is recorded. > > Best regards > Peter > Brian West schrieb: >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >> >>> Try an alias on the sip profile. >>> >>> 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] odbc FLAG_MULTI_STATMENTS
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445) On Nov 26, 2009, at 2:22 PM, Frank @ Impact wrote: > “GREAT SCOTT!!! Cannot execute batched statements! > If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and > enable FLAG_MULTI_STATEMENTS” > > I realize a bit off of list topic… > > But I do have mysql 3.51.18 and higher but for the life of me , I cannot seem > to get the DSN config setup so that the odbc connector seems to tell FS that > it can do multi statements. > > Anyone have any insight on how and where to set this flag? > > > ___ > 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] dialplan rule to send the caller to voice mail when same extension is called.
Of course. Please read through the default configs and the getting started guide and xml dialplan information on the wiki. Mike On Nov 26, 2009, at 12:38 PM, Orien Love wrote: > Is there any way to build a dial plan so that when an extension calls > itself the call is automatically put to that users voice mail? > > Example, extension 1001 calling 1001 and is sent to voice mail (to > receive messages). > I know that there is a * code to get to voice mail, I cannot recall > which one right now but my phones want to dial their extension to get to > voice mail.I can modify the voice mail button but this works only for > the first line registered at that 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] Problems with nat
In this case you should not need 2 profiles either. On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote: > It's a windowsserver which is behind a router. > > Which profile should local-network-acl be specified on? > > When I bridge calls to the outside world, should I use > sofia/internal/@gateway or sofia/external/@gateway? > > > On Thu, Nov 26, 2009 at 4:42 PM, Brian West wrote: > Are you doing this all on a linux box thats acting as your router too? If > not you don't need two profiles... you also don't need to set the > local-network-acl on ANY profile that isn't do anything with nat. > > /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] Callback to the user in ESL
Your using outbound socket and you hangup the call, so it tells you it
is done with the server disconnected message and drops the
connection. This is all as expected. I guess I don't understand what
you think is the problem. This code is doing exactly what I would
expect it to do.
Mike
On Nov 26, 2009, at 4:27 AM, lakshmanan ganapathy
wrote:
Hi, Any help or suggestion regarding my previous post. Especially
"I also noted that, if I don't receive any events, especially
"SERVER_DISCONNECTED", then the connection is in established state,
but once I receive the "SERVER_DISCONNECTED" event, the connection
is closed. Is it correct??"
Here is the program by which I confirmed the above!
require ESL;
use IO::Socket::INET;
my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort =>
'8447', Proto => 'tcp', Listen => 2, Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;
my $con;
my $type = "user/";
for(;;) {
# wait for any client to connect, a new client will get
connected when a new call comes in the dialplan.
my $new_sock = $sock->accept();
# Do fork and let the parent to wait for more clients.
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
# Extract the host of the client.
my $host = $new_sock->sockhost();
# file descriptor for the socket.
my $fd = fileno($new_sock);
print "Host name is $host\n";
# Create object for the ESL connection package to access the
ESL functions.
$con = new ESL::ESLconnection($fd);
# Gets the info about this channel.
my $info = $con->getInfo();
my $uuid = $info->getHeader("unique-id");
printf "Connected call %s, from %s to %s\n", $uuid, $info-
>getHeader("caller-caller-id-number"), $info->getHeader("caller-
destination-number");
# Answer the channel.
$con->execute("answer");
# Set the event lock to tell the FS to execute the
instructions in the given order.
$con->setEventLock("true");
# Play a file & Get the personal number from the user.
$con->execute("playback","/usr/local/freeswitch/sounds/en/us/
callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
$con->execute("hangup");
while($con->connected())
{
my $e=$con->recvEvent();
my $ename=$e->getHeader("Event-Name");
print $e->serialize();
print "$ename\n";
print "Connection exists\n";
sleep(1);
}
print "Bye
\n--
\n";
close($new_sock);
}
I've not registered for any events.
In the above program I'm receiving the SERVER_DISCONNECTED event.
Output when receiving event:
Host name is 192.168.1.222
Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000
to 9097
Event-Name: SERVER_DISCONNECTED
SERVER_DISCONNECTED
Connection exists
Bye
When I comment the recvEvent line, I got the following output.
Host name is 192.168.1.222
Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000
to 9097
Connection exists
Connection exists
Connection exists
Connection exists
Connection exists
On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy > wrote:
I've tried the following program as per the suggestion that you've
told. But it seems, no success. Once the connection is closed, I
created a new connection and I send originate to originate a new
call. But it is not working.
require ESL;
use IO::Socket::INET;
use Data::Dumper;
my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort =>
'8447', Proto => 'tcp', Listen => 2, Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;
my $make_call;
my $con;
my $type = "user/";
for(;;) {
my $new_sock = $sock->accept();
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
my $host = $new_sock->sockhost();
my $fd = fileno($new_sock);
$con = new ESL::ESLconnection($fd);
my $info = $con->getInfo();
my $uuid = $info->getHeader("unique-id");
printf "Connected call %s, from %s to %s\n", $uuid, $info-
>getHeader("caller-caller-id-number"), $info->getHeader("caller-
destination-number");
$con->filter("Unique-Id", $uuid);
$con->events("plain", "all");
$con->execute("answer");
$con->setEventLock("true");
my $number=$con->execute("read","2 4 /usr/local/freeswitch/
sounds/en/us/callie/conference/8000/conf-pin.wav accnt_number 5000
#");
while($con->connected())
{
my $e=$con->recvEvent();
my $ename=$e->getHeader("Event-Name");
my $app=$e->getHeader("Application");
if(
Re: [Freeswitch-users] ESL command completion
There are execute_complete events. I can't recall everything that is in them but they should always be fired. Mike On Nov 25, 2009, at 8:38 PM, Josh Rivers wrote: > Is there a way of determining if a call-command sent to a session via ESL has > completed? Is there a return event which is always fired? Is there a > identifier I can use to verify that the return event matches my command? > > Thanks, > Josh > ___ > 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] Handling the 302 Moved Temporarily response from JavaScript
from http://svn.freeswitch.org/svn/freeswitch/trunk/conf/sip_profiles/internal.xml It appears this never made the wiki, could someone please get it on there. Thanks Mike On Nov 25, 2009, at 6:21 PM, John Platts wrote: > > How do I turn on dialplan processing of 302 responses? I can solve my problem > if I can process 302 responses in my dialplan. > > >> From: m...@jerris.com >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> To: freeswitch-users@lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily response >> from JavaScript >> >> In trunk there is a sofia profile setting to allow dialplan processing of >> 302 responses. This won't get you back into your same javascript, but you >> can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily response if >>> I use JavaScript? >>> >> >> ___ >> 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 > > _ > Bing brings you maps, menus, and reviews organized in one place. > http://www.bing.com/search?q=restaurants&form=MFESRP&publ=WLHMTAG&crea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 > ___ > 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] Grandstream gateways
On Nov 25, 2009, at 5:18 PM, Adam Ford wrote: > Samuel, > > FreeSWITCH has a Skype module that uses Skype client instances to connect to > the Skype network, you can read about it at > http://wiki.freeswitch.org/wiki/Skypiax > > As far as an official Skype module for non-Asterisk PBX-es, it looks like it > is in beta right now - > http://www.skype.com/business/products/pbx-systems/sip/ > > -AF If by in beta you mean they turned off all the servers the beta testers could talk to, then yes, it is indeed. 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] Bypass_media and re_invite
"something that is not available in that lib at this time." Mike On Nov 25, 2009, at 2:47 PM, srinivasula reddy wrote: > can please tell me how can i exchange session state into sip library. > > Thanks > srinivas > > On Wed, Nov 25, 2009 at 11:47 PM, Michael Jerris wrote: > For that you would need to fully exchange session state into the sip library, > something that is not available in that lib at this time. > > > On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > >> HI, >> thanks for your reply, my requirement is i am doing failover stuff with >> freeswitch. i dont want cut the calls when freeswitch dies, when failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to end the >> call at that time i want to close the call for the other end also. >> > > ___ > 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 > > > > > -- > Srinivasula Reddy K > ___ > 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] Bypass_media and re_invite
For that you would need to fully exchange session state into the sip library, something that is not available in that lib at this time. On Nov 25, 2009, at 12:55 PM, srinivasula reddy wrote: > HI, > thanks for your reply, my requirement is i am doing failover stuff with > freeswitch. i dont want cut the calls when freeswitch dies, when failover > happens mean one freeswitch dies we are going to start the second freeswitch, > i dont want close call intiated by the first freeswtich, they are > communicating with meida(bypass media). when one endpoing try to end the call > at that time i want to close the call for the other end also. > ___ 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 find whether the destination extension supports encryption
You can send the call with secure enabled and if it supports it it will use it. Mike On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: > Hello, > > We have a mix of phones that support RTP encryption and those that do not. > I have to support both types in the meanwhile, and would like to have > encryption enabled on the relevant leg, even if the other leg does not > support it (why? one of our ATAs either must have it unencrypted or have it > encrypted, but cannot have both). > > How do I find whether the destination supports encryption? I do not want to > manage an additional table in the database... > ___ 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] remote_media_ip variable not set
It's possible it does not. I just added some code to set it on auto-adjust so it might be there sometimes now. You might need to add some code in mod_sofia to add it other times. Maybe it makes sense to move that var setting down to switch_rtp.c. Patches for this would be welcome. Thanks Mike On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: > Hi, > > In the case of proxy_media=true, does it gets set at all 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] Handling the 302 Moved Temporarily response from JavaScript
In trunk there is a sofia profile setting to allow dialplan processing of 302 responses. This won't get you back into your same javascript, but you can probably do something clever from there. Mike On Nov 24, 2009, at 5:04 PM, John Platts wrote: > > I have considered writing JavaScript code to bridge two calls together. > However, I would like to perform custom handling of the 302 Moved Temporarily > response. How do I handle the 302 Moved Temporarily response if I use > JavaScript? > ___ 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 kick to abort invitations
Its a feature we don't have, patches welcome. Mike On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: > Hi members, > I’m controlling freeswitch with the conference module via xmlrpc. > > Is it desired that the kick command can only kick users that are connected to > the conference? > Is there no chance abort an invitation? > The kick command has no effect until the person I invited with the dial > command is connected. ___ 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 and re_invite
FreeSWITCH will kill the calls when you shut it down, if you intentionally kill the network without shutting down FreeSWITCH the only thing you can do is enable session timers or rtp timers in the soft phones to kill the call when FreeSWITCH dies or when the call is over. Mike On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: > Hi All, > > goodmorning to all, i have a scenario, two pjsua clients are connected with > Freeswitch and they are in call and bypass_media=true. i close the > Freeswitch server, still they are in call, again i started the Freeswitch, > and registerd these two endpoints, now how can i end the call(estabilished by > the first Freeswitch)? if i call re_invite will it estabilish the call > between two endpoints? > any idea? ___ 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 NOTIFY MWI when registering via proxy.
e_destroy: entering >>nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized >>nua: nua_stack_set_params: entering >>nta: sent 401 Unauthorized for REGISTER (7) >>nta: timer set to 32000 ms >>nua(0x7fd5d409c8f0): recv signal r_destroy >>nta_leg_destroy((nil)) >>nta: received REGISTER sip:sip1.mydomain.com >><http://sip1.mydomain.com> SIP/2.0 (CSeq 6) >>nta: REGISTER (6) going to a default leg >>nua: nua_stack_process_request: entering >>nua: nh_create: entering >>nua: nh_create_handle: entering >>nua: nua_stack_set_params: entering >>nua(0x905a80): event i_register 100 Trying >>nua: nua_application_event: entering >>nua: nua_respond: entering >>nua(0x905a80): sent signal r_respond >>nua: nua_handle_destroy: entering >>nua(0x905a80): recv signal r_respond 401 Unauthorized >>nua(0x905a80): sent signal r_destroy >>nua: nua_stack_set_params: entering >>nua: nua_handle_magic: entering >>nua: nua_handle_destroy: entering >>nta: sent 401 Unauthorized for REGISTER (6) >>nua(0x905a80): recv signal r_destroy >>nta_leg_destroy((nil)) >>nta: received PUBLISH sip:1...@sip1.mydomain.com >><mailto:sip%3a...@sip1.mydomain.com> SIP/2.0 (CSeq 3) >>nta: PUBLISH (3) going to a default leg >>nua: nua_stack_process_request: entering >>nua: nh_create: entering >>nua: nh_create_handle: entering >>nua: nua_stack_set_params: entering >>nua(0x905f10): event i_publish 100 Trying >>nua: nua_application_event: entering >>nua: nua_respond: entering >>nua(0x905f10): sent signal r_respond >>nua: nua_handle_magic: entering >>nua: nua_handle_destroy: entering >>nua(0x905f10): recv signal r_respond 200 OK >>nua: nua_stack_set_params: entering >>nua(0x905f10): sent signal r_destroy >>nta: sent 200 OK for PUBLISH (3) >>nua(0x905f10): recv signal r_destroy >>nta_leg_destroy((nil)) >>nta: received SUBSCRIBE sip:mod_so...@192.168.178.200:5062 >><http://sip:mod_so...@192.168.178.200:5062> SIP/2.0 (CSeq 2) >>nta: canonizing sip:mod_so...@192.168.178.200:5062 >><http://sip:mod_so...@192.168.178.200:5062> with contact >>nta: SUBSCRIBE (2) going to existing leg >>nua: nua_stack_process_request: entering >>nta: sent 200 OK for SUBSCRIBE (2) >>nua(0x905560): event i_subscribe 200 OK >>nua: nua_application_event: entering >>nta: received REGISTER sip:sip1.mydomain.com >><http://sip1.mydomain.com> SIP/2.0 (CSeq 8) >>nta: REGISTER (8) going to a default leg >>nua: nua_stack_process_request: entering >>nua: nh_create: entering >>nua: nh_create_handle: entering >>nua: nua_stack_set_params: entering >>nua(0x7fd5dc073ba0): event i_register 100 Trying >>nua: nua_application_event: entering >>nua: nua_respond: entering >>nua(0x7fd5dc073ba0): sent signal r_respond >>nua(0x7fd5dc073ba0): recv signal r_respond 200 OK >>nua: nua_stack_set_params: entering >>nua: nua_handle_destroy: entering >>nua(0x7fd5dc073ba0): sent signal r_destroy >>nua: nua_handle_magic: entering >>nua: nua_handle_destroy: entering >>nta: sent 200 OK for REGISTER (8) >>nua(0x7fd5dc073ba0): recv signal r_destroy >>nta_leg_destroy((nil)) >>nta: received REGISTER sip:sip1.mydomain.com >><http://sip1.mydomain.com> SIP/2.0 (CSeq 7) >>nta: REGISTER (7) going to a default leg >>nua: nua_stack_process_request: entering >>nua: nh_create: entering >>nua: nh_create_handle: entering >>nua: nua_stack_set_params: entering >>nua(0x8fc3d0): event i_register 100 Trying >>nua: nua_application_event: entering >>nua: nua_respond: entering >>nua(0x8fc3d0): sent signal r_respond >>nua(0x8fc3d0): recv signal r_respond 200 OK >>nua: nua_handle_destroy: entering >>nua: nua_stack_set_params: entering >>nua(0x8fc3d0): sent signal r_destroy >>nua: nua_handle_magic: entering >>nua: nua_handle_destroy: entering >>nta: sent 200 OK for REGISTER (7) >>nua(0x8fc3d0): recv signal r_destroy >>nta_leg_destroy((nil)) >>nta: received SUBSCRIBE sip:1...@sip1.mydomain.com >><mailto:sip%3a...@sip1.mydomain.com>;user=phone SIP/2.0 >>(CSeq 1) >>nta: SUBSCRIBE (1) going to a default leg >>nua: nua_stack_process
Re: [Freeswitch-users] How to connect SIP phone to freeswitch
http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Nov 25, 2009, at 1:36 AM, ovvenkat wrote: > Hi . > > Could you please tell me, How to connect sip phone (which one is more > friendly with freeswitch) to freeswitch. How I can check whether connection > is properly established or not? > ___ 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] Problem while playing more than 10 voice files using playback
"you should use execute_complete events to tell when a command you tried to execute has finished and not poll the channel for a variable to be set because FreeSWITCH is an asynchronous application in the mode you are describing and you can never be sure of the timing." You are STILL polling for the variable. If you want help, perhaps you should at least attempt what is being suggested? Mike On Nov 25, 2009, at 1:18 AM, Thangappan.M wrote: > The example script is there in the following link > http://pastebin.com/f332f2fda > > In the previous post I have attached it. But it was not shown. > > 2009/11/25 Thangappan.M > FreeSWITCH version: freeswitch 1.0.4 > I am using ESL library > I attached the example Perl script which does the same steps that I posted > already. ( Sample.pl) > I supplied the log , Here I attached the output of the ESL log. (Output.txt) > > Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. > But in the output I got only 2,4,5,4 ( DTMF 1 is missed) > > Output of Perl code could be like > > Wait for response time out > EVENT [COMMAND] > Wait for response time out > EVENT [DTMF] > DTMF digit 2 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 5 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > Buffer: 2454 > BYE > > Why the first digit(1) is missed here? > In ESL log there is no digit called 1 why? > Why the COMMAND event is received instead of DTMF? > How can I get all DTMF digits? > > > > > > > > > > > > > > > > > On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M > wrote: > The reason for waiting only for DTMF event is to handle the time outs in the > IVR concept like response and inter digit time out. Using our own logic we > 10 voice files in each play back if the voice files are more than 10. Now it > works fine. > > Now the new problem has been raised. The problem is we are filtering only for > DTMF events but we are getting COMMAND event . Because of this the DTMF > digits are missing at the time . I am not able to proceed further. We are > in the critical situation. > > Why this command event is occurring? > How can I restrict this? > What are the information it has? > How can I get all the information in it ? ( If command event has info) > > Help me > > > On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M > wrote: > I am waiting only for DTMF events. That's why I am setting freeswitch > variable for knowing whether the playback has done. > > My question is "why this freeswitch variable is not setting properly when I > play back more than 10 files using playback_delimiter option?". > > When I play back lesser than ten voice files the variable has been set > properly. What could be the reason? > > > > -- Forwarded message -- > From: Thangappan.M > Date: Sat, Nov 21, 2009 at 2:52 PM > Subject: Problem while playing more than 10 voice files using playback > To: freeswitch-users > > > Dear all, > > I am in the process of implementing IVR using event outbound socket > (async mode). > I have implemented using Perl language. > > I did the following steps: >=> Set the playback_delimiter variable >=> Set the playback_sleep_val variable >=> Set the event lock as true > => Set the freeswitch ( my own) variable as zero > => Wait in the loop until the variable is been set as zero >=> Playback the voice files ( Here I combined the voice > files with the delimiter value if more than one voice files are there) >=> Set the freeswitch(my own) variable as true ( This is > used to identify whether the voice files are played > successfully). >=> Wait in the loop until the variable is been set as one. >=> Set the Event lock as false > >=> Trying to get the DTMF digits ( Have a assurance that > all the voice files are played). > >The problem is, > > The above steps are working fine when the voice file count is > lesser than or equal to 10. After the voice files are played only the > variable(my own freeswitch) is set. Based on the variable I am doing further > things. > > But when I tried to give the voice files count of more than 10 > the variable has been set while starting to play back the first voice file > itself . Because of this I am not able to proceed further. > > DID I MAKE ANY MISTAKE IN THE ABOVE STEPS? > > NOTE: I also referred mod_file_string documentation. In that they specified > 128 files can be used to play back the voice files using playback_delimiter > option. > > Please help me? > Thanks in advance. > >
Re: [Freeswitch-users] Call Transfer Help Please
On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote: > Hi, > > I'm trying to setup call transfer for a phone without a transfer button. I > was on IRC last night and got some pointers to how this is setup in > dialplan.xml and features.xml and what "bind meta app" does. > > Once it became clear how the transfer is initiated and that the transfer, in > the default config, can only be initiated by the "b" leg of the call, I was > able to make this work as configured in the defaults, i.e, to initiate a > transfer (for an internal call) from the dialled extension to a new extension. > > Now the problem . . . > > I have an incoming PSTN line that rings a group of extensions, what I want to > be able to do is to give whoever answers the PSTN call ability to transfer > the call on to another extension. > > There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch > extension of 1000, it rings the extension phones in the group. > > I'd hoped that the default transfer setup would handle this without > modification - the incoming call on extension 1000 would be the "a" leg, the > answering extension would be the "b" leg and a transfer from "b" would work > as per the default config. This does not work for me though. > > I'm struggling a bit with the "bind meta app" options and can't seem to make > it do what I want. > > Could someone please confirm that what I'm trying to do is feasible and > perhaps suggest the right parameters to use in dialplan.xml and features.xml > please ? > > Relevant section in the "is_transfer" section in features.xml > > > And in default.xml from > to > > I've tried posting a call log to the Pastebin (11252/3) but there was an > error - it looks like the dump was too big. Not sure what the maximum size on > pastebin dumps is ? > > > My understanding (or lack of) of "a" and "b" are in the scenario described is > not helping ... > > Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Yes, the calling leg > Is the answering extension the "b" leg ? Yes > What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? I don't understand this question > What is the correct "transfer" data string in features.xml ? > ditto > Or am I totally on the wrong track here ? > You should just need to make sure that the bind meta is called in this scenario so the b leg is able to do it, thats it. > If it is possible to do what I want, and changes are required to the > dialplan.xml and/or features.xml files, is it possible to have different > logic in there such that the actions are different whether it is the "a" leg > or "b" leg that's requesting the transfer ? > > regards > Dave > > FreeSwitch Version 1.0.4 (14460) also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an issue that has already been fixed. 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] DTMF Event is not coming while using playback terminators.
1. can you supply a trace of this esl communications. 2. is it inband or rfc2833 dtmf ? MIke On Nov 24, 2009, at 3:59 AM, velusamy velu wrote: > Yes, I am using async mode only.. > > On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris wrote: > async? > > On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > > > Dear All, > > I am using Perl ESL::IVR module to develop a simple IVR. I have > > filtered DTMF events. I have also set playback_terminators to cut the > > playback when giving the digits. I have faced problem that DTMF event has > > not come if DTMF given while playing voice files. I have received 'COMMAND' > > event. I have the following questions. > > > >Why the 'COMMAND' event came event filter is on? > >How to avoid this event in ESL? > ___ 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] need help !! Problem with freeswitch & uniMRCP
What does this have to do with uniMRCP? Mike On Nov 24, 2009, at 9:54 AM, Imthiyaz Ahmed wrote: > Hi > > Can we enable passive recording in freeswitch ,wanpipe ,openzap , we > are using a sangoma tapping system with freeswitch. ___ 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] DTMF Event is not coming while using playback terminators.
async? On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: > Dear All, > I am using Perl ESL::IVR module to develop a simple IVR. I have > filtered DTMF events. I have also set playback_terminators to cut the > playback when giving the digits. I have faced problem that DTMF event has not > come if DTMF given while playing voice files. I have received 'COMMAND' > event. I have the following questions. > >Why the 'COMMAND' event came event filter is on? >How to avoid this event in ESL? ___ 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] DTMF javasript
Your not telling anything to call your callback.
On Nov 24, 2009, at 1:03 AM, Baskar wrote:
> Hi,
>
> I want to check value given to the javascript with conditions whether it is
> voicefile, extension or mobile Number when i press the dtmf value.
>
> Steps i need to check in javascript:
>
> When i Press the DTMF value 1 it should check the 3 condition
>
> If the Value for argv[2]=vfsurya means it is a voice file so it should play
> the Voice file
> If the Value for argv[2]=1001 means it is a extension. The call should Bridge
> the extension
> If the Value for argv[2]=9841799874 means it is a Mobile number. The call
> should Bridge the Mobile number
>
> var exit = false;
> var dtmf_digits = "";
> var repeat = 0;
> var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number
>
>
> function onInput( session, type, data, arg )
> {
> if ( type == "dtmf" )
> {
> console_log( "info", "Got digit " + data.digit + "\n" );
> if ( data.digit == "1" )
> {
> if(argv[2].startswith("vf"))
> {
> var voice2=voice.substring(2)+""
>
> session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/"+voice2+".wav",
> onInput );
> }
> else if(argv[2].length==4)
> {
> console_log( "info", "Got voicefile " + argv[2] + "\n"
> );
> session.execute("bridge",
> "sofia/internal/"+argv[2]+"%192.168.1.2", onInput );
> }
> else
> {
> session.execute("bridge",
> "sofia/default/sip:"+argv[2]+"@192.168.1.135:5066", onInput );
> }
> }
> }
> }
>
> But if 1 is pressed there is no event trigger but it get the dtmf value as 1
> in freeswitch console.
>
> can any one specify what is the error or correct me where i am wrong.
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