March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Diversion header
by Jonas Gauffin 23 Jun '10

23 Jun '10

Hello, I got a incoming call from my provider which goes to a javascript. The javascript transfers the call to an external destination (goes back through my provider). A diversion headers is added by FreeSWITCH on the outbound call. How can I remove it (or if not possible, how do I change it)? javascript: session.setCallerData("caller_id_number", aNumber); session.setVariable("gate_caller_id_number", aNumber); session.setVariable("gate_site_id", SITE_ID); session.execute("transfer", this.dtmfBuffer + " XML internal"); send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200: ------------------------------------------------------------------------ INVITE sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com>SIP/2.0 Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS Max-Forwards: 67 From: "0236661XXX" <sip:0243795362@212.247.XX.XX>;tag=BHvytKtv47r2a To: <sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com> > Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5 CSeq: 127654862 INVITE Contact: <sip:gw+dalaconnecttele2@212.247.XX.XX :5060;transport=udp;gw=dalaconnecttele2> User-Agent: Gateon Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 273 Diversion: 0 <sip:0@212.151.Y.Y ;user=phone>;reason=unconditional;counter=1;privacy=off X-FS-Support: update_display Remote-Party-ID: "0236661XXX" <sip:0243795362@212.247.XX.XX >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX s=FreeSWITCH c=IN IP4 212.247.XX.XX t=0 0 m=audio 32712 RTP/AVP 8 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20

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Lua script hangup detect
by Chia-Yen Wu 10 Jun '10

10 Jun '10

Hello, i tried to use Lua script to replace xml macro in dialplan, but I found out that Lua wont terminate if client hangup, ,so the session is still on but client is already hangup, is there a way to avoid this ?

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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xml_cdr http post differentiation between legs
by Adam Ford 03 Jun '10

03 Jun '10

Is there any way to tell whether it is the a-leg or b-leg when receiving the POST data from mod_xml_cdr? I wrote a parser in Python which parses the a_*.cdr.xml log files, and only grabs the info in the associated b-leg log files if it needs additional information(for example - in the case of a call that ends with mod_fifo). I want to adapt this to using the POST features of mod_xml_cdr so that it can be real time. Any suggestions are greatly appreciated, -Adam

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Simple IAX2 setup - help with converting from asterisk to freeswitch
by Matthew Law 26 May '10

26 May '10

Hi, I have compiled freeswitch trunk from svn on debian. I am trying to convert the following simple working asterisk config to freeswitch and I would be really grateful if someone could point me in the right direction: pbx:/etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=my.ip.address jitterbuffer=yes disallow=all allow=ulaw allow=alaw context=deadend [voiptalk] type=peer username=XXXXXX secret=XXXXXX host=iax.voiptalk.org [0843XXXXXX] type=friend username=08433XXXXXX context=incoming requirecalltoken=auto [1000] type=friend host=dynamic mailbox=1000 secret=XXXXXX context=phones requirecalltoken=auto pbx:/etc/asterisk# cat extensions.conf [general] autofallthrough=yes [outgoing] exten => _0[1-9].,1,Dial(IAX2/XXXXXX@voiptalk/44${EXTEN:1}) exten => _00.,1,Dial(IAX2/XXXXXX@voiptalk/${EXTEN:2}) [internal] exten => 901,1,VoiceMailMain() exten => 901,2,Hangup() exten => 902,1,MeetMe(1234,cdM) exten => 902,2,Hangup() [incoming] exten => XXXXXXXX,1,Dial(IAX2/1000,30) exten => XXXXXXXX,2,VoiceMail(1000@internal) [phones] include => internal include => outgoing pbx:/etc/asterisk# cat voicemail.conf [internal] 1000 => XXXX,Matt,me(a)myomain.com format=wav49 maxsilence=0 Sorry to be a bonehead, but I'm struggling with the wiki docs (especially with regard to IAX2) and also concerned about security. I installed the samples when I compiled freeswitch but wondering if that is a security risk as the PBX box is on the public internet? Many thanks, Matt

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freeswitch, openzap, dahdi issues...
by Tom Christensen 27 Apr '10

27 Apr '10

In my quest to replace an unstable and deadlock prone asterisk solution I was able to get calls working over my PRI last night using freeswitch, native openzap, and dahdi 2.2.1. However, in order to make it work, I had to disable all echo cancellation in dahdi. If any of the 4 echo cancellation algo's were specified in /etc/dahdi/system.conf then there was no audio at all on the calls. Signaling worked, and calls could be made/received, but no audio in either direction. As soon as I disabled EC, audio worked (although, there was a good amount of echo). Also, I had intermittent (like 1 out of 2 or 1 out of 3) calls with very poor voice quality (robotic sounding, distortions, dropping in and out). This experience leads me to believe that possibly openzap isn't ready for prime time? Would I have been better served to include libpri in the technology stack? At this stage my "solution" looks like leaving asterisk where it is, and using it as a media gateway, passing calls off to freeswitch via SIP or IAX and using freeswitch as the app/registration/feature server. I don't know if that will necessarily be better, but I've had good luck with asterisk and call quality tuning (the PRI here has always had pretty bad echo, but I've been able to get asterisk EC working quite well on it), and it seems like asterisk has most of its issues with stability in transferring/parking/hold/voicemail/music on hold... I'm running a current FS SVN from 3 days ago, dahdi 2.2.1, and openzap using a digium te110p. The PRI is an NI2 t1 if that matters... Any ideas or recommendations of things to try would be greatly appreciated. I'd really like to avoid the dual server situation if at all possible. Thanks, -Tom

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Re: [Freeswitch-users] Grandstream gateways
by Samuel Mukoti 23 Apr '10

23 Apr '10

Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory(a)voipsupply.com> wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews(a)sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request(a)lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris <mike(a)jerris.com> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users(a)lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B(a)jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> 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. >> >>

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Nibblebill for NoSQL
by Dome Charoenyost 22 Apr '10

22 Apr '10

Dear All, I use nibllebill + postgresql for longtime. it' work wel when concurrent about 500 calls.. Now i have 1200 calls . postgresql got many problem and slow. so i have 2 choice. 1. upgrade hardware , add more memory may be 32 GB and change HDD to SSD (or IOdrive ). 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant) I choose 2. and use tokyo tyrant. because it's persistant and support memcache API. Today i create new module base on nibblebill but use memcacahe API connect to tokyo tyrant. it's look fast. and work without any problem. If someone want to use my module I'l acttach to JIRA and create WIKI page. Please let's me know if you want. BG Dome C.

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Error when recording
by Maciej Aniserowicz 14 Apr '10

14 Apr '10

Hello, I have a problem with recording sessions. The scenario is: 1) dial 1st call and park 2) dial 2nd call and park 3) bridge them 4) start recording 2nd call 5) receive error message "Can not record session. Media not enabled on channel" The log is here: http://pastebin.freeswitch.org/12566. Both clients are FreeSWITCH extensions that play some wav file using playback application defined in dialplan. I get this error in ~25% of recorded sessions. What am I doing wrong? Thanks, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4817081.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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SIP Overlap support?
by Dennis 06 Apr '10

06 Apr '10

hi there, i would like to ask, if fs has support for something like "SIP Overlap"? instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side? thanks and kind regards dennis

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