December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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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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compilation error of skypiax_protocol.c
by Jingwei Yang 25 Feb '10

25 Feb '10

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 â€˜X11_errors_handlerâ€™: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function â€˜skypiax_send_messageâ€™: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function â€˜skypiax_do_skypeapi_thread_funcâ€™: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code make[5]: *** [skypiax_protocol.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_skypiax-install] Error 1 make[2]: *** [install-recursive] Error 1 I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well?

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Presence Change Distribution
by Jerry Richards 01 Feb '10

01 Feb '10

Is there a setting to control how fast FS distributes presence changes to subscribers? Currently, it appears to take several minutes before I see presence changes. I would like to see them almost instantaneously, if possible. Thanks and Best Regards, Jerry

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Bypass Media True Disables MOH
by Jerry Richards 20 Jan '10

20 Jan '10

When I uncomment the following tag, internally held calls no longer hear MOH. <param name="inbound-bypass-media" value="true"/> Is there a way to have the above uncommented and still provide MOH to held calls? Best Regards, Jerry

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zrtp endpoints have different sas through fs 1.0.4
by Harondel J. Sibble 19 Jan '10

19 Jan '10

I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client, 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian s60) and an O2 Xda Flame (windows mobile 5). All 3 endpoints are registered with FS using the default extensions of 1000- 1003 With global_setvar zrtp_secure_media=true the zrtp negotiation between end points happens but the SAS never matches,below is console output for a call between 2 of the endpoints 2009-08-23 14:10:17.643073 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1003(a)10.12.13.45! 2009-08-23 14:10:21.257568 [NOTICE] sofia.c:3794 Channel [sofia/internal/sip:1000@10.12.13.166:5062] has been answered 2009-08-23 14:10:21.275521 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1003(a)10.12.13.45] has been answered 2009-08-23 14:10:22.232053 [WARNING] mod_sofia.c:810 We were told to use ptime 20 but what they meant to say was 80 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-08-23 14:11:34.496118 [NOTICE] sofia.c:322 Hangup sofia/internal/sip:1000@10.12.13.166:5062 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-23 14:11:34.512100 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1003(a)10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1086 Session 16 (sofia/internal/sip:1000@10.12.13.166:5062) Ended 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1000@10.12.13.166:5062 [CS_DESTROY] 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1086 Session 15 (sofia/internal/1003(a)10.12.13.45) Ended 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1003(a)10.12.13.45 [CS_DESTROY] Of note, with the endpoints registered through the Ekiga sip server, the sas DOES match on both ends. With global_setvar zrtp_secure_media=false, the endpoints can't detect a zrtp peer. Reading the list archives hasn't enlightened me. I see this comment from 2008 http://www.nabble.com/Freeswitch-and-Twinkle-and-ZRTP- td18518140.html#a18518343 On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote: > it should in bypass_media or proxy_media modes. in the other modes we > are in the media path and would not know how to handle the encrypted > packets. > > Mike Is this still relevant? Or is there some other setting not covered here http://wiki.freeswitch.org/wiki/ZRTP to make this work properly? I ask firstly about this in the context of a peer 2 peer zrtp communication between the endpoints, then secondly in the case of FS acting as a trusted middleman as in section 2 here http://www.zfoneproject.com/docs/asterisk/man/html/u_guide.html#passthrough Lastly how does one implement the security enrollment as noted above with FS -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help(a)pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice)

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GSM modem for Skype
by 大泥人 13 Jan '10

13 Jan '10

Hello All, I had a GSM VoIP Gateway which can be connected to service providers who use SIP protocal. I remember somebody in Freeswitch community mentioned there is a software which can connect GSM network with Skype network. I'm not sure if this software can work with my GSM VOIP gateway or not. If somebody can send me such software that would be much appriciated and even license fee required would be accepted. I hope I can have a testing on this or I can send this gateway to those guys who have this kind of software for testing. Thanks and Merry Christmas. Daniel Zeng _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

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Hardware echo cancellation.
by Tim Uckun 13 Jan '10

13 Jan '10

I am about to build a new machine as a VOIP server. I am going to get either a quad core intel or a six core AMD processor with at least eight gigabytes of RAM in it. Given that much horsepower I am wondering if there is any need to purchase hardware with echo cancellation (I am thinking about redfone devices).. I can save some money by not getting the echo cancellation. So is it worth saving that money? Is it always better to have hardware echo cancellation? Is a quad core capable of dealing with echo cancellation needs of an IVR which is going to take lots of simultaneous calls? Thanks.

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[ANN] Spice Telephony 0.9.5 released (FS based callcenter)
by Andrew Thompson 12 Jan '10

12 Jan '10

So, it's been a while since I mentioned this project, but its finally nearing the point where it's going to be able to go into production (and replace my old asterisk-based platform) so I decided to dredge it up again. Briefly, spice telephony is a call/contact center platform that leverages FS for VoIP, IVRs, call recording, etc. It also supports handing email/voicemail contacts (chat is planned, too). Here's some features: * Skill based routing * Priority, unified queues * Web based administration, agent interaction (using the dojo toolkit) * Supervisory drag and drop interface for managing agents/call flow * Queue 'recipes' - ability to play announcements, send to voicemail, modify skills or priority based on certain conditions (queue time, media type, hour of day, # of available agents, etc). * Integration API for importing agents/clients out of a CRM/AD/whatever * Detailed CDRs recording every step of a call (IVR, Queue, Ring, Transfer, Wrapup, etc). The project is implemented in Erlang (erlang.org) and thus allows spice-telephony to be deployed as a distributed system (multiple nodes aggregated into a single system). Calls can come into any node and, skills permitting, can be offered to any agent on the local node or any of the remote nodes. Nodes can also operate independantly if isolated by a netsplit or simply deployed standalone. CDRs and config files are stored in erlang's distributed database, mnesia, and CDRs can be output in parallel to any node configured to do so (so you can have all your call data in multiple places without having to do SQL replication). Erlang's fault tolerant nature also allows the platform to be very robust, entire subsystems can fail at runtime and be automatically restarted by supervisor process, and the entire erlang node can be automatically restarted if the node crashes. There's a lot more than mentioned above, so I'd encourage anyone interested to grab the latest release from: http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz and look at the install guide: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony_Install_Guide You'll need an erlang version >= R13B01 and ruby's 'rake' installed, you shouldn't need much of anything else. It *does* work on windows but I don't recommend it (I can try to help you get it working though). There's also some more information available here: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony The documentation is a little sparse, but I'll do my best to answer any questions. Any feedback is appreciated. Andrew

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