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Blind transfer fails in FreeSWITCH, even if proxying and media bypass are enabled
by John Platts 01 Dec '09

01 Dec '09

I have tried to do a blind transfer from a phone that is registered with FreeSWITCH, and it will fail, even when proxying and media bypass are enabled. Details about this issue can be found here: http://jira.freeswitch.org/browse/MODENDP-272 _________________________________________________________________ Get gifts for them and cashback for you. Try Bing now. http://www.bing.com/shopping/search?q=xbox+games&scope=cashback&form=MSHYCB…

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Re: [Freeswitch-users] User logon/logout from analog phones
by Ryanny Lin 01 Dec '09

01 Dec '09

Dear Michael: Yes, I want to distribute a real phone number to each analog phone (direct inward dialing). One FXS one analog phone. I guess the user maybe want to add a number mapping this FXS port. Thank you, Michael. mod_xml_curl is really a powerful module. :D ---------- Forward ---------- From: Michael Collins <msc(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Date: Tue, 1 Dec 2009 11:04:43 -0800 I'm not entirely sure that I understand your question, so I am going to ask a few questions to clarify. Are you looking to have analog telephones receive incoming calls, like in a call center? Is that why the user of the analog phone would need to log in and log out? I would recommend checkout out mod_xml_curl for the dynamic dialplan. -MC On Tue, Dec 1, 2009 at 10:40 AM, Ryanny Lin <ryannyl(a)gmail.com> wrote: > Dear All: > > I try to register from a feature code of an analog phone like Elastix. > It is useful for DID. > There is an idea that I use dynamic dialplan to implement it and it's not > really register to FS. > And I need to run script to insert or delete dialplan to the database when > dialed.(Input logon's ExtNumber and Password) > Is that right? or any recommendation? > > Thanks in advance. > -- > Sincerely regards, > Ryanny

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Problem with compiling revision 15739
by John Platts 01 Dec '09

01 Dec '09

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…

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Re: [Freeswitch-users] Passing incoming remote-party-id from called to caller
by Anthony Minessale 01 Dec '09

01 Dec '09

Just set the variables effective_callee_id_name and effective_callee_id_number in your dp before you answer the call On Dec 1, 2009 12:08 AM, "Yehavi Bourvine" <yehavi.bourvine(a)gmail.com> wrote: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing & OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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User logon/logout from analog phones
by Ryanny Lin 01 Dec '09

01 Dec '09

Dear All: I try to register from a feature code of an analog phone like Elastix. It is useful for DID. There is an idea that I use dynamic dialplan to implement it and it's not really register to FS. And I need to run script to insert or delete dialplan to the database when dialed.(Input logon's ExtNumber and Password) Is that right? or any recommendation? Thanks in advance. -- Sincerely regards, Ryanny

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Passing incoming remote-party-id from called to caller
by Yehavi Bourvine 01 Dec '09

01 Dec '09

Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing & OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi:

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Holiday routing examples
by Andrew Thompson 30 Nov '09

30 Nov '09

Tony committed my patch for doing 'week of month' conditions in the XML dialplan along with some holiday routing examples to the default dialplan. Now you can detect all the major US holidays in pure dialplan XML without having to do any nasty math or anything (I did it all for you). I've also added a page to the wiki describing how to use it for other dates (like non-US holidays): http://wiki.freeswitch.org/wiki/Holiday_Routing Hope this helps some people. Andrew

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Questions on ISDN support for Freeswitch
by Albano Daniele Salvatore - Lavoro 30 Nov '09

30 Nov '09

Hi to all, shortly i'll make a pbx for a customer that uses a couple of isdn bri lines and, looking for hardware, i've seen that not too much expensive isdn cards that works well are the ones that uses hfc-4/8s controller (specifically i'll use a OpenVox B200P that has 2 ISDN ports and use an HFC-4S controller). I've seen that FreeSwitch doesn't support mISDN but uses openzap (trough ozmod_isdn.so). I got some serious troubles using OpenZAP on analogical lines (bad dtmf recognition on fxs ports (like press 4 and get 44), annoying noise, busy/hangup tone unrecognized [tones, in italy, differs by cadency and not frequency], and more). Using tone_detect i bypassed the tone recognition problem and the noise was however acceptable: the blocking problem was the bad dtmf recognition. In the end (hope god forgive me) i put asterisk as (*1*)(*2*)sip proxy between zaptel and freeswich: i need to fix in config hangup cause detection but it seems to works fine. So my questions are: - Do freeswitch supports mISDN? - If it doesn't support mISDN, tone and dtmf recognition will be done by the isdn card, the kernel module or will be done by openzap? - There are alternative ways to use (*3*) mISDN with freeswitch, apart put asterisk as proxy? Thank for your support! Best Regards, Daniele --- (*1*) At beginning i tried using IAX but freeswitch segfaults when it try to answer the call and when i reload the module (trought reload mod_iax) shutdown routing didn't get called(the used iax library, smartly, start using another port without saying anything), however i need to do more testing: hope to open some tickets on jira in short. (*2*) I've seen that mod_iax config file support a "context" variable, but it isn't used so i wrote a small fix to use it if context isn't specified in the iax request (*3*) I need to use mISDN, rather other things, because i should use octasis soft echo cancellation and them supports only mISDN and Zaptel modules

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Sangoma RTP TAP
by Helmut Kuper 30 Nov '09

30 Nov '09

Hello, has anyone of you tried the RTP TAP function of sangoma`s wanpipe driver? It is described here: http://wiki.sangoma.com/wanpipe-voice-rtp-tap On my side wanrouter log says that RTP TAB is configured and enabled, but I can't detect any udp packets received by the remote server (which is described by RTP_TAP_IP, RTP_TAP_MAC and RTP_TAP_PORT). I've latest driver and double checked the wanpipe.conf config. I tried to send some udp packets from wanpiping server to remote server, where the packets were shown up via tcpdump. So there is no FW problem involved. Each try to do some kind of printf debugging in wanpipe-driver doesn't succeed. Any ideas?

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Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700
by John Platts 30 Nov '09

30 Nov '09

I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. I have looked over the SIP traces of the failing scenarios. I have caught the following problems in the failing scenarios: - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not handling this correctly when calls are transferred. - The wrong codec is being negotiated in re-INVITE to the transferred number in the scenario when media proxying is enabled but media bypass is disabled. - In the scenario where media bypass is used, the re-INVITE actually appears to contain the correct details, and we are receiving the correct responses from our IP to IP gateway, but FreeSWITCH is not handling the media streams properly. Example of SDP descriptor coming from IP phone (with SDP descriptor modified by SIP ALG): v=0 o=- 123576 123576 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 16406 RTP/AVP 18 0 8 2 9 104 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:104 L16/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv Notice that the c= line has the correct public IP address and the m= line containing the correct port. Example of incorrect SDP descriptor being sent by FreeSWITCH in re-INVITES: v=0 o=- 121397 121398 IN IP4 192.168.1.4 s=- c=IN IP4 192.168.1.4 t=0 0 m=audio 16404 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=ptime:20 Note that the c= line contains the wrong IP address, but the m= line contains the correct RTP port. Example of wrong re-INVITE message being sent to the number that the call was being transferred to: INVITE sip:19729831777@168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj Max-Forwards: 69 From: "John Platts" <sip:19725357722@168.75.202.212>;tag=c61Drt38KF72m To: <sip:19729831777@ipipgw.ipdimensions.com>;tag=2B1339E0-1A2C Call-ID: 1c095553-5741-122d-33a8-00185167f91d CSeq: 123615824 INVITE Contact: <sip:mod_sofia@168.75.202.212:5062> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 183 X-FS-Support: update_display Remote-Party-ID: "John Platts" <sip:19725357722@168.75.202.212>;party=calling;screen=yes;privacy=off v=0 o=- 123576 123577 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 30186 RTP/AVP 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 Here is the correct re-INVITE for the call that was unsuccessfully transferred (after the transfer was completed): INVITE sip:19729555871@168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKgaDHFKZrc06vD Max-Forwards: 16 From: <sip:19725357722@168.75.202.212>;tag=BX8mpZj5p6ggS To: <sip:19729555871@168.75.202.246>;tag=2B12D184-BEC Call-ID: 15A1F95-DBD611DE-8C95D9DF-3419A306(a)168.75.202.246 CSeq: 123615820 INVITE Contact: <sip:19725357722@168.75.202.212:5062;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 222 X-FS-Support: update_display v=0 o=- 121397 121399 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 26106 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. 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