Re: [Freeswitch-users] Multitenant dialplans
Thanks Brian. I did have both force-register-domain and force-register-db-domain commented in both the internal.xml and internal-ipv6.xml. The phones appear to register to the company1 domain, as shown in sofia status profile company1; however I have noticed that when I try to make a call to another a phone in the same domain, the system is trying to call sofia/internal/1...@company1 -- this is when we get the message, user not registered. If I can the phones to just register to the IP address of the machine, they call fine and is shows sofia/internal/sip:1...@phonesgatewayipaddress. Is this a dialplan problem? In both cases I am just using the sample dialplan. On 12/22/2009 8:13 AM, Brian West wrote: The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote: I have Freeswitch setup and working as a single tenant system mostly using the default configuration. Trying to convert to a multitenant environment, I have used both the Multi-tenant and Multiple Companies wiki's. I get the phone to register, can call out using the external profile to a ITSP, can call music on hold; however I can not call other users in the company. It appears that when logged in with single company and default context it sucessfully calls other internal phones with bridge to sofia/internal/sip:exters...@public-ip:translated-port; however when I log into Company1 with the phones, it tries sofia/internal/dialed-extens...@company1 ... I also get User not Registered. The dialplans are the same either way. Any ideas? Thanks John ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Multitenant dialplans
One point of clarification, currently all the phones are behind NAT, so it appears that when the phones are in a Non-multitenant scenario, they use SIP:dialed_num...@ip-address-of-their-gateway. On 12/22/2009 9:16 AM, John wrote: Thanks Brian. I did have both force-register-domain and force-register-db-domain commented in both the internal.xml and internal-ipv6.xml. The phones appear to register to the company1 domain, as shown in sofia status profile company1; however I have noticed that when I try to make a call to another a phone in the same domain, the system is trying to call sofia/internal/1...@company1 -- this is when we get the message, user not registered. If I can the phones to just register to the IP address of the machine, they call fine and is shows sofia/internal/sip:1...@phonesgatewayipaddress. Is this a dialplan problem? In both cases I am just using the sample dialplan. On 12/22/2009 8:13 AM, Brian West wrote: The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote: I have Freeswitch setup and working as a single tenant system mostly using the default configuration. Trying to convert to a multitenant environment, I have used both the Multi-tenant and Multiple Companies wiki's. I get the phone to register, can call out using the external profile to a ITSP, can call music on hold; however I can not call other users in the company. It appears that when logged in with single company and default context it sucessfully calls other internal phones with bridge to sofia/internal/sip:exters...@public-ip:translated-port; however when I log into Company1 with the phones, it tries sofia/internal/dialed-extens...@company1 ... I also get User not Registered. The dialplans are the same either way. Any ideas? Thanks John ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Click-to-call and click-to-dial
How can I perform click-to-call or click-to-dial in FreeSWITCH? Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Click-to-call and click-to-dial
You've made my day. From: jpitc...@nuvio.com To: freeswitch-users@lists.freeswitch.org Date: Wed, 16 Dec 2009 08:11:05 -0800 Subject: Re: [Freeswitch-users] Click-to-call and click-to-dial John, To do a click to call in FS you need to have some app that connects to the ESL or Event Socket Layer and runs one of the calls diagramed here ... http://wiki.freeswitch.org/wiki/Mod_commands#originate For use with the ESL just prepend api in front of the originate so your call looks something like: $command = 'api originate u...@domain bridge(u...@domain)'; As for programs able to do that from a Microsoft Product, That I am not sure of. Jonathan Pitcher On 12/16/09 9:59 AM, John Platts wrote: How can I perform click-to-call or click-to-dial in FreeSWITCH? Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Update to MODENDP-272
I have uploaded the dialplan and JavaScript files used to process calls to MODENDP-272. I have even done a make current to revision 15755, and the blind transfer is still failing. _ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7secslideid=1media=aero-shake-7secondlistid=1stop=1ocid=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] can't register Inphonex
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 from http://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] Problem with compiling revision 15739
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=7secslideid=1media=aero-shake-7secondlistid=1stop=1ocid=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] Blind transfer fails in FreeSWITCH, even if proxying and media bypass are enabled
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+gamesscope=cashbackform=MSHYCBpubl=WLHMTAGcrea=TEXT_MSHYCB_Shopping_Giftsforthem_cashback_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
Re: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700
To clarify the problem, the invite message is incorrect because comfort noise is being negotiated in the re-invite instead of G.711 or G.729: INVITE sip:19729831...@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:19725357...@168.75.202.212;tag=c61Drt38KF72m To: sip:19729831...@ipipgw.ipdimensions.com;tag=2B1339E0-1A2C Call-ID: 1c095553-5741-122d-33a8-00185167f91d CSeq: 123615824 INVITE Contact: sip:mod_so...@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:19725357...@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 How do I get it to negotiate G.711, G.729, or other codec instead of comfort noise? Our IP phones, our FXS gateways, and our IP to IP gateways expect G.711, G.729, iLBC (if supported by the endpoints), G.722 (if supported by the endpoints), or G.726 (if supported by the endpoints) be negotiated. From: john_pla...@hotmail.com To: freeswitch-users@lists.freeswitch.org Date: Sat, 28 Nov 2009 23:34:24 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 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
Re: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript
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=restaurantsform=MFESRPpubl=WLHMTAGcrea=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] Patch to allow gateways to be defined without the password parameter set
I have modified sofia.c in mod_sofia so that I can define gateways without having to specify the password parameter. This is because I am using a SIP gateway that does not require SIP registration. The modified version still requires the password to be set on any gateway for which register is set to true. Attached is the diff file for these changes. _ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurantsform=MFESRPpubl=WLHMTAGcrea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 sofia_password_patch.diff Description: Binary data ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Call forwarding problem
I was having trouble doing call forwarding from my SIP phone that is connected to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved Temporarily responses, but my SIP gateway does not support 302 Moved Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages? Here is the SIP debug from our gateway: Received: INVITE sip:+19725357...@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0 Record-Route: sip:65.243.172.245:5060;lr v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236 record-route: sip:63.77.76.236;lr f: sip:+19729555...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3ec95a-3ad03068-3ec95a t: sip:+19725357...@63.77.76.236:5060;user=phone i: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Max-Forwards: 18 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 m: sip:199.173.101.208:5060;transport=UDP c: application/SDP l: 210 P-Asserted-Identity: sip:9729555...@63.77.76.236;user=phone Privacy: none v=0 o=- 540754816 540754816 IN IP4 199.173.111.141 s=- c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 From: sip:+19729555...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3ec95a-3ad03068-3ec95a To: sip:+19725357...@63.77.76.236:5060;user=phone Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 24 15:08:00.367 CST: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357...@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212 Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 Supported: timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 1208058493-3631485406-2892683743-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259096880 Contact: sip:19729555...@168.75.202.246:5060 Expires: 180 Allow-Events: telephone-event Max-Forwards: 17 P-Asserted-Identity: sip:19729555...@168.75.202.246 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 2925 1780 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.141 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 Nov 24 15:08:00.367 CST: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212 Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 CSeq: 101 INVITE Timestamp: 1259096880 0.000342 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Content-Length: 0 Nov 24 15:08:00.419 CST: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212;tag=49aF8vtgHme2c Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=16;text=NORMAL_CLEARING Content-Length: 0 P-Asserted-Identity: 19725357722 sip:19725357...@168.75.202.212 Nov 24 15:08:00.427 CST: //-1//SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357...@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212;tag=49aF8vtgHme2c Date: Tue,
Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
I actually checked out revision 15654 today, and I was still getting problems with proxy media and bypass media in FreeSWITCH. From: m...@jerris.com Date: Tue, 24 Nov 2009 03:39:16 -0500 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. Mike On Nov 23, 2009, at 11:33 PM, John Platts wrote: I was using revision 15586. From: br...@freeswitch.org Date: Mon, 23 Nov 2009 18:25:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH What rev exactly? /b On Nov 23, 2009, at 6:19 PM, John Platts wrote: I actually checked out the latest version of FreeSWITCH in the SVN repository. I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml: _ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript
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? _ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurantsform=MFESRPpubl=WLHMTAGcrea=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
Re: [Freeswitch-users] Call forwarding problem
Is there any way to tell FreeSWITCH to do the following when 302 Moved Temporarily is sent to FreeSWITCH: - End the session between FreeSWITCH and the phone - Bridge the original session with the number that the call is forwarded to From: br...@freeswitch.org Date: Tue, 24 Nov 2009 15:32:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Call forwarding problem You'll have to hairpin the media thru your machine usually if they won't accept either of those. /b On Nov 24, 2009, at 3:05 PM, John Platts wrote: How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in
/usr/local/freeswitch/conf/dialplan/default.xml:
extension name=setup_media continue=true
condition field=${sip_nat_detected} expression=true
action application=set data=proxy_media=true /
action application=set data=bypass_media=false /
anti-action application=set data=proxy_media=false /
anti-action application=set data=bypass_media=true /
/condition
/extension
I have the following configured in /usr/local/freeswitch/conf/vars.xml:
X-PRE-PROCESS cmd=set
data=global_codec_prefs=G729,i...@20i,G722,PCMU,PCMA/
X-PRE-PROCESS cmd=set
data=outbound_codec_prefs=G729,i...@20i,G722,PCMU,PCMA/
Here is the SIP trace for the failing call:
Nov 23 17:55:05.245 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
INVITE
sip:+19725357...@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246
SIP/2.0
v: SIP/2.0/UDP
65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24
Record-Route: sip:65.211.120.237:5060;lr
v: SIP/2.0/UDP
63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236
record-route: sip:63.77.76.236;lr
f:
sip:+19729831...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a
t: sip:+19725357...@63.77.76.236:5060;user=phone
i: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585
CSeq: 1 INVITE
Max-Forwards: 16
k: 100rel, replaces
allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208
m: sip:199.173.101.208:5060;transport=UDP
c: application/SDP
l: 210
P-Asserted-Identity: sip:9729831...@63.77.76.236;user=phone
Privacy: none
v=0
o=- 641026559 641026559 IN IP4 199.173.111.147
s=-
c=IN IP4 199.173.111.147
t=0 0
m=audio 33344 RTP/AVP 18 0 8 101
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 17:55:05.257 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP
63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208
From:
sip:+19729831...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a
To: sip:+19725357...@63.77.76.236:5060;user=phone
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Nov 23 17:55:05.257 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:19725357...@168.75.202.212:5062 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: sip:19729831...@168.75.202.246;tag=105BD148-201C
To: sip:19725357...@168.75.202.212
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
Supported: timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1961129755-3619819998-2727664095-874095366
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1259020505
Contact: sip:19729831...@168.75.202.246:5060
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 15
P-Asserted-Identity: sip:19729831...@168.75.202.246
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314
v=0
o=CiscoSystemsSIP-GW-UserAgent 5041 5861 IN IP4 168.75.202.246
s=SIP Call
c=IN IP4 199.173.111.147
t=0 0
m=audio 33344 RTP/AVP 18 0 8 101
c=IN IP4 199.173.111.147
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
Nov 23 17:55:05.261 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: sip:19729831...@168.75.202.246;tag=105BD148-201C
To: sip:19725357...@168.75.202.212
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Timestamp: 1259020505 0.000345
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Content-Length: 0
Nov 23 17:55:05.309 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: sip:19729831...@168.75.202.246;tag=105BD148-201C
To: sip:19725357...@168.75.202.212;tag=DFKSy9Q5DK1Na
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Contact: sip:19725357...@168.75.202.212:5062;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
I was using revision 15586. From: br...@freeswitch.org Date: Mon, 23 Nov 2009 18:25:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH What rev exactly? /b On Nov 23, 2009, at 6:19 PM, John Platts wrote: I actually checked out the latest version of FreeSWITCH in the SVN repository. I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Need help configuring our FreeSWITCH instance
I have installed FreeSWITCH on our server, and need some help configuring our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance are in the format: 1NPANXX (where NPA is the area code, and NXX are the last 7 digits of the phone number). I need the following configuration: Calls coming from our IP to IP gateway into our FreeSWITCH instance needs to be routed to the endpoint that is registered with FreeSWITCHCalls coming from any of the registered SIP endpoints need to be sent to the appropriate destination. The appropriate destination for any number that is not registered with FreeSWITCH is our IP to IP gateway.Our IP to IP gateway does not require any SIP registration or authentication.G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to be enabledSIP registrar enabled for registering endpoints other than our IP-IP gatewaySIP traffic needs to be accepted to and from both the IP-IP gateway and from the registered SIP endpoints. How do I get the above configured in FreeSWITCH? _ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Precompiled Windows Binaries
Brian West wrote: I looked out my window... but I didn't see pigs flying... did I miss something! :P /b On Nov 4, 2009, at 11:22 AM, Giovanni Maruzzelli wrote: ...and will get more people using the x64 version of Windows! ;) -gm When their own commercials say that there old software is prone to crashing and hangs, why should I trust there new software? -- JohnM ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Accessing a global variable from lua
How do you get a system variable from within a lua startup script? Specifically I want domain_name from vars.xml ... normally I'd use session:getVariable, however there is no session in this case. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] Accessing a global variable from lua
You can execute global_getvar api call. Thanks ... I've updated the wiki. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] Originating a call from lua with rudimentary error checking
What's the recommended way to check if the session constructor was
successful (i.e. the number could be dialed)?
check that s is nil.
Doesn't work ... s is never nil. Type shows it as userdata
even if Session failed. Specifically my test was:
local s = freeswitch.Session (
{ignore_early_media=true,origination_caller_id_name= ..
caller .. }loopback/ .. destination .. /default/XML)
stream:write (type(s))
if s == nil then
stream:write (-ERR call failed\n)
return
end
and I dialed an unreachable number.
and that s.ready() is true
Checking s.ready() results in:
[ERR] freeswitch_lua.cpp:102 session is not initalized
if Session failed.
What I'm looking for is a way to try to originate a call which doesn't
throw ERR messages if the attempt fails.
Explicitly calling session.originate seems to allow you to check if
the call was successful ... is there a particular reason it's discouraged?
I'm happy to avoid it if a better approach is available, however I'm
having trouble finding one.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495 | |
-
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Re: [Freeswitch-users] Live Upgrade Techniques
On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote:
On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:
On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
Well, if you're running multiple machines, waiting for it to drainstop
isn't that big of a deal unless you're in some sort of hurry, right?
Give it an hour or so to drainstop, then kill 'em.
Yes that's exactly what I'm trying to do. The problem is some people will
only try one IP address.
Clients that don't properly implement SRV/NAPTR and fail over need to be
smacked. :) (not customers but software that fails to do that)
Yes I'm sure much of their software can do this but it has been set up for
static numeric IPs. And getting the IP changed is a week-long process for
some customers!
Would it not be simpler to try to do something with re-invites or REFER,
assuming your endpoints support it?
That was actually plan A. I already added a property in sip_profile called
failover_redirect, which specifies another server to try if FS can't
allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.),
by sending back a SIP 302 Moved Temporarily response, instead of 503 Max
Calls In Progress.
You can't send a 302 to a call thats already established.
Yes and I don't want to touch established calls - those calls can stay
there until they drop. This is sent to new requests when
switch_core_session_request fails in mod_sofia.
Turns out not all my endpoints support it :(
AKA broken endpoints. :)
Some are broken. Some just have this feature disabled. For 'security
reasons'. You know the drill.
{P^/
John
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[Freeswitch-users] No VoIP audio after upgrading to latest svn
Upgraded from Apr 3 svn to svn 13769. Calling from openzap to (music on hold) works. Calling from openzap to 9995 (5 sec echo test) works. Calling from openzap to vmail works. Calling from Grandstream to (music on hold) works. Calling from Grandstream to 9995 (5 sec echo test) doesn't work ... call goes through however silence is heard. Calling from Grandstream to vmail doesn't work ... call goes through however vmail disconnects apparently due to receiving silence. Calling from Grandstream to openzap doesn't work ... call goes through and the Grandstream can hear what is said on the openzap side, however openzap hears silence from the Grandstream. Calling from Grandstream to Grandstream doesn't work ... call goes through however both sides hear silence. Suggestions on how to proceed? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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 VoIP audio after upgrading to latest svn
BTW: in all cases show channels says PCMU 8000 is being used for the read and well as write codec. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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 VoIP audio after upgrading to latest svn
Yet more information ... a packet trace of a openzap to Grandstream call shows: SourceDestination Packet FreeSWITCHGrandstream SIP Request: INVITE ... Grandstream FreeSWITCH SIP Status: 100 Trying Grandstream FreeSWITCH SIP Status: 180 Ringing Grandstream FreeSWITCH SIP Status: 200, with session description FreeSWITCHGrandstream SIP Request: ACK ... FreeSWITCHGrandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCHGrandstream RTP Payload type=ITU-T G.711 PCMU ... ... FreeSWITCHGrandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCHGrandstream SIP Request: BYE ... Grandstream FreeSWITCH SIP Status: 200 OK The interesting thing is I don't see the Grandstream attempt to send audio. Is there something that FreeSWITCH needs to say to the Grandstream in order for the phone to send audio? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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 VoIP audio after upgrading to latest svn
I think possibly that the configs changed, specially the auto-nat stuff Yep ... a closer look at the packet trace showed FreeSWITCH settings the Contact as 10.10.10.1 instead of the actual IP address of the machine. If you have modified those two files then I recommend looking at the new default config versions of those two files and integrating your changes into the new ones. Yep ... that's my SOP. Looking at the internal.xml supplied with the new FS I see: param name=ext-rtp-ip value=auto-nat/ param name=ext-sip-ip value=auto-nat/ Once I commented out those entries everything worked fine. I'm kind of surprised that this default changed ... the older FS came with these commented out and worked fine in the simple configuration where the server and phones are on the same network segment. In any case my config has been adjusted, things are working, it's Friday, and I get to go home so life is good. :-) -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] Live Upgrade Techniques
OK thanks that is what I thought the general way of doing it would be. But
it seems a bit wasteful to have that SIP proxy there the whole time
especially when I am using FS in the role of an SBC.
The problem with the graceful restart of course is that you have to wait
for the calls count to get to zero, which may never happen. It's 3:30am
here in Sydney now and I just checked FS: 20 calls in progress still!
So what I plan to do is add a '--upgrade' cmd line arg to FS. This will
make the new instance contact the old one on a unix socket and receive a
dup of its SIP socket fd(s) via a SCM_RIGHTS sendmsg. It will use those
for sending and the unix socket for receiving. Meanwhile the old instance
will pass any packets with unknown Call-Ids over the unix socket to the
new instance, instead of handling them itself. When the old instance has
no calls left, it shuts down. The new instance detects the unix socket is
closed and switches to reading from the SIP socket (which would have
buffered any unread packets - so nothing is lost).
Sound good? I realise this will be 90% in libsofia but I've read teh code
and it seems very do-able. Anyone interested in my changes will of course
be most welcome to them.
The runner-up approach I considered was to make a kernel module that
extends iptables with a filter that can extract the Call-Id and look it up
in a table that is somehow populated from FS. Maybe this exists already?
Kind of a SIP proxy lite that can be enabled on the server machine when
needed. Anyway that lost out as it's more work and even less portable.
{P^/
John
On Thu, 11 Jun 2009 at 11:54 -0500, Anthony Minessale wrote:
or you can put a sip proxy in front of 2 boxes where you can control the
flow of traffic.
when you want to upgrade one, take all the traffic off of it by forcing all
calls to the other box, upgrade it then shift the traffic to the new one.
if that goes well, upgrade the other one too.
On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki
michal.bieli...@halo2.plwrote:
Am 11.06.2009 um 05:04 schrieb John Dalgliesh:
Hi,
I am slowly gaining confidence using FreeSWITCH in production, but there
is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current calls?
So far I have been upgrading in the dead of night, after pausing for 5
minutes then dropping the stragglers, but this is hardly ideal.
What I would like to do is to run an upgraded instance of FreeSWITCH on
the same machine, and have it handle all new call packets, whereas the old
instance continues to handle the existing call packets, until there are no
more old calls left.
I can think of about seven ways to accomplish this, but before I dive into
the code I thought I'd better ask what everyone else has been doing :)
(The only standard way I can think of doing this is to have a SIP proxy
sitting in front of FS the whole time, just to handle these upgrade
windows. It seems like a bit of a waste.)
So how are you handling your FS software upgrades?
{P^/
John
We use freeswitch on solaris and just upgrade it to a new zfs which gets
remounted to the old place and freeswitch gracefully restartet. On failure
we can allways do a rollback, which takes between 2 and 10 seconds, so the
dwntime is pretty acceptable.
Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. i...@halokwadrat.de
e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de
Hauptgeschäftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 153539
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] Live Upgrade Techniques
I assume he's talking about hardware failures here :P
But to answer the question: crashes are easy to deal with. With a crash
you have lost the calls that are in progress anyway; you don't have to
manage a gradual transition.
Currently, since FS is quite quick to start up, I am just relaunching it
immediately.
But when I have a second box up and running what I'll do is just add the
IP of the dead machine as another IP of the second box, and then it will
take all the old machine's traffic. That is the plan anyway. I've seen
some commercial boxes that use a similar trick.
(I've only seen one crash that wasn't my fault. Something to do with
terminating a bridge: when the first leg gets a hangup it hangs up the
other leg on its own thread... which can cause problems if the other leg
was doing something funky at the time. Leads to a heap corruption. Doesn't
happen with MALLOC_CHECK_ set so I'm just leaving it set for now :)
{P^/
On Thu, 11 Jun 2009 at 00:41 -0400, Mathieu Rene wrote:
By reporting it on Jira so it doesn't crash anymore :D
On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote:
How are you handling your FS box crashing?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of John Dalgliesh
Sent: Wednesday, June 10, 2009 9:04 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Live Upgrade Techniques
Hi,
I am slowly gaining confidence using FreeSWITCH in production, but
there
is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current
calls?
So far I have been upgrading in the dead of night, after pausing for 5
minutes then dropping the stragglers, but this is hardly ideal.
What I would like to do is to run an upgraded instance of FreeSWITCH
on
the same machine, and have it handle all new call packets, whereas
the old
instance continues to handle the existing call packets, until there
are no
more old calls left.
I can think of about seven ways to accomplish this, but before I
dive into
the code I thought I'd better ask what everyone else has been doing :)
(The only standard way I can think of doing this is to have a SIP
proxy
sitting in front of FS the whole time, just to handle these upgrade
windows. It seems like a bit of a waste.)
So how are you handling your FS software upgrades?
{P^/
John
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Re: [Freeswitch-users] Live Upgrade Techniques
On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo m...@giagnocavo.netwrote:
Exactly. You probably want to have something like this anyways, so that
when someone accidentally unplugs the system, or the disks/CPU/RAM crash,
you’re not stuck.
That is, until FreeSWITCH can record its internal state to some
inter-machine memory so we can have hot failover. ;)
I think that's going to be in 1.0.5. :)
I'm still too much of a noob to be certain that's a joke :) ... but FS
core already does record much of its internal state... to a DB, right? It
just has to not clear that out on startup and problem solved!
OTOH there will be a bit of trouble getting the internal state out of all
those modules and libraries... in particular sofia :D
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Re: [Freeswitch-users] Caller id when doing transfers
It appears from some limited testing that the original caller id is always
shown when the call is transfered. Is there some way to have the person
making the transfer show up as the caller id?
To answer my own question it appears that the information is available
in the sip_h_Referred-By variable. E.g.:
extension name=system25_park
condition field=destination_number expression=^\*5$ /
condition field=${sip_h_Referred-By} expression=^sip:([0-9]{4})@.*$
allows the station id making the transfer to be known when a call is
transfered to *5. The station id can then be used to park the call in
the proper fifo.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495 | |
-
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Re: [Freeswitch-users] Live Upgrade Techniques
Hi,
On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
Well, if you're running multiple machines, waiting for it to drainstop
isn't that big of a deal unless you're in some sort of hurry, right?
Give it an hour or so to drainstop, then kill 'em.
Yes that's exactly what I'm trying to do. The problem is some people will
only try one IP address.
Would it not be simpler to try to do something with re-invites or REFER,
assuming your endpoints support it?
That was actually plan A. I already added a property in sip_profile called
failover_redirect, which specifies another server to try if FS can't
allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.),
by sending back a SIP 302 Moved Temporarily response, instead of 503 Max
Calls In Progress.
Turns out not all my endpoints support it :(
I considered REFER too but there seems to be even less support for that.
If I can't get the socket-sharing upgrade working then I will fall back to
this - and peers which don't support the 302 response (or more likely,
don't authorise it) will just get no service during the upgrade.
-Michael
{P^/
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of John
Dalgliesh
Sent: Thursday, June 11, 2009 12:14 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Live Upgrade Techniques
I assume he's talking about hardware failures here :P
But to answer the question: crashes are easy to deal with. With a crash
you have lost the calls that are in progress anyway; you don't have to
manage a gradual transition.
Currently, since FS is quite quick to start up, I am just relaunching it
immediately.
But when I have a second box up and running what I'll do is just add the
IP of the dead machine as another IP of the second box, and then it will
take all the old machine's traffic. That is the plan anyway. I've seen
some commercial boxes that use a similar trick.
...
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[Freeswitch-users] Finding all active calls belonging to the same phone
To duplicate our old PBX park functionality I need for a user who's
on a call to be able to pick up a second line and dial a number to
park the other call which is on his phone. I have something working,
however am curious if there's a better way to accomplish this.
Specifically I'm curious if there's a recommended way to find all the
calls to / from the same phone / channel.
I ended up configuring *5 to call a javascript program which:
a) Gets the uuid from the session and uses it to search show channels
to find the channel name.
b) Normalizes the channel name and uses it to search show channels
to find an uuid associated with the channel which is different from
the one that invoked *5.
c) Uses the uuid from b to search show calls to find the peer
uuid.
d) Uses uuid_setvar to set hangup_after_bridge=false and uuid_transfer
to transfer the peer uuid to the proper fifo.
One of the problems I ran into is the channel name has slightly different
formats depending on whether it is an inbound or outbound channel. E.g.:
sofia/internal/1...@xxx.xxx.xxx.xxx
sofia/internal/1...@xxx.xxx.xxx.xxx:5060
sofia/internal/sip:1...@yyy.yyy.yyy.yyy:5060;transport=udp;...
where XXX is the freeswitch box and YYY is the phone. I created the
following function to normalize the channel name for comparison:
function normalize_channel_name (name, direction, ip_addr)
{
var re = /^sofia\//g;
var length = name.search (re);
var new_name = name;
if (length == -1)
return new_name;
if (direction == inbound) {
re = /@.*$/g;
new_name = name.replace (re, @ + ip_addr);
}
else if (direction == outbound) {
re = /\/sip:(@[^:]*):.*$/g;
new_name = name.replace (re, /$1);
}
return new_name;
}
Suggestions for a better approach? Keep in mind that my existing user
population expects (for better or worse) to use *5 to park the call on
their phone so I'm somewhat limited in what I can do.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495 | |
-
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[Freeswitch-users] Live Upgrade Techniques
Hi,
I am slowly gaining confidence using FreeSWITCH in production, but there
is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current calls?
So far I have been upgrading in the dead of night, after pausing for 5
minutes then dropping the stragglers, but this is hardly ideal.
What I would like to do is to run an upgraded instance of FreeSWITCH on
the same machine, and have it handle all new call packets, whereas the old
instance continues to handle the existing call packets, until there are no
more old calls left.
I can think of about seven ways to accomplish this, but before I dive into
the code I thought I'd better ask what everyone else has been doing :)
(The only standard way I can think of doing this is to have a SIP proxy
sitting in front of FS the whole time, just to handle these upgrade
windows. It seems like a bit of a waste.)
So how are you handling your FS software upgrades?
{P^/
John
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[Freeswitch-users] How to tell if 100 Trying received
Hi,
I am trying to use FS to make outgoing SIP calls. I have a number of
gateways that can make the call. However, if one of them is down or has
some other problem then I would like to detect that quickly.
I intended to use the provisional '100 Trying' message for this... if it
hasn't been received in a couple of seconds then go on and try the next
gateway.
But I can't find a flag/event/state which corresponds to receipt of this
message. Can anyone tell me where I should be looking? I put a debug print
in sofia_event_callback for every event but there doesn't seem to be one
fired for this condition.
Thanks in advance.
{P^/
John
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Re: [Freeswitch-users] How to tell if 100 Trying received
Hi Anthony,
Thanks for the reply!
While waiting for my question to appear on the list yesterday (6H delay at
yoda.ostag.org... is first post moderated?) I went deep into the SIP stack
and figured out the solution: You just have to give NTATAG_PASS_100(1) as
one of the tags for nua_create. Then you get a sofia event for it. I guess
the author has made it easier since your last discussion.
I have changed my mod_sofia to do this. I also added a channel flag which
is set if any response has been received from the remote end (be it 100,
18X, 2XX, etc.). The flag is now tested by switch_ivr_originate to
time-out a call quickly.
Would you/anyone be interested in a patch to do this? If so please let me
know the procedure for posting patches etc.
{P^/
On Tue, 21 Apr 2009 at 10:03 -0500, Anthony Minessale wrote:
That 100 trying is handled deep in the sip stack.
The author of sofia said it would be a big job to bring that up to the even
callback.
Someone may be able to persuade him to allow you to pass a global timeout
waiting for 100
or something but no solution exists atm
On Tue, Apr 21, 2009 at 3:32 AM, John Dalgliesh jo...@defyne.org wrote:
Hi,
I am trying to use FS to make outgoing SIP calls. I have a number of
gateways that can make the call. However, if one of them is down or has
some other problem then I would like to detect that quickly.
I intended to use the provisional '100 Trying' message for this... if it
hasn't been received in a couple of seconds then go on and try the next
gateway.
But I can't find a flag/event/state which corresponds to receipt of this
message. Can anyone tell me where I should be looking? I put a debug print
in sofia_event_callback for every event but there doesn't seem to be one
fired for this condition.
Thanks in advance.
{P^/
John
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[Freeswitch-users] need help getting ISDN talking to Cisco 3845
On Thu, Apr 9, 2009 Michael Collins wrote: Just curious - why are you using zaptel at all? Does it provide something for you that the wanpipe drivers do not? The wanpipe API mode isn't available on my platform (which is to say zaptel is the only game in town if you are using FreeBSD). See: http://wiki.sangoma.com/wanpipe-freebsd-drivers for futher information. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] need help getting ISDN talking to Cisco 3845
Okay, a few things. First off, the wanpipe2.conf file has a booboo. Don't think so. This line is WRONG: TDMV_DCHAN = 0 Not exactly. My understanding is you can use either: wanpipeX.conf: TDMV_DCHAN = 0 zaptel.conf: dchan = 24 (or in our case 48 since it's the second span) which means use zaptel to handle the d-channel hdlc or wanpipeX.conf: TDMV_DCHAN = 24 zaptel.conf: hardhdlc = 24 (or in our case 48 since it's the second span) which means use wanpipe to handle the d-channel hdlc assuming the wanpipe driver has the necessary support (wanpipe on my platform doesn't). Also, I recommend changing this line: wbg1 = wanpipe2, , TDM_VOICE, Comment To this: wbg1 = wanpipe2, , TDM_VOICE_API, Comment The sangoma voice API interface isn't available on my platform and shouldn't be necessary when using zaptel. assuming that this is what you want then you will need to use ozmod_libpri because the default OpenZAP PRI stack does not currently support being the network side. Are you sure? Openzap appears to contain implementations for both NT and TE. The configuration file supports specifying either user or network for the mode. Is the NT support currently nonfunctional? I had tried configuring the Cisco as the NT with similar results. I don't see where timing is specified It's the same T1 which was being used for RBS between FreeSWITCH and the Cisco so that timing (etc) should be okay. No errors are showing up at the physical level and the Cisco reports Layer 1 as active. The trace on the Cisco seems to show Layer 2 coming up (timestamps 22:53:44.264 through 22:54:21.760), then there's a long pause during which no Receive Ready frames are received from FreeSWITCH. At this point the Cisco gets unhappy and marks Layer 2 as down. If nothing obvious comes to anyone's mind, then I'll simply need to trace through the FreeSWITCH ISDN code and see what's going on. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] need help getting ISDN talking to Cisco 3845
Our FreeSWITCH setup has an existing T1 using RBS to talk to a digital modem pack in a Cisco 3845. I'm interested in changing from RBS to ISDN. I changed both sides, restart things, and see FreeSWITCH report: 2009-04-07 18:53:15 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:54:40 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:36 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:45 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:46 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:47 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:48 [NOTICE] Span:0 Q.921() I frame in invalid state ignored I've attached the configs and Cisco debug below. This is using the native ISDN support in FreeSWITCH with a Sangoma A104d on FreeBSD 6.4. I unfortunately don't currently speak ISDN (though I'm starting to pick up a little as a result of this exercise) ... suggestions / hints regarding what's going on and how to resolve it would be welcomed. -- John -- wanpipe2.conf --- [devices] wanpipe2 = WAN_AFT_TE1, Comment [interfaces] wbg1 = wanpipe2, , TDM_VOICE, Comment [wanpipe2] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 5 PCIBUS = 5 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 2 TE_CLOCK= MASTER TE_REF_CLOCK= 1 TE_HIGHIMPEDANCE= NO TE_RX_SLEVEL= 120 LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 2 TDMV_DCHAN = 0 TDMV_HW_DTMF= YES [wbg1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES -- zaptel.conf - #Sangoma A104 port 2 [slot:5 bus:5 span:2] wanpipe2 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 -- openzap.conf [span zt] ; A104D FE 2 1-6 MICA name = Cisco Digital Modem trunk_type = t1 number = 2487 b-channel = 25-47 d-channel = 48 --- openzap.conf.xml --- pri_spans span id=2 !-- Log Levels: none, alert, crit, err, warning, notice, info, debug -- param name=q921loglevel value=info/ param name=q931loglevel value=info/ param name=mode value=net/ param name=dialect value=national/ param name=dialplan value=XML/ param name=context value=default/ /span /pri_spans -- Cisco config controller T1 1/0 framing ESF linecode b8zs cablelength short 220 pri-group timeslots 1-24 interface Serial1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice modem isdn calling-number 2487 no cdp enable -- Cisco debug - #show isdn stat Global ISDN Switchtype = primary-ni ISDN Serial1/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x807F Number of L2 Discards = 2, L2 Session ID = 117 Total Allocated ISDN CCBs = 0 Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: User TX - SABMEp sapi=0 tei=0 Apr 7 22:53:44.312: ISDN Se1/0:23 Q921: User RX - UAf sapi=0 tei=0 Apr 7 22:53:44.312: %CSM-5-PRI: add PRI at 1/0:23 (index 0) Apr 7 22:53:44.312: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0:23, TEI 0 cha nged to up Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User RX - RRp sapi=0 tei=0 nr=0 Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User TX - RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User RX - RRp sapi=0 tei=0 nr=0 Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User TX - RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User TX - UAf sapi=0 tei=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User RX - RRp sapi=0 tei=0 nr=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User TX - RRf sapi=0 tei=0 nr=0 Apr 7 22:54:51.760: ISDN Se1/0:23 Q921: User TX - RRp sapi=0 tei=0 nr=0 Apr 7 22:54:52.760: ISDN Se1/0:23 Q921: User TX - RRp sapi=0 tei=0 nr=0 Apr 7 22:54:53.760: ISDN Se1/0:23 Q921: User TX - RRp sapi=0 tei=0 nr=0 Apr 7 22:54:54.760: ISDN Se1/0:23 Q921: User TX - RRp sapi=0 tei=0 nr=0 Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: User TX - SABMEp sapi=0 tei=0 Apr 7
Re: [Freeswitch-users] does anyone have a working FS / aastra config
Figured out the phone was sending packets that were too large, and the receiving system was not reassembling the fragmented packet. This can be fixed on the Aastra by enabling basic codecs: Go to the phone web-UI -- global SIP -- Codec Preference List -- Codec 1 -- change all to basic, save settings and restart the phone. Or in cfg files for aastra set: sip use basic codecs: 1 regards- John On Wed, Feb 4, 2009 at 10:06 PM, John Hyde jacre...@gmail.com wrote: I am having problems getting an Aastra 57i to make calls through FS. the phone registers fine, but all calls fail. If i use xlite or a nokia sip phone, i have no problems. Here is a packet capture of an attempted call: http://pastebin.freeswitch.org/7039 notice packet 9, it should have been a SIP INVITE, but it turned out to be a Fragmented IP protocol The phone and FS are both on the same lan subnet, and the phone connects fine with an asterisk server on the same subnet. Is there a known config for aastra phones that I can reference, or does anyone know why I am having this issue? -- john -- - j ___ Freeswitch-users mailing list Freeswitch-users@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] BT IPExchange Interoperability Testing
Brian West wrote: Yes search the mailing list people have interoped with BT in record time. On another note you hijacked the DTMF not being recognized by clicking reply, deleting the text and changing the subject. Please try not to do that in the future, click new message input freeswitch-users@lists.freeswitch.org then type your subject and message then click send. Your email client echo's back the headers that causes the mailing list server and many email clients to thread the message properly. Whoops, sorry! User IQ Error. jd -- John Daragon argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK Registered in England Company Number 02947782 v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ Freeswitch-users mailing list Freeswitch-users@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] Newbie - point me in the right direction
Hi, I am a real newbie. I have been building Asterisk based applications for a couple of years now. I am looking at migrating these apps to FreeSwitch - eventually. I want to do this gradually - I need to keep things running in the meantime. I have two Asterisk boxes, A1 A2, each running a separate telephony app. We have an external SIP service with DID's N200 - N299. We want to direct the incoming SIP calls so that the DID's N200 - N219 go to Asterisk server A1 and N220 - N299 to Asterisk server A2. Yes we really just want the calls switched on the DID. I'm struggling to know where to start - can someone point me in the right direction? Regards, John ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] VMWare voice quality
Michael Collins wrote: If anyone figures this out please post it to this thread. I am working on a wiki page for the VMWare appliance and I would like to be able to inform people on how to handle this situation. I had some issues under vmware fusion. They were resolved by adding clock=pit [1] to the kernel boot params and switching to host-only networking, and running natd + ipfw on the host system. The vmware natd would probably also work. haven't tried myself. The clock=pit is the big kicker. Also, recompiling the kernel with HZ=100 might help to reduce the load on the host system. [2] Though, with a small number of vms/vcpus on decent hw the number of context switches probably won't have much of an effect. [1] www.vmware.com/pdf/vmware_timekeeping.pdf [2] http://communities.vmware.com/docs/DOC-3580 Also, IIUC, those running VMWare Fusion on Macs are not experiencing this, correct? What about those using a hypervisor like ESXi? Any known issues? Thanks, MC On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice kr...@suspicious.org wrote: On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote: Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base ___ Freeswitch-users mailing list Freeswitch-users@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] LDAP Integration
Vinicius Kobashi wrote: hi ppl. i tried hard to make it work, but still i couldnt find a complete openldap scheme that provides these information, and i still could't find out where to put these configuration... can anyone help me? thankz! vinicius escreveu: thankz! ill set my openldap to provide these information.. but these about these binding settings... where should i set them? best regards John Skopis (Lists) wrote: vinicius wrote: hi ppl.. i tried to find something at google, but i couldnt manage to find anything. i still dont know what to do to make the mod_xml_ldap work. i couldnt find information about how to build a config file for the module, and where to store it... can anyone give me a help? Be advised mod_xml_ldap is probably not production quality and will undoubtedly change, eventually at least. Here is what I used once: bindings binding name=directory !--%s is populated with the extension -- param name=filter value=(FSid=%s) bindings=directory/ !--basedn for the searches %s is replaced with domain-- param name=basedn value=ou=people,dc=example / param name=url value=ldap://172.16.75.129; / param name=binddn value=cn=admin,dc=example / param name=bindpass value=secret / trans !-- we need to translate these attrs into FS attrs -- tran name=id mapfrom=FSid / tran name=mailbox mapfrom=FSmailbox / tran name=password mapfrom=FSPassword / tran name=vm-password mapfrom=FSvm-password / tran name=email-addr mapfrom=FSemail-addr / tran name=vm-email-all-messages mapfrom=FSvm-email-all-messages / tran name=vm-delete-file mapfrom=FSvm-delete-file / tran name=vm-attach-file mapfrom=FSvm-attach-file / /trans /binding binding name=configuration param name=filter value=(%s=%s) bindings=configuration/ param name=basedn value=name=%s,dc=example / param name=url value=ldap://172.16.75.129; / param name=binddn value=cn=admin,dc=example / param name=bindpass value=secret / /binding /bindings which should/probably/might work with ldap objects like these: dn: cn=John Skopis,ou=people,dc=example objectClass: person objectClass: inetOrgPerson objectClass: organizationalPerson objectClass: FreeSWITCH-Exten-Object objectClass: top cn: John Skopis sn: Skopis givenName: John FSid: 1001 FSmailbox: 1001 FSpassword: 1234 FSvm-password: 1001 FSemail-addr: john...@skopis.com FSvm-email-all-messages: TRUE FSvm-delete-file: TRUE FSvm-attach-file: TRUE dn: SIPIdentityUserName=1001,ou=h350,dc=example objectClass: person objectClass: SIPIdentity objectClass: top cn: 1001 sn: 1001 SIPIdentitySIPURI: sip:1...@172.16.75.129 SIPIdentityRegistrarAddress: 172.16.75.128 SIPIdentityProxyAddress: 172.16.75.128 SIPIdentityPassword: 1234 SIPIdentityUserName: 1001 SIPIdentityServiceLevel: premium Again, the module is not production quality. Hopefully I will conjurer the time and know-how to put something decent together eventually. To load configuration for any fs module you need to define the XML configuration element under the section configuration. A good starting point is the file $PREFIX/conf/freeswitch.xml http://wiki.freeswitch.org/wiki/Freeswitch.xml Also take a look at $PREFIX/logs/freeswitch.xml.fsxml to load mod_xml_ldap you would need to add something like this to modules.conf.xml load module=mod_xml_ldap / and create an xml_ldap.conf.xml in $PREFIX/autoload_configs/xml_ldap.conf.xml configuration name=xml_ldap.conf ... /configuration The ITU is doing some work called h.350: http://www.itu.int/ITU-T/studygroups/com16/h350/index.html Here is what I was working with: attributetype ( 1.3.6.1.4.1.65535.2.1.1 NAME 'FSid' DESC 'FreeSWITCH Extension ID' EQUALITY caseIgnoreIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) attributetype ( 1.3.6.1.4.1.65535.2.1.2 NAME 'FSmailbox' DESC 'FreeSWITCH Extension Mailbox' EQUALITY caseIgnoreIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) attributetype ( 1.3.6.1.4.1.65535.2.1.3 NAME 'FSpassword' DESC 'FreeSWITCH Password' EQUALITY caseExactIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.4 NAME 'FSa1hash' DESC 'FreeSWITCH Crypted Password' EQUALITY caseExactIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.5 NAME 'FSvm-password' DESC 'FreeSWITCH VoiceMail Password' EQUALITY integerMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.27 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.6 NAME 'FSemail-addr' DESC 'E-mail address to send
[Freeswitch-users] another switch_ivr_set_user() can't find user
a) Should sip_auth_realm be set by FreeSWITCH to the value associated
with force-register-domain
You have to remember the default assumes a lot. You go to changing
things you have to then change the way things are assumed.
I appreciate that. Let me ask the question slightly differently.
sofia_reg_parse_auth contains the following logic:
if (!switch_strlen_zero(profile-reg_domain)) {
domain_name = profile-reg_domain;
} else {
domain_name = realm;
}
where profile-reg_domain is set from force-register-domain.
It then calls switch_xml_locate_user using domain_name.
It looks like force-register-domain is intended to make
FreeSWITCH believe that the user is in domain specified by
force-register-domain.
Later there's:
switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
sip_auth_realm, realm);
switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
domain_name, realm);
Shouldn't the add_header for domain_name contain the value for
the actual domain used to locate the user?
And ideally shouldn't the rest of FreeSWITCH (including examples
intended to get you started) work in the same fashion for consistency
sake (i.e. when trying to locate a user reference the domain used by
sofia_reg_parse_auth to locate the user instead of blindly using
sip_auth_realm)?
My thought is if sofia_reg_parse_auth set things up properly,
then the rest of FreeSWITCH shouldn't know or even care that
force-register-domain is in use ... it should be as if the
VoIP phone had in fact registered using the domain specified
by force-register-domain.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495 | |
-
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[Freeswitch-users] another switch_ivr_set_user() can't find user
You don't have a default user in domain 192.168.14.10, in the default config I used this so that you can set some vars on every call with Thanks for pointing it out and explaining the purpose. It looks like the domain is coming from set_domain in default.xml which gets it from sip_auth_realm. I guess the question is if force-register-domain is being used then: a) Should sip_auth_realm be set by FreeSWITCH to the value associated with force-register-domain b) or should set_domain in default.xml simply check for force-register-domain when setting domain? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] LDAP Integration
vinicius wrote: hi ppl.. i tried to find something at google, but i couldnt manage to find anything. i still dont know what to do to make the mod_xml_ldap work. i couldnt find information about how to build a config file for the module, and where to store it... can anyone give me a help? Be advised mod_xml_ldap is probably not production quality and will undoubtedly change, eventually at least. Here is what I used once: bindings binding name=directory !--%s is populated with the extension -- param name=filter value=(FSid=%s) bindings=directory/ !--basedn for the searches %s is replaced with domain-- param name=basedn value=ou=people,dc=example / param name=url value=ldap://172.16.75.129; / param name=binddn value=cn=admin,dc=example / param name=bindpass value=secret / trans !-- we need to translate these attrs into FS attrs -- tran name=id mapfrom=FSid / tran name=mailbox mapfrom=FSmailbox / tran name=password mapfrom=FSPassword / tran name=vm-password mapfrom=FSvm-password / tran name=email-addr mapfrom=FSemail-addr / tran name=vm-email-all-messages mapfrom=FSvm-email-all-messages / tran name=vm-delete-file mapfrom=FSvm-delete-file / tran name=vm-attach-file mapfrom=FSvm-attach-file / /trans /binding binding name=configuration param name=filter value=(%s=%s) bindings=configuration/ param name=basedn value=name=%s,dc=example / param name=url value=ldap://172.16.75.129; / param name=binddn value=cn=admin,dc=example / param name=bindpass value=secret / /binding /bindings which should/probably/might work with ldap objects like these: dn: cn=John Skopis,ou=people,dc=example objectClass: person objectClass: inetOrgPerson objectClass: organizationalPerson objectClass: FreeSWITCH-Exten-Object objectClass: top cn: John Skopis sn: Skopis givenName: John FSid: 1001 FSmailbox: 1001 FSpassword: 1234 FSvm-password: 1001 FSemail-addr: john...@skopis.com FSvm-email-all-messages: TRUE FSvm-delete-file: TRUE FSvm-attach-file: TRUE dn: SIPIdentityUserName=1001,ou=h350,dc=example objectClass: person objectClass: SIPIdentity objectClass: top cn: 1001 sn: 1001 SIPIdentitySIPURI: sip:1...@172.16.75.129 SIPIdentityRegistrarAddress: 172.16.75.128 SIPIdentityProxyAddress: 172.16.75.128 SIPIdentityPassword: 1234 SIPIdentityUserName: 1001 SIPIdentityServiceLevel: premium ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after transfer
No. I wish it were that simple. I'm doing all of my testing on an internal network. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Wednesday, December 10, 2008 5:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer I smell a NAT... is there any NAT involved? On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford [EMAIL PROTECTED] wrote: Okay. I just tried this. Now we're getting the audio going one way, but not the other. So, I can hear the person that I just transferred to, but they can't hear me. Anyone have any other ideas? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raymond Chandler Sent: Wednesday, December 10, 2008 3:41 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer try blocking ICMP packets TO the MSS i had this exact same problem a few months ago MSS starts sending RTP to FS before FS is ready to accept so the OS catches the port not open and returns an ICMP 3:3 back to the MSS which in turn chokes on the queued up RTP and refuses to send anymore... -Ray John Rutherford wrote: I just emailed it to him. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Wednesday, December 10, 2008 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford [EMAIL PROTECTED] wrote: I have a pcap, but I'm not able to see anything obviously wrong with it. We find that some equipment (in fact a lot of equipment) have features that cause issues to be quite non-obvious, so perhaps you could give the pcap to Brian for him to review. He's a total ace when it comes to bug hunting. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 10, 2008 12:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer would be most helpful to capture a pcap of the entire thing by itself start to finish. /b On Dec 10, 2008, at 11:51 AM, John Rutherford wrote: No. I realize that's it's a B2BUA and that's exactly what we want. Everything with the transfer seems to work fine, except that there is no audio. ___ Freeswitch-users mailing list Freeswitch-users@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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after transfer
Sent. Let me know if you see anything. I'm not able to see anything wrong. Thanks, John From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Thursday, December 11, 2008 9:39 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer can you send a pcap of sip and rtp with the new problem? -Ray John Rutherford wrote: No. I wish it were that simple. I'm doing all of my testing on an internal network. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, December 10, 2008 5:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer I smell a NAT... is there any NAT involved? On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford j...@loopfx.com mailto:j...@loopfx.com wrote: Okay. I just tried this. Now we're getting the audio going one way, but not the other. So, I can hear the person that I just transferred to, but they can't hear me. Anyone have any other ideas? From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Wednesday, December 10, 2008 3:41 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer try blocking ICMP packets TO the MSS i had this exact same problem a few months ago MSS starts sending RTP to FS before FS is ready to accept so the OS catches the port not open and returns an ICMP 3:3 back to the MSS which in turn chokes on the queued up RTP and refuses to send anymore... -Ray John Rutherford wrote: I just emailed it to him. Thanks! -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, December 10, 2008 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford j...@loopfx.com mailto:j...@loopfx.com wrote: I have a pcap, but I'm not able to see anything obviously wrong with it. We find that some equipment (in fact a lot of equipment) have features that cause issues to be quite non-obvious, so perhaps you could give the pcap to Brian for him to review. He's a total ace when it comes to bug hunting. -MC From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, December 10, 2008 12:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer would be most helpful to capture a pcap of the entire thing by itself start to finish. /b On Dec 10, 2008, at 11:51 AM, John Rutherford wrote: No. I realize that's it's a B2BUA and that's exactly what we want. Everything with the transfer seems to work fine, except that there is no audio. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo
[Freeswitch-users] No audio after transfer
Sorry to repost, but I haven't heard anything back on this in a little while. I checked out the trunk last week. I'm on revision 10597. Thanks, John From: John Rutherford Sent: Monday, December 08, 2008 4:36 PM To: freeswitch-users@lists.freeswitch.org Subject: No audio after transfer I'm trying to get an attended transfer work with freeSWITCH, but it's not quite working. I have Microsoft Speech Server on one side and Televantage on the other. MSS is originating a call, which freeSWITCH is bridging to Televantage. That calls connects just fine. Then, MSS sends a re-INVITE to Televantage to put the call on hold. This works too. Then, MSS originates another call to freeSWITCH, which is again bridged to Televantage. This works fine too. Then, MSS sends a REFER to freeSWITCH to do the transfer. The transfer should be complete, but there is no audio between the two calls-just silence. I have looked at pcaps and the freeSWITCH logs, but I'm not seeing anything obviously wrong. After the REFER, I can see audio for both calls going between freeSWITCH and Televantage, so it seems that the only thing missing is freeSWITCH routing the audio from one call to the other call and vice-versa. Any help would be greatly appreciated. I have a pcap and the freeSWITCH logs, and I can easily reproduce this. Thanks! John ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after transfer
I just emailed it to him. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Wednesday, December 10, 2008 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford [EMAIL PROTECTED] wrote: I have a pcap, but I'm not able to see anything obviously wrong with it. We find that some equipment (in fact a lot of equipment) have features that cause issues to be quite non-obvious, so perhaps you could give the pcap to Brian for him to review. He's a total ace when it comes to bug hunting. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 10, 2008 12:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer would be most helpful to capture a pcap of the entire thing by itself start to finish. /b On Dec 10, 2008, at 11:51 AM, John Rutherford wrote: No. I realize that's it's a B2BUA and that's exactly what we want. Everything with the transfer seems to work fine, except that there is no audio. ___ Freeswitch-users mailing list Freeswitch-users@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] No audio after transfer
Okay. I just tried this. Now we're getting the audio going one way, but not the other. So, I can hear the person that I just transferred to, but they can't hear me. Anyone have any other ideas? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raymond Chandler Sent: Wednesday, December 10, 2008 3:41 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer try blocking ICMP packets TO the MSS i had this exact same problem a few months ago MSS starts sending RTP to FS before FS is ready to accept so the OS catches the port not open and returns an ICMP 3:3 back to the MSS which in turn chokes on the queued up RTP and refuses to send anymore... -Ray John Rutherford wrote: I just emailed it to him. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Wednesday, December 10, 2008 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a pcap, but I'm not able to see anything obviously wrong with it. We find that some equipment (in fact a lot of equipment) have features that cause issues to be quite non-obvious, so perhaps you could give the pcap to Brian for him to review. He's a total ace when it comes to bug hunting. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 10, 2008 12:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer would be most helpful to capture a pcap of the entire thing by itself start to finish. /b On Dec 10, 2008, at 11:51 AM, John Rutherford wrote: No. I realize that's it's a B2BUA and that's exactly what we want. Everything with the transfer seems to work fine, except that there is no audio. ___ Freeswitch-users mailing list Freeswitch-users@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] No audio after transfer
I'm trying to get an attended transfer work with freeSWITCH, but it's not quite working. I have Microsoft Speech Server on one side and Televantage on the other. MSS is originating a call, which freeSWITCH is bridging to Televantage. That calls connects just fine. Then, MSS sends a re-INVITE to Televantage to put the call on hold. This works too. Then, MSS originates another call to freeSWITCH, which is again bridged to Televantage. This works fine too. Then, MSS sends a REFER to freeSWITCH to do the transfer. The transfer should be complete, but there is no audio between the two calls-just silence. I have looked at pcaps and the freeSWITCH logs, but I'm not seeing anything obviously wrong. After the REFER, I can see audio for both calls going between freeSWITCH and Televantage, so it seems that the only thing missing is freeSWITCH routing the audio from one call to the other call and vice-versa. Any help would be greatly appreciated. I have a pcap and the freeSWITCH logs, and I can easily reproduce this. Thanks! John ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after transfer
Sorry. I forgot to mention that. I checked out the trunk last week. I have revision 10597. John From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, December 08, 2008 4:48 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No audio after transfer Are you on SVN trunk? If not what rev? /b On Dec 8, 2008, at 3:36 PM, John Rutherford wrote: Any help would be greatly appreciated. I have a pcap and the freeSWITCH logs, and I can easily reproduce this. Thanks! John ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] switch_ivr_set_user() can't find user
I can call between VoIP phone ext 1001 and 1003 fine. I can call from VoIP phone ext 1003 over a winkstart line into the PBX fine. Before updating to the current SVN I could also call from the PBX over a winkstart line to VoIP ext 1003 fine. Now what I see is: [NOTICE] switch_channel.c:551 switch_channel_set_name() New Channel OpenZAP/4:1/1003 [d99f59f2-52bb-dd11-80a4-001fc6ab49e2] [WARNING] switch_ivr.c:1832 switch_ivr_set_user() can't find user [EMAIL PROTECTED] [WARNING] mod_dptools.c:2024 user_outgoing_channel() Can't find user [EMAIL PROTECTED] [ERR] switch_ivr_originate.c:1067 switch_ivr_originate() Can not create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] It appears openzap sees the request from the PBX fine ... somehow FreeSWITCH can't connect the openzap inbound call to 1003 with the VoIP phone on ext 1003. Suggestions / pointers? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] switch_ivr_set_user() can't find user
make sure you have the domain_name variable set so the call can go to
the right domain name.
Set where? I'm pretty much using the stock sample files. vars.xml contains:
X-PRE-PROCESS cmd=set data=domain=$${local_ip_v4}/
It was my understanding that domain defaults to the IP address of the
server's interface.
openzap.conf.xml contains:
param name=context value=default/
in each of the analog_spans / analog_em_spans. Is something else needed
to specify the domain for processing inbound openzap calls?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495 | |
-
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[Freeswitch-users] switch_ivr_set_user() can't find user
No domain != domain_name they are different.
Well ... yes and no. default.xml has:
extension name=set_domain continue=true
condition field=${domain_name} expression=^$/
condition field=source expression=mod_sofia/
condition field=${sip_auth_realm} expression=^$
action application=set data=domain_name=$${domain}/
anti-action application=set data=domain_name=${sip_auth_realm}/
/condition
/extension
so it seems that it's intended for domain_name to equal domain if
a realm isn't supplied. If I use:
extension name=set_domain continue=true
condition field=${domain_name} expression=^$/
condition field=source expression=mod_sofia
action application=set data=domain_name=${sip_auth_realm}/
anti-action application=set data=domain_name=$${domain}/
/condition
/extension
then things work.
Thoughts:
1) It may be desirable to be able to specify the domain associated with
an openzap line in the openzap.conf.xml file.
2) The comment by set_domain says:
!--
Try to get the domain from the sip_auth_realm otherwise it will
default domain in vars.xml for cases it can't figure it out.
--
However it appears that the logic is wrong. It fails to handle
cases where the source isn't mod_sofia.
What JIRA category should I file this under?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495 | |
-
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[Freeswitch-users] switch_ivr_set_user() can't find user
This is the key to why its not working source is mod_openzap so might Yes ... I understand that. I'm pretty much running the supplied sample configuration which apparently doesn't handle openzap. Is there a specific JIRA category I should use to log this issue? I took a quick glance and wasn't sure what to use. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] javascript access to conf/directory/default
Okay if I do: xml_locate directory domain name 192.251.93.2 at the CLI I get XML. However, if I run the script: var d = apiExecute (xml_locate, directory domain name 192.251.93.2); console_log (err, D + d + \n); it appears that d is empty. Though: var d = apiExecute (status, ); console_log (err, D + d + \n); works fine. What's the proper way to invoke xml_locate from javascript? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] javascript access to conf/directory/default
I've written a javascript program which allows a caller to lookup an extension by entering the person's name using DTMF. Currently the list of name people / extensions is hardcoded. How do I iterate through conf/directory/default from javascript in order to dynamically build a table containing everyone's effective_caller_id_name and effective_caller_id_number instead of hardcoding it? Does FreeSWITCH load this into a registry of some type at startup and if so how do I access the registry from javascript? Likewise is there an easy way from javascript to determine the voicemail path (currently it is also hardcoded)? I use it in order to play the recorded_name (if present) for an extension when doing the lookup. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] Newb question on dialplan config
Hello, I am a FreeSwitch newb but have been using asterisk for a while now. I have a project for which I think FreeSwitch will be the best answer, so I need to learn. Have been reading the docs and followed the example at: http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk when I call from a Polycom on the asterisk box to a polycom on the freeswitch box all is good. When id do the reverse I.E. call the ast polycom from the freeswitch polycom I get only the following in the freswitch CLI: 2008-10-10 13:33:24 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [521c96a2-5205-bf46-9f9f-31124757b0ef] 2008-10-10 13:33:24 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing John Millican-2002 in context default 2008-10-10 13:33:24 [NOTICE] switch_ivr.c:1098 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-10 13:33:24 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/[EMAIL PROTECTED] [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 12 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] It would seem that the line: 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting is telling me my problem but I do not yet know why freeswitch does not have a route. I am certain that I have not correctly set the dial plan but haven't a clue what to look at. Both machines are on the 192.168.100.0 net, firewall is off on both the freeswitch box which is running on a VMware installation of WinXP SP3 and the asterisk box. I am using the default configs with the additions per the above page. I did have to change the following from the defaults in vars.xml to get 2 way audio when I call from asterisk to freeswitch: X-PRE-PROCESS cmd=set data=bind_server_ip=192.168.100.16/ X-PRE-PROCESS cmd=set data=external_rtp_ip=192.168.100.16/ X-PRE-PROCESS cmd=set data=external_sip_ip=192.168.100.16/ Any ideas? Is there something else I need to post to help decipher what I have done wrong or have not yet done? Thanks, JohnM ___ Freeswitch-users mailing list Freeswitch-users@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] Newb question on dialplan config
Brian West wrote:
Its looking for extension 2002 in context default on FreeSWITCH and
one doesn't exist so you get the NO ROUTE message.
Add a route to map 2002 so that it points at the Asterisk box.
/b
On Oct 10, 2008, at 1:00 PM, John Millican wrote:
Processing John Millican-2002 in context defau
I currently have this in default.xml in the context default:
extension name=ast_extens
condition field=destination_number expression=^(2\d{3})$
action application=set data=hangup_after_bridge=true/
action application=bridge data=sofia/external/[EMAIL PROTECTED]/
action application=hangup/
/condition
/extension
Is this not a routemap?
I apologize for such simple questions, but I am learning.
JohnM
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Re: [Freeswitch-users] Newb question on dialplan config
Yep that was it! Now I just need to add a matching gateway, as I am
getting the error no matching gateway found, which I think I can
figure out.
Thank you for such quick and accurate help.
JohnM
Brian West wrote:
I'm going to guess you added it at the very bottom of the default.xml?
It needs to be above this line:
X-PRE-PROCESS cmd=include data=default/*.xml/
/b
On Oct 10, 2008, at 1:22 PM, John Millican wrote:
I currently have this in default.xml in the context default:
extension name=ast_extens
condition field=destination_number expression=^(2\d{3})$
action application=set data=hangup_after_bridge=true/
action application=bridge data=sofia/external/[EMAIL PROTECTED]
mailto:sofia/external/[EMAIL PROTECTED]/
action application=hangup/
/condition
/extension
Is this not a routemap?
I apologize for such simple questions, but I am learning.
JohnM
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Re: [Freeswitch-users] Newb question on dialplan config
Thanks, will do that right now. JohnM Brian West wrote: The new default configs in SVN trunk have a HUGE warning at the very bottom along with more documentation. I highly recommend you check it out. http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/dialplan/default.xml?r=9935 /b On Oct 10, 2008, at 1:46 PM, John Millican wrote: Yep that was it! Now I just need to add a matching gateway, as I am getting the error no matching gateway found, which I think I can figure out. Thank you for such quick and accurate help. JohnM ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- John Millican Director of Technology Sentinel Communications, LLC PO Box 9 Wentworth, NH 03282 Phone (603) 764-9163 Fax (603) 764-7213 ___ Freeswitch-users mailing list Freeswitch-users@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] help with record_session
Hi, I'm having trouble getting record_session to work. I added record_session to my dialplan and now all calls are rejected with 488. The error logged says sofia_glue.c :1476 sofia_glue_tech_set_codec() No audio codec available. I have tried several different file extensions including wav, gsm, au, etc., and I get this same error with all of them. Has anyone run into this before? I installed from the openSUSE 10.3 rpm. Thanks! John ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] help with record_session
No. I have 'inbound-proxy-mode' set to true and I have 'inbound-late-negotiation' set to true as well. John From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Friday, October 03, 2008 12:27 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] help with record_session Are you doing no-media mode? /b On Oct 2, 2008, at 7:14 PM, John Rutherford wrote: Hi, I'm having trouble getting record_session to work. I added record_session to my dialplan and now all calls are rejected with 488. The error logged says sofia_glue.c :1476 sofia_glue_tech_set_codec() No audio codec available. I have tried several different file extensions including wav, gsm, au, etc., and I get this same error with all of them. Has anyone run into this before? I installed from the openSUSE 10.3 rpm. Thanks! John ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] help with record_session
That makes sense. I disabled those options and it's working now. Thanks! John From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Friday, October 03, 2008 1:21 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] help with record_session Can't record with those options set because we don't have a codec or the real media you can access. /b On Oct 3, 2008, at 12:14 AM, John Rutherford wrote: No. I have 'inbound-proxy-mode' set to true and I have 'inbound-late-negotiation' set to true as well. John ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] 1U servers and Sangoma A102 Card
Anyone had any luck getting the Sangoma A 102 card into a 1 U box? I'm looking to build 2 freeswitch servers for redundancy, and was wondering if anyone has had any luck getting these cards to work with 1U riser cards and if so with what brand of case/mobo? Any recommended bare bones kits would be awesome too. I'm not afraid to install a ESB motherboard or anything its just the last time I did it, it ended up involving a vise and a hacksaw, and the CEO isn't amused to walking in on such scenes in our tech department. ___ Freeswitch-users mailing list Freeswitch-users@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] T1 RBS Support Revisited
Is there support in FreeSWITCH for MF? To answer my own question ... Perhaps I can add a ZAP_COMMAND_SEND_MF case to zap_channel_command patterned after ZAP_COMMAND_SEND_DTMF and just use teletone_set_tone to set the proper frequencies??? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] conference.freeswitch.org via ipv6 (Testing)
Brian West wrote: http://www.tunnelbroker.net If you need/want ipv6 tunnels. sixxs also http://www.sixxs.net/pops/occaid/ ___ Freeswitch-users mailing list Freeswitch-users@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] Performance bottleneck
Michael Collins wrote: That begs the question… is there a mechanism in sqlite or Linux that allows for the RAM drive to be backed up periodically? That would be a cool feature to get documented for those power users like Ken! ;) Interesting thought: http://tldp.org/HOWTO/LVM-HOWTO/snapshots_backup.html -MC *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Ken Rice *Sent:* Tuesday, August 12, 2008 11:07 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Performance bottleneck The Disk IO on sqlite can be quite a bit... One work around for this is to create a ram drive of sufficient size and mount it to /usr/local/freeswitch/db (or whatever your db dir is for freeswitch) this helps out greatly... But anything in the db will not be saved across system reboots unless you do something about that yourself K *From: *Michael Jerris [EMAIL PROTECTED] *Reply-To: *freeswitch-users@lists.freeswitch.org *Date: *Tue, 12 Aug 2008 13:59:13 -0400 *To: *freeswitch-users@lists.freeswitch.org *Subject: *Re: [Freeswitch-users] Performance bottleneck It's going to be the disk io from sqlite. The presense states are all stored in sqlite (or odbc) data source. Mike On Aug 12, 2008, at 1:53 PM, UV wrote: Turning the presence off did the trick, although it would be important (to me, at least) to understand why as it changes the performance significantly. Is the presence mechanism waiting for some response from the network? I’m assuming it’s waiting on something external because I couldn’t find any CPU activity… *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony Minessale *Sent:* Wednesday, August 13, 2008 12:55 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Performance bottleneck 9996 is not a good test extension because it does not generate any audio unless it gets some. 9998 that generates a tone or make up an ext that plays a file is a better one. Processing of the sip calls can be delayed by the presence stuff which is very intensive, you can try turning it off and see if you get more calls. Also you should compare it to what happens with the test exten first in the dial plan. On Tue, Aug 12, 2008 at 2:58 AM, UV [EMAIL PROTECTED] wrote: I'm trying to determine the FS resource bottleneck when operating under load (in windows environment), but can't get the FS to load for some unseen reason. FS environment (a weak PC on purpose): CPU 2x Intel Pentium 4 3GHz RAM 2x 512MB DDR II RAM Chipset - Intel E7221 (Copper River) chipset ICH6R + FWH + BCM5721 LAN 1x Broadcom Giga LAN Windows 2003 Server – Service pack 2 FS version 9235 Running Release build on highest priority Load script: A different machine running sipP Running rtp_echo load, 50 cps, limit of 1000 calls, 30sec call duration, extension 9996 (echo test): sipp -rtp_echo -r 50 -l 1000 -d 3 -s 9996 -sf auc.xml -mp 25000 -i 192.168.1.1 http://192.168.1.1 -mi 192.168.1.1 http://192.168.1.1 192.168.1.2 http://192.168.1.2 Results: Test ran for 9.5 hours Total of 48828 calls - all successful No timeouts, retransmissions or unexpected messages. ___ Freeswitch-users mailing list Freeswitch-users@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] Comparison matirx
Grey Man wrote: [snip] One suggestion I'd have for another row is Security Fix Rate. For example while the Asterisk community's approach to handling security releases is commendable the rate at which they happen is a real pain when you have to potentially upgrade a production system for each one. Although the pain comes from having to worry about whether the version of Asterisk that you need to upgrade to will be one of the stable or dud versions! I would certainly agree the security is important. Responsiveness to security flaws is one thing. I think another point of valuation would be average bugs per year or month, weighted accordingly (pre-auth remote command execution should have a greater weight than an xss in the built-in web server). Though, that might turn into a whole other book. ;] ___ Freeswitch-users mailing list Freeswitch-users@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] Where do the trixswitch devs meet?
Mike Fedyk wrote: Hi, I'm taking a look at the installation of freeswitch in the trixswitch project and see a few things I'd like to help with, but I don't see any SRPMs, or even posts from the developers on their forums. Anything to point me in the right direction would be great. fbsd-64# find . -name *.spec ./libs/apr/apr.spec ./libs/speex/Speex.spec ./libs/sofia-sip/packages/sofia-sip-1.12.9.spec ./libs/apr-util/apr-util.spec ./libs/voipcodecs/voipcodecs.spec ./libs/curl/packages/Linux/RPM/curl.spec ./libs/curl/packages/Linux/RPM/curl-ssl.spec ./libs/curl/packages/AIX/RPM/curl.spec ./libs/libsndfile/libsndfile.spec ./libs/js/nsprpub/pkg/linux/sun-nspr.spec ./freeswitch.spec I have never actually used freeswitch packages. I think it would be great if they would offer daily debug/prod builds. Also, irc is a great resource. HTH -john ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] safe_freeswitch (like safe_asterisk): restarting FS automatically?
Birgit Arkesteijn wrote:
Hi all,
We've got an older version of FreeSWITCH (Trunk 7948) running on a Linux
x86_64 machine. At the moment it's crashing few times a day, making our
services very unreliable.
At the moment we don't have the time to rebuild this version, so I'm
looking for an equivalent of the safe_asterisk script. This script
runs Asterisk in a loop, restarting it when it goes down.
I couldn't find any equivalent script, but maybe I using the wrong
keywords in my search.
Does anyone know if such a script is available, and (if so) could you
point out where I can find it?
Thanks, Birgit
not too go against the freeswitch crew, especially since I agree you
should probably update. (just backup fs first).
I am not sure of the performance implications running fs like this (-nf)
but you can do something like (and this it's quite a nasty hack):
#!/bin/sh
PREFIX=/usr/local/freeswitch
stop() {
local count=0
kill -9 `cat /var/run/freeswitch_loop.pid` /dev/null 21
while [ $count -lt 10 ]; do
[ $count -gt 0 ] sleep 5
count=`expr $count + 1`
pidof freeswitch /dev/null killall freeswitch || count=100
done
return $?
}
start() {
while true; do $PREFIX/bin/freeswitch -nc -nf /dev/null 21; done
echo $! /var/run/freeswitch_loop.pid
return 0
}
case $1 in
start) start ;;
stop) stop ;;
restart) killall -HUP freeswitch;;
reload) killall -HUP freeswitch;;
esac
exit $?
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Re: [Freeswitch-users] trunk fails on FreeBSD
Adrian Gschwend wrote:
Adrian Gschwend wrote:
I can't compile latest trunk on FreeBSD 7.0-STABLE, bootstrap works,
configure fails here:
[...]
I could solve that problem at least:
My BSD has the following autoconf installed:
autoconf-2.61_2
autoconf-2.62
- autoconf-2.62 is the problem, bootstrap decides to take that one in
my case and that doesn't work. Doing it with 2.61 solves the issue.
However, mod_spidermonkey doesn't want to compile now, I get plenty of
errors like:
making all mod_spidermonkey
./config/autoconf.mk, line 112: Need an operator
./config/autoconf.mk, line 113: Need an operator
./config/autoconf.mk, line 114: Need an operator
./config/autoconf.mk, line 120: Need an operator
./config/autoconf.mk, line 121: Need an operator
./config/autoconf.mk, line 122: Need an operator
./config/autoconf.mk, line 127: Need an operator
./config/autoconf.mk, line 128: Need an operator
./config/autoconf.mk, line 129: Need an operator
Makefile, line 52: Need an operator
Makefile, line 55: Missing dependency operator
Makefile, line 56: Need an operator
Makefile, line 58: Need an operator
Makefile, line 59: Need an operator
Makefile, line 60: Need an operator
Makefile, line 61: Need an operator
./config/rules.mk, line 73: Need an operator
./config/rules.mk, line 75: Need an operator
[...]
The Makefile looks like this:
you need to use gmake on *BSD.
fbsd-64# make mod_spidermonkey-clean
`libfreeswitch.la' is up to date.
making clean mod_spidermonkey
fbsd-64# make mod_spidermonkey-install
/bin/sh /usr/local/src/freeswitch/quiet_libtool --mode=install
/usr/bin/install -c 'libfreeswitch.la'
'/usr/local/freeswitch/lib/libfreeswitch.la'
/usr/bin/install -c .libs/libfreeswitch.so.1
/usr/local/freeswitch/lib/libfreeswitch.so.1
(cd /usr/local/freeswitch/lib { ln -s -f libfreeswitch.so.1
libfreeswitch.so || { rm -f libfreeswitch.so ln -s libfreeswitch.so.1
libfreeswitch.so; }; })
(cd /usr/local/freeswitch/lib { ln -s -f libfreeswitch.so.1
libfreeswitch.so || { rm -f libfreeswitch.so ln -s libfreeswitch.so.1
libfreeswitch.so; }; })
/usr/bin/install -c .libs/libfreeswitch.lai
/usr/local/freeswitch/lib/libfreeswitch.la
/usr/bin/install -c .libs/libfreeswitch.a
/usr/local/freeswitch/lib/libfreeswitch.a
chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
ranlib /usr/local/freeswitch/lib/libfreeswitch.a
--
Libraries have been installed in:
/usr/local/freeswitch/lib
If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
- add LIBDIR to the `LD_LIBRARY_PATH' environment variable
during execution
- add LIBDIR to the `LD_RUN_PATH' environment variable
during linking
- use the `-Wl,--rpath -Wl,LIBDIR' linker flag
See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
--
making install mod_spidermonkey
Compiling mod_spidermonkey.c...
Creating mod_spidermonkey.so...
./config/autoconf.mk, line 112: Need an operator
./config/autoconf.mk, line 113: Need an operator
./config/autoconf.mk, line 114: Need an operator
./config/autoconf.mk, line 120: Need an operator
./config/autoconf.mk, line 121: Need an operator
./config/autoconf.mk, line 122: Need an operator
./config/autoconf.mk, line 127: Need an operator
./config/autoconf.mk, line 128: Need an operator
./config/autoconf.mk, line 129: Need an operator
Makefile, line 52: Need an operator
Makefile, line 55: Missing dependency operator
Makefile, line 56: Need an operator
Makefile, line 58: Need an operator
Makefile, line 59: Need an operator
Makefile, line 60: Need an operator
Makefile, line 61: Need an operator
./config/rules.mk, line 73: Need an operator
./config/rules.mk, line 75: Need an operator
./config/rules.mk, line 77: Need an operator
./config/rules.mk, line 79: Need an operator
./config/rules.mk, line 81: Need an operator
./config/config.mk, line 55: Need an operator
./config/config.mk, line 57: Need an operator
./config/config.mk, line 59: Need an operator
./config/config.mk, line 61: Need an operator
./config/config.mk, line 77: Need an operator
./config/config.mk, line 80: Need an operator
./config/config.mk, line 84: Missing dependency operator
./config/config.mk, line 86: Need an operator
./config/config.mk, line 88: Missing dependency operator
./config/config.mk, line 90: Need an operator
./config/config.mk, line 91: Missing dependency operator
./config/config.mk, line 94: Need an operator
./config/config.mk, line 95: Missing dependency operator
./config/config.mk, line 97: Missing dependency operator
./config/config.mk, line 99: Need an operator
./config/config.mk, line 101:
Re: [Freeswitch-users] trunk fails on FreeBSD
Adrian Gschwend wrote: Hi group, I can't compile latest trunk on FreeBSD 7.0-STABLE, bootstrap works, configure fails here: === configuring in libs/libsndfile (/usr/home/ktk/freeswitch.trunk/libs/libsndfile) configure: running /bin/sh ./configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' --cache-file=/dev/null --srcdir=. checking build system type... i386-unknown-freebsd7.0 checking host system type... i386-unknown-freebsd7.0 checking target system type... i386-unknown-freebsd7.0 ./configure.lineno: 1997: Syntax error: word unexpected (expecting ) I was already discussing that two nights ago on IRC. Does that work for anyone on FreeBSD with the latest trunk code? My BSD has the following autoconf installed: autoconf-2.61_2 autoconf-2.62 I can't take 1.0.0 as this fails in another module :) cu Adrian It worked for me on 7.0-RELEASE on amd64. Not sure if it has anything to do with autconf-wrapper. autoconf-2.61_2 Automatically configure source code on many Un*x platforms autoconf-wrapper-20071109 Wrapper script for GNU autoconf automake-1.9.6_2GNU Standards-compliant Makefile generator (1.9) automake-wrapper-20071109 Wrapper script for GNU automake Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 9065 Node Kind: directory Schedule: normal Last Changed Author: anthm Last Changed Rev: 9065 Last Changed Date: 2008-07-16 22:04:31 + (Wed, 16 Jul 2008) ___ Freeswitch-users mailing list Freeswitch-users@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] send_dtmf problems
Try a sleep after the answer. Will do ... just curious as to why and for how long? 1) When I dial the extension from a Grandstream GXP-2000 ... How do I configure FreeSWITCH to send both RTP digits and inband audio? The VoIP phone has no reason to reproduce any DTMF. I understand that the VoIP phone has no reason to create DTMF tones from the RTP digits sent by FreeSWITCH. What I was asking is how to you configure FreeSWITCH to generate inband DTMF tones in addition to the RTP digits. I.e. this particular phone can be configured to send DTMF digits to FreeSWITCH as any combination of inband audio, RTP digits, and / or SIP info. Going in the opposite direction I see where in the FreeSWITCH sofia xml configuration type you can set the dtmf-type as either rfc2833 or as SIP info however I don't see an option for using both nor do I see an option for generating inband DTMF audio. Note: I was just using send_dtmf with the VoIP phone for testing purposes ... this application doesn't actually need to send DTMF to a VoIP phone. 2) When I dial the extension from a FXO port on a Sangoma A204D it connects, Is this using OpenZAP? Yes. In my particular application the System 25 PBX connects to FreeSWITCH using OpenZAP running on a Sangoma A204DX card. At the end of a call sent to voicemail I need FreeSWITCH to send some DTMF tones to the PBX in order to turn on / off the message waiting indicator. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] send_dtmf problems
action application=gentones data=1234567890/ Yep, this works through the A204d FXO openzap lines, though sometimes there's little odd click / hiccup in the middle of playing the tones. I'm a little confused as to why using send_dtmf didn't seem to work well, however no matter. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] slow hangup detection using FXO into voicemail application
If I call into the voicemail application from a VoIP phone and hangup while the welcome message is playing FreeSWITCH reports the hangup right away. If I call into the voicemail application from a A204DX FXO line and hangup while the welcome message is playing FreeSWITCH doesn't report the hangup until 33 seconds have elapsed. I've emailed Sangoma however thought to post it here in case it rings a bell. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] send_dtmf problems
I need to send some dtmf tones after handling vmail for the System 25 so I've set up (for testing purposes) an extension which should just answer the phone and play dtmf: extension name=system25_vmail condition field=destination_number expression=^(5590)|(55[1-8])$ action application=answer/ action application=send_dtmf data=[EMAIL PROTECTED]/ action application=sleep data=5000/ !--action application=javascript data=system25vmail.js/-- /condition /extension 1) When I dial the extension from a Grandstream GXP-2000 a network trace shows the digits being sent from FreeSWITCH as RTP events, however no dtmf audio tones are sent. How do I configure FreeSWITCH to send both RTP digits and inband audio? 2) When I dial the extension from a FXO port on a Sangoma A204D it connects, after five seconds it sends two tones, and then disconnects. How do I configure FreeSWITCH to send all the dtmf tones? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] outbound fxo line pooling
I currently have:
extension name=outgoing-fxo
condition field=destination_number expression=^(4[0-9][0-9])$
action application=set data=dialed_ext=$1/
action application=export data=dialed_ext=$1/
action application=bridge data=openzap/1/1/${dialed_ext}/
/condition
/extension
which routes calls to extension 4XX out the first openzap line.
How do I set up a pool of openzap lines and route calls to extension
4XX to any available openzap line?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495 | |
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[Freeswitch-users] T1 RBS Support?
Just curious as to the state of / plan for T1 RBS support. Our System 25 PBX doesn't support ISDN, however it does support RBS. It would be nice to be able to run more lines into our FreeSWITCH box using a T1 instead of analog lines. -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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 / dialplan / Sangoma A204 questions
Can you open an Jira on this so we don't loose track of it.. http://jira.freeswitch.org I took a look, however I don't see openzap listed there ... do you want filed under endpoint modules? BTW: Here's the build error: src/zap_ss7_boost.c: In function `zap_ss7_boost_run': src/zap_ss7_boost.c:849: error: syntax error before rfds src/zap_ss7_boost.c:864: warning: implicit declaration of function `FD_ZERO' src/zap_ss7_boost.c:864: error: `rfds' undeclared (first use in this function) src/zap_ss7_boost.c:864: error: (Each undeclared identifier is reported only once src/zap_ss7_boost.c:864: error: for each function it appears in.) src/zap_ss7_boost.c:865: error: `efds' undeclared (first use in this function) src/zap_ss7_boost.c:866: warning: implicit declaration of function `FD_SET' src/zap_ss7_boost.c:873: warning: implicit declaration of function `select' src/zap_ss7_boost.c:878: warning: implicit declaration of function `FD_ISSET' make: *** [src/zap_ss7_boost.o] Error 1 -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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 / dialplan / Sangoma A204 questions
Can you try upgrading to the latest svn build of FS. There are several fixes to openzap in there that I know will fix your issue. Yes, that works much better. A couple minor changes required so the openzap code would compile on FreeBSD 6.2: a) The header for select needed to be included. b) A comment needed to be fixed. I'm now able to dial out through the fxo as well as have incoming calls answered. What would my dialplan look like so that dialing 551 just bridges the call to the tip ring line from the PBX? I.e. dialing 551 gets me a PBX dialtone without actually dialing an extension on the PBX. -- John ---8---8--- *** libs/openzap/src/include/openzap.h.ORIGINAL Tue Jul 1 19:07:52 2008 --- libs/openzap/src/include/openzap.h Tue Jul 1 19:20:12 2008 *** *** 127,132 --- 127,133 #include strings.h #endif #include assert.h + #include sys/select.h #include zap_types.h #include hashtable.h #include zap_config.h *** libs/openzap/src/isdn/Q931.c.ORIGINAL Tue Jul 1 19:07:51 2008 --- libs/openzap/src/isdn/Q931.cTue Jul 1 19:17:25 2008 *** *** 346,352 /* Protocol Discriminator */ m-ProtDisc = Mes[IOff++]; ! /* CRV */add m-CRVFlag = Mes[IOff + 1] 0x80; m-CRV = Q931Uie_CRV(pTrunk, Mes, m-buf, IOff, ISize); --- 346,352 /* Protocol Discriminator */ m-ProtDisc = Mes[IOff++]; ! /* CRV add */ m-CRVFlag = Mes[IOff + 1] 0x80; m-CRV = Q931Uie_CRV(pTrunk, Mes, m-buf, IOff, ISize); - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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 / dialplan / Sangoma A204 questions
Can you try upgrading to the latest svn build of FS. There are several fixes to openzap in there that I know will fix your issue. Note I do get the following messages: [WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:2 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 17 Any idea what they mean? Should I be concern? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] openzap / dialplan / Sangoma A204 questions
I have a A204 with hardware echo cancelling and two FXO modules
on FreeBSD 6.2 connected to tip-ring lines from a PBX. ztcfg reports:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
Hooking a plain old telephone to the tip-ring lines from the PBX
works fine. On startup freeswitch reports:
[DEBUG] zap_io.c:1951 load_config() found config for span
[DEBUG] zap_io.c:1978 load_config() created span 1 of type zt
[DEBUG] zap_io.c:1991 load_config() span 1 [name]=[OpenZAP]
[DEBUG] zap_io.c:1991 load_config() span 1 [number]=[551]
[DEBUG] zap_io.c:1991 load_config() span 1 [fxo-channel]=[1]
[DEBUG] zap_io.c:2020 load_config() setting trunk type to 'FXO' start(KEWL)
[WARNING] zap_zt.c:135 zt_open_range() this ioctl fails on older zaptel but
is harmless if you used ztcfg
[device /dev/zap/channel chan 1 fd 26 (Inappropriate ioctl for device)]
[INFO] zap_zt.c:170 zt_open_range() configuring device /dev/zap/channel
channel 1 as OpenZAP device 1:1 fd:25
Ultimately I want freeswitch to run a script when any of the FXO lines
receive a call. Playing around produced some questions:
1) I have a dialplan of:
extension name=outgoing-fxo
condition field=destination_number expression=^55[1-4]$
action application=set data=dialed_ext=482/
action application=bridge data=openzap/1/1/${dialed_ext}/
/condition
/extension
which I'm assuming will cause freeswitch to use the fxo to dial 482
on the PBX when routing a call to 551. When I dial 551 from a VoIP
phone I see:
[NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel
OpenZAP/1:1/482 [edf17e96-0247-dd11-9800-001fc6ab49e2]
[WARNING] zap_analog.c:52 analog_fxo_outgoing_call() VETO Changing state on
1:1 from DOWN to DIALING
[WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:1
however I don't hear anything on the VoIP phone (i.e. no ringing) and
extension 482 which is right next to the VoIP doesn't ring.
2) What would my dialplan look like so that dialing 551 bridges the call
to the FXO with the FXO just going off hook ... not dialing?
I.e. dialing 551 just gets me a PBX line with dialtone.
3) What condition would I use in my dialplan to match an FXO line ringing?
I.e. when the FXO line rings I want to invoke javascript.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495 | |
-
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Re: [Freeswitch-users] config / scripting questions from voice mail integration attempt
Anthony Minessale writes: apiExecute is for FSAPI calls voicemail is a dialplan application. you would want to do one of the following: session.execute(voicemail, voicemail args); session.execute(transfer, some ext that leads to voicemail); Yep ... that was exactly the command I was looking for. I take it that execute is more like the C function system rather than the system call exec? I.e. control is returned to the javascript after the application finishes? Your plan to set the variables is the best one to avoid overhead of leaving JS running. The only thing you can't do is set a variable in one action of your dialplan and then expect that variable to be set to do more condition tags inside that same dialplan. The inband DTMF contains information regarding whether the caller is checking their voicemail or leaving a message for someone else which means that based on the DTMF I need to call the voicemail application with different parameters. How can I have the script just set variables if I can't use those variables with condition tags later in the dialplan to control how to call the voicemail application? -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] config / scripting questions from voice mail integration attempt
if you do session.execute(transfer, some ext); and exit the script. That goes back to the dialplan for a *new* lookup so now your variables can be used in conditions. Thanks, -- John - | Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] | |John Wehle| Fax: 1-215-540-5495 | | - ___ Freeswitch-users mailing list Freeswitch-users@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] [Fwd: Openser + FreeSwitch Integration.]
Brian West wrote: On Jun 18, 2008, at 1:11 AM, Aadilkhan Maniyar wrote: Thanks for the reply Brian. So what you mean to say is that I need not configure mod_spidermonkey_odbc at all in order to store registration data in the MySQL database?. Correct this is just a way to access odbc from Spidermonkey. If I configure FS with --enable-core-odbc and set the /odbc-dsn/ parameter in one of the sip profile xml, i should be able to store the registrations in the MySQL database. Do you see any errors during startup? Does isql work with your dsn? Brian West wrote: Considering that mod_spidermonkey_odbc isn't for this that would explain why that wasn't working. You need to configure freeswitch with --enable-core-odbc-support In addition to what Brian said you need to configure FS to use the odbc dsn. grep in the conf dir: grep -RHin dsn * to see how. ___ Freeswitch-users mailing list Freeswitch-users@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] LDAP Internal FreeSwitch
Nicola wrote: Thanks ... now works ... the problem was that the server ldap lacked the parameter: dial-string. Last problem: How come when I call some internal, the voice can not go? Could it be a firewall issue? Do you see anything in your logs? ___ Freeswitch-users mailing list Freeswitch-users@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] LDAP Internal FreeSwitch
Nicola wrote:
I have checked ...
authentication is successful on the server ...
I have only one problem: if two internal name is registered on LDAP,
answered the VoiceMail even though the interiors are online.
I attach logs:
2008-06-05 12:10:31 [WARNING] mod_dptools.c: 372 set_user_function () can
not find user [EMAIL PROTECTED]
As if the system proves to call default .. and not the internal desired.
You need to make sure that 'default' is a valid extension. Look at the
Local Extension portion of the public and default dialplans.
Also, and I am not sure if something changed recently, but you need to
have a dial-string, and probably user_context, for the XML to be
considered valid.
Here's what I used to test:
dn: cn=1235,ou=172.16.75.129,dc=example
objectClass: top
objectClass: sipCred
objectClass: inetOrgPerson
cn: 1235
sn: 1235
idname: 1235
param: password
param: vm-password
param: dial-string
paramvalue: 1234
paramvalue:
paramvalue:
[EMAIL
PROTECTED],transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/[EMAIL
PROTECTED])}
variable: accountcode
variable: user_context
variable: effective_caller_id_name
variable: effective_caller_id_number
variablevalue: 1234
variablevalue: default
variablevalue: test
variablevalue: 0
dn: cn=1234,ou=172.16.75.129,dc=example
objectClass: top
objectClass: sipCred
objectClass: inetOrgPerson
cn: 1234
sn: 1234
idname: 1234
param: password
param: vm-password
param: dial-string
paramvalue: 1234
paramvalue:
paramvalue:
[EMAIL
PROTECTED],transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/[EMAIL
PROTECTED])}
variable: accountcode
variable: user_context
variable: effective_caller_id_name
variable: effective_caller_id_number
variablevalue: 1234
variablevalue: default
variablevalue: test
variablevalue: 0
hth,
john
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Re: [Freeswitch-users] LDAP Internal FreeSwitch
Nicola wrote: Hello, I am writing from italy ... I'm trying to use FreeSwitch with ldap ... So far I managed, after many attempts to create LDAP server on the schema. That schema was meant as an example only. I think the only attrs the module uses are: param paramvalue variable variablevalue Only those attrs need to be in the schema. FreeSwitch still leans to the various files.Xml (1000.xml, 1001.xml) etc. to manage the internal How can I make it clear to freeswitch that must use LDAP as internal SIP? I set in conf/directory/default.xml: domain name=foo Since there is no sofia domain configured of that name (there shouldn't be at least) the default config is gone. Of course you could also just rm -rf conf/directory ;] -john ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] LDAP Internal FreeSwitch
Nicola wrote: thanks for your answer the problem is that FreeSwitch NOT MADE no query the LDAP server. Consequently, I can not know whether the query is wrong or not. I tried to follow the instructions you have given me kindly ... but freeswitch not going to make any queries on LDAP: In fact, when I try to register with a cliet SIP, debugging freeswitch tells me: User non-existent! Grazie... It sounds like possibly the module is not loaded? add an entry to conf/autoload_configs/modules.conf.xml: load module=mod_xml_ldap/ Also, make sure you have conf/autoload_configs/xml_ldap.conf.xml If the query is not successful the mod will let you know. prego ___ Freeswitch-users mailing list Freeswitch-users@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 Ldap Integration
Michael Jerris wrote: For sip auth we do pass all the hash information from the auth headers when it does the lookup, so you are able to do the auth in your module if you care to. We are unable to pass up the raw password as we never actually have that information. Mike I remember now... sip phones != mail clients. but if they were... Thanks for reminding me Mike. I am not exactly sure how difficult it is to do digest auth in openldap or AD. If I manage to find some time I will look into it though, -john ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch Ldap Integration
Hi, I had some free time and decided to add sasl support to mod_xml_ldap. I only tested kerberos5 auth though. I also added support for search filters. The problem with the xml_ldap directory stuff is that since freeswitch never actually sends the hash (or cleartext password) to the module it can't bind as the extension to compare. I would have much rather registered a module as an authentication provider vs a xml provider. As I understand this portion of code is somewhere in between the core and mod_sofia so it would need to be done as an extension to mod_sofia. All it does is connect to ldap as a privileged user, search for a freeswitch extension, generate xml, and send the xml to fs. If I find some more free time I may add support for requested attrs in the ldap query. If you take a look at the code it currently grabs all attrs and then joins on key=keyval. I did this so ldap schema didn't have to be extended everytime the fs xml schema is extended. The reason would be rather than requesting 100 attrs and iterating through them all just request the ~4 required to generate the xml and iterate through them. Also startls/ssl support might be nice ;] -john PS Until someone adds the config to tree (I cant write the config file =\) here it is: http://rafb.net/p/A37tLo10.html Faraz R. Khan wrote: Thanks a lot. I intend to use it mostly as a SIP user directory. For the dial-plan I dont mind parsing and syncing XML file across servers (if there were a small cluster). The main deal is AUTHENTICATION. The authentication scheme I wish to keep is Kerberos (with SASL in Ldap for binding). This way all my credentials are centralized, be it SIP or mail. This would be a great achievement for me and many enterprises having thousands of identities. The dialplan stays fairly static once developed so I dont mind that being in a XML file. The dynamic stuff (user credentials) I wish to keep in a centralized store such as LDAP. On Wed, 2008-05-28 at 20:12 -0500, John Skopis (Lists) wrote: At one point I was very interested in this...then I got a job. =[ I thought mod_ldap was more of a PoC than anything. It might work (I couldn't get it working and unfortunately don't remember exactly why..) but there really isn't much point. I would have to do at least 5 ldap queries (if not more) to get the most of the same functionality as the XML dialplan. Also, the elegance of stackable functionality in the XML dialplan is hard to imitate, at least with the any amount of efficiency. If you don't need to stack actions a regular expression will almost certainly be better. Attached is the schema, config, and sample ldif I used to get the xml stuff working. With a little effort it could work with an existing schema (possibly the ITU recomended LDAP schema that ser uses). I am not sure how easy it would be to get the same flexibility as key/value pairs (like the FS xml uses) though. -John Anthony Minessale wrote: We have a concept called the directory interface not to be confused with the user directory. The directory interface is a pluggable abstract API that looks and feels like LDAP only you can plug in anything you want to implement the functions. mod_ldap is a module that registers to this interface and connects LDAP to it. So essentially you load mod_ldap then you use the freeswitch directory interface as you would have used the ldap code and it will carry over. There is a mod_dialplan_directory who uses the directory interface to ask for a dialplan, and installs the results into an extension. In the case of mod_ldap obviously it allows you to get your dialplan from LDAP. Now also in mod_ldap, there is some code someone recently contributed to tie our XML interface to LDAP, This is more interesting because then when anything in FreeSWITCH tries to lookup a user, dialplan entry or anything else in our XML config, all the important details are passed to LDAP where it can make a query, pull the info out of LDAP and deliver it back to FreeSWITCH as the XML it was looking for. This allows you to make all of the registrations, dialplan etc real time driven by LDAP, you can also bind a perl or lua script to this operation as well as mod_xml_curl who will turn the request into an HTTP post to a web server to fetch the data. On Wed, May 28, 2008 at 8:09 AM, Michael Jerris [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Currently the directory interface is only used for that dialplan, I would like to enhance that in the future. The directory dialploan uses a filter of exten=destination number, and then has name/value pairs, I will see if I can find the schema we used back when we developed it, short of that, the code is the best reference on that. Another option is to use mod_xml_curl and have your cgi back end to ldap of your choice. This will give you more flexibility to use
Re: [Freeswitch-users] Freeswitch Ldap Integration
At one point I was very interested in this...then I got a job. =[ I thought mod_ldap was more of a PoC than anything. It might work (I couldn't get it working and unfortunately don't remember exactly why..) but there really isn't much point. I would have to do at least 5 ldap queries (if not more) to get the most of the same functionality as the XML dialplan. Also, the elegance of stackable functionality in the XML dialplan is hard to imitate, at least with the any amount of efficiency. If you don't need to stack actions a regular expression will almost certainly be better. Attached is the schema, config, and sample ldif I used to get the xml stuff working. With a little effort it could work with an existing schema (possibly the ITU recomended LDAP schema that ser uses). I am not sure how easy it would be to get the same flexibility as key/value pairs (like the FS xml uses) though. -John Anthony Minessale wrote: We have a concept called the directory interface not to be confused with the user directory. The directory interface is a pluggable abstract API that looks and feels like LDAP only you can plug in anything you want to implement the functions. mod_ldap is a module that registers to this interface and connects LDAP to it. So essentially you load mod_ldap then you use the freeswitch directory interface as you would have used the ldap code and it will carry over. There is a mod_dialplan_directory who uses the directory interface to ask for a dialplan, and installs the results into an extension. In the case of mod_ldap obviously it allows you to get your dialplan from LDAP. Now also in mod_ldap, there is some code someone recently contributed to tie our XML interface to LDAP, This is more interesting because then when anything in FreeSWITCH tries to lookup a user, dialplan entry or anything else in our XML config, all the important details are passed to LDAP where it can make a query, pull the info out of LDAP and deliver it back to FreeSWITCH as the XML it was looking for. This allows you to make all of the registrations, dialplan etc real time driven by LDAP, you can also bind a perl or lua script to this operation as well as mod_xml_curl who will turn the request into an HTTP post to a web server to fetch the data. On Wed, May 28, 2008 at 8:09 AM, Michael Jerris [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Currently the directory interface is only used for that dialplan, I would like to enhance that in the future. The directory dialploan uses a filter of exten=destination number, and then has name/value pairs, I will see if I can find the schema we used back when we developed it, short of that, the code is the best reference on that. Another option is to use mod_xml_curl and have your cgi back end to ldap of your choice. This will give you more flexibility to use other caller information in your ldap lookup. Mike On May 28, 2008, at 1:58 AM, Faraz R. Khan wrote: First of all- Amazing project. Tired of asterisk deadlocking all the time we have been deploying asterisk with OpenSER as the registrar. Freeswitch is a huge relief! This is an extremely important feature we have been looking for. Asterisk realtime ldap integration is very flaky. I found this page: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_Directory But there are no clues as to the schema, the queries generated and what exactly it can hold in Ldap. I am also curious to know whether sofia's sip registrations, gateways etc can be kept in LDAP. We are basically developing an extensive plugin based control panel and a Asterisk module is already ready. However, we are writing asterisk .conf files for managing asterisk. We would be quite pleased to develop a FreeSwitch Ldap plugin to manage users,sip gateways, groups, features, etc. Though the XML configuration file is extremely easy to parse and write, pure LDAP integration would be amazing. Any pointers on this would be appreciated. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz http://www.emergen.biz ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@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 mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
