October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Digest Authentication
by Masood Ahmed 14 Oct '11

14 Oct '11

Dear , I had successfully installed and configured radiusclient. I do not know that what next lines should i do to get users registered on my freeswitch using digest authentication with my radius server, Like I have my radsius server and billing system. Now what lines should i add in sofit conf or in dialplans that every time client hits server request should be sent to radius server and what lines should i add to register them if i get proper return code and how to do that. Your Help is needed -----

1 0

From param
by Chad Vogel 14 Oct '11

14 Oct '11

Does anyone know of a way to add a param to the from header in a dialplan? I would like to do something like a isup-oli=6 (From: <sip:restricted@10.100.63.137;OTG=777;isup-oli=6>) Chad

2 3

Re: [Freeswitch-users] Call/Transaction Does Not Exist - RESOLVED
by Joe Flemmings 13 Oct '11

13 Oct '11

This has been resolved by upgrading to the latest version Thank You On Thu, Oct 13, 2011 at 2:39 PM, Joe Flemmings <joe.jflemmings(a)gmail.com>wrote: > Actually just updated to the latest Freeswitch and its working. > > If i don't write back then consider this to be working. > > Thank you > > > On Thu, Oct 13, 2011 at 2:21 PM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > >> 4 things to try in order of likelihood to help but you should do all 4 >> regardless. >> >> 1) edit your sofia profile and add <param name="NDLB-force-rport" >> value="true"/> >> 2) Update your polycom to latest firmware >> 3) Update to latest FreeSWITCH >> 4) make sure your network router does not have any SIP ALG enabled. >> >> >> On Wed, Oct 12, 2011 at 8:34 PM, Joe Flemmings <joe.jflemmings(a)gmail.com> >> wrote: >> > Thanks for the help Anthony, >> > >> > Bellow is the pastebin URL >> > >> > http://pastebin.freeswitch.org/17518 >> > >> > Joe >> > >> > On Wed, Oct 12, 2011 at 8:51 AM, Anthony Minessale >> > <anthony.minessale(a)gmail.com> wrote: >> >> >> >> collect a full trace and put it on pastebin.freeswitch.org >> >> >> >> console loglevel debug (/log debug on fs_cli) >> >> sofia global siptrace on >> >> >> >> >> >> >> >> >> >> On Tue, Oct 11, 2011 at 4:29 PM, Joe Flemmings < >> joe.jflemmings(a)gmail.com> >> >> wrote: >> >> > When i call from one extension and hangup, the other call does not >> >> > hangup >> >> > and continues ringing. Please see log bellow. >> >> > >> >> > Any ideas? >> >> > >> >> > >> >> > >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > INVITE sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> >> > <sip:213@10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: >> >> > <sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, >> CANCEL, >> >> > OPTIONS, INFO, MES >> >> > SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: >> >> > en..Supported: >> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: >> >> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- >> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN >> IP4 >> >> > 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18 >> >> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 >> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 >> >> > telephone-event/8000.. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 100 Trying..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> >> > <sip:213@10.10.78.36;user=phone> >> >> > ..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 1 >> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba >> 2011-05-23 >> >> > 22-51-43 -0500..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@6 >> >> > 6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 1 INVITE..User-Agent: >> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 >> >> > -0500..Accept: >> >> > appl >> >> > ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, >> >> > UPDATE, >> >> > INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, >> >> > precondition, path, replaces..Allow-Events: talk, hold, prese >> >> > nce, dialog, line-seize, call-info, sla, >> include-session-description, >> >> > presence.winfo, message-summary, refer..Proxy-Authenticate: Digest >> >> > realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac >> >> > ", algorithm=MD5, qop="auth"..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > ACK sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <si >> >> > p:213@10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 >> ACK..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: >> >> > <sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, >> CANCEL, >> >> > OPTION >> >> > S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, >> REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: >> >> > en..Max-Forwards: 70..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > INVITE sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom >> One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> >> > <sip:213@10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: >> >> > <sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, >> CANCEL, >> >> > OPTIONS, INFO, MES >> >> > SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: >> >> > en..Supported: >> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz >> >> > ation: Digest username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth, >> >> > cnonce="KnfhF00Oi0GfJCi", nc=00000001, >> >> > uri="sip:213@10.10.78.36:5060;user=phone", >> >> > response="82a75edb7ba97bfe7dea585684701d0c", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- >> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone >> >> > ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 >> 8 >> >> > 18 >> >> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 >> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 >> >> > telephone-event/8000.. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 100 Trying..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom >> One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> >> > <sip:213@10.10.78.36;user=phone> >> >> > ..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 >> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba >> 2011-05-23 >> >> > 22-51-43 -0500..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52783 >> >> > INVITE sip:pbxjker_213@10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP >> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: >> 69..From: >> >> > "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81 >> >> > rv8Fp..To: <sip:pbxjker_213@10.10.78.24:52783>..Call-ID: >> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact: >> >> > <sip:mod_sofia@10.10.78.36:5060>..User-Agent: >> FreeSWITCH-mod_sofia/1.0. >> >> > head-git-1086cba 2011-05-23 22-51-43 -0500..Allow: INVITE, ACK, >> BYE, >> >> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, >> >> > PUBLISH, >> >> > SUBSCRIBE..Supported: timer, precondition, path, replaces.. >> >> > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, >> >> > sla, >> >> > include-session-description, presence.winfo, message-summary, >> >> > refer..Content-Type: application/sdp..Content-Disposition: session.. >> >> > Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID: >> >> > "Polycom One" >> >> > >> >> > <sip:pbxjker_2564@10.10.78.36 >> >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH >> >> > 1318333156 1318333157 IN IP4 >> >> > 10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio >> 29124 >> >> > RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98 >> >> > bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320 >> >> > 00..a=fmtp:18 annexb=no..a=rtpmap:101 >> telephone-event/8000..a=fmtp:101 >> >> > 0-16..a=ptime:20.. >> >> > # >> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060 >> >> > SIP/2.0 100 Trying..Via: SIP/2.0/UDP >> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two" >> >> > <sip:pbxjker_213@99.65.2 >> >> > 17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID: >> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: >> >> > <sip:pbxjker_213@10.10.78.24:52783>..User-Agent: >> >> > PolycomSoundPointIP-SPIP_320-UA/3.2 >> >> > .2.0477..Accept-Language: en..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060 >> >> > SIP/2.0 180 Ringing..Via: SIP/2.0/UDP >> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two" >> >> > <sip:pbxjker_213@99.65. >> >> > 217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 >> INVITE..Call-ID: >> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: >> >> > <sip:pbxjker_213@10.10.78.24:52783>..User-Agent: >> >> > PolycomSoundPointIP-SPIP_320-UA/3. >> >> > 2.2.0477..Allow-Events: talk,hold,conference..Accept-Language: >> >> > en..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 180 Ringing..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom >> One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> >> > <sip:213@10.10.78.36;user=phone >> >> > >;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 INVITE..Contact: >> >> > <sip:213@10.10.78.36:5060;transport=udp>..User-Agent: >> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2 >> >> > 3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE, >> >> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, >> >> > PUBLISH, >> >> > SUBSCRIBE..Supported: timer, precondition, path, replaces..A >> >> > llow-Events: talk, hold, presence, dialog, line-seize, call-info, >> sla, >> >> > include-session-description, presence.winfo, message-summary, >> >> > refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker >> >> > _213>;party=calling;privacy=off;screen=no.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 >> >> > 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: >> >> > <sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, >> >> > OPTIONS, >> >> > INFO, MESSAGE, SUBSCR >> >> > IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: >> Digest >> >> > username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-462 >> >> > 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, >> >> > uri="sip:213@10.10.78.36:5060;user=phone", >> >> > response="d47848babd8c396d8da15145503875d0", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Length: 0.. >> >> > .. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060 >> >> > SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> <sip:213@66.11 >> >> > 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 >> CANCEL..Content-Length: >> >> > 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 >> >> > 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: >> >> > <sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, >> >> > OPTIONS, >> >> > INFO, MESSAGE, SUBSCR >> >> > IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: >> Digest >> >> > username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-462 >> >> > 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, >> >> > uri="sip:213@10.10.78.36:5060;user=phone", >> >> > response="d47848babd8c396d8da15145503875d0", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Length: 0.. >> >> > .. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060 >> >> > SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> <sip:213@66.11 >> >> > 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 >> CANCEL..Content-Length: >> >> > 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 >> >> > 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: >> >> > <sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, >> >> > OPTIONS, >> >> > INFO, MESSAGE, SUBSCR >> >> > IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: >> Digest >> >> > username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-462 >> >> > 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, >> >> > uri="sip:213@10.10.78.36:5060;user=phone", >> >> > response="d47848babd8c396d8da15145503875d0", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Length: 0.. >> >> > .. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060 >> >> > SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: >> <sip:213@66.11 >> >> > 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 >> CANCEL..Content-Length: >> >> > 0.... >> >> > # >> >> > >> >> > >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users(a)lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Anthony Minessale II >> >> >> >> FreeSWITCH http://www.freeswitch.org/ >> >> ClueCon http://www.cluecon.com/ >> >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> >> AIM: anthm >> >> MSN:anthony_minessale@hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> >> >> FreeSWITCH Developer Conference >> >> sip:888@conference.freeswitch.org >> >> googletalk:conf+888@conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users(a)lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> googletalk:conf+888@conference.freeswitch.org >> pstn:+19193869900 >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >

1 0

TLS problems
by Ben Naylor 13 Oct '11

13 Oct '11

Hi all I was wondering if anyone can help me with a problem I am having with using TLS SIP when NAT routers are involved? The scenario is as follows - SIP UA --- > NAT firewall --- > Internet --- > Freeswitch --- > Internet --- > NAT Firewall --- > SIP UA When I use UDP as the transport protocol, everything works fine. This is down to (correct me if I am wrong) the NAT router being able to inspect the SIP traffic, and changing the SIP INVITE message so that the public IP is used for the RTP stream. This is seen in the following ( 1.2.3.4 = SIP UA, 50.50.50.50 = Freeswitch server) - recv 1042 bytes from udp/[1.2.3.4]:34019 at 08:40:24.302496: ------------------------------------------------------------------------ INVITE sip:1004@50.50.50.50 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:34019;branch=z9hG4bKD8xhgo0smbsOQMUy;rport Contact: <sip:1002@1.2.3.4:34019> Max-Forwards: 70 From: "1002" <sip:1002@50.50.50.50>;tag=A6E182EBB96E21919A44C50399F1BF27 Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY Supported: replaces, path User-Agent: Acrobits Softphone Business/1.8.8 To: <sip:1004@50.50.50.50> Content-Type: application/sdp Call-ID: 2926496CCA1C6189C90A52D821B06B5E9DBA2B32 CSeq: 1 INVITE Proxy-Authorization: Digest username="1002",realm="50.50.50.50",algorithm=MD5,uri="sip:1004@50.50.50.50" ,nonce="09195369-6474-4310-b3a6-60e7d9c780f1",qop=auth,cnonce="1b774f833fd01 b5b29954c2ddc2b5457",nc=00000002,response="07bda5b4c5fa5e17f653a3e9c7a2f05e" Content-Length: 241 v=0 o=- 10805 6353 IN IP4 1.2.3.4 < --- Public IP address used so that the other party knows to send the RTP stream to this address. s=xynwwjx c=IN IP4 1.2.3.4 t=0 0 m=audio 34962 RTP/AVP 102 3 0 8 9 101 a=rtpmap:101 telephone-event/8000 a=rtpmap:102 ILBC/8000 a=fmtp:102 mode=30 a=fmtp:101 0-15 a=ptime:30 a=sendrecv This was obtained from the Siptrace on the Freeswitch server. The Freeswitch server then forwards these details to the other party. The problem occurs when TLS is used, as the NAT firewalls are unable to inspect the SIP packets. Here is another SIPtrace on the Freeswitch server for TLS example (again I have changed the public IPs to the same as the above example) - recv 1048 bytes from tls/[1.2.3.4]:64403 at 16:00:42.271063: ------------------------------------------------------------------------ INVITE sip:1002@50.50.50.50 SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:64403;branch=z9hG4bKz6qAkQj3XsYuJ5Os;rport Contact: <sip:1004@1.2.3.4:64403;transport=tls> Max-Forwards: 70 From: "Ben" <sip:1004@50.50.50.50>;tag=F28D08D5E039D8FE021D08022A72F982 Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY Supported: replaces, path User-Agent: Acrobits Softphone Business/1.8.8 To: <sip:1002@50.50.50.50> Content-Type: application/sdp Call-ID: 32166133760BDDAD823725B563F6A1B3989090D6 CSeq: 1 INVITE Authorization: Digest username="1004",realm="50.50.50.50",algorithm=MD5,uri="sip:1002@50.50.50.50" ,nonce="5af9829e-716d-4472-924f-84eb41d9e78e",qop=auth,cnonce="d641702f981e1 0f3a41e074061bdb5b3",nc=00000003,response="a9f577ac37f8f367b2f839a58d43e1e0" Content-Length: 240 v=0 o=- 71766 47529 IN IP4 172.16.234.176 < --- Using Private IP for the RTP details s=fscmvnc c=IN IP4 172.16.234.176 t=0 0 m=audio 64628 RTP/AVP 3 102 101 a=rtpmap:101 telephone-event/8000 a=rtpmap:102 ILBC/8000 a=fmtp:102 mode=30 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ As you can see, the SIP UAs are going to try and use a non-existent private IP address to send the RTP stream to. I can also confirm that the UAs are trying to do this, by looking at the SIP logs built in to the SIP UA client software. My (Freeswitch related) question is, am I able to set something in Freeswitch to change the details to be the public IP of the firewall? The SIP messages include the correct IP, can the Freeswitch server use this address and replace the Private IP with this one? I have heard about TLS proxies/STUn servers etc etc, but would much rather use Freeswitch to do this. Any suggestions are welcome, also please let me know if I have misunderstood any of the concepts behind SIP/VOIP and firewalls - I am rather new to the VOIP game! Kind regards Ben Ben Naylor Network Support Engineer <http://www.sirran.com/> Description: Description: cid:image001.png@01CC4312.3962A460 T: +44 (0)207 096 1648` M: +44 (0)7924 349 113 F: +44 (0)207 150 595 E: <mailto:bnaylor@sirran.com> bnaylor(a)sirran.com W: <http://www.sirran.com/> www.sirran.com A: SiRRAN Communications Ltd, The Hawk Business Park, Easingwold, YO61 3FE Registered Office: SiRRAN Communications Ltd,. 6-9 Trinity Street Dublin 2, Ireland _____________________________________________________________ The information contained in this e-mail and any attached documents may be privileged, confidential and protected from disclosure. If you are not the intended recipient you may not read, copy, distribute or use this information. If you have received this communication in error, please notify the sender immediately by replying to this message and then delete it from your system.

2 6

prepaid solutions
by reza a 13 Oct '11

13 Oct '11

Freeswitch users, Does anyone use freeswitch for prepaid applications? Do you have a walk through on setting up an environment like this? I'd like to play with freeswitch a bit, I know there alot of modules out there but I'm wondering what's the best cohesive set of modules/admin tools to use for a prepaid style applications. Thanks a lot

5 9

Re: [Freeswitch-users] continue_on_fail
by Madovsky 13 Oct '11

13 Oct '11

did you resolve it ? after some test I'm facing the same problem if for example the callee decides to send a user busy so FS tries with the second bridge and makes a second call which can be interpreted as harrassment ;) ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Monday, October 10, 2011 11:28 PM Subject: Re: [Freeswitch-users] continue_on_fail maybe modify continue_on_fail as a leg var on the first bridge ----- Original Message ----- From: Boris Kovalenko To: FreeSWITCH Users Help Sent: Monday, October 10, 2011 11:12 PM Subject: Re: [Freeswitch-users] continue_on_fail Hello! No, the question was how to FAILOVER :) I need to stop processing dialstring if user at carrier1 is busy. the question was about mod_lcr. it returns something like: sofia/gateway/carrier1/14105780999|sofia/gateway/ carrier2/14105780999|sofia/gateway/ carrier3/14105780999 The question was how to make it when the first returns "user_busy" to not failover and try the other ones.-Avi On Mon, Oct 10, 2011 at 6:40 PM, Michael Collins <msc(a)freeswitch.org> wrote: Well, now we're talking about something more than just continuing_on_fail. We'll need to look at the actual dialstring to answer any more specific questions. -MC On Mon, Oct 10, 2011 at 8:54 AM, Avi Marcus <avi(a)avimarcus.net> wrote: It seems from testing, and I was told on channel: continue_on_fail is per bridge string, not per leg. So it doesn't actually help with the question.. Unless I'm mistaken. -Avi On Mon, Oct 10, 2011 at 5:12 PM, Michael Collins <msc(a)freeswitch.org> wrote: You will need to specify the error conditions on which to continue. See the example here: http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail -MC P.S. - Check out the "enumeration" column on this page to get an idea of the kinds of failures you may not be aware of: http://wiki.freeswitch.org/wiki/Hangup_causes On Sat, Oct 8, 2011 at 2:54 PM, Avi Marcus <avi(a)avimarcus.net> wrote: I've been wondering how to do this. When I asked in channel, I was basically told this couldn't be done in a single bridge string... I hope that's not actually the case. -Avi On Fri, Oct 7, 2011 at 4:40 PM, Boris Kovalenko <boris(a)tagnet.ru> wrote: Hello! Have read docs but misundertood :( I use mod_lcr which return two or more gateways to the same destination. And when the user is busy with the first gateway the call continues with the second and so on. How may I to not continue gateways if user is busy? -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Boris ---------------------------------------------------------------------------- FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

3 7

Is Bgapi threadsafe?
by John Rose 13 Oct '11

13 Oct '11

Does anyone know if calling Bgapi on an ESL connection is threadsafe? Im calling it from ManagedESL via C# and have a lock around it but would like to remove the lock if possible John

3 6

ESL Ruby - sched_api / task id issue
by Claudio Álvarez 13 Oct '11

13 Oct '11

Hi all, I'm currently experimenting with sched_api to schedule a ringback sound from a Ruby script. My code looks like this: ringback_result = con.bgapi('sched_api +8 none uuid_broadcast ' + leg1_uuid + ' /var/www/mi/audio/ringback.mp3 aleg') puts "Serialized result:" puts ringback_result.serialize() When I run the script, my output looks like: Serialized result: Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK%20Job-UUID%3A%20cec047cc-97d7-4320-811a-e15e454843f9 Job-UUID: cec047cc-97d7-4320-811a-e15e454843f9 The problem here is that I'm not getting the scheduled task ID returned by sched_api I expect. I need the task ID so as to run sched_del should I need to abort the task. Trying the same api_sched command on fs_cli I get an output confirming that the task has been added, together with the assigned task ID. sched_api +8 none uuid_broadcast 41b482a9-8595-47d4-a3fd-3c6a0d32e942 /var/www/mi/audio/ringback.mp3 aleg +OK Added: 34 Can anyone help me out with this one? Thanks! -- Claudio Álvarez, Ph.D. Arquitecto de Sistemas / Systems Architect Medularis S.p.A. - ClickFono Tel: 56 2 934 40 74 Cel: 56 9 9079 4951 claudio(a)clickfono.com

2 2

Digest Authentcation with radius server
by Masood Ahmed 13 Oct '11

13 Oct '11

Dear , I had successfully installed and configured radiusclient. I do not know that what next lines should i do to get users registered on my freeswitch using digest authentication with my radius server, Like I have my radsius server and billing system. Now what lines should i add in sofit conf or in dialplans that every time client hits server request should be sent to radius server and what lines should i add to register them if i get proper return code and how to do that. -- *Regards, Masood Ahmed *

1 0

Audio is not flowing , log is mod_sofia.c:1117 Asynchronous PTIME not supported, changing our end from 30 to 20
by 占平陈 13 Oct '11

13 Oct '11

I use FS ,codec is ulaw ,when other softswitch call in,the audio is not flowing and distortion. the log is : http://pastebin.freeswitch.org/17517 I change my codec ulaw packet time to 30 ms (PCMU@30i) in external.xml,but the audio is not flowing also. I add <param name="rtp-autofix-timing" value="false"/> to external.xml, it no useful. Someone can tell me ,Why is it?

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