October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to troubleshoot "503 Maximum Calls In Progress"?
by Chris Fowler 05 Jan '12

05 Jan '12

Hi, On our Production FreeSWITCH box: FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500) UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, 751 microseconds 12519 session(s) since startup 2 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 At 14:30:37 a reloadxml command was issued. At 14:30:43 the box started rejecting all calls with 503 error. At 14:38 it started working again. Looking at the code I see three conditions can trigger this behavior: if (sess_count >= sess_max || !sofia_test_pflag(profile, PFLAG_RUNNING) || !switch_core_ready()) { nua_respond(nh, 503, "Maximum Calls In Progress", SIPTAG_RETRY_AFTER_STR("300"), TAG_END()); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n"); goto done; } Logs show it wasn't lack of available sessions. How can sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and busy the system? Thoughts? Thx, Chris.

3 2

LUA transfer result:0
by Anestis Mavro 24 Dec '11

24 Dec '11

Hi, I have tried two different ways of transfer in LUA: session:transfer(number,"XML","default") And Session:execute("transfer","12345 XML default") And I can't make it working. I get "switch_cpp.cpp:797 transfer result: 0" The same transfer from within the dialplan works fine The call is not answered before the transfer. Does anybody have an idea? Thanks Anestis

4 4

Feedback
by Chad Vogel 23 Dec '11

23 Dec '11

Hello, I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users. Here are some of the changes we have made: * We converted FS core to C++ and having it compiling to .NET 4.0 CLI * FS now runs inside its own namespace * We replace FS_CLI with a PowerShell shell app * Supports SQL Server support via the Native SQL Server Client API * Support for windows clustering (up to 32 node active/active cluster) * Ties more directly into the Win32 API and has less reliance on open source Libs * Replaces OpenSSL sockets with Windows encrypted sockets * Added windows performance monitoring * No longer need to use mod_managed for managed modules * Memory management relies on .Net garbage collection * Added support for Microsoft Speech * VoiceXML 3.0 and SCXML support * No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages) * Fixed RTP clock timing issues in virtualized environments * New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

18 27

FreeSWITCH cluster with IM/Presence
by Even André Fiskvik 22 Dec '11

22 Dec '11

Hi. I currently have a project where I'm researching how to establish a clustered and scalable platform with: - registration - presence/blf between eyebeam/bria and Linksys SPA962 clients - IM between eyebeam/bria clients. - calling between the clients The first scenario I want to test is with two FreeSWITCH servers and shared ODBC db running in a "cluster". How the registrations and calls etc. are balanced between them is not decided yet, but I'm thinking that a UltraMonkey setup or DNS SRV records could to the trick here? The main challenge in this setup is sharing presence, and doing call-setup and IM between two FreeSWITCH servers (which can be scaled by adding n-nodes). I'm interested in user's that can confirm wether this solution is possible or not with FreeSWITCH and it's current state, and what it would take to make something like this work. If there's any part of FS that is lacking some implementation to make such a scenario work, we are interested in providing the funding to support those features. I've seen that the core db has added support for ODBC, but I'm not sure wether this is something that would be helpful in this scenario or not. This is only conceptual right now, so I'm interested in paying someone that could help me configure the two FS servers to support this scenario on an hourly basis to get a quick proof-of-concept up and running. The proof-of-concept has to be completed by March 19th, after which we will evaluate what technologies and setup we want to commit to. Some other variations I'm concidering is Opensips/Kamailio -> Asterisk, Opensips/Kamailio -> FreeSWITCH, Opensips/Kamailio -> Asterisk and Openfire + Asterisk-im plugin for IM/presence. I'm also open to other suggestions, but I'd like this topic to stay on how FreeSWITCH could facilitate such a platform. Best regards, Even André Fiskvik CTO - Oyatel AS

2 1

Linksys PAP2-T and echo
by Boris Kovalenko 21 Dec '11

21 Dec '11

Hello! I'm wondering if it is possible to solve echo with Linksys PAP2-T devices. Does anybody here uses it? I read many articles and still can't understand how to eleminate echo :( I read http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found that Asterisk has its own EC. Does Freeswitch has it too? I found this article http://docs.freeswitch.org/echo_can_page.html but can't understand - has Freeswitch EC or not? If yes - how to turn it on and off? -- Regards, Boris

5 13

Call/Transaction Does Not Exist
by Joe Flemmings 21 Dec '11

21 Dec '11

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.... #

3 6

Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

5 8

Outbound calling with recorded message.
by Dave 08 Dec '11

08 Dec '11

Hi all, I have been using freeSWITCH for a while for inbound calls in which a person registers for a seminar via IVR. I simply use the DID, CID number and name to identify the person and put them in the Database. That part of the application is working wonderfully. The issue I'm presented with now is that we need automate making calls to a few of these registrants, after each event, whos caller_id_name comes in as "Unknown" or "Wireless Caller", Play a recording to let them know we need the name, and that they can record the information right then, or call the office. I am limited to using a Windows 7 machine. What would be the best tool to use to automate these calls with the use of an IVR? Dave Goodwin

3 3

Need help
by Sam 08 Dec '11

08 Dec '11

Hi, I need some help with an idea I am trying to implement with Freeswitch. I am trying to work with a partner company which will forward a sip call to my freeswitch server and my freeswitch server is suppose to playback annoucements to the caller and then once they are finished listening they should be forwarded back to the partner company so that the call can continue without my freeswitch being in the call path. So does any one have any suggestions/ideas on the best way to implement this?

4 4

get cepstral volume lower
by Madovsky 05 Dec '11

05 Dec '11

I'm trying the SSML example from mod_cepstral wiki <action application="speak" data="cepstral|david|<prosody volume=13>Thank you for holding</prosody>"/> but the voice doesn't say the sentence but says something like "slash blablabla...". Any idea why I can't embed SSML tag in the cepstral sentence ? Thanks

3 3