- FreeSWITCH-users - FreeSWITCH Mailing Lists

FS behind NAT encounter no audio
by Jimmy Chang 28 Mar '13

28 Mar '13

Hi, I could make a call within NAT without any problem. When I made a call from internet, I got no audio. I have read the document http://wiki.freeswitch.org/wiki/NAT_Traversal and set the parameters. vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.server.ip"/> internal.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> Any advice? Thanks. Jimmy

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Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
by Siri MM 28 Mar '13

28 Mar '13

Hi All, I am trying a simple setup as follows: * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered * Open ACL When I try to make a call from X-Lite to SNOM phone, i get a CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the configurations are at: http://pastebin.freeswitch.org/20722 Where am I going wrong?

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How is -ncwait supposed to work?
by Nathan Neulinger 27 Mar '13

27 Mar '13

I normally start with -nc, but have periodically tried using -ncwait, and what I find is that freeswitch terminates after it's fully up and running. I can diagnose further, but my understanding was that -ncwait was supposed to be equivalent to -nc, except it wouldn't fork until freeswitch was fully ready for fs_cli connections. Is that not correct? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Re: [Freeswitch-users] IP authentication
by Andre 27 Mar '13

27 Mar '13

HI, How do I set my Freeswitch up to take either username and password or Ip Authentication. I looked at this page http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/003067.htm l But I didn't get it to work, then I found this page http://wiki.freeswitch.org/wiki/Acl so now i'm confused and I ended up breaking my dialing plan so now I'm starting over again. My goal is to be able to IP authentication or username or password. If I can only have one I'll take IP auth. Any help is great. I'm new to Freeswitch so I will need step by step instructions if at all possible. Thanks Andre

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fax goes out at 9600 but not at 14400
by Luis Daniel Lucio Quiroz 27 Mar '13

27 Mar '13

Hello Look 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (49) The call dropped prematurely. 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:505 Remote station id: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-03-26 19:12:42.602751 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== Bug if transfer rate is 9600 it goes for good. Currently Im passing these variables, but seems it is not working. 2013-03-26 19:12:08.632735 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [disable-v17]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_v17_disabled]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_disable_v17]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_transfer_rate]=[9600] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [absolute_codec_string]=[PCMU] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [origination_caller_id_name]=[anonymous] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [origination_caller_id_number]=[16138007358] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_enable_t38]=[false] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_attempts]=[32] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_limit]=[32] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_retry_sleep]=[180] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_verbose]=[true] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_use_ecm]=[off] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [fax_enable_t38_request]=[false] 2013-03-26 19:12:08.632735 [DEBUG] switch_event.c:1569 Parsing variable [execute_on_fax_success]=[system /usr/local/bin/email2fax.unqueue /var/spool/push.fax/1981467.bl4Q53kE2n(a)everardo.okay.com.mx test.pdf.tiff.to.16136997792(a)faxofaxo.com.job] Does anyone have an idea? LD

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Call UDP -> TLS
by Chusov Alexsander 27 Mar '13

27 Mar '13

Hello all, I'm trying to deploy FreeSWITCH as a back-end for Opensips ( http://wiki.freeswitch.org/wiki/Opensips ). TLS -> Opensips -> UDP -> FreeSWITCH TLS work fine end point is registered. But when call phone FreeSWITCH send invite use TLS instead of UDP. Register exsample: 172.20.0.24 - opensips 172.20.0.22 - freeswitch 172.20.0.20 - phone REGISTER sip:172.20.0.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK3641.9fabd823.0;i=15 Via: SIP/2.0/TLS 172.20.0.20:5060;received=172.20.0.20;branch=z9hG4bK1590064540;rport=47050;alias From: <sip:1001@172.20.0.24:5061>;tag=1518243149 To: <sip:1001@172.20.0.24:5061> Call-ID: 1616382919-5060-1(a)BHC.CA.A.CA CSeq: 2505 REGISTER Contact: <sip:1001@172.20.0.20:5060;transport=tls>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B823DB6B2>" Authorization: Digest username="1001", realm="172.20.0.24", nonce="102dacd4-959f-11e2-8317-67a135a6f66b", uri="sip:172.20.0.24:5061", response="8f80bc7f8fb5fe5a895a5b39f9b5cde6", algorithm=MD5, cnonce="12386720", qop=auth, nc=00000005 Max-Forwards: 30 User-Agent: Grandstream GXP1405 1.0.5.10 Supported: path Expires: 300 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Path: <sip:172.20.0.24;r2=on;lr;received=sip:172.20.0.20:47050> X-AUTH-IP: 172.20.0.20 Call-ID: 1616382919-5060-1(a)BHC.CA.A.CA User: 1001(a)172.20.0.22 Contact: "user" <sip:1001@172.20.0.20:5060;transport=tls;fs_path=%3Csip%3A172.20.0.24%3Br2%3Don%3Blr%3Breceived%3Dsip%3A172.20.0.20%3A47050%3E> Agent: Grandstream GXP1405 1.0.5.10 Status: Registered(TLS)(unknown) EXP(2013-03-26 01:01:18) EXPSECS(309) Host: 172.20.0.22 IP: 172.20.0.24 Port: 5060 Auth-User: 1001 Auth-Realm: 172.20.0.24 MWI-Account: 1001(a)172.20.0.22 sofia/internal/sip:1001@172.20.0.20:5060;transport=tls;fs_path=%3Csip%3A172.20.0.24%3Br2%3Don%3Blr%3Breceived%3Dsip%3A172.20.0.20%3A47050%3E I'm not familiar with C + + to test made small changes I understand it's not right but it works in src/mod/endpoints/mod_sofia/sofia_glue.c 1233, 1348 change } else if (!strncasecmp(str, "tls", 3)) { return SOFIA_TRANSPORT_TCP_TLS; } to } else if (!strncasecmp(str, "tls", 3)) { return SOFIA_TRANSPORT_UDP; } Can anyone tell how to configure FreeSWITCH for normal UDP->TLS TLS->UDP work. Or maybe I'm doing something wrong? Sorry for my english

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It's been almost 3 years and things have changed...
by Usama Zaidi 27 Mar '13

27 Mar '13

Hey, Count me in. Regards. -youjelly -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, March 27, 2013 7:58 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 292 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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FreeBSD thread issues?
by Tim St. Pierre 27 Mar '13

27 Mar '13

Hello, I have a moderately sized production environment where I'm running 1.2 stable on FreeBSD 8.2 After either about 60,000 calls, or about a week of operation, I run a very high risk of getting into some sort of state where FS can't create any more threads. Lately, I'm seeing instances where Sofia stops processing requests for a minute or so, and channels will get stuck (you can't kill them with uuid_kill). I have opened JIRA tickets on these issues, but the response I got was that it looks like another FreeBSD specific threading issue, and it won't be fixed any time soon. I have about 350 SIP devices registered, and usually don't see more than 50 sessions at a time, yet I'm having a very hard time running a stable production system. Are there others out there using FreeBSD, and if so, are you having the same problems? I'm doing a test install with CentOS to see if it solves my probles, but I'm not nearly as familiar with it and would rather stay with BSD if I can. Thanks for the input. -Tim

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Outbound calls from CLI
by Amit Kumar 27 Mar '13

27 Mar '13

I am trying to call my Cellphone using the attached PSTN number from the FXO. This is the command I am trying originate sofia/gateway/outbound/09xxxxxxxx4 1002 This does not seem to work. However, if I use a software SIP phone(Zoiper), and use the account 1001 to call the same cellphone number, it works. So something if definitely wrong in the command I am using, but I can't pinpoint the issue.

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Execute extension after rxfax
by Eugene Prokopiev 27 Mar '13

27 Mar '13

Hi, Is it posible to execute some extension after rxfax? I tried to use: <action application="set" data="api_hangup_hook=execute_extension fax-report-$1 XML common"/> but I got only: 2013-03-27 15:32:29.156525 [DEBUG] switch_core_state_machine.c:608 Hangup Command with no Session execute_extension(fax-report-3912 XML common): INVALID COMMAND! Is it possible with XML only and without external lua script file? -- Regards, Eugene Prokopiev

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