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voicemail to email question
by Sean Devoy 26 Apr '14

26 Apr '14

Hi All, I have just in the last 2 days succeeded in getting FS to use SENDMAIL and send me my voicemails. However, I do mean ME and almost nobody else. It appears that my Microsoft Office 365 mailbox has NO restrictions on emails it will accept. Almost any spoofing is FINE. I think my underlying problem is the network and sendmail setup on my server. (FYI: I am windows guy so this is pretty arcane stuff to me). The email headers of the message I have received look BAD since the system name is "bizfocused.local" and the IP address reported is 127.0.0.1. Here is an excerpt: Received: from BY2PR01MB251.prod.exchangelabs.com (10.242.233.20) by [SNIP] Received: from bizfocused.local (66.241.102.41) by [SNIP] Received: (from root@localhost) by bizfocused.local (8.13.8/8.13.8/Submit) id s3Q0JHkO012279; From: root <root(a)bizfocused.local> To: Undisclosed recipients:; Return-Path: root(a)bizfocused.local Received-SPF: None (: bizfocused.local does not designate permitted sender hosts) So it would seem I have 4 options: 1. Properly configure eth0 and sendmail and add an SPF record to match. While this is the "proper" option, it will likely take several hours of googling and trials. It may in fact be that this is very simple and I am just not aware - I hope someone jumps in on this. 2. Maybe configure SENDMAIL to use my actual mail server? 3. Create a shell script to translate the voicemail to wav and submit a curl job to my server for emailing. Creating the WAV file is the only unknown for me here. I have another script already curl'ing to my server which emails it out clean as a whistle. Anyone help with the tmp file to WAV file conversion? 4. Configure some other mail client and Freeswitch for this task. (BOO HISS) So feel free to jump in with ideas. And as always THANK YOU. sean

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URGENT - users getting [INCOMPATIBLE_DESTINATION] through gateway
by Sean Devoy 25 Apr '14

25 Apr '14

Hi, SOME of my extensions are failing when dialing OUT through my Vitelity Gateway. The log shows : 2014-04-25 13:22:46.039025 [DEBUG] sofia.c:5815 Channel sofia/external_noauth/14108084699 entering state [terminated][488] 2014-04-25 13:22:46.039025 [NOTICE] sofia.c:6659 Hangup sofia/external_noauth/14108084699 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2014-04-25 13:22:46.039025 [DEBUG] switch_channel.c:3187 Send signal sofia/external_noauth/14108084699 [KILL] [SNIP] 2014-04-25 13:22:46.039025 [DEBUG] switch_ivr_originate.c:3657 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] Vitelity says it is my FS that is refusing the call. Please Help. Sean

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Build errors with --enable-core-pgsql-support
by Phil Quesinberry 25 Apr '14

25 Apr '14

Latest git pull of v1.2 stable on CentOS 6.5. Im getting the following errors during make: src/switch_pgsql.c: In function âswitch_pgsql_next_result_timedâ: src/switch_pgsql.c:389: error: âPGRES_SINGLE_TUPLEâ undeclared (first use in this function) src/switch_pgsql.c:389: error: (Each undeclared identifier is reported only once src/switch_pgsql.c:389: error: for each function it appears in.) src/switch_pgsql.c:400: error: âPGRES_COPY_BOTHâ undeclared (first use in this function) src/switch_pgsql.c: In function âswitch_pgsql_handle_exec_string_detailedâ: src/switch_pgsql.c:531: error: âPGRES_SINGLE_TUPLEâ undeclared (first use in this function) make[1]: *** [libfreeswitch_la-switch_pgsql.lo] Error 1 make: *** [all] Error 2 Speaking with a few folks on irc gave me the impression that postgres support for FS isnt quite ready for prime time on CentOS, but I wanted to see if there was something obvious that I was missing. The following relevant packages are installed: postgresql92-9.2.8-1PGDG.rhel6.x86_64 postgresql-8.4.20-1.el6_5.x86_64 postgresql92-odbc-09.02.0100-1PGDG.rhel6.x86_64 postgresql-libs-8.4.20-1.el6_5.x86_64 postgresql92-libs-9.2.8-1PGDG.rhel6.x86_64 postgresql92-server-9.2.8-1PGDG.rhel6.x86_64 postgresql-odbc-08.04.0200-1.el6.x86_64 postgresql-devel-8.4.20-1.el6_5.x86_64 postgresql92-devel-9.2.8-1PGDG.rhel6.x86_64 pgdg-centos92-9.2-6.noarch PATH includes /usr/pgsql-9.2, /usr/pgsql-9.2/bin, /usr/pgsql-9.2/lib, /usr/pgsql-9.2/include, and PKG_CONFIG_PATH is /usr/local/lib/pkgconfig which contains libpq.pc. Attempting the configure with LDFLAGS: ./configure --enable-zrtp --enable-core-pgsql-support LDFLAGS="-L/usr/pgsql-9.2/lib" CPPFLAGS="-I/usr/pgsql-9.2/include" still generates the following errors: src/switch_pgsql.c: In function âswitch_pgsql_next_result_timedâ: src/switch_pgsql.c:391: error: stray â#â in program src/switch_pgsql.c:391: error: âiâ undeclared (first use in this function) src/switch_pgsql.c:391: error: (Each undeclared identifier is reported only once src/switch_pgsql.c:391: error: for each function it appears in.) src/switch_pgsql.c:391: error: expected â;â before âendifâ cc1: warnings being treated as errors src/switch_pgsql.c:387: error: enumeration value âPGRES_TUPLES_OKâ not handled in switch src/switch_pgsql.c:288:1: error: unterminated #ifdef make[1]: *** [libfreeswitch_la-switch_pgsql.lo] Error 1 make: *** [all] Error 2 Is this issue readily resolvable on this platform or are those of us running RHEL/CentOS mostly out of luck? I would be happy to file a Jira on this if it will help. Thanks, - Phil

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The webrtc can not call into freeswitch
by rockxsj 25 Apr '14

25 Apr '14

I use sipml5 call to freeswitch, the most time it work well.But sometimes if too many calls at a time, it did not work. When I capture package on client and server, I found that my client cannot recive package from the server.The below message is the cpu info when I cannot call, and I found the server can not send data to the socket. BUG: soft lockup - CPU#14 stuck for 60s! [freeswitch:10399] CPU 14: Modules linked in: be2iscsi ib_iser rdma_cm ib_cm iw_cm ib_sa ib_mad ib_core ib_addr iscsi_tcp bnx2i cnic ipv6 xfrm_nalgo crypto_api uio cxgb3i libcxgbi cxgb3 8021q libiscsi_tcp libiscsi2 scsi_transport_iscsi2 scsi_transport_iscsi dm_mirror dm_multipath scsi_dh video backlight sbs power_meter hwmon i2c_ec i2c_core dell_wmi wmi button battery asus_acpi acpi_memhotplug ac parport_pc lp parport joydev sr_mod cdrom sg tpm_tis tg3 tpm tpm_bios pcspkr dm_raid45 dm_message dm_region_hash dm_log dm_mod dm_mem_cache ahci libata shpchp megaraid_sas sd_mod scsi_mod ext3 jbd uhci_hcd ohci_hcd ehci_hcd Pid: 10399, comm: freeswitch Not tainted 2.6.18-348.el5 #1 RIP: 0033:[<0000003d26c0d55b>] [<0000003d26c0d55b>] RSP: 002b:00002b87166103d8 EFLAGS: 00000246 RAX: 0000000000000002 RBX: 0000000000000004 RCX: 0000003d26c0e021 RDX: 0000000000000000 RSI: 0000000000000002 RDI: 0000000000000000 RBP: 00002b8716612000 R08: 00002b8716611940 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000000 R13: 0000000000003303 R14: 00002b8716610480 R15: 0000000000000004 FS: 00002b8716611940(0000) GS:ffff810107ba9cc0(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 00002b877542b000 CR3: 00000002aba9c000 CR4: 00000000000006a0 Call Trace: When it send failure,it will re-try,the below picture is the screenshot of capture pacakte. http://yazi.whoknow.me/wp-content/uploads/2014/04/cpuinfo.jpg Somebody can help me? Is there some wrong with webrtc support?

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Compiling Freeswitch for Android?
by GillesToo 25 Apr '14

25 Apr '14

Hello Searching the archives only seems to return information on how to use SIPDroid as an SIP client to connect to a Freeswitch server. Since Android is built on Linux, and Freeswitch has been compiled for compact devices, I was wondering if someone had tried compiling it for Android? I could use a small IVR on a smartphone to handle incoming calls. Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Andr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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v1.2 or v1.4 for production
by mbo 25 Apr '14

25 Apr '14

Hello, I need to install a new production server, usually I always install the lastest available tag. Has Freeswitch version 1.4 with tag v1.4.2 already left beta state and is it as satiable as latest tag v1.2.23 from version 1.2? Thanks for your recommendations Markus

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Siproutes incompatible_destination - solved
by Avi Marcus 25 Apr '14

25 Apr '14

If you've been getting incompatible_destination on your siproutes calls, it seems some underlying carrier can no longer handle type 13 from CNG. If you have a gateway file for siproutes, you can simply add: <param name="supress-cng" value="true"/> (It's also accessible in the advanced options in fusionpbx) They also recommend setting <param name="auto-rtp-bugs" value="clear"/> because of sonus, but I can't find on the wiki what that does. -Avi Marcus

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Send users wav of voicemail via sendmail
by Sean Devoy 25 Apr '14

25 Apr '14

Hi All, Can someone explain (or point to documentation) to get voice mail messages emailed to the end user. I have been over and over the mod_voicemail wiki page at https://wiki.freeswitch.org/wiki/Mod_voicemail. It is a nice reference page as it lists lots and lots of voicemail variables and parameters. However, I cannot determine what the minimum rwquirements are. Here is what I have so far: In /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml I have: <param name="mailer-app" value="sendmail"/> <param name="mailer-app-args" value="-t"/> <param name="dump-cores" value="yes"/> In the user directory entry I have: <param name="vm-mailto" value="sdevoy(a)bizfocused.com"/> <param name="vm-mailfrom" value="voicemail(a)bizfocused.com"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true" /> I can send emails from SSH when logged in as root. mail -s "Test mail from testuser1" sdevoy(a)bizfocused.com Hello this is the test mail . I can't find any errors in the console log, but no email gets sent. Have I missed something about creating a freeswitch user in sendmail??? Thanks, Sean

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Bypass media with callback
by FSX 25 Apr '14

25 Apr '14

I'm trying to use GV callback feature in FS and having trouble with making it work in bypass media mode. Incoming calls are working well. Call comes to public context, then it's transferred to local domain context. Then it goes via: <action application="set" data="bypass_media=true"/> <action application="bridge" data="..."/> RTP streams directly between remote server (caller party) and local SIP client, as it should be by design. Calls have better audio quality and don't use recourses of local SIP server. Now, I try to do the same when I use GV callback feature. Local dial plan (as recommended in Google Voice API <https://wiki.freeswitch.org/wiki/Google_Voice_API> ) sets hash with current "uuid" and runs system call to make callback. Then in public context dial plan call is accepted, using "intercept" application with saved "uuid". Exactly as it's recommended. It's working well, but I can't get bypass mode whatever I try to do. * I tried to set "bypass_media=true" before run system call to make callback * I tried to set it before executing "intercept" application * I tried to apply global setting to make sure that FS is always using bypass mode ("inbound-proxy-media" set to "true" in "sip_profiles\internal.xml") Nothing helps so far... FS always ignore those settings and proxies RTP stream. How to make bypass media mode work in this scenario? Thank you in advance -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Bypass-media-with-callback-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Dual WAN router suggestions
by Lloyd Aloysius 25 Apr '14

25 Apr '14

Hello All This is not a freeswitch question. I am sorry. But looking for a Dual WAN router suggestion? I try the pfSense and find a huge limitation, when WAN interface down. It keeps the firewall states and phones never able to connect through the second wan, unless someone clear the firewall rules manually. Any suggestions appreciated Thank you LLoyd

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