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fs_path question
by Andre DeMattia 25 Feb '15

25 Feb '15

Hello, I’m using fs_path to a proxy server with OpenSIPs. When I add ;[fs_path=sip:ip:5060] if I don’t add the [] brackets I get 403 errors. When I added the brackets [], the call works however the CDR’s are now having odd results. <origination> <origination_caller_profile> <callee_id_number and <destination_number are both 5060]. It’s like a bracket is missing. I’ve tried to change the brackets to \” and the results are 5060” Could this be a bug or am I doing something wrong? Andre

3 2

invalidating entries in http_cache?
by Neil Patel 25 Feb '15

25 Feb '15

Hi All, We are using mod_http_cache to store remote media for playback over IVR (lua). Once in a while the file at source gets updated/overwritten, so we want FS to be able to detect an invalid cache entry and re-download from the given URL. Is there a way to dirty/clear a single http_cache entry? I don't believe there is a checksum mechanism, but that would do the trick. Is there a way to force-download a URL? Neither http_pre_fetch nor http_get does this. I don't see any way in the docs, so am considering managing this independently and manually removing files from the http_cache folder when we overwrite a source file. Thanks, Neil

3 2

accepted level for parameters on freeswitch
by Achintha 25 Feb '15

25 Feb '15

hi we are experiencing voice quality issue with one of our vpn connection.please let me know what are the accepted level for bellow cdr parameters. rtp_audio_in_jitter_min_variance rtp_audio_in_jitter_max_variance rtp_audio_in_jitter_loss_rate rtp_audio_in_jitter_burst_rate rtp_audio_in_quality_percentage -- Best Regards.. Achintha Udukumbura

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purpose of realm parameter in the gateway configuration
by Sergey Safarov 25 Feb '15

25 Feb '15

Can anyone help me understand the purpose "realm" parameter in the gateway configuration if all of its functions are implemented parameter "from-domain"

1 0

Where to go from here?
by T Fred Farmington 25 Feb '15

25 Feb '15

After my attempt to Connect with the 'outside' SIP line, I did notice in the log the following: [DEBUG] sofia.c:8732 IP <my SIP IP here> Rejected by acl "domains". Falling back to Digest auth. With that in mind, and going by what I had read before, I edited the conf/autoload_configs/acl_conf.xml file and made the following entries <list name="lan" default="allow"> <node type="allow" cidr="<my SIP IP here>/32"/> </list <list name="domains" default="deny"> <node type="allow" cidr="<my SIP IP here>/32"/> <list> But I still got a BUSY on the next attempt. And the freeswitch.log is still showing: [DEBUG] sofia.c:8732 IP <my SIP IP here> Rejected by acl "domains". Falling back to Digest auth. What am I still missing? Thanks

2 1

Where to go from here?
by T Fred Farmington 24 Feb '15

24 Feb '15

"I assume that this is a reply to my previous message, but since you created a new message it broke the thread." My apologies, I am still getting used to the Account settings in this list. On other lists I can typically set options (or default to) to receive email notifications for my own postings and optionally not others (which, as a novice, is what I want for now) But as it was, I was not receiving any email notifications for this posting. That is why I could not Reply. I changed my account setting, so maybe I guessed correctly this time. ==================== Regardless, I just now edited my conf/dialplan/public/00_inbound_did.xml file I added an entire 'extension' definition for my 'outside' SIP DID based on what is appearing in my freeswitch.log file <extension name="velocity_did"> <condition field="${sip_to_user}" expression="^(specific phone number here)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> Which I HOPE will tell FreeSWITCH to receive the 'outside' call and route it to the default extension 1000 But with only that change, I still only get a BUSY when I dial it on the 'outside' SIP line. The freeswitch.log shows the call being 'seen' as: sofia/internal/<specific phone number here>@<the SIP line IP> Do I also need to modify conf/vars.xml and/or conf/sip_profiles/<one of the xml files> as well to get the connection? Thanks

2 1

Build FS/Master fails on ZRTP
by Victor Medina 24 Feb '15

24 Feb '15

Hi guys. Im getting errors when trying to build FS_MASTER with libZRTP. Im using Ubuntu LTS. libtool: link: ( cd ".libs" && rm -f "libcryptomath.la" && ln -s "../ libcryptomath.la" "libcryptomath.la" ) gcc -DHAVE_CONFIG_H -I. -I./crypto/include -fPIC -Wall -O4 -fexpensive-optimizations -funroll-loops -I./src -Icrypto/include -I./include -I./crypto/include -fPIC -Wall -O4 -fexpensive-optimizations -funroll-loops -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT aes_tables.o -MD -MP -MF .deps/aes_tables.Tpo -c -o aes_tables.o `test -f 'tables/aes_tables.c' || echo './'`tables/aes_tables.c mv -f .deps/aes_tables.Tpo .deps/aes_tables.Po /bin/bash ./libtool --tag=CC --mode=link gcc -fPIC -Wall -O4 -fexpensive-optimizations -funroll-loops -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -L. -lssl -lcrypto -o aes_tables aes_tables.o libcryptomath.la -lcrypto -lcrypto -lcrypto libtool: link: gcc -fPIC -Wall -O4 -fexpensive-optimizations -funroll-loops -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -o aes_tables aes_tables.o -L. ./.libs/libcryptomath.a -lssl -lcrypto make[1]: Leaving directory `/home/ubuntu/tmp/freeswitch-master/libs/srtp' CC libs/spandsp/src/libfreeswitch_spandsp_la-plc.lo CC libs/spandsp/src/libfreeswitch_spandsp_la-alloc.lo CC libs/spandsp/src/libfreeswitch_spandsp_la-bit_operations.lo CCLD libfreeswitch_spandsp.la *** Warning: Linking the shared library libfreeswitch_spandsp.la against the *** static library libs/libzrtp/third_party/bnlib/libbn.a is not portable! cd libs/libzrtp && make make[1]: Entering directory `/home/ubuntu/tmp/freeswitch-master/libs/libzrtp' Makefile:743: src/.deps/libzrtp_a-zrtp.Po: No such file or directory Makefile:744: src/.deps/libzrtp_a-zrtp_crc.Po: No such file or directory Makefile:745: src/.deps/libzrtp_a-zrtp_crypto_aes.Po: No such file or directory Makefile:746: src/.deps/libzrtp_a-zrtp_crypto_atl.Po: No such file or directory Makefile:747: src/.deps/libzrtp_a-zrtp_crypto_ec.Po: No such file or directory Makefile:748: src/.deps/libzrtp_a-zrtp_crypto_ecdh.Po: No such file or directory Makefile:749: src/.deps/libzrtp_a-zrtp_crypto_hash.Po: No such file or directory Makefile:750: src/.deps/libzrtp_a-zrtp_crypto_pk.Po: No such file or directory Makefile:751: src/.deps/libzrtp_a-zrtp_crypto_sas.Po: No such file or directory Makefile:752: src/.deps/libzrtp_a-zrtp_datatypes.Po: No such file or directory Makefile:753: src/.deps/libzrtp_a-zrtp_engine.Po: No such file or directory Makefile:754: src/.deps/libzrtp_a-zrtp_iface_cache.Po: No such file or directory Makefile:755: src/.deps/libzrtp_a-zrtp_iface_scheduler.Po: No such file or directory Makefile:756: src/.deps/libzrtp_a-zrtp_iface_sys.Po: No such file or directory Makefile:757: src/.deps/libzrtp_a-zrtp_initiator.Po: No such file or directory Makefile:758: src/.deps/libzrtp_a-zrtp_legal.Po: No such file or directory Makefile:759: src/.deps/libzrtp_a-zrtp_list.Po: No such file or directory Makefile:760: src/.deps/libzrtp_a-zrtp_log.Po: No such file or directory Makefile:761: src/.deps/libzrtp_a-zrtp_pbx.Po: No such file or directory Makefile:762: src/.deps/libzrtp_a-zrtp_protocol.Po: No such file or directory Makefile:763: src/.deps/libzrtp_a-zrtp_responder.Po: No such file or directory Makefile:764: src/.deps/libzrtp_a-zrtp_rng.Po: No such file or directory Makefile:765: src/.deps/libzrtp_a-zrtp_srtp_builtin.Po: No such file or directory Makefile:766: src/.deps/libzrtp_a-zrtp_string.Po: No such file or directory Makefile:767: src/.deps/libzrtp_a-zrtp_utils.Po: No such file or directory Makefile:768: src/.deps/libzrtp_a-zrtp_utils_proto.Po: No such file or directory Makefile:769: test/.deps/cache_test-cache_test.Po: No such file or directory Makefile:770: test/cmockery/.deps/cache_test-cmockery.Po: No such file or directory Makefile:771: third_party/bgaes/.deps/libzrtp_a-aes_modes.Po: No such file or directory Makefile:772: third_party/bgaes/.deps/libzrtp_a-aescrypt.Po: No such file or directory Makefile:773: third_party/bgaes/.deps/libzrtp_a-aeskey.Po: No such file or directory Makefile:774: third_party/bgaes/.deps/libzrtp_a-aestab.Po: No such file or directory Makefile:775: third_party/bgaes/.deps/libzrtp_a-sha1.Po: No such file or directory Makefile:776: third_party/bgaes/.deps/libzrtp_a-sha2.Po: No such file or directory make[1]: *** No rule to make target `third_party/bgaes/.deps/libzrtp_a-sha2.Po'. Stop. make[1]: Leaving directory `/home/ubuntu/tmp/freeswitch-master/libs/libzrtp' make: *** [libs/libzrtp/libzrtp.a] Error 2 Thanks in advance. -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

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Re- Lua feature session:setAutoHangup(false), can this be done inside the dialplan xml?
by Andrew Keil 24 Feb '15

24 Feb '15

To FreeSWITCH users, Quick question. Is there a way as the first dialplan action (inside the actual dialplan xml) to call a similar function to what I can call inside Lua (ie. session:setAutoHangup(false))? So that all the dialplan applications or actions can be completed without remote disconnect (hangup) interrupting the xml dialplan flow? Obviously I could call a Lua script to perform "session:setAutoHangup(false)", however it would be nice to see it inside the dialplan itself for clarity. Kind Regards, Andrew Keil

3 3

disabling DTMF info packets via dtmf_type or dtmf-type
by Robert Stevens 24 Feb '15

24 Feb '15

Hello, During my testing, I have noticed, that if you do not set dtmf_type as a channel variable or dtmf-type as a sofia profile parameter, and if the SDP does not support rfc2833, then FS will failover to INFO packets. This of course is fine on the face of it. According to https://freeswitch.org/confluence/display/FREESWITCH/Variables, you can set dtmf_type to either info, rfc2833, or none as a channel variable. I have noticed however, that if dtmf_type is set to 'none', FS will still failover to INFO packets when the SDP does not contain a rfc2833 media attribute. The dtmf-type sofia profile parameter does work as expected when I set it as a profile parameter, but I would like to set it in the dialplan and the only way to do this, I would presume, is via dtmf_type. However, it currently is not working (I always get the 'no rfc2833 found in sdp switching to info' log entry) and see info packets when the sofia siptrace is enabled). I would think this is a bug. I tried several workarounds to mitigate but was not successful. Specifically I tried setting dtmf-type in the sofia profile params to 'rfc2833' thinking that it would be hard coded to be rfc2833 and not do the switchover to INFO, but alas, it still did the switchover. Does anyone have any insight on this? I think it is a bug and will file a JIRA on it shortly. Best regards, Robert Stevens rstevens(a)robcoit.com

1 0

Where to go from here?
by T Fred Farmington 24 Feb '15

24 Feb '15

It appears to me that the vanilla configuration files that come with FreeSWITCH allow you to run in a very limited mode. As I indicated before I can get my in-house softphones to successfully call into the FreeSWITCH and reach one of the default extensions. When I do this I hear the associated FreeSWITCH default voice recordings But when I try to call in via my outside SIP line, I cannot get the connection. I see the call arrive in the FS_CLI and in the log files, but I only get a Busy signal and no connection is made. Editing my sip_profiles - internal & external xml files did not change that behavior. Either: 1. I made the wrong edits in those files 2. I did not edit enough files 3. or I have no idea what I am doing. I would rather understand what should be done than just 'groping in the dark' making edits here and there based on a partial understanding (or lack of understanding) of some web reference. I thought that the FreeSWITCH 1.2. book would help, but nowhere does it say: If you want to receive 'outside' calls, do this...... My current SIP line is configured very simply. * No SIP authentication required * A single IP address on Port 5060 upd * DTMF type is set to RFC2833 * Rtp will be from 208.85.184.41 udp ports 10,000 to 59,000 (although we are doing voice only at this time) I have a number of things that I would like the FreeSWITCH phone switch to do, but I feel that I need to take 'baby steps'. So my First goal is to just successfully receive an 'outside' call to one of the existing default extensions - similar to how the softphones work now. My second 'baby step' goal would be to to use one of my softphones to place a call OUT on the same SIP line. After that I can look at beginning to understand how to configure the various internal extensions to do various tasks. Thanks for your continued patience and support.

2 1