April 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Override sip hangup cause - transfer bridge call
by Milan Milosavljević 12 Apr '17

12 Apr '17

If a bridged call fails I transfer it to 'to503' extension to overwrite hangup cause. Some carriers reject `bridge` call with hangup code like normal_temporary_failure which i need to override with 503/34. Here is the dialplan and extension to transfer to. <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data=" ...bridge data... "/> <action application="transfer" data="to503"/> <extension name="to503"> <condition field="${proto_specific_hangup_cause}" expression="^(.*)$"> <action application="set" data="sip_ignore_remote_cause=true"/> <action application="respond" data="503"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> It works fine but in saved xml_cdr file that i process later 'sip_network_ip' variable is wrong, its taken from some of the other incoming calls and looks always random. How can i please get right ip from originated call? If i set leg to "transfer" data="-bleg to503" i get NORMAL_CLEARING as hangup code, with -both i get what i need but IP is incorrect. Maybe there is a way to respond with 503 without transferring the call to extension. If i simply respond with data='503' without transfer i get 503 but 41 instead of 503/34 NORMAL_CIRCUIT_CONGESTION. Thanks in advance

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FreeSWITCH 100% CPU utilization
by Örn Arnarson 12 Apr '17

12 Apr '17

Hello FS users, I'm running a FS server with 8 CPU cores and around 3000 users, ~50 concurrent calls. FS is proxying the media. I'm running into about a 1-1.5 second delay when forwarding SIP responses back to the UAC (200 OK for example). It seems that almost all of the processing is being done by one CPU core, while the other 7 sit idle. I found this on the old wiki: libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles There's no mention of this on the new one. Has this changed at all? In any case, I'm assuming that SIP and RTP co-exist on the thread, seeing as I don't have that many SIP requests per second, and yet it is taking more than 1 second to forward the requests. I guess my question is; Is it possible to offload the RTP handling off to different cores somehow? Or force FreeSWITCH to use more than one thread per sofia profile? Regards, Örn

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python-ESL for Python3
by Deepika Yadav 11 Apr '17

11 Apr '17

Hi, I need to python-ESL with python version 3. However, by default ESL library get installed which is compatible for python2, I have various other modules that run only on python3. How can I create ESL for python3 When I try to run "pip3 install python-ESL", I get the following error: error: command 'swig' failed with exit status 1 ---------------------------------------- Cleaning up... Command /usr/bin/python3 -c "import setuptools, tokenize;__file__='/tmp/pip_build_sangoshthi/python-ESL/setup.py';exec(compile(getattr(tokenize, 'open', open)(__file__).read().replace('\r\n', '\n'), __file__, 'exec'))" install --record /tmp/pip-pzzj15oz-record/install-record.txt --single-version-externally-managed --compile failed with error code 1 in /tmp/pip_build_sangoshthi/python-ESL Storing debug log for failure in /home/sangoshthi/.pip/pip.log Regards Deepika

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Verto WSS Connection Refused
by simba msongelwa 11 Apr '17

11 Apr '17

So I installed verto with no errors, and it's running on my secure site. Can anyone please tell me how to get a secure web socket (wss) connection? Netstat shows my server is listening on wss port 8082 but connection is rfefused both firefox & chrome, and I did restart fs many times, nginx too.The port is open in my firewall so i am stuck on what to try next. http://prntscr.com/ev5ywx http://prntscr.com/ev5zdi http://prntscr.com/ev60h9Thanks,Simba

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External Profile Fail Registration
by Deepika Yadav 11 Apr '17

11 Apr '17

Hi, My Freeswitch suddenly started showing error : MySIP Failed Registration with status Service Unavailable [503]. failure sofia status shows: external::MySIP gateway sip:iiitdmon@as1.doorvaani.com FAIL_WAIT How can I resolve this? Regards Deepika

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Job Opportunity
by Brian West 10 Apr '17

10 Apr '17

FreeSWITCHers, https://freeswitch-solutions-llc.workable.com/jobs/466300 Apply via that site. -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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Saving fax files to s3 servers via rxfx
by Hüseyin TUNÇ 10 Apr '17

10 Apr '17

Hi everyone, We’re having some problems about saving fax files to s3 when getting faxes with rxfx. All is well when we give a local path for file but when it comes to give s3 url it doesnt work at all. Is that even possible? Best regards. -- Hüseyin TUNÇ Sent with Airmail

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Stereo calls recording.
by Mimiko 10 Apr '17

10 Apr '17

Hello. Hope there is time to give some ideas for new release of FS. In FS I've found very useful recording a conversation in stereo with each calling party on its channel: one caller on left channel and one on right channel. Its like you a sitting between them. And what about stereo calls? I think if two parties talk with stereo microphones FS should record all channels in a file in different channels. I see this using DD or AC3 with 4 channels. The callers could be recorded like this: Front Caller 1 Left Mic - FL FR - Caller 2 Right Mic Caller 1 Left listener Right Caller 2 Caller 1 Right Mic - SL SR - Caller 2 Left Mic Rear So, listening to this record, one will feel like is sitting between two callers and will feel the position in space of each one. One good thing about decoders is that if listener have only headset, then one can choose listening only for front channels (left and right) hearing each party, or down mix rear channels to front ones. What do you think about this idea? -- Mimiko desu.

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Wrong download?
by Luis Daniel Lucio Quiroz 10 Apr '17

10 Apr '17

Are you aware of this? making all mod_event_zmq make[4]: Entering directory `/builddir/build/BUILD/freeswitch-1.6.16/src/mod/event_handlers/mod_event_zmq' cd /builddir/build/BUILD/freeswitch-1.6.16/libs && /bin/sh /builddir/build/BUILD/freeswitch-1.6.16/build/getlib.sh http://download.zeromq.org zeromq-2.1.9.tar.gz || cd /builddir/build/BUILD/freeswitch-1.6.16/libs && /bin/sh /builddir/build/BUILD/freeswitch-1.6.16/build/getlib.sh http://download.zeromq.org/historic zeromq-2.1.9.tar.gz % Total % Received % Xferd Average Speed Time Time Time Current Dload Upload Total Spent Left Speed 0 95 0 95 0 0 69 0 --:--:-- 0:00:01 --:--:-- 840 curl: (60) Peer certificate cannot be authenticated with known CA certificates More details here: http://curl.haxx.se/docs/sslcerts.html curl performs SSL certificate verification by default, using a "bundle" of Certificate Authority (CA) public keys (CA certs). If the default bundle file isn't adequate, you can specify an alternate file using the --cacert option. If this HTTPS server uses a certificate signed by a CA represented in the bundle, the certificate verification probably failed due to a problem with the certificate (it might be expired, or the name might not match the domain name in the URL). If you'd like to turn off curl's verification of the certificate, use the -k (or --insecure) option. cannot find zeromq-2.1.9.tar.gz % Total % Received % Xferd Average Speed Time Time Time Current Dload Upload Total Spent Left Speed 0 9 0 9 0 0 39 0 --:--:-- --:--:-- --:--:-- 77 gzip: zeromq-2.1.9.tar.gz: not in gzip format /bin/gtar: This does not look like a tar archive /bin/gtar: Exiting with failure status due to previous errors -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH

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user context being ignored
by Peter Villeneuve 10 Apr '17

10 Apr '17

Hi all, I'm confused by some strange behavior I'm seeing on my vanilla FS installation. If I have <param name="apply-inbound-acl" value="domains"/> set in my internal sip profile, even my registered user is unable to make any calls since he gets rejected by the 2017-04-07 17:13:10.085105 [WARNING] sofia.c:9983 IP xx.xx.xx.xx Rejected by acl "domains" message in the logs. Shouldn't FS fall back to regular digest authentication like it used to? I know the user (let's call him 1000) is registered properly since running sofia status profile internal reg in the cli shows that he is indeed registered as expected. I made sure in user 1000's directory entry that <variable name="user_context" value="default"/> is set. If 1) I I apply <param name="apply-inbound-acl" value="domains"/> to the internal profile then calls from user 1000 get rejected by acl domains as explained above, with no fallback to digest authentication (don't know why it doesn't challenge the user) if 2) I disable <param name="apply-inbound-acl" value="domains"/> then the call hits the public dialplan instead of the default one even though <variable name="user_context" value="default"/> is set in user 1000's directory xml entry. I haven't used FS in awhile but I recall in the past not having this much trouble trying to get it working as expected. I suppose I could set the context in the internal profile to default but then all kinds of bad guys will be able to make calls on my dime (it's happened before). I want authenticated users only to make calls but this seems to be eluding me. Is this a known bug (my debian installation was built from git 02c0860 2017-03-03 23:35:25Z 32bit) or am I doing something silly? Thanks, Peter

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