October 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch and twilio
by Michael Nielsen 02 Oct '15

02 Oct '15

I'm trying to setup twilio as gateway following this guide: https://www.twilio.com/docs/sip-trunking/sample-configuration#freeswitch In their example of twiliotrunk.xml they have twilio-inbound and twilio-outbound. What is the difference between the two gateways? I know their names indicate inbound and outbound calling, but how can Freeswitch tell the difference between them. Especially how can Freeswitch now to use twilio-inbound for incoming calls from twilio and not just twilit-outbound as the only difference is register-proxy and expire-seconds between the two? Make sense?

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Audio delays and jitter
by Danny Gershman 02 Oct '15

02 Oct '15

We are using the OPUS codec with Chrome and WebRTC to make SIP calls into FS conferences. We have packet captures at the edge of network that sound clear, but when they get to FreeSWITCH they sound distorted via delays and clipping. I've tried several combinations of jitterbuffer settings, disabled/enabling RTP timers, conference. I have a mechanism I use in which I replay the RTP packets using SIPP and try them out. Is there a way to dynamically adjust the jitterbuffer so that the right size is being used? Right now I have it set at 180:540:540. Maybe there is some other setting I could try that would help? Perhaps there's a way to force OPUS to use less bandwidth at the expense of quality? I'm willing to share the PCAPs some examples. Thanks in advance. --Danny Gershman

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Reg exp in tutorial
by Michael Nielsen 02 Oct '15

02 Oct '15

In the doc's, should the 10_gateway_inbound.xml contain: expression="^[a-z\.]" instead of: expression="^[a-z]" So it can catch the domain and context int.example.net.

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Setting loglevel to warning removes log in fs_cli
by Michael Nielsen 02 Oct '15

02 Oct '15

I've set <param name="loglevel" value="warning"/> in switch.conf.xml. This removes a lot of output from the fs_cli. But if I want to try and log things to debug I'm not able to from fs_cli, I need to change the setting in switch.conf.xml and restart FreeSWITCH. How can I easily change debug/log level from fs_cli? I've tried /debug and /log, without any luck.

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not enough buffer space for required resample operation!
by Tim Meade 02 Oct '15

02 Oct '15

Intermediate transcoding server. Debian 8 FreeSWITCH Version 1.7.0+git~20150827T040104Z~eefe21536d~64bit (git eefe215 2015-08-27 04:01:04Z 64bit) We have ONE cell phone number that triggers this. All other numbers in testing have worked fine. I can't seem to find much on this error. Before I open a JIRA I thought I would ask if anyone else had seen this. We are in the process form moving from long term cents to debian. Maybe something was missed. Thanks Tim 2015-10-02 11:52:10.942998 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/2125551212(a)162.213.51.55! 2015-10-02 11:52:10.942998 [NOTICE] switch_ivr_originate.c:3526 Channel [sofia/internal/2125551212(a)162.213.51.55] has been answered 2015-10-02 11:52:11.303018 [NOTICE] switch_core_io.c:1191 Activating write resampler 2015-10-02 11:52:11.342999 [NOTICE] switch_core_io.c:1191 Activating write resampler 2015-10-02 11:52:11.342999 [CRIT] switch_core_io.c:1261 sofia/internal/2125551212(a)162.213.51.55 not enough buffer space for required resample operation! 2015-10-02 11:52:11.342999 [NOTICE] switch_core_io.c:1263 Hangup sofia/internal/2125551212(a)162.213.51.55 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 15798 max locked memory (kbytes, -l) 64 max memory size (kbytes, -m) unlimited open files (-n) 65536 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) 15798 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited

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Non regression tests?
by fayçal noushi 02 Oct '15

02 Oct '15

Hi all, A long time ago, I opened a ticket for an issue that I had. I wanted to extract SDP from multipart messages (this was in a SIP-I scenario case). To do so, I've had to modify a function in nua_session.c. I'm not used to C programming, but I made sure the modifications only impacted the SDP multipart case which isn't supported anyway. The thing is, it was an important step to make a SIP-I call go through FreeSwitch. I've discussed this in verto with you, people. But, as I understood it, this couldn't be merged since it had to undergo regression testing. My question is this: how can I make those tests? Since then, the pull request got declined since I had to redo the same thing with FS 1.6. I didn't do it since I don't know it's really useful. Thanks, -- NOUSHI Fayçal

1 0

How to test a Pull Request! aka how to help us out!
by Brian West 02 Oct '15

02 Oct '15

Checkout freeswitch master first! git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git cd freeswitch.git git config --add remote.origin.fetch '+refs/pull-requests/*/from:refs/remotes/origin/pr/*' git config alias.pr '!f() { git checkout pr/$1; git config branch.pr/$1.merge refs/pull-requests/$1/from;} ; f' git pr <PR_NUMBER> build, test, report back. You can view the list of pull requests here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests Questions? Comments? Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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kamailio not starting
by Vikas Kumar 02 Oct '15

02 Oct '15

Hi all, Getting error : 0(32420) ERROR: db_mysql [km_my_con.c:114]: db_mysql_new_connection(): driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) 0(32420) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(32420) ERROR: usrloc [dlist.c:654]: register_udomain(): failed to open database connection 0(32420) ERROR: registrar [reg_mod.c:563]: domain_fixup(): failed to register domain 0(32420) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:609 0(32420) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:611 0(32420) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:615 Any idea to resolve this. -- Regards, Vikas Kumar

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Is there a channel variable available to a dialplan that contains the current number of active sessions? ie. matches the command: "show channels count"
by Andrew Keil 02 Oct '15

02 Oct '15

To FreeSWITCH users, I just wondered if there is a way to return a count of the number of active sessions as a channel variable for use within a diaplan? >From the FreeSWITCH console or FS_CLI the command "show channels count" returns <x> total (where <x> is the current number of active sessions and it is the value of <x> that I would like to use). Please note: I do not require to check against any limits, I just need this to expose the number of concurrent calls (including the session that is running) to the dialplan script. Thanks in advance, Andrew Keil Visytel Pty Ltd

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Detect source IP of a call
by Dominique Jeannerod 01 Oct '15

01 Oct '15

Hello, I think it's a kind of newbie question, but i'm trying to find the simplest way of detecting the source of a call (public trunks), which of course rely on the source IP of the call. I already defined one ACL per trunk, to authorize only allowed IP adresses, and test the different acls in the dialplan to identify the source trunk of a call, and set the associated accountcode. Example : <condition field="${acl(${network_addr} T_CNX_VPN)}" expression="true" break="on-true"> <action inline="true" application="set" data="accountcode=T_CNX"/> </condition> It works but the dialplan needs to be modified for each new trunk. I am wondering if there is a best way to do that, using for example a channel variable, containing the ACL which matched for the call. I tested sip_acl_authed_by, but this seems to be available only when the call is authenticated (?) I'm sure a lot of people had the same question What would be your advice ? D. Jeannerod

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