June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_dingaling & inbound audio
by Locutis of Borg 14 Sep '12

14 Sep '12

Followed instructions on wiki. Enabled mod_dingaling to register over at GV. Set external rtp to right address. Outbound calling terminates, and outbound audio works, but seems that I can not get return audio. FS is on DMZ - same problem Moved to internal (behind NAT, port forwarding) - same problem Moved back to DMZ and killed iptables and router spi - same problem In and out calling on SIP to provider works fine in both cases. So my questions are: Can mod_dingaling be used for GV or not? Since it uses TLS, NAT shouldn't be an issue? and Does GV not like us FS users? Thank you for you help. "Resistance is Fruitful"

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Variable continue on fail
by Виталий Давудов 13 Sep '12

13 Sep '12

Hi all! In my dialplan I've included variable continue on fail: <extension name="Dial Out SIP-M200 Mobile numbers"> <condition field="destination_number" expression="^(89\d{9})$"> <action application="export" data="t38_passthru=true"/> <action application="set" data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> <action application="info"/> <action application="bridge" data="sofia/gateway/SIP_M200/$1"/> <action application="limit_execute" data="hash limit SMG-1016M 60 bridge sofia/gateway/SMG-1016M/$1"/> </condition> </extension> And if FS recieve from first gateway USER_BUSY, then FS try to bridge this call to another gateway. Although in line <action application="set" data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> there is no code Q.850 = 17. How resolve this issue? -- Best regards, Vitaly Davudov "VIP-TELECOM-SERVICE" Ltd. ("ETERIA" Group of companies) http://www.vts24.ru

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freeswitch - sangoma CAS T1 - a101 configuration
by copycall 07 Sep '12

07 Sep '12

hi, i'm having trouble getting instructions to configure freeswitch with a sangoma a101 T1 card for CAS, B8ZS, ESF. the first application i seek to solve is using this setup as a gateway (replacing a dialogic dmg 2000) for sip from my itsp to the tdm endpoint: a legacy centigram - mitel voicemail server. any help will be greatly appreciated. thank you, dave

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How to set a variable for a call going from gateway?
by "Гришин В.С." 04 Sep '12

04 Sep '12

After I read a http://wiki.freeswitch.com/wiki/Sofia.conf.xml#Variables I configured FS [root@freeswitch1 external]# more /usr/local/freeswitch/conf/sip_profiles/external/SMG-1016M.xml </include> <gateway name="SMG-1016M"> <param name="realm" value="10.10.10.1"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="25"/> <variables> <variable name="SMG-1016M_inbound_var_name" data="this_is_inbound_call" direction="inbound"/> <variable name="SMG-1016M_outbound_var_name" data="this_is_outbound_call" direction="outbound"/> <variable name="SMG-1016M_both_var_name" data="this_on_any_direction_call"/> </variables> </gateway> </include> [root@freeswitch1 external]# I insert the info application into public context of dialplanbefore a transfer application? make call but don't see variables (SMG-1016M_inbound_var_name,SMG-1016M_outbound_var_name,SMG-1016M_both_var_name) in fs_cli. I want to see variable = <gateway name="SMG-1016M"> into dialplan for each gateway in (single) external sip profile. How it to make? Vladislav Grishin

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Open Source unified autodialer software released
by tahir almas 28 Aug '12

28 Aug '12

Pleased to announce the release of open source Fax , SMS and Voice broadcasting software solution ICTDialer http://www.ictdialer.org developed over reknown Drupal Conent Mnagment System and powerfull Plivo Communication framework , Your contribution and suggestions are welcome Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT **************************************************************************************************************** NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation.

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caller_id_number
by Boris Kovalenko 16 Aug '12

16 Aug '12

Hello! When the call is incoming for my network I want CID is translated by my rules. So I translate caller_id_number and set effective_caller_id_number. But there is a little problem - when the call arrived to a analog phone (for example Panasonic KTX-TCD540) I see two number on display - original caller_id_number and effective_caller_id_number. Is there a way to not show caller_id_number in CID? Or may I modify caller_id_number instead of setting effective_caller_id_number? -- Regards, Boris

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External profile & NAT
by ocset 03 Aug '12

03 Aug '12

Hi I would like to allow a person with a softphone app to register with Freeswitch (extension 1111) when they are outside the office network. I would like to to implement this using an external profile. According to the wiki page (http://wiki.freeswitch.org/wiki/External_profile) I just need to create a copy of the "conf/sip_profiles/external.xml" file and change the port number to 5090 (and also open the port on the modem). When I do a "sofia status", this new gateway does not show up and to be honest, I don't quite understand how that works and what else to read to make sense of what to do. What are the ways in which other Freeswitch admins are granting access to external devices? ps. just to confirm, do I set external_rtp_ip and external_sip_ip to the fix IP address that my ISP has given me - the one that is shown by www.whatsmyip.org? Please help Thanks O.

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ESL problem
by Thangappan.M 31 Jul '12

31 Jul '12

Dear all, In the previous post, I got the information that using event outbound socket we can implement the IVR and also see the example in libs/esl/perl/server2.pl. I have seen it and understood the flow of the script.But when I was running that script it tells the following error. Can't locate loadable object for module ESL in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at ESL.pm line 11 Compilation failed in require at server2.pl line 1. Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) But in perl directory there is no directory called ESL. What would be the issue?. Is ESL necessary is necessary for implementing IVR using event outbound socket? -- Regards, Thangappan.M

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EVENT_TRAP: IP change detected -> no Write unlock -> Error Creating SIP UA for profile: internal
by Nestor A Diaz 31 Jul '12

31 Jul '12

Hi Guys. Today my standard sip profile decide not to work anymore, in fact it was related with the fact i had put some ip alias on the same internal network interface a few days ago, by searching the logs, this is what i got: 2012-02-28 14:50:56.126576 [INFO] mod_sofia.c:5155 EVENT_TRAP: IP change detected 2012-02-28 14:50:56.126576 [INFO] mod_sofia.c:5156 IP change detected []->[] [::1]->[fe80::225:90ff:fe3a:2e1a] 2012-02-28 14:50:56.306534 [NOTICE] sofia_glue.c:5295 Reload XML [Success] [...] 2012-02-28 14:50:57.926576 [DEBUG] sofia.c:2121 Write lock doublenat 2012-02-28 14:50:57.926576 [DEBUG] sofia.c:2132 Write unlock doublenat [...] 2012-02-28 14:50:58.026533 [DEBUG] sofia.c:2121 Write lock internal-ipv6 2012-02-28 14:50:58.026533 [DEBUG] sofia.c:2132 Write unlock internal-ipv6 [...] 2012-02-28 14:50:58.026533 [ERR] sofia.c:1916 Error Creating SIP UA for profile: internal-ipv6 [...] 2012-02-28 14:50:58.026533 [DEBUG] sofia.c:2121 Write lock internal but no unlock internal profile and no creating sip ua for profile: internal. Later final users reported the problem and i tried to start the profile: $ sofia profile internal start and i got: 2012-02-28 17:08:30.586507 [ERR] sofia.c:1916 Error Creating SIP UA for profile: internal So i deactivated the ip aliases and restart freeswitch and everything comes to live. But I suspect a problem with sofia because it wasn't able to unlock the internal profile, i checked the status of sofia via command line when this happens and there was no sofia internal profile listed. I don't know how the ip change event was triggered since i added the ip alias some days ago, not today when the problem was triggered. Maybe there will be some bug there. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor(a)tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

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RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a
by Hoexum, Edwin 25 Jul '12

25 Jul '12

Gents, I am trying to do some load testing on a freeswitch for transcoding from SILK to G711a. The question is: Is the limit for RTP sessions limited to CPU power or are there also limits on the freeswitch application and can I tune freeswitch to do the job for 1500 sessions or more. At the moment freeswitch is the best transcoding device for SILK to G711a that I know of. Also compared to the big companies who deliver transcoding devices. SETUP: 3 X G7 HP DL 380 with 16 G mem 6 core 12 CPU's 1 for SIPP UAC (1Gb interface for media and Signaling) 1 for Freeswitch (trancoding function SILK to G711a) 1Gb interface for media and Signaling 1 for SIPP UAS (1Gb interface for media and Signaling) First tuning OS (with help from the wiki and some LINUX sysctl tuning) SIPP UAC <-> Freeswitch root@mnd-rc0001-srvot02:~# iperf -c 172.23.125.46 -u -d -b 400m ------------------------------------------------------------ Server listening on UDP port 5001 Receiving 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ ------------------------------------------------------------ Client connecting to 172.23.125.46, UDP port 5001 Sending 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ [ 4] local 172.23.125.42 port 46443 connected with 172.23.125.46 port 5001 [ 3] local 172.23.125.42 port 5001 connected with 172.23.125.46 port 39259 [ ID] Interval Transfer Bandwidth [ 4] 0.0-10.0 sec 483 MBytes 405 Mbits/sec [ 4] Sent 344796 datagrams [ 3] 0.0-10.0 sec 483 MBytes 406 Mbits/sec 0.001 ms 0/344828 (0%) [ 3] 0.0-10.0 sec 1 datagrams received out-of-order [ 4] Server Report: [ 4] 0.0-10.0 sec 483 MBytes 405 Mbits/sec 0.004 ms 261/344795 (0.076%) [ 4] 0.0-10.0 sec 1 datagrams received out-of-order root@mnd-rc0001-srvot02:~# Linux mnd-rc0001-srvot06 2.6.32-28-vserver #55~ppa1-Ubuntu SMP Fri Feb 4 21:25:09 UTC 2011 x86_64 GNU/Linux Ubuntu 10.04 LTS Welcome to Ubuntu! * Documentation: https://help.ubuntu.com/ Last login: Thu Jun 21 15:01:35 2012 from 172.21.88.65 root@mnd-rc0001-srvot06:~# iperf -s -u -i1 ------------------------------------------------------------ Server listening on UDP port 5001 Receiving 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ ------------------------------------------------------------ Client connecting to 172.23.125.42, UDP port 5001 Sending 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ [ 5] local 172.23.125.46 port 39259 connected with 172.23.125.42 port 5001 [ 3] local 172.23.125.46 port 5001 connected with 172.23.125.42 port 46443 [ ID] Interval Transfer Bandwidth [ 5] 0.0- 1.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 0.0- 1.0 sec 48.4 MBytes 406 Mbits/sec 0.004 ms 262/34754 (0.75%) [ 5] 1.0- 2.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 1.0- 2.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34476 (0%) [ 5] 2.0- 3.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 2.0- 3.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34480 (0%) [ 5] 3.0- 4.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 3.0- 4.0 sec 48.3 MBytes 406 Mbits/sec 0.004 ms 0/34483 (0%) [ 5] 4.0- 5.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 4.0- 5.0 sec 48.3 MBytes 406 Mbits/sec 0.005 ms 0/34482 (0%) [ 5] 5.0- 6.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 5.0- 6.0 sec 48.3 MBytes 405 Mbits/sec 0.004 ms 0/34474 (0%) [ 5] 6.0- 7.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 6.0- 7.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34477 (0%) [ 5] 7.0- 8.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 7.0- 8.0 sec 48.3 MBytes 405 Mbits/sec 0.004 ms 0/34480 (0%) [ 5] 8.0- 9.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 8.0- 9.0 sec 48.3 MBytes 406 Mbits/sec 0.005 ms 0/34482 (0%) [ 5] 9.0-10.0 sec 48.3 MBytes 406 Mbits/sec [ 5] 0.0-10.0 sec 483 MBytes 406 Mbits/sec [ 5] Sent 344829 datagrams [ 3] 0.0-10.0 sec 483 MBytes 405 Mbits/sec 0.004 ms 261/344795 (0.076%) [ 3] 0.0-10.0 sec 1 datagrams received out-of-order [ 5] Server Report: [ 5] 0.0-10.0 sec 483 MBytes 406 Mbits/sec 0.000 ms 0/344828 (0%) [ 5] 0.0-10.0 sec 1 datagrams received out-of-order UDP traffic looks fine. SIPP (UAC) inject SILK RTP (PCAP ) and send this to Freeswitch. Freeswitch send this G711a to the SIPP( UAS). I doing some testing and until 700 sessions it look fine (Packed loss <0.1% and delay <40ms) If I do more sessions 10% of packet have no packet loss but delay of more the 100ms. SIPP uac side: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 50.0(0 ms)/1.000s 5060 230.53 s 2000 172.23.125.12:5060(UDP) Call limit reached (-m 2000), 0.000 s period 0 ms scheduler resolution 0 calls (limit 4500) Peak was 2000 calls, after 40 s 0 Running, 1654 Paused, 0 Woken up 0 dead call msg (discarded) 6459 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg Pause [ 2000ms] 2000 0 INVITE ----------> 2000 0 0 100 <---------- 2000 0 0 0 183 <---------- 0 0 0 0 400 <---------- 0 0 0 0 401 <---------- 0 0 0 0 403 <---------- 0 0 0 0 404 <---------- 0 0 0 0 405 <---------- 0 0 0 0 406 <---------- 0 0 0 0 408 <---------- 0 0 0 0 415 <---------- 0 0 0 0 433 <---------- 0 0 0 0 480 <---------- 0 0 0 0 481 <---------- 0 0 0 0 484 <---------- 0 0 0 0 485 <---------- 0 0 0 0 486 <---------- 0 0 0 0 487 <---------- 0 0 0 0 488 <---------- 0 0 0 0 489 <---------- 0 0 0 0 500 <---------- 0 0 0 0 502 <---------- 0 0 0 0 503 <---------- 0 0 0 0 180 <---------- 2000 0 0 0 200 <---------- E-RTD1 2000 0 0 0 ACK ----------> 2000 0 Pause [ 500ms] 2000 0 [ NOP ] BYE <---------- 2000 0 0 0 200 ----------> 2000 0 ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2012-06-28 20:17:17:312 1340907437.312927 Last Reset Time | 2012-06-28 20:21:07:857 1340907667.857562 Current Time | 2012-06-28 20:21:07:857 1340907667.857769 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:000 | 00:03:50:544 Call Rate | 0.000 cps | 8.675 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 2000 Total Call created | | 2000 Current Call | 0 | SIPP UAS side: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5060 65.10 s 2000 UDP 0 new calls during 1.001 s period 1 ms scheduler resolution 2000 calls Peak was 2000 calls, after 44 s 0 Running, 2002 Paused, 4 Woken up 0 dead call msg (discarded) 3 open sockets 2021 Total echo RTP pckts 1st stream 5.327 last period RTP rate (kB/s) 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate (kB/s) Messages Retrans Timeout Unexpected-Msg ----------> INVITE 2000 0 0 0 <---------- 100 2000 0 [ 500ms] Pause 2000 0 <---------- 180 2000 0 <---------- 200 2000 0 0 ----------> ACK 2000 0 0 0 [ 3:08] Pause 2000 0 <---------- BYE 0 0 0 ----------> 200 0 0 0 0 ------------------------------ Sipp Server Mode ------------------------------- ############################################################################## MPSTAT -A 5 on freeswitch: 8:29:50 CPU %usr %nice %sys %iowait %irq %soft %steal %guest %idle 18:29:55 all 24.84 0.00 25.63 0.00 0.02 7.46 0.00 0.00 42.05 18:29:55 0 43.52 0.00 26.72 0.00 0.00 4.05 0.00 0.00 25.71 18:29:55 1 27.88 0.00 39.18 0.00 0.00 3.12 0.00 0.00 29.82 18:29:55 2 27.70 0.00 33.40 0.00 0.00 4.17 0.00 0.00 34.72 18:29:55 3 25.57 0.00 35.98 0.00 0.00 3.03 0.00 0.00 35.42 18:29:55 4 27.36 0.00 27.56 0.00 0.00 2.56 0.00 0.00 42.52 18:29:55 5 25.45 0.00 24.44 0.00 0.00 2.22 0.00 0.00 47.88 18:29:55 6 24.77 0.00 31.28 0.00 0.00 2.98 0.00 0.00 40.97 18:29:55 7 23.05 0.00 23.81 0.00 0.00 2.67 0.00 0.00 50.48 18:29:55 8 22.22 0.00 23.03 0.00 0.20 2.02 0.00 0.00 52.53 18:29:55 9 4.84 0.00 5.71 0.00 0.00 62.86 0.00 0.00 26.59 18:29:55 10 23.36 0.00 17.01 0.00 0.00 2.46 0.00 0.00 57.17 18:29:55 11 20.21 0.00 15.11 0.00 0.00 2.98 0.00 0.00 61.70 18:29:50 CPU intr/s 18:29:55 all 30433.80 18:29:55 0 0.00 18:29:55 1 0.00 18:29:55 2 0.00 18:29:55 3 0.00 18:29:55 4 0.00 18:29:55 5 0.00 18:29:55 6 0.00 18:29:55 7 0.00 18:29:55 8 0.00 18:29:55 9 0.00 18:29:55 10 0.00 18:29:55 11 0.00 18:29:50 CPU 0/s 1/s 8/s 18:29:55 0 0.00 0.00 0.00 18:29:55 1 0.00 0.00 0.00 18:29:55 2 0.00 0.00 0.00 18:29:55 3 0.00 0.00 0.00 18:29:55 4 0.00 0.00 0.00 18:29:55 5 0.00 0.00 0.00 18:29:55 6 0.00 0.00 0.00 18:29:55 7 0.00 0.00 0.00 18:29:55 8 0.00 0.00 0.00 18:29:55 9 0.00 0.00 0.00 18:29:55 10 0.00 0.00 0.00 18:29:55 11 0.00 0.00 0.00 I tested it with G711a to G711a and got the same result. I tested also some RTP parameters like <param name="rtp-autoflush-during-bridge" value="true"/> or false <param name="rtp-autoflush" value="true"/> or false The lot of the module are not activated like CDR stuff. mvg, Edwin ________________________________ This message is confidential and may be privileged. Any review, retransmission, dissemination or other use of, taking any action with reference to this information by persons other then the intended recipient is prohibited. If you receive this message in error, please notify the sender by reply e-mail and delete this message from all computers. Please note that e-mails are susceptible to change. The sender will not accept liability for the improper or incomplete transmission of the information contained in this message. Spaar het milieu door deze e-mail niet te printen/Please consider the environment before printing this email.

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