February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Generating calls from external source
by Nik Middleton 03 Feb '09

03 Feb '09

Hi Guys, As a long time Asterisk user, I'm looking into freeswitch as an alternative mainly due to (list multiple reasons here) Can anyone give me a pointer as to how I would achieve the following? I need to replicate an emergency broadcast system currently running under Asterisk. At the moment, I run through a Mysql database and using the manager API, issues an Originate command to dial a number. When the call is answered, a message is played, and the recipient has the option of hitting a digit to confirm receipt. I then call an AGI script to update the database. Is this fairly easy to do in Freeswitch? Not looking for code, just some pointers as to what's available to do the above / Regards,

8 11

Conference dialing and uuid
by Sias Mey 03 Feb '09

03 Feb '09

Hi, Im trying to build a web based conference control system. Got most of it sorted with some help from the list but I seem to have run into some strangeness. I use a conference dial call to pull extra users into the conference. I couldent find a way of setting channel variables or executing javascript directly on the conference dial since it expects and endpoint and the {} syntax produced an error. So now I am using the Loopback inteface to register some values. One of the functions these script fullfill is to register the uuid of the new channel in the database, however after thinking for a while all of this is working fine upon further testing I found that going through loopback generated 3 channels. I was saving the uuid of the a leg of the call to loopback into the database. However to manipulate the call in the conference I need the uuid of the leg bridgeing to the conference. I have tried some queries against the core database in via javascript however there seems to be some delay as to when the needed leg gets inserted into the channels table. I have tried with execute_on_ring and execute_on_answer. But I suspect that the call only gets added after it is actually answered. Is there some way for me to find the uuid of this call? I can use api calls via conference list to find all the calls in the conference, however if two people get added to the same conference in rapid succesion it will be quite circumstantial as to which one is which since I dont have a way of directly relating the call back to the channels that originally spawned it without a lot of costly text comparisons. Any help from someone who understands this beter than I do will be greatly appreciated. Thanks in advance, Sias

4 14

debuild breaks since the last few days
by Leon de Rooij 03 Feb '09

03 Feb '09

Hi all, I've been trying to build new debs, but debuild seems to break.. I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages with: debuild -i -us -uc -b (which worked before) And now it breaks at openzap with: cc1: warnings being treated as errors src/ozmod/ozmod_isdn/ozmod_isdn.c: In function 'writeQ931PacketToPcap': src/ozmod/ozmod_isdn/ozmod_isdn.c:220: warning: implicit declaration of function 'pcap_dump_flush' make[7]: *** [src/ozmod/ozmod_isdn/ozmod_isdn.o] Error 1 make[7]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap' make[6]: *** [../libopenzap.so] Error 2 make[6]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap/mod_openzap' make[5]: *** [all] Error 1 make[5]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap/mod_openzap' make[4]: *** [../../libs/openzap/mod_openzap-all] Error 1 make[4]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src' Making all in build make[3]: Entering directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/bin/make install + +----------------------------------------------+ make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' make: *** [build-stamp] Error 2 debuild: fatal error at line 1247: debian/rules build failed Does anyone know how to fix this ? thanks, Leon

4 3

fifo problem
by Tamas Cseke 03 Feb '09

03 Feb '09

Hello, We have a problem with mod_fifo. we monitor fifo push event on event socket, call consumer with originate & fifo out nowait Similar like fifo_outbound works, but we have an external strategy for consumer selection (eg.: skill-based routing) The problem is when a caller stops waiting in fifo, the originated calls kepp ringing the consumer, and when the consumer answer the call, he or she may grab somebody else from the fifo, which is a problem because the callers are identified and some data (eg name, phonenumber is shown for the consumer). so it can happen these data will be wrong. We tried to resolve this issue by a call tracking in the external script using event socket. we pushes a variable into the CHANNEL_ORIGINATE event calling the consumer containing the caller uuid. and if the caller aborts the fifo, we hangup the consumer call with (uuid_kill) But it's not prefect becasue it can happen that the consumer pop another caller from the fifo. and we hangup this call, so as a side-effect we loosing another caller. Could anybody advise a solution for this please? we thinking about to have a fifo_caller_uuid variable, that we set before calling fifo with the out method. and if this uuid is in the top of the fifo then pop it else don't pop anybody. it seems to be a hack anyway.... Thanks in advance, Tamas

2 1

Strange Performance when using as SBC
by rod 03 Feb '09

03 Feb '09

Dear list, I've been playing with freeswitch for some time (2 months) and the fact is that I'm very pleased with the functionnalities of this software. I'd like to use FS as a SBC handling media and I'm doing some tests with sipp to load the machine but I'm unable to bridge more than 60 calls without seeing the CPU being loaded at 100%. I'm sure something is going wrong with my setup but I'm unable to see what. The test machine has the following specs: Athlon XP 3500+ with 2GB of memory (I know this is not a high end machine :p) Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 95 model name : AMD Athlon(tm) 64 Processor 3500+ stepping : 2 cpu MHz : 2199.973 cache size : 512 KB fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic cr8_legacy bogomips : 4402.97 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc I installed FS on a fresh debian 64: Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 x86_64 GNU/Linux I set the ulimit parameters like those on the website: freeswitch@internal> ... Freeswitch:/opt/free-svn/bin# ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited real-time priority (-r) 0 stack size (kbytes, -s) 244 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited My network setup is the following: SIPP machine (10.10.10.1/24)----------------vlan 55 ----------(10.10.10.254/24) FS (10.10.20.254/24)-------------- vlan56 -------------------(10.10.20.100/24) OTHER STOCK FS I launched sipp with: sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i 10.10.10.1 -mp 25000 10.10.10.254:5060 The dialplan on FS is very simple: <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default"> <extension name="hold_music_bridge"> <condition field="destination_number" expression="^9999$"> <action application="answer"/> <action application="bridge" data="sofia/external/9999(a)10.10.20.100"/> </condition> </extension> </context> </include> FreeSWITCH Version 1.0.trunk (11560M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[100] SQL [Enabled] The test is very simple: sipp dial 9999 that matches in my FS dialplan and this is bridged to an other FS machine playing music on hold. When I launch "top" I see after 30 to 40 s that FS consumes all the CPU ressources (with a mean of 50-60 % before), with 80 calls. When I set 70 calls, I have to wait 70-80 s before seeing the same issue. Presence is set to false on the 2 profile. I have the same issue with FS 1.0.2 that' s why I tried FS 11560. When I use the FS machine as a router to test the packet per second performance, I'm reaching 100Mbps with 8000pps in each direction (from vlan 55 to vlan56) with less than 12% CPU. So that I don't think there's an issue with the network. Here is an "mpstat -P ALL 1" to show you what's happening suddenly with 70 bridge calls: 12:31:26 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 0,00 89,00 6241,00 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 0,00 89,00 6241,00 12:31:27 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 0,00 22,22 6035,35 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 0,00 22,22 6035,35 12:31:28 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 0,00 0,00 5483,17 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 0,00 0,00 5483,17 The CPU is going from 89% idle to 0% in less than 2 seconds. I know that I don't have to expect too much from this kind of hardware, but it seems strange that the CPU power vanished so suddenly. Thanks a lot for the guys that have read this long mail :p kind regards, rod

9 27

Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value.
by clif＠eugeneweb.com 02 Feb '09

02 Feb '09

Hi Gang, I've been struggleing with this also. Actually I can get it to bind to my address, the problem is it randomly drops my calls. :-( I have a FS running on a box with a static IP and I can start a call between two extensions and it will go for hours. Then I add anther interface say eth0:0 with a new static IP and reconfigure my phones and FS to use that, and the calls drop after about 15-20 mins. Though it's pretty random. Here is my setup. I have Debian Linux 2.6.23.1 kernel, and freeswitch-1.0.1. Here is my /etc/network/interfaces: # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) # The loopback interface auto lo iface lo inet loopback # The first network card - this entry was created during the Debian installation auto eth0 eth0:0 iface eth0 inet dhcp iface eth0:0 inet static address 192.168.0.249 netmask 255.255.255.0 gateway 192.168.0.254 The only change I made to the FS config is in Vars.xml. I added this line close to the top: <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/> Here is the console log of the call being dropped: freeswitch@archive> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia@67.171.158.226:5080 RUNNING (0) internal profile sip:mod_sofia@192.168.0.249:5060 RUNNING (2) nat profile sip:mod_sofia@67.171.158.226:5070 RUNNING (0) default alias internal ALIASED outbound alias external ALIASED 192.168.0.249 alias internal ALIASED ================================================================================================= 3 profiles 3 aliases freeswitch@archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM Reloaded 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 sofia_read_frame() Hangup sofia/internal/1003@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/1001(a)192.168.0.249 [CS_EXECUTE] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/1003@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1003@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 5 (sofia/internal/1001(a)192.168.0.249) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1001(a)192.168.0.249 [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [192.168.0.249] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [default] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started Profile internal [sofia_reg_internal] 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [outbound] for profile [external] 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile external [sofia_reg_external] 2009-02-01 13:23:20 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile nat [sofia_reg_nat] sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia@67.171.158.226:5080 RUNNING (0) internal profile sip:mod_sofia@192.168.0.249:5060 RUNNING (0) outbound alias external ALIASED 192.168.0.249 alias internal ALIASED nat profile sip:mod_sofia@67.171.158.226:5070 RUNNING (0) default alias internal ALIASED ================================================================================================= 3 profiles 3 aliases There is an older thread that says one should set <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/> but in this (later) thread is says only Jingleling usese that variable. ie. see: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg00695.… http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg07345.… So what do you think causes this? What is the correct way? ;-) Thanks, Clif

2 1

Call Variable not available when call hangup
by shehzad p 02 Feb '09

02 Feb '09

Hi all, I need to process some CDR variables in Dialplan, like call duration, Answered time etc. but when I place info application after bridge, it is not listing them properly as below: =========================================== Caller-Channel-Created-Time: [1233573341672157] Caller-Channel-Answered-Time: [1233573342712939] Caller-Channel-Hangup-Time: [0] ========================================== Here Hangup time is 0, So how can I find actual values? --I know that we can use xml_cdr or cdr_csv, but my current need is to get those values from dialplan itself so that can be passed to some script... thanks, msp -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp217885… Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 5

Phonebooth?
by J. G. 02 Feb '09

02 Feb '09

I got a group email from Anders Brownworth this weekend regarding him donating Phonebooth to the FreePBX project? Wonder what the impact will be.. -- ----- Jason Gehman General Manager North Voice Communications www.NorthVC.com

1 0

LDAP Integration
by Vinicius Kobashi 02 Feb '09

02 Feb '09

7 16

Outbound Proxy configuration
by Boris Lansky 02 Feb '09

02 Feb '09

What way should I configure Freeswitch to get out all the outbound calls through a specific proxy server? Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky(a)interwise.com <mailto:Simon.shaw@att.com>

5 9