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Minimal configuration for new projects
by Stanislav Sinyagin 01 Oct '13

01 Oct '13

I got bored with cleaning up the vanilla configuration one more time, so I created this Github repo as a base for new projects: https://github.com/xlab1/freeswitch_conf_minimal This is a FreeSWITCH configuration which is completely free from features and predefined stuff (with a small exception in the public context). feel free to use as you like.

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CALL-ID
by Miha 01 Oct '13

01 Oct '13

HI, I need a little help. I have opensips and behind opensips a hew few Freeswitch servers which are for media handling. Opensips is also doing acc with radius. Here is what is happening and I do not know how can I prevent FS do doing so: A CALLS B. A(call-id)->Opensips(call-id)->fs(call-id)->opensips(fs change call-id!!!)->B(changed call-id). How to prevert so that FS will not change call-id of the call? thx! miha

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Help needed with audio test setup
by Victor Chukalovskiy 30 Sep '13

30 Sep '13

Hello Community, The problem is that a 3-rd party SIP hardware causes audio isssues in less then 1% of the cases. Need to do 100's of manual test calls to identify it and bring to the vendor. Is there a quick recepie to make the following FS test setup? Should be as simple as possible. If someone has something ready-made to offer, please contact me off-list. Here is the idea: -FS generates test SIP call and keeps it up for N minutes. -One side initates the call and sends test tone at X Hz -another side (also FS) answers the call and sends test tone at Y Hz -both sides listen to the tone coming from the other side. -If tone in either direction degrades into siginificant noise or silence, FS leaves the call up indifentely long AND sends email notification "got the sucker!" -If in N minutes tone in both directions did not degrade, FS hangs up the test call and initates a new one. Infinite testing cycle until we turn it off. Thank you, Victor

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estimate the sustainable load
by crazygabry 30 Sep '13

30 Sep '13

Hello everybody! i'm setting up a multi tenant environment on the cloud. In order tu understand how many companies can i put on a single server, i need to evaluate how many simultaneous calls can i manage. Can you please suggest a way to do so? Thanks in advice -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/estimate-the-sustainable-load… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Automatically don't respond to abnormal registrations
by Sayyed Mohammad Emami Razavi 30 Sep '13

30 Sep '13

Can anyone describe a way in FS to get rid of crawlers and hackers testing your PBX second by second on the internet?! What is the problem? This is ls of my FS's sqlite db after some annoying behaviours of hackers and crawlers violently done: [root@freeswitch db]# ll total 705580 -rw-r--r-- 1 root root 7168 Sep 29 13:20 callcenter.db -rw-r--r-- 1 root root 14336 Sep 23 10:29 call_limit.db -rw-r--r-- 1 root root 257024 Sep 29 13:56 core.db -rw-r--r-- 1 root root 5120 Sep 29 13:20 fifo.db -rw-r--r-- 1 root root 98304 Sep 24 11:34 sofia_reg_sipinterface_2.db -rw-r--r-- 1 root root 367616 Sep 25 18:31 sofia_reg_sipinterface_3.db -rw-r--r-- 1 root root 500726784 Sep 29 13:56 sofia_reg_sipinterface_5.db -rw-r--r-- 1 root root 221003312 Sep 29 13:56 sofia_reg_sipinterface_5.db-journal -rw-r--r-- 1 root root 16384 Sep 23 10:29 voicemail_default.db [root@freeswitch db]# sofia_reg_sipinterface_5.db belongs to the interface listening on 5060 port. As you see sofia_reg_sipinterface_5.db exceed 500MB of size! and this db has crashed! and leads to drop port 5060 of interfaces. when i ask fs_cli about profiles it tells me: +OK log level [7] freeswitch@internal> sofia status Name Type Data State ================================================================================================= sipinterface_3 profile sip:mod_sofia@192.168.2.73:5080 RUNNING (0) voicemail_1 alias sipinterface_2 ALIASED voicemail_2 alias sipinterface_2 ALIASED sipinterface_2 profile sip:mod_sofia@192.168.2.73:5070 RUNNING (0) ================================================================================================= 2 profiles 2 aliases Which open source firewalls can handle automatically weird sip requests to ban?! Sincerely yours,

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TLS with Cisco SPA112
by Nick Vines 30 Sep '13

30 Sep '13

Has anyone successfully setup the spa112 or spa122 to use TLS? I tried setting my spa112 up following the generic TLS guide<http://wiki.freeswitch.org/wiki/SIP_TLS>, but I am getting the following repeated error when I use `sofia loglevel all 9`: tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fb6d40054c0): events IN tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fb6d40054c0): new secondary tport 0x7fb6d40e9b00 tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fb6d40e9b00): new connection from tls/my_ip_address:my_port/sips tport_tls.c:869 tls_connect() tls_connect(0x7fb6d40e9b00): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0x7fb6d40e9b00): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2084 tport_close() tport_close(0x7fb6d40e9b00): tls/my_ip_address:my_port/sips I double checked that TLS would work using fsclient to connect to my server, and that connected instantly. Thanks, Nick

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Transfer A-Leg in a callcenter queue bridged to an agent to another queue
by Mimiko 30 Sep '13

30 Sep '13

Hello. I am wondering if this is possible at all. For example the caller selects and enters a queue Que1 in a callcenter application. This queue has recording of conversation enabled. Sometime agent which was bridged to this call need to transfer the caller to another queue, due to caller's error on selecting queue in IVR. Initially I enabled simple transfer with *1N# from default dialplan example. This is working fine to transfer to some other number in the system directly, and while first agent terminates conversation, the recording of conversation continues already with the new B-Leg connection. Its good, because we can track whole conversation of the client. But in case of transferring to another queue there will be two recording files, one started from Que1 and another started from Que2 to which the caller was transferred. How can the caller be transferred to other queue and disconnect from first queue? Other queue also needs recording. The caller can use IVR to get to Que2 for example. Thank you. -- Mimiko desu.

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(no subject)
by Rafal Gwizdala 30 Sep '13

30 Sep '13

Hi Guys, i have another question regarding mod_httapi Mod_httapi works in 'batches' - it requests a batch of work from an external http application and executes it. When done, it requests another batch from the application and the procedure repeats until hangup. Now my question is if it's possible to break a batch in a middle and make mod_httapi request a new batch immediately. By 'breaking in a middle' I mean an immediate halt/interrupt to any playback in progress or other operations that can be running (pause, recording, etc). My goal is to provide a way to interrupt ivr script execution as soon as some event occurs and start another script in that place. Can you provide some ideas? Thanks RG

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Abandoned Channel/Sessions/Threads after hold/unhold
by Frank Park 30 Sep '13

30 Sep '13

Latest 1.2.stable is having issues with resuming a previously held calls. I am experiencing that if I try blind transferring an unheld call, the call drops. After staring at the logs for awhile, I am seeing that calls are dropping because the call's UUID no longer "exists", even though channel data still exists in the DB and the process thread is still alive (looking at the process list). To reproduce this problem, answer a call, put them on hold, pick it back up, then blind transfer the call to another extension. Call should drop, the "show calls" on CLI will continue to show you the call as active. On a same note, if you hold/resume a call more than 3 times, a caller no longer hears MOH, which I think it's being caused by the same issue. Any help would be awesome! Frank

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Impossible to Run the Conference call part
by Labolinux | Sonerep 30 Sep '13

30 Sep '13

Dear Friends, I'm interested by voip solution and especially by FreeSwitch and FusionPbx since some days. So I installed Freeswitchand Fusionpbx on Freebsd 9.0. and everything is ok. I already configured internal extensions and logged on with softphones. AndInternal calls are going well. Now I want to configure a conference room but found some problems doing it. When I make a call to the conference room, the call is going automatically to the the conference room but enabling the "Status Active Calls" instead of Active Conference. I was really blocked and i don't know how solving or making it run well. Please, somebody can explain me what is wrong in my configuration or help me to find how solving this problem? I joined to this email in attached file the log of my conference room. Thanks for your understanding and assistance. Sylvain Gotri Labolinux 14, Rue des Callitris, Tokoin Solidarité Lomé - Togo E-Mail: noc(a)sonerep.com

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