March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Ghost Sessions in CLI after a longterm test
by rod 04 Mar '09

04 Mar '09

Hi All, I ran some longer tests with FS 1.0.3 acting as an SBC. The test machine has the following specs: - Intel Quad Core Q9550 - 8GB RAM (far too much from what I saw) After 3 days running SIPP with 750 simultaneous calls (1500 channels) at 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. In the CLI, using status command I got this: freeswitch@internal> status UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, 607 microseconds 15817560 session(s) since startup 22 session(s) 0/500 But when I use "show channels" or "show calls", I see nothing. So I'm wondering where are these 22 sessions ? FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. Successful call --> 5271434 Failed call ---> 1554 (less than 0.03%) regards, rod. complete SIPP summary: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 10.10.10.254:5060(UDP) 0 new calls during 0.856 s period 7 ms scheduler resolution 0 calls (limit 750) Peak was 750 calls, after 15 s 0 Running, 34 Paused, 0 Woken up 15544 out-of-call msg (discarded) 1 open sockets 9213070274 Total echo RTP pckts 1st stream 0.000 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 ----------> 5273022 0 0 100 <---------- 5273022 0 1554 180 <---------- 0 0 0 183 <---------- 0 0 0 200 <---------- E-RTD1 5271434 0 0 ACK ----------> 5271434 0 Pause [ 35.0s] 5271434 0 BYE ----------> 5271434 0 0 200 <---------- 5271434 0 0 ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2009-02-27 09:11:31 Last Reset Time | 2009-03-02 07:49:10 Current Time | 2009-03-02 07:49:11 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:857 | 70:37:39:429 Call Rate | 0.000 cps | 20.739 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 5273022 Total Call created | | 5273022 Current Call | 34 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 5271434 Failed call | 0 | 1554 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000 | 00:00:00:240 Call Length | 38:32:13:386 | 00:00:36:131 ------------------------------ Test Terminated --------------------------------

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Mandatory IE missing
by Ali Al-Rubaie 04 Mar '09

04 Mar '09

Hi, During proxy authentication, I got "mandatory IE missing" error in the response as shown below. How this error can be resolved? recv 585 bytes from udp/[209.82.10.250]:1898 at 15:31:39.933782: ------------------------------------------------------------------------ INVITE sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Contact: sip:1002@209.82.10.250:1091 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 INVITE Content-Length: 187 Content-Type: application/sdp User-Agent: HelpCaster SoftPHONE Supported: com.hearme.mux v=0 o=HelpCaster 153553387 153553387 IN IP4 192.168.10.31 s=HelpCaster c=IN IP4 209.82.10.250 t=0 0 m=audio 8002 RTP/AVP 0 4 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.934100: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Content-Length: 0 ------------------------------------------------------------------------ send 746 bytes to udp/[209.82.10.250]:1091 at 15:31:39.938913: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=K83mZmZH8g3Hr Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="209.82.10.235", nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54" Content-Length: 0 ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.949517: ------------------------------------------------------------------------ ACK sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=K83mZmZH8g3Hr Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 584 bytes from udp/[209.82.10.250]:1898 at 15:31:39.953137: ------------------------------------------------------------------------ INVITE sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Contact: sip:1002@209.82.10.250:1091 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 INVITE Content-Length: 187 Content-Type: application/sdp User-Agent: HelpCaster SoftPHONE Supported: com.hearme.mux Proxy-Authorization: Digest username="1002",realm="209.82.10.235",nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response=" d1614cd024acab0b794751285f5cb1fe",uri="sip:9999@209.82.10.235" ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.953336: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Content-Length: 0 ------------------------------------------------------------------------ send 696 bytes to udp/[209.82.10.250]:1091 at 15:31:39.956016: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=mHXD1FgN5SS4K Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" Content-Length: 0 ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.964632: ------------------------------------------------------------------------ ACK sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=mHXD1FgN5SS4K Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 ACK Content-Length: 0 Thanks!

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Qt portaudio interface
by Mathieu Rene 04 Mar '09

04 Mar '09

Hi all, Anyone interested in contributing to a Qt interface in order to make a decent softphone using FS please reply to this thread. (also give your availability so we can have a conference call to decide stuff) Thanks, Math

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First time test set up FreeSwitch and SPA3102/SPA3000
by Aplayful Idiot 03 Mar '09

03 Mar '09

I have no background in telephony but probably need to use a PBX. FreeSwitch was recommended by a casual contact so I would like to start first by setting up a small test. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and it works ok but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as <variable name="effective_caller_id_number" value="line1"/>. Looking back over the FS wiki, I'm now wondering if the SPA should of been set-up as a gateway but reading it is confusing at least to me. Sometimes I think the analogue-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? api

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Detecting the origin of voice activity using VAD
by Cameron Sorlie 03 Mar '09

03 Mar '09

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hide caller_id_number, transfer and then change caller_id_number
by Peter P GMX 02 Mar '09

02 Mar '09

Hello, I have the following problem while providing callback (mod_eventsocket is used): 1) I want to call a certain destination number A with a suppressed caller_id_number (this works fine with some vars in the origination string) 2) The destination number A picks up the phone and enters a target number B by DTMF 3) freeswitch then forwards the call to target number B by DTMF and I want to show the number A. I do this with uuid_setvar. The problem is that it still shows unknown. This is all with SIP. uuid_setvar however worked if I did not set the caller_id_number to unknown. Per default this is then "00000000000" and can then be changed with uuid_setvar to the number of A. But if I set caller_id_number to unknown I can no longer change it to A. Any hint? Best regards Peter

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Howto config early dial
by Sergio Alecha 02 Mar '09

02 Mar '09

In asterisk, with the parameter AMPBADNUMBER = FALSE it is possible to use "early dial" Grandstream telephones. How do Freeswitch in? thank you very much.

1 0

Sangoma USB FXO U100 + Asus notebook?
by Fred 02 Mar '09

02 Mar '09

Thanks guys for the feedback. So, the OpenZap driver isn't ready for production yet?

2 1

First time setting up FreeSwitch and SPA3102 / SPA3000
by Aplayful Idiot 02 Mar '09

02 Mar '09

Hi. I have little background in telephony and need to use a PBX but would like to start first with a small test set-up. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and this works to a point but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as <variable name="effective_caller_id_number" value="line1"/>. Looking back over the FS wiki, I'm now wondering if the SPA was registered or set-up in FS correctly but reading the documentation is confusing me a bit. Sometimes I think the analogue-phone-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? Thanks. api

1 0

Orginate: getting status of call fail
by Nik Middleton 02 Mar '09

02 Mar '09

Hi Guys, I've been running a test script written in lua which now works very well thanks to Anthony's fix to stream file. Right now I'm using an event socket to initiate the call and passing the name of the script along with originate thus: $dialstring = "originate {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/ Mygw/phonenum '&lua(helloworld.lua )'"; $result = $obj ->bgapi_command($dialstring); The script gets fired (it would appear) on answer. However, if the number is invalid , timed out or was busy, I'm not sure the script gets executed or am I wrong? I want to be able to fire an event back on what happed to the call in the event that it failed for whatever reason. I know I can simply call the originate and pass the number as an argument and execute the dial within the script but I'm led to believe that's not very efficient, or am I completely wrong? Looking for the most FS friendly way here Regards,

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