June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

XML-RPC
by Santhosh 12 Jun '09

12 Jun '09

Hi, Is there any where I can find more documentation on the XML-RPC interface of freeswitch. I am trying to initiate a conference and add in users to it from flex. I would appreciate any insights. Thanks San

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Unregister extension?
by Lars Zeb 12 Jun '09

12 Jun '09

How can I unregister a softphone's registration? I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I changed the second one to 1000. Now when I do 'sofia status profile internal' all three show up. How do I get rid of the 1001 extension? I shutdown and restarted FS but that didn't do it. I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is blocking the Polycom at that same extension and that is the reason the Polycom is not showing. Thanks, Lars Registrations: ============================================================================ ===================== Call-ID: 3c267015ab6b-bd6gioq5ytor User: 1010(a)192.168.10.29 Contact: "1010" <sip:1010@192.168.10.104:2048;line=dg4k4xql> Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1010 Auth-Realm: 192.168.10.29 Call-ID: 3c267015afa6-6v0sw4o3qei3 User: 1001(a)192.168.10.29 Contact: "1001" <sip:1001@192.168.10.104:2048;line=co52ym3a> Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1001 Auth-Realm: 192.168.10.29 Call-ID: OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. User: 1019(a)192.168.10.29 Contact: "1019" <sip:1019@192.168.10.11:19040;rinstance=5394acb4dfa00c0a> Agent: Bria Professional release 2.4.3 stamp 50906 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) Host: fs IP: 192.168.10.11 Port: 19040 Auth-User: 1019 Auth-Realm: 192.168.10.29 Call-ID: 3c270d667ff5-47fq2p6n1ou1 User: 1000(a)192.168.10.29 Contact: "1000" <sip:1000@192.168.10.104:2048;line=gzdwwjqr> Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1000 Auth-Realm: 192.168.10.29 ============================================================================ =====================

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Database and "Too many open files" Problem
by Helmut Kuper 12 Jun '09

12 Jun '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I updated to latest trunk. Everything compiled well. FS starts up without errors. Calls are executed successfully. But after a few minutes with very few call activity I get this lines on console: 2009-06-10 18:21:16.52527 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) 2009-06-10 18:21:16.152501 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) 2009-06-10 18:21:16.252481 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) ... 2009-06-10 18:22:37.121405 [CRIT] switch_core_sqldb.c:209 SQL thread unable to commit transaction, records lost! I modified switch_core_sqldb.c to output the actual sql statement as well to get an idea which database is affected. It's the core-db. Mixed with that lines above I got lines like this: 2009-06-10 18:21:32.656605 [ERR] mod_sndfile.c:194 Error Opening File [/opt/app/voip/ippbx/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav] [System error : Too many open files in system.] This may points to the ulimit thing, cause I run FS as non root (on centos 5.3). ulimit -a shows: [ippbx@ippbx-test-node0 ~]$ 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) unlimited max locked memory (kbytes, -l) 32 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) 10240 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited /etc/security/limits.conf contains: ippbx - nofile 999999 ippbx - core unlimited ippbx - data unlimited ippbx - fsize unlimited ippbx - sigpending unlimited ippbx - msgqueue unlimited ippbx - nproc unlimited ippbx - locks unlimited So has anyone an idea what's wrong? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKL+454tZeNddg3dwRAqcCAJsFSEWuI2X1fvOF2rmIFoSmRdBZzQCggD8B MQiS7RZkOoL9lhV8LV+pL7s= =v7QY -----END PGP SIGNATURE-----

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Dialplan XML phone number match
by Lars Zeb 12 Jun '09

12 Jun '09

I have a match expression for outbound calls as "\d{10}". It's fine for unformatted numbers. Not knowing any better, I created another extension to handle numbers formatted like XXX-XXX-XXXX, which is easier to read and exists in one hard phone's phonebook. It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many extensions for different formats. There's got to be a better way. Any suggestions? Thanks, Lars

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Status Event
by Nik Middleton 11 Jun '09

11 Jun '09

Not sure where enhancement requests should be posted, but here it is anyway I would dearly love to be able to send a status event that returns an event style output that provides machine readable output rather than the wordy human readable response. (I hate parsing) Is there such an event already? Regards

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Orphaned calls
by Nik Middleton 11 Jun '09

11 Jun '09

Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 sessions remaining like the one below (number and ISP changed) Anyone have an idea why these 6 sessions remain? I also had 120 calls that I didn't get a hang-up for, but that might be me not processing the events fast enough. That said, FS was handling a steady concurrent call level of around 350 which was awesome !! Regards UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 microseconds 86913 session(s) since startup f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 18:38:23,1244741903,sofia/external/0XXXXXXXX@gk.myISP.net,CS_NEW,,,,,,,, ,,,,,

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re direct calls
by NOx-WHV 11 Jun '09

11 Jun '09

Hello Freeswitch User! I am using FS since a few weeks. My intent is to have clients who uses TLS and SRTP for a full encrypted call. I just managed it, that calls are encrypted with TLS and SRTP. My second aim ist to redirect this calls for reduce the processing of the server. I only use the FS for calls between users of this freeswitch. I just tested a "redirect" in the dialplan (without TLS and SRTP) but it doesn´t work. How i have to configure the dialplan for redirect the call to the other user. I use a SNOM hardphone and a phonerlite softphone. Thanks for sour help! NOX -- View this message in context: http://www.nabble.com/redirect-calls-tp23982889p23982889.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Caller id when doing transfers
by John Wehle 11 Jun '09

11 Jun '09

> It appears from some limited testing that the original caller id is always > shown when the call is transfered. Is there some way to have the person > making the transfer show up as the caller id? To answer my own question it appears that the information is available in the sip_h_Referred-By variable. E.g.: <extension name="system25_park"> <condition field="destination_number" expression="^\*5$" /> <condition field="${sip_h_Referred-By}" expression="^<sip:([0-9]{4})@.*$"> allows the station id making the transfer to be known when a call is transfered to *5. The station id can then be used to park the call in the proper fifo. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Dialplan extension using Caller-ID-Name not matching condition
by Larry Marshall 11 Jun '09

11 Jun '09

http://pastebin.freeswitch.org/9365 I do not know what I am doing wrong. I am trying to set the effective_caller_id_name and _number depending on the originating extension. I tried: <condition field=${caller_id_number}$ expression="^100[09]$" break="on-true"> and <condition field=${Caller-ID-Name}$ expression="^100[09]$" break="on-true"> and <condition field=${sip_from_user}$ expression="^100[09]$" break="on-true"> But each got substituted with the name of the extension in the log: Dialplan: sofia/internal/1000(a)192.168.10.29 Regex (FAIL) [Long Distance - flowroute] () =~ /^100[09]$/ break=on-true where the extension looks like: <extension name="Long Distance - flowroute"> Info from the log shows variable_sip_from_user: [1000] Caller-Caller-ID-Number: [1000] Can anyone help? Thanks, Lars

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Can't hear outbound calls
by Lars Zeb 11 Jun '09

11 Jun '09

I had a working FS installation which I messed up by doing a fresh install. I tried to integrate all my custom changes, but I'm sure I screwed something up. The symptom is on an outbound call, sometimes I can hear ringing, other times I cannot. Finally I can see FS connects via a softphone, but I hear only silence. The other side of the conversation hears static. I pasted a siptrace of the external profile. The Contact, Via and SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 FS exists on a LAN behind a NAT firewall along with all its clients. There is a SwitchVox system which predates the FS. I had to use an external sip = 5090 for FS. Also I think I had to use a different WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came up with (xxx.xxx.xxx.82), but I can't figure out where I set this address. I would appreciate any help. Thanks, Lars

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