January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

database disk image is malformed ISSUE
by Nikolai Geordzhev 18 Jan '10

18 Jan '10

Hi Guys, I have an issue playing with FS as a registrar server. I have made some tests with 3000 subscribers registering every 600 seconds running for days and everything went fine. Then I have tried with 6000 subscribers registering every 3600 seconds and after some time ( between 1 and 2 days) I received [ERR] switch_core_sqldb.c:662 SQL ERR [database disk image is malformed] message in the freeswitch.log file. The only solution I have found is deleting the db folder ( mounted in tmpfs) and restarting the application. When I measure the packets/sec rates of both setups i see 13 pack/s for the first setup and 7 pack/s for the second one. Can somebody advise what can cause this Error and if there is some kind of a solution. Thanks and best regards

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voicemail->email
by François Legal 18 Jan '10

18 Jan '10

Sorry to come back on this topic, but I could not manage to fix this, and I don't know what to do. voicemail to email was working on FS 1.0.3, but not anymore since upgrade to 1.0.4 (now running1.0.5-20100112-0400 (hacked)) When FS sneds the message, sendmail segfaults. Running FS as root or standard user does not change this. Modifying FS config to run a script like exec tee -a /tmp/fsmail.log | /usr/sbin/sendmail -O LogLevel=7 -t >> /tmp/fsmail.log 2>&1 make sendmail segfaults too. then doing su -l freeeswitch then cat /tmp/fsmail.log | /usr/sbin/sendmail -f freeswitch(a)mydomain.com -t calle(a)mydomain.com succeeds. Don't know what to look for now. The system was not (except freeswitch) updated between running with FS 1.0.3 and 1.0.4. Anybody can help ? François

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Mod_Fax
by Costa Zikalala 18 Jan '10

18 Jan '10

Hi All Has anyone worked on a email2fax script for mod_fax? If not how much would it cost for some genius here to quickly whip-up one? Ideally both email2fax and fax2email should come standard with mod_fax or is this not possible? Thanks Costa

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Multiple registrations: Only calling the first element in the list in this mode
by Peter P GMX 18 Jan '10

18 Jan '10

Hello, I habe the following behaviour when I call a user which is registered twice with 2 phones via bridge user/100@domain both phones are ringing. This is correct as I allow multiple registrations in a profile However when I call multiple endpoints via bridge user/100@domain,user/101@domain,user/102@domain only one phone with number100 is ringing. Console log shows "Only calling the first element in the list in this mode.": 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable string 0 = [presence_id=100@domain] 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable string 1 = [transfer_fallback_extension=100] 2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only calling the first element in the list in this mode. 2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel sofia/internal/sip:100@10.11.12.203:2048 [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61] Is there any way to work around this? I need all phones to be ringing in this scenario. Best regards Peter

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eavesdrop problem?
by Nikolay Kondratyev 18 Jan '10

18 Jan '10

Hi all, I want to use eavesdrop application. Playing with it I found that when one tries to eavesdrop caller the feature works ok. But when trying to eavesdrop callee eavesdrop attempt failes. I just updated to the latest version from http://latest.freeswitch.org <http://latest.freeswitch.org/> [freeswitch@freeswitch log]$ fs_cli -x version FreeSWITCH Version 1.0.5-20100115-0400 (16318M) My setup is as following: I have 3 internal extensions: 2853, 2840, 2879 (all are xlite). 1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected. 2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. I attached logs for both cases. I don't believe it's intended behavior. Can anybody please advise if it is a configuration or a software problem? Thanks and regards, Nikolay.

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Questions about mod_fifo
by Jingwei Yang 18 Jan '10

18 Jan '10

Hi Guys, I'm implementing an ACD system using ESL and mod_fifo. Based on what Anthony suggested in this post: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg01652.… *You can make an event socket application that listens for FIFO events and keeps track of what FIFOs are currently busy and when there are people waiting you can have that script generate a call to a group of SIP phones so when the first one answers, it sends them in as an agent where they can field the calls. * 1. How should I handle the concurrent issues? If I bridge a user to two agents and both of them answers, how does FS take care of this situation? Will a slower agent get a busy tone automatically? 2. If the socket application is brought up after some users have called in, what command should I use to check the busy queues? fifo list? 3. Am I using fifo list and fifo count correctly? here's the testing dialplan: <extension name="in_fifo"> <condition field="destination_number" expression="^779$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="answer"/> <action application="fifo" data="*myq *in"/> </condition> </extension> when a call comes in and gets queued, these are the results of some commands I tried. freeswitch(a)localhost.localdomain> fifo list API CALL [fifo(list)] output: <fifo_report> <fifo name="cool_fifo(a)192.168.1.104" consumer_count="0" caller_count="0" waiting_count="0" importance="0"> <callers></callers> <consumers></consumers> </fifo> </fifo_report> freeswitch(a)localhost.localdomain> fifo list myq API CALL [fifo(list myq)] output: <fifo_report></fifo_report> freeswitch(a)localhost.localdomain> fifo count myq API CALL [fifo(count myq)] output: none It seems *myq* doesn't get created at all? Please enlighten. Thanks and best regards, -Jingwei

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IMPORTANT -- Call for bug updates
by Michael Jerris 18 Jan '10

18 Jan '10

To all FreeSWITCH users- If you have bugs open on http://jira.freeswitch.org please login today and post a status update on these bugs (even if they appear to be awaiting comment by the development team). If you have a patch, please update these patches to svn trunk so that they may be reviewed. I know many patches have been sitting for quite some time but I will make a strong effort to review and merge patches that are ready to go in. If you have a bug, please update to svn trunk and comment if the issue still exists or is now fixed in trunk. If you bug has a comment requesting more information, please provide it. If you don't have any open bugs, and are not currently using recent svn trunk, I would appreciate it if you could carve a little bit of time out of your days and test out trunk. Feel free to look through jira and find a bug you are interested in and test it as well. We are working towards having the most stable and feature rich release of FreeSWITCH yet and we need your support and assistance to do so. As always, if you have a new bug to file, please do so as soon as possible and try to get as much information as possible on the bug. There are some good guidelines for reporting at http://wiki.freeswitch.org/wiki/Reporting_Bugs As always, many thanks to the community for all your hard work and support. Mike

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Re: [Freeswitch-users] Trouble with Originate/Bridge from ESL [SOLVED]
by Max Bridgewater 18 Jan '10

18 Jan '10

Given that I'm using Java, I had to escape the ringback value's quote twice: originate {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52) On Sun, Jan 17, 2010 at 3:51 PM, Max Bridgewater <max.bridgewater(a)gmail.com>wrote: > One more piece of information: the call is being terminated by Freeswitch > with the event: > > Event-Name: CHANNEL_HANGUP > Hangup-Cause: NO_ANSWER > > Which is strange because B leg doesn't even have the time to answer. > > > > On Sun, Jan 17, 2010 at 1:55 PM, Max Bridgewater < > max.bridgewater(a)gmail.com> wrote: > >> Hi, >> >> The following command works great from the command line: >> originate >> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52 >> &bridge(sofia/internal/1005%74.63.243.52) >> >> But this one isn't working from the ESL: >> api originate >> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52 >> &bridge(sofia/internal/1005%74.63.243.52) >> >> The observed behavior is following: The A leg is dialed, then B leg is >> also dialed but immediately followed by a hangup. That is, the B user >> doesn't even has time to answer. >> >> Any idea what I'm doing wrong? >> >> Thanks, >> Max. > > >

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No media after Originate
by Mohamed Hassan 18 Jan '10

18 Jan '10

Hi Freeswitch users thank you in advance as i am new to freeswitch iam facing a problem while writing a Callback service i tried to bridge the two legs through one sip provider the call estaplished successfully but without media passing and after some searching i found that there is a bug with sofia loopback so i shoulkd handle the two legs with different sip gateways and after trying i found the same error using cli : originate {ignore_early_media=true}sofia/gateway/prov/00xxxxxxxxxxx 9999 XML callback and here the 9999 context: <context name="callback"> <extension name="9999"> <condition field="destination_number" expression="9999"> <action application="sleep" data="5000"/> <action application="answer"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="[ignore_early_media=false]sofia/gateway/prov/00xxxxxxxxxx"/> </condition> </extension> </context>

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Change Domain Freeswitch
by Edmar Cruz 18 Jan '10

18 Jan '10

Dear All, How can i change the domain of my freeswitch 52.236.125.12 to sip.grandminister.com to be able to detect the presence of the user whos online or not... Thanks Edmar -- View this message in context: http://old.nabble.com/Change-Domain-Freeswitch-tp27104680p27104680.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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