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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 23 Sep '08

23 Sep '08

Well what I really should say is that adding many variables to the bridge data, such as this: data="[origination_caller_id_number=45161061,my_var1=blah,my_other_var=w hatever]sofia/gateway/45161061/$1"/> would be very nice. My problem is that I cannot verify that the variables are passed by the info application in an execute_on_answer code done when the call has been answered. The variable is not defined at that stage? Is it lost? /Jon

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 23 Sep '08

23 Sep '08

Resolved: The problem has been solved after updating to the last trunk: 1.0.1, 9624. The <action application="bridge" data="[origination_caller_id_number=45161061]sofia/gateway/45161061/$1"/ > works, but it would be nice to be able to add more variables than this only one. Jon

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store sofia profile_name and hostname in sip_registrations
by Leon de Rooij 23 Sep '08

23 Sep '08

Hi all, I'd like to keep track of on what switch and sofia profile_name a UA is registered. By default, FS automatically creates a table sip_registrations without these fields, so I started looking in mod_sofia to see whether the fields could be added, and to my surprise, I saw they should already be inserted (along with presence_hosts) ! sofia_reg.c in function sofia_reg_handle_register at line 774: sql = switch_mprintf("insert into sip_registrations " "(call_id ,sip_user ,sip_host ,presence_hosts ,contact ,status ,rpid ,expires,user_agent,server_user,server_host,profile_name,hostname) " "values ('%q','%q', '%q','%q','%q','%q', '%q', %ld, '%q', '%q', '%q', '%q', '%q')", call_id, to_user, reg_host, profile->presence_hosts ? profile->presence_hosts : reg_host, contact_str, reg_desc, rpid, (long) switch_timestamp(NULL) + (long) exptime * 2, agent, from_user, guess_ip4, profile->name, mod_sofia_globals.hostname); (Is that the right spot ?) So I tried just adding the fields as VARCHAR(255)'s to the table, and restarted the switch. It coredumped (for the first time btw), so I guessed the NULL values for the already present registrations were to blame, so I emptied the sip_registrations table and restarted. It started alright, but then gave errors on registrations this: 2008-09-23 17:57:26 [ERR] sofia_glue.c:2880 sofia_glue_actually_execute_sql() ERR: [insert into sip_registrations values ('446BD213B8D1111(a)172.16.42.124', '00A0C5D4C03A-1','scarlet.nl','"user" <sip: 00A0C5D4C03A-1@213.204.223.35:63747;fs_nat=yes>','Registered(UDP- NAT)', 'unknown', 1222192646, 'dunno', '00A0C5D4C03A-1', '213.204.232.18')] [STATE: HY000 CODE 1136 ERROR: [unixODBC][MySQL][ODBC 3.51 Driver] [mysqld-5.0.51a-3ubuntu5.1]Column count doesn't match value count at row 1 ] How can this be ? This query is very different from the one before... Is it already possible to save the hostname and sip_registrations (and presence_hosts) somehow ? thanks, Leon de Rooij leon(a)scarlet-internet.nl

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Slightly OT: SIP Article
by Michael Collins 23 Sep '08

23 Sep '08

Guys, I know how much you all just *love* SIP, so I thought you might be interested in chiming in on this article: http://blogs.zdnet.com/carroll/?p=1877 -MC

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Re: [Freeswitch-users] Problems with configure script
by Jon Bruel 23 Sep '08

23 Sep '08

Resolved. Thanks, Michael, for the help. For the sake of the community the problem likely was erroneous trunk download. /Jon.

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SQLite Locked Database
by Jim Flowers 23 Sep '08

23 Sep '08

I get '[ERR] switch_core_sqldb.c:95 switch_core_db_persistant_execute_trans() SQL ERR [database is locked]' just following '[NOTICE] switch_loadable_module.c:393 switch_loadable_module_process() Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1]' on freeswitch 1.0.trunk(9609) startup. Is there anything I can do to correct this? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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Execute on answer only executes once?
by Jon Bruel 23 Sep '08

23 Sep '08

I have set up the internal extensions to execute on answer. When I from A call B, it works OK. If I make an unattended transfer from B to C, thus setting up a connection between A and C, it does not execute when C answers. It this intentional, and is there a way around it to get it to execute every time? /Jon

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Re: [Freeswitch-users] Problems with configure script
by Jon Bruel 23 Sep '08

23 Sep '08

I have tried several times to bootstrap. The result is the same every time. Could it be some errors related to different copies of the programs used - some under /usr/local/share, some under /usr/local? And from other users: have you experienced the same problem? /Jon

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Re: [Freeswitch-users] Problems with configure script
by Jon Bruel 23 Sep '08

23 Sep '08

I do have c++ installed, and the release version 1.0.1 did install OK. On the other hand 1.0.1 came with the configure script, in this version configure was build by bootstrap. I can't answer what the difference of the trunk version compared to version 1.0.x is. Anyhow I have the same problem with the snapshot from 20-09-2008. Somehow I think the AX_COMPILER_VENDOR is undefined when called, but I'm not at all into the autoconf-world, and I waste time trying to figure out what's going on.. /Jon

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Freeswitch with Audiocode Mediant 2000
by Gopal krishnan 23 Sep '08

23 Sep '08

Hi, I followed the below link to configure the Audiocode Mediant 2000 with Freeswitch http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/11… but the above link is for FXO line, where I am using digital PRI line. when I try to dial I am getting call failed, the traffic from freeswitch were hitting audiocode the log as follows, attached with this email, *some sample SIP header as follows,* d:2h:17m:7s INVITE sip:9894929942@172.20.176.254<sip%3A9894929942(a)172.20.176.254>SIP/2.0 Via: SIP/2.0/UDP 172.20.176.31;rport;branch=z9hG4bKKmB9HrNr22HZQ Max-Forwards: 69 From: "Extension 1002" <sip:9894929942@172.20.176.31<sip%3A9894929942(a)172.20.176.31> >;tag=j9a4e9Q4ycvtr To: <sip:9894929942@172.20.176.254 <sip%3A9894929942(a)172.20.176.254>> Call-ID: 7702517d-0413-122c-efab-0019d150d051 CSeq: 104969298 INVITE Contact: <sip:mod_sofia@172.20.176.31:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9596M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 347 Remote-Party-ID: "Extension 1002" <sip:9894929942@172.20.176.31<sip%3A9894929942(a)172.20.176.31> >;screen=yes;privacy=off 1d:2h:17m:7s ( sip_stack)(212 ) ?? [WARNING] AcSIPParser: Unrecognized Header was detected at line: 12 1d:2h:46m:9s ( lgr_TrnkGrp)(344 ) !! [ERROR] #1:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 9894929942 1d:2h:46m:9s ( lgr_psbrdif)(345 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:9894929942 Source:9894929942 SourceIp:ac14b01f 1d:2h:46m:9s ( lgr_psbrdif)(346 ) TrunkBoard::GetEndPoint- Current trunk status:0010 1d:2h:46m:9s ( lgr_call)(347 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number 9894929942 *Freeswitch log* *as follows* http://pastebin.freeswitch.org/5635 So how to proceed in this stage. -- Thank you with regards, Gopal,

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