December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_xml_ldap compile issue.
by Keith Laaks 18 Dec '09

18 Dec '09

Hi, I am having an issue getting mod_xml_ldap to compile properly.... <cut-cut> making all mod_xml_cdr making all mod_xml_ldap Creating mod_xml_ldap.la... /usr/bin/ld: /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_xml_ldap.log: No such file or directory make[5]: *** [mod_xml_ldap.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 ------------------------------------------------------------------------ I notice the openldap library has been bumped up to .19 - not sure if that may have anything to do with it. At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. mod_ldap compiles OK, but mod_xml_ldap fails as per the above. What am I doing working here ? Best Regards Keith

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How can I join two freeswitch on two servers?
by Edmar Cruz 18 Dec '09

18 Dec '09

I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Destination Formats Expression
by Edmar Cruz 18 Dec '09

18 Dec '09

Hi Everyone, Is there a link or tutorial for the expressions format. Example: <condition field="destination_number" expression="^(10[01][0-9])$"> 10 - default number [01[ - second number that start only on 0 or 1; [0-9] - 0 to 9 can be use Is there any? Thanks, Edmar -- View this message in context: http://old.nabble.com/Destination-Formats-Expression-tp26840010p26840010.ht… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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how does FS failover or load balance outbound calls between tow proxy
by Lei Tang 18 Dec '09

18 Dec '09

Hi All I have a FS cluster behind two OpenSIPS proxy, the incoming calls is load balance and failover to FS cluster by OpenSips, It works well. The problem is, the outbound calls from FS must also route throw then OpenSIPS servers. So, does FS servers can loadbalance the outbound calls between the two OpenSIPS servers and failover if one of the Opensips server is down? -- Lei.Tang lei.tlfly(a)gmail.com

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Fwd: incoming call
by srinivasula reddy 18 Dec '09

18 Dec '09

Hi, i have up the freeswitch with domain(eg sipserver.domain.com) name instead of local ip, two clints are regitered with freeswitch using domain name(eg sipserver.domain.com) one client is making a call to other one, other clint receiving a invite request like this 173927 3120.658532 10.91.154.108 10.91.154.80 SIP/SDP Request: INVITE sip:1010@10.91.154.80:5061, with session description, but usually it should come with INVITE sip:1010@sipserver.domain.com<sip%3A1010(a)sipserver.domain.com>? what is changes i need to do for this? any idea? Regards-- Srinivasula Reddy K -- Srinivasula Reddy K

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Creating Default Accounts on Directory
by Edmar Cruz 18 Dec '09

18 Dec '09

Hi Sir, I want to create a new xml file on the default directory of freeswitch where 1000.xml is located, sample i created 9387821.xml and copy the contents of the 1000.xml. The problem is when I used the account 9387821.xml and call 1000.xml it doesn't work the message in freeswitch it always CS_DESTROY... Please help me this with issue thanks... Edmar -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p268… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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sip message logging and analysis
by Frank ＠ Impact 18 Dec '09

18 Dec '09

I bit off topic but. Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis?

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Handling REFER...
by Kristian Kielhofner 18 Dec '09

18 Dec '09

Hello everyone, I've got two profiles running: s2s and trunk. The context for s2s is defined as s2s-in. The context for trunk is defined as trunk-in. trunk is bound to 192.168.168.3. recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: ------------------------------------------------------------------------ REFER sip:mod_sofia@192.168.168.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 To: "NONAME" <sip:19415551212@192.168.168.3>;tag=BagvZeKSrj7yH From: <sip:9412848354@192.168.168.76:5065;transport=udp>;tag=203332153_1430350929_10 Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Max-Forwards: 70 Refer-To: <sip:6463959906@192.168.168.3> Contact: <sip:ssp@192.168.168.76:5065;transport=udp> Content-Length: 0 ------------------------------------------------------------------------ send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 From: <sip:9412848354@192.168.168.76:5065;transport=udp>;tag=203332153_1430350929_10 To: "NONAME" <sip:9415551212@192.168.168.3>;tag=BagvZeKSrj7yH Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Contact: <sip:mod_sofia@192.168.168.3:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 FS routed this to the s2s-in context, even though it was sent to the trunk profile. Shouldn't it have ended up in trunk-in? For the time being I wrote some crazy dialplan for s2s-in to transfer the call to trunk-in but I'm wondering what could be going on here. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Voicemail->Email
by Oliver Schönbeck 17 Dec '09

17 Dec '09

Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>&1 [[ -t 1 ]] && echo "Writing to logfile '$LOG'." exec > $LOG 2>&1 exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver

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Performance Tuning
by Ujjval Karihaloo 17 Dec '09

17 Dec '09

Looking at Performance Tune my Freeswitch http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Is refers to the following": Turn off every module you don't need Turn presence off in the profiles libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles mod_cdr_csv is slower than mod_xml_cdr How do I change each one ....any references on Wiki?

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