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Unable to compile FS this morning
by Francois Delawarde 23 Apr '14

23 Apr '14

Am I the only one? make[5]: Entering directory `/home/francois/svn/cm/packages/trunk/src/freeswitch/freeswitch.git/src/mod/xml_int/mod_xml_rpc' CC mod_xml_rpc_la-mod_xml_rpc.lo mod_xml_rpc.c: In function 'websocket_hook': mod_xml_rpc.c:594:2: error: too few arguments to function 'ws_init' In file included from mod_xml_rpc.c:72:0: ws.h:102:5: note: declared here make[5]: *** [mod_xml_rpc_la-mod_xml_rpc.lo] Error 1 Thanks! -- Francois Delawarde <fdelawarde(a)wirelessmundi.com> Wireless Mundi S.L.

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Friday JIRA Rampage 2PM Central / April 25h
by Brian West 23 Apr '14

23 Apr '14

This Friday we’ll be continuing our weekly JIRA rampage, I would like anyone with outstanding issues in JIRA that they would like to work on to join in. If you file a JIRA please pay attention, respond to our questions and TEST GIT MASTER, ‘Yes’ or ‘No’ or some other random string in the git master hash field when filing a JIRA is not acceptable. If the hash in that field is over 24 hours old then you didn’t do your part to help us help you. Help us help you PRETTY PLEASE! I really don’t like asking for more details then no reposes for 3 weeks… thats really not fair. I have OCD and that twerks my OCD badly :) Want to give back but can’t code? We have ways to pitching and help out, The FS Docs project and the FS Bug marshal program, Contact me if you wish to help, this may even be a gateway task to learning how to code in C! Its what happened to me when I was doing the same bug marshal tasks while working on Asterisk. We already have a few members of the community leading up The FS Docs project, we meet on Tuesdays at 2PM Central in 888 we can touch on this Friday too. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org:81/key.txt (AB93356707C76CED)

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mod_xml_curl and SUBSCRIBER_ABSENT
by Kenn Leth Hansen 23 Apr '14

23 Apr '14

Hi all, I am using mod_xml_curl for serving directory XML based on database lookups. I can register phones using mod_xml_curl without problems, however I am having trouble calling/transfering/bridging calls to registered users. If I register using the default 1000 user there are no issues. XML returned to mod_xml_curl: <?xml version="1.0" encoding="UTF-8"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="example.com"> <params> <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> </params> <groups> <group name="agents"> <users> <user id="test"> <params> <param name="password" value="test"/> </params> </user> </users> </group> </groups> </domain> </section> </document> setting up a call to the default 1000 user using "originate users/1000 echo" works flawlessly, if user is not registered FS returns "-ERR USER_NOT_REGISTERED" When trying "originate users/test echo" or "originate users/test(a)example.comecho", the CLI waits for timeout and then returns: -ERR SUBSCRIBER_ABSENT 2014-04-23 08:52:24.175708 [DEBUG] switch_ivr_originate.c:2061 Parsing global variables 2014-04-23 08:52:24.215756 [WARNING] mod_dptools.c:3943 Can't find user [ test(a)example.com] 2014-04-23 08:52:24.215756 [NOTICE] switch_ivr_originate.c:2662 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2014-04-23 08:52:24.215756 [DEBUG] switch_ivr_originate.c:3639 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] When looking up registered users in the Sofia profile the user seems to be registered: freeswitch@internal> sofia status profile Profile reg test Registrations: ================================================================================================= Call-ID: e546c814b5c44f598286b94f923af64e User: test(a)example.com Contact: "test" <sip:04268973@192.168.1.112:64622> Agent: Blink 0.8.0 (Windows) Status: Registered(UDP)(unknown) EXP(2014-04-23 09:06:21) EXPSECS(578) Host: fs01 IP: 192.168.1.111 Port: 5060 Auth-User: test Auth-Realm: example.com MWI-Account: test(a)example.com Total items returned: 2 ================================================================================================= Using FreeSWITCH Version 1.2.16+git~20131218T195141Z~6968ef88fa~64bit (git 6968ef8 2013-12-18 19:51:41Z 64bit) Regards, Kenn Leth Hansen

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Re: [Freeswitch-users] Consulting help
by andretodd＠verizon.net 23 Apr '14

23 Apr '14

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SRV in sip_route_uri
by Tjardick van der Kraan 23 Apr '14

23 Apr '14

Hi, I was wondering if using SRV in the sip_route_uri is permitted. It seems to sort of work as when providing a host/domain record without any A records it does seem to lookup a proxy, and send traffic to it. Though mid transaction (after receiving a 200 OK) it seems to re-lookup the same record in stead of using the cache response. In case there are multiple proxies in the SRV records, this breaks the transaction in some cases as it picks a different proxy from the SRV set to send the ACK to. So before opening a jira issue, I was wondering if the above is indeed a bug or should the sip_route_uri only use IP or real hostnames ? (in which case i guess looking up the SRV might be a bug) Regards, Tj

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Freeswitch Week in Review(Master Branch) April 13th-19th
by Kathleen King 22 Apr '14

22 Apr '14

Hello, again. This week in the FreeSWITCH master branch we had 25 commits. It was a relatively quiet week with various bugs and tweaks to various things. REMINDER: If you register for ClueCon before May 1st you can save $100 dollars by locking in last year's price using the code CLUECON2013! And speaker proposals are still being accepted! If you have a topic you would like to tell us about email a proposal to marketing(a)cluecon.com <mailto:marketing@cluecon.com>. The following bugs were squashed: b0c2c6d FS-6447 --resolve return subscriber-absent platform-code 20 if <dial> is attempted to user that is not registered Jira: http://jira.freeswitch.org/browse/FS-6447 8ce6198 fixed a bug in ESL dispatch 6e8729a OPENZAP-222 fixed caller_ton in libpri 190c241 FS-6452 fixed no reaction on events without variable_domain_name in mod_spy Jira: http://jira.freeswitch.org/browse/FS-6452 bf093e9 Dispatch will call you on register, and call you back if you hangup on next register ae69c5a FS-6407 --resolve support SIP URI caller ID w/ display name in mod_rayo Jira: http://jira.freeswitch.org/browse/FS-6407 New features that were added: 79ebcb1 add enable-chat sofia profile param which defaults to true when not present, set to false to turn off the chat interface in sofia 295964f FS-6457 --resolve Upgrade to OpenSSL 1.0.1g in Windows build (Visual Studio 2010 and 2012) Jira: http://jira.freeswitch.org/browse/FS-6457 0526381 add parsing of user params on invite in mod_sofia 265de3d added the variable variable_sip_to_user as a username to allow eavesdropping on the local endpoint in mod_spy Jira: http://jira.freeswitch.org/browse/FS-6082 b99ef3a add SWITCH_MESSAGE_HEARTBEAT_EVENT indication to execute on a session when session heartbeat fires a4a7924 add generic keepalive system and implement it in sofia to send MESSAGE or INFO packets in-dialog at specified interval. Improvements in cross platform build supports: bb22179 FS-6459: --resolve fix srcdir build running modcheck Jira: http://jira.freeswitch.org/browse/FS-6459 In terms of stability these were the use cases that were fixed: 93a84cf FS-6460 --resolve fix leaked RTP ports that cause mod_skinny to fail after N calls Jira: http://jira.freeswitch.org/browse/FS-6460 748b7de Get rid of a small memory leak when the configs are reloaded in mod_enum 96523a9 FS-6460 --resolve fix leaked RTP ports that cause mod_skinny to fail after N calls Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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DID providers - any thoughts? (jan 2013)
by Cal Leeming [Simplicity Media Ltd] 22 Apr '14

22 Apr '14

Hello everyone, I've been recently reviewing several DID providers, and looking for some feedback. I appreciate this isn't totally on topic for FreeSWITCH, but over the last few months I've found the community feedback to be invaluable - so I'm hoping it might open up a good discussion. I would be really interested to hear other peoples experiences with their DID providers, as much detail as possible. There are several providers we've looked at, but they are either reselling others services, or do not deliver the service quality we are looking for. The criteria I've been looking for is; * Able to provide either US or UK numbers * Exposes an API that allows control of purchasing and routing * Guarantees premium call quality, good latency, and inbound CLI. * Does NOT operate on a market/exchange basis * Trading as a telecoms provider for more than 5 years * Has a reputable name within the community or industry So far, I have only found the following providers that meet this criteria; * Hello Telecom - UK numbers only, ITSPA registered, no charges for minutes, rental or concurrent channels. No minimum charges. Excellent call quality/latency, no CLI problems, and almost no downtime in the last 6 months. However, they do not offer a sane purchasing API which makes automation difficult, and interop with FS has been somewhat painful. * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but still very fair. Currently at inter-op stage, so can't comment further - but they seem very good so far. * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, difficult to justify unless you already have existing traffic. * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over 9000 pages of inter-op and around £10k in set up fees. Thanks in advance Cal

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transfer registration
by Helena Garcia-Nieto 22 Apr '14

22 Apr '14

Hi all, I’m trying to transfer the registration to a gw. I want freeswitch to send the registration message received from the external interface (phone via internet) to the internal interface (to the internal gw, private interface) Could someone point me in the direction of how to do it? I am reading sofia module and some examples but I cannot find the right way. Thanks! Helena

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Re: [Freeswitch-users] Exporting Diversion Header Twice
by Sanjiv Kumar 22 Apr '14

22 Apr '14

Anyone ?? At least let me know if this cannot be done. I am trying to export sip_h_Diversion header twice but I only see the last attempt on leg-B. It appears that second export attempt overwrites the first one. How can I export it twice? <extension name="public_did"> <condition field="destination_number" expression="^9999(\d{10})$"> <action application="set" data="ocn=18005551212"/> <action application="set" data="rdn=18005552323"/> <action application="export" data=" nolocal:sip_h_Diversion =<sip:${ocn}@${network_addr} <sip:$%7bocn%7d@$%7bnetwork_addr%7d> >;reason=user-busy;counter=3;privacy=off"/> <action application="export" data=" nolocal:sip_h_Diversion =<sip:${rdn}@${network_addr} <sip:$%7brdn%7d@$%7bnetwork_addr%7d> >;reason=no-answer;counter=1;privacy=off"/> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="continue_on_fail=true"/> <action application="transfer" data="1001 XML default"/> </condition> </extension>

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SSL3_READ_BYTES:sslv3 alert handshake failure
by Assaf Dahary 22 Apr '14

22 Apr '14

Hi, I've successfully installed a FS server with TLS using PsitiveSSL and it is working great. Few days ago I've followed the same installation on another standalone machine with the same FS-1.2.22 and PsitivieSSL CA but this time I cannot connect over TLS. It seems that FS has no cipher to response with and it fails on negotiations. The PositiveSSL is OK because I verified it locally with "openssl s_client" and from the internet using browser/https. My ssl/ pem files are made with (like I did with the first server - OK): #cat mysite_com.crt myserver.key > agent.pem #cat PositiveSSLCA2.crt AddTrustExternalCARoot.crt > cafile.pem # chown freeswitch.freeswitch *.pem #chmod 640 *.pem When issuing "$ sslscan myfs.com:5091 | grep Accepted " I get no single cipher. I get long list of 'Rejected' ciphers. When I'm running the same command for my first server I get a list of supported ciphers - which is OK. When [root@www ~]# openssl s_client -connect myfs.com:5091 CONNECTED(00000003) depth=2 C = SE, O = AddTrust AB, OU = AddTrust External TTP Network, CN = AddTrust External CA Root verify return:1 depth=1 C = GB, ST = Greater Manchester, L = Salford, O = COMODO CA Limited, CN = PositiveSSL CA 2 verify return:1 depth=0 OU = Domain Control Validated, OU = PositiveSSL, CN = myfs.com verify return:1 140160541112136:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure:s3_pkt.c:1256:SSL alert number 40 140160541112136:error:140790E5:SSL routines:SSL23_WRITE:ssl handshake failure:s23_lib.c:177: I've already re-installed FS with clean config files. Centos 6.x 64, OpenSSL 1.0.1e-fips 11 Feb 2013. I would appreciate any help/tip on this TLS fail issue. Regards Assaf

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