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URI dialplan setup
by Mark Campbell-Smith 30 May '14

30 May '14

Hi Voip provider Ippi allow calls to/from skype users. For me, this will work quite well in my home pabx, and I do not want to implement the SipThreeSkype solution. I have created the gateway and successfully registered to Ippi's sip server at ippi.fr. Ippi requires that to dial a skype user, the following URI format must be dialed: my-skype-address(a)skype.ippi.com. (this successfully calls my skype address using a softphone) I am trying to create a dialplan that will allow me to call a skype user through the Ippi gateway. This is my gateway setup (notice ippi uses ippi.fr): <include> <gateway name="ippi.fr"> <param name="username" value="username"/> <param name="password" value="password"/> <param name="proxy" value="ippi.fr"/> <param name="realm" value="ippi.fr"/> <param name="auth-username" value="username"/> <param name="register" value="true"/> </gateway> </include> I have the following in the dialplan: <extension name="ippi-skype"> <condition field="destination_number" expression="^5000$"> <action application="set" data="call_timeout=90"/> <action application="set" data="hangup_after_bridge=true"/> <action application="info"/> <action application="bridge" data="sofia/gateway/ ippi.fr/my-skype-address(a)skype.ippi.com"/> </condition> </extension> Now I realize that according to the wiki, I should not put sofia/gateway/ ippi.fr/my-skype-address(a)skype.ippi.com (ie the @skype.ippi.com), but I need a way to dial out through the ippi gateway (ippi.fr), but to a user in domain skype.ippi.com. I did try the two following options: <action application="set" data="sip_invite_to_uri= my-skype-address(a)skype.ippi.com"/> <action application="export" data="sip_invite_to_uri= my-skype-address(a)skype.ippi.com"/> Wireshark gives me the following for a successful call using Jitsi softphone: Session Initiation Protocol Request-Line: INVITE sip:my-skype-address@skype.ippi.com SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Call-ID: 891b23da468c6fdf05434ac89f123cf1@0:0:0:0:0:0:0:0 CSeq: 1 INVITE Sequence Number: 1 Method: INVITE From: "username" <sip:username@ippi.fr>;tag=35bcc007 SIP Display info: "username" SIP from address: sip:username@ippi.fr SIP tag: 35bcc007 To: <sip:my-skype-address@skype.ippi.com> SIP to address: sip:my-skype-address@skype.ippi.com Via: SIP/2.0/TCP 192.168.1.6:62110 ;branch=z9hG4bK-373232-f15d6a66ae2d195ba99f1d308e43e61d Transport: TCP Sent-by Address: 192.168.1.6 Sent-by port: 62110 Branch: z9hG4bK-373232-f15d6a66ae2d195ba99f1d308e43e61d Max-Forwards: 70 Contact: "username" <sip:username@192.168.1.6:62110 ;transport=tcp;registering_acc=ippi_fr> Contact Binding: "username" <sip:username@192.168.1.6:62110 ;transport=tcp;registering_acc=ippi_fr> User-Agent: Jitsi2.5.5137Windows 7 Content-Type: application/sdp Content-Length: 910 What do I use in the bridge statement so that it calls my-skype-address(a)skype.ippi.com, using gateway ippi.fr? Suggestions appreciated!

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NetBSD/OpenBSD/DragonFlyBSD Ports
by Brian West 30 May '14

30 May '14

FreeSWITCHers, Looking for interested parties to help build and maintain ports of FreeSWITCh on these platforms, I'm already working with someone on FreeBSD to get those updated. Please email me off list we can schedule a call to discuss this in a community project as we do with the FS-Docs team, We'll just have a FS-Portability team community meeting ;) Again you don't have to be a wizard with C or autotools, just proficient on the target platform with a interested in helping spread FreeSWITCH. Thanks, -- Brian West brian(a)freeswitch.org 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/key.txt (AB93356707C76CED)

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Dialplan detect user on call
by Sean Devoy 29 May '14

29 May '14

Hi, I have an issue that is the result of Cisco shortsightedness. The Cisco SPA5xx phones support auto-answer. However, if the user is on the phone, the caller is slammed on hold and the auto-answer picks up. The person on hold is left wondering why they got thrown away with no warning. At a minimum, Cisco should allow you to select which behavior you want. Anyway, how can I detect (in an XML dialplan) if the extension is currently on a call? For those that don't know me, I have limited scripting skills. Can a script be used to detect this and return a value to a dialplan condition? Thanks, Sean

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WebRtc video
by Daniel Ivanov 29 May '14

29 May '14

To be honest, the state of video singalling is disguisting right now when it comes to handling webrtc. I tried the sipml5 demos and they either break my browser or do nothing. What is the most reliable way to have video transferred over webrtc?

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SQL connection problem with mod_nibblebil
by Dmitry Erankevich 29 May '14

29 May '14

Dear all. I'm testing mod_nibblebill and it looks like the mod can't connect to the mysql db: reload mod_nibblebill +OK Reloading XML +OK module unloaded +OK module loaded 2014-05-29 05:35:05.627682 [INFO] mod_enum.c:876 ENUM Reloaded 2014-05-29 05:35:05.627682 [INFO] switch_time.c:1191 Timezone reloaded 530 definitions 2014-05-29 05:35:05.627682 [NOTICE] switch_loadable_module.c:978 Deleting Application 'nibblebill' 2014-05-29 05:35:05.627682 [NOTICE] switch_loadable_module.c:1033 Deleting API Function 'nibblebill' 2014-05-29 05:35:05.627682 [CONSOLE] switch_loadable_module.c:1876 Stopping: mod_nibblebill 2014-05-29 05:35:05.627682 [NOTICE] switch_event.c:2073 Event Binding deleted for mod_nibblebill:SESSION_HEARTBEAT 2014-05-29 05:35:05.627682 [CONSOLE] switch_loadable_module.c:1896 mod_nibblebill unloaded. 2014-05-29 05:35:05.627682 [INFO] mod_nibblebill.c:222 odbc_dsn is freeswitch 2014-05-29 05:35:05.627682 [CONSOLE] switch_loadable_module.c:1409 Successfully Loaded [mod_nibblebill] 2014-05-29 05:35:05.627682 [NOTICE] switch_loadable_module.c:267 Adding Application 'nibblebill' 2014-05-29 05:35:05.627682 [NOTICE] switch_loadable_module.c:313 Adding API Function 'nibblebill' Trying to make call: 2014-05-29 05:38:20.606838 [ERR] switch_core_sqldb.c:1131 SQL ERR: [SELECT cash AS nibble_balance FROM accounts WHERE id='12'] no such table: accounts 2014-05-29 05:38:20.606838 [ERR] mod_nibblebill.c:420 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE id='12'] But: [root@localhost default]# isql freeswitch +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> SELECT cash AS nibble_balance FROM accounts WHERE id='12' +-------------------------+ | nibble_balance | +-------------------------+ | 5 | +-------------------------+ SQLRowCount returns 1 1 rows fetched FreeSWITCH Version 1.2.24+git~20140528T155016Z~e19860eef4~64bit (git e19860e 2014-05-28 15:50:16Z 64bit) I've already tried different parameters for DB connection in nibblebill.conf.xml (odbc-dsn, db-dsn, etc.) with no success. Would be appreciated for any help. Dmitry.

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Errors during make for v1.4 latest a/o 5/27/2014
by bedgar＠vseinc.com 29 May '14

29 May '14

Hello All, Am trying to build latest v1.4 release and encountering a problem during make. I am running CentOS release 6.5 (Final) yum'd up. Helpful suggestions please. Brian /usr/lib64/libnss3.so: undefined reference to `PR_FindSymbol' /usr/lib64/libnss3.so: undefined reference to `PR_RWLock_Rlock' /usr/lib64/libssl3.so: undefined reference to `PR_OpenAnonFileMap' /usr/lib64/libnss3.so: undefined reference to `PR_UnloadLibrary' /usr/lib64/libnss3.so: undefined reference to `PL_InitArenaPool' /usr/lib64/libnss3.so: undefined reference to `PR_NewRWLock' /usr/lib64/libnss3.so: undefined reference to `PR_RWLock_Wlock' /usr/lib64/libnss3.so: undefined reference to `PR_LoadLibrary' /lib64/libldap-2.4.so.2: undefined reference to `PR_GetEnv' /usr/lib64/libnssutil3.so: undefined reference to `PR_LoadLibraryWithFlags' /usr/lib64/libnssutil3.so: undefined reference to `PL_ClearArenaPool' /usr/lib64/libnss3.so: undefined reference to `PR_DestroyRWLock' /usr/lib/gcc/x86_64-redhat-linux/4.4.7/../../../../lib64/libcurl.so: undefined reference to `PR_NewTCPSocket' /lib64/libldap-2.4.so.2: undefined reference to `PR_SetEnv' /lib64/libldap-2.4.so.2: undefined reference to `PR_GetLibraryName' /usr/lib64/libssl3.so: undefined reference to `PR_ErrorInstallTable' /usr/lib64/libssl3.so: undefined reference to `PR_ExportFileMapAsString' /usr/lib64/libnssutil3.so: undefined reference to `PR_GetLibraryFilePathname' /usr/lib64/libnss3.so: undefined reference to `PR_FindFunctionSymbol' /usr/lib64/libsmime3.so: undefined reference to `PL_NewHashTable' /lib64/libldap-2.4.so.2: undefined reference to `PR_ErrorToString' /usr/lib64/libnss3.so: undefined reference to `PR_RWLock_Unlock' /usr/lib64/libssl3.so: undefined reference to `PR_ImportFileMapFromString' /lib64/libldap-2.4.so.2: undefined reference to `PR_GetDirectorySeparator' collect2: ld returned 1 exit status make[2]: *** [freeswitch] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [all] Error 2

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Webrtc and video
by Alexander Haugg 29 May '14

29 May '14

Hi, i try to configure the fs for using webrtc with video. Voice with the sipml5 works fine. 1. I can't find the mod_vp8 in the mod folder, but in the module.conf.xml is it on by default (gives an error message if fs is stating). On the wiki there is a dead link In the solution I can't find it too. 2. Where can I find a tiny description for it? The fs version is -> Version 1.5.13b+git~20140524T091421Z~1fe6fb6d46~64bit (Today) I use the sipml5 with chrome (newest version too) Thanks freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 172.16.103.1,172.16.103.1 Dialplan XML Context public Challenge Realm auto_from RTP-IP 172.16.103.1 SIP-IP 172.16.103.1 URL sip:mod_sofia@172.16.103.1:5060 BIND-URL sip:mod_sofia@172.16.103.1:5060;transport=udp,tcp WS-BIND-URL sip:mod_sofia@172.16.103.1:5066;transport=ws HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM,VP8 CODECS OUT OPUS,G722,PCMU,PCMA,GSM,VP8 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 4 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 0 REGISTRATIONS 2

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480 no user responding
by Miha 29 May '14

29 May '14

Hi, is it possible to tell fs to send 408 if user does not responed to call? Right now it is sending 480 which our provider (to which are call are send throught) represente as IVR messege "number is temporary unavalible". Other softswitch is sending 408. So is it possible to send 408 instead of 480? tnx miha

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NibbleBill
by Chris Martineau 29 May '14

29 May '14

Hi, Having trouble doing concurrent billing with nibblebill. I have latest master but the wiki seems out of date. If I set up 2 calls to the same account they both take the current account balance and bill to the end of that balance. At no time does it seem to attempt to bill on the fly or allow for the fact that there are multiple calls using the balance. At the end of the calls the full value is written to the db resulting in obviously a negative balance as if heartbeat was set to 0. Increment is set to 60 (i am assuming this is seconds) and heartbeat is also 60. Is says in the document that this is possible but not sure if this is the case. Any help you could offer would be appreciated. Regards CHris

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Cisco 7942 realm / multi tenant setup
by Remco 29 May '14

29 May '14

Hi all, I'm running into an issue with the Cisco 7942G phones. The phones will not send the realm in the request URI, but use the IP address instead. So, instead of: INVITE 100(a)pbx.company.com they send INVITE 100(a)1.1.1.1 This behavior causes issues with a multi tenant setup. Is there anyone with insight into these black boxes that can tell if there is some sort of configuration option to fix this? Thanks, Remco.

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