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How to make callcenter not auto-answer
by Jimmy Chang 17 Sep '13

17 Sep '13

Hi, We have configured a IVVR. Here is the call flow: 1) user makes a call to FS 2) FS bridges the call to IVVR 3) IVVR plays welcome video and brdiges the call back to FS callcenter My problem is when FS callcenter recieves the call, it would answer the call and IVVR would stop playing welcome video. At that monent, agents don't answer the call yet. Did I miss configured something and how to disable callcenter auto answer? Thanks in advance. Jimmy

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Build error Mac OS X Mountain Lion
by Chris Cachor 17 Sep '13

17 Sep '13

Hello, I'm trying to build Freeswitch on my machine and when I run the make command, I get an error towards the end: /var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:58:suffix or operands invalid for `lea' /var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:63:suffix or operands invalid for `movq' make[7]: *** [gsm0610_rpe.lo] Error 1 make[6]: *** [all] Error 2 make[5]: *** [all-recursive] Error 1 make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 I've built Freeswitch on this same machine about 2 months ago, so not sure why I'm running into this issue. Has anyone had luck with recent builds on OS X Mountain Lion? - Chris

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Bandwidth Dialplan for inbound calls
by James Mortensen 17 Sep '13

17 Sep '13

Hello, I've gone through the dialplan documentation several times and am having trouble getting inbound calls from the PSTN to land on my extension 1000. I've followed the example here as best I can, but what I'm seeing are 482 Request Merged and Bandwidth continuing to hammer the server with INVITE after INVITE. I've verified the firewall is open and can make outbound calls and calls between two Chrome browsers, but seem to have trouble setting up the dialplan for PSTN to Chrome. sip_profiles/external/bandwidth.com.xml: <include> <gateway name="bandwidth.com"> <param name="username" value="your user name"/> <param name="password" value="your password"/> <param name="from-user" value="fromuser"/> <param name="from-domain" value="67.X.X.X"/> <param name="proxy" value="67.X.X.X"/> <param name="expire-seconds" value="600"/> <param name="register" value="false"/> <param name="retry_seconds" value="30"/> <param name="extension" value=""/> <param name="context" value="public"/> <param name="avp" value="true"/> </gateway> </include> /dialplan/public/bandwidth.com.xml: The way I understand this is that I match the number **being called in the public context** and then that is transferred to the default context. The number I'm calling is 503389xxxx and in my INVITE it shows as +1503389xxxx. I've tried matching the destination number as XXXxxxxxxx, 503389xxxx, and \+1(5033896xxx) but I get the same result every time. Can someone please explain how to tell which extension context my calls are going through? Putting the console in DEBUG mode doesn't show me how these patterns are being matched... <include> <extension name="bandwidth.com inbound"> <condition field="context" data="public"/>   <condition field="destination_number" expression="^(XXXxxxxxxx)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="$1 XML default"/> </condition> </extension> </include> /dialplan/default/bandwidth.com.xml: In the default context, you first see my outbound config, which works like a charm. But the second block, taken from Example 9 in the documentation, is something I don't quite understand. I'm pattern matching a 10 digit number, which is what I'm calling... I've tried matching this to XXXxxxxxxx, 503389xxxx and \+1503389xxxx with no change.... <include> <extension name="bandwidth.com Outbound 9.10d">   <condition field="destination_number" expression="^9(\d{10})$"> <action application="bridge" data="sofia/gateway/bandwidth.com/+1$1 "/> </condition> </extension> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_9:_Routing_DID_to_an_e…> <extension name="Local_Extensioi2"> <condition field="destination_number" expression="^(XXXxxxxxxx)$"> <action application="set" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/1000@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> </include> In the logs, I just see looping over and over and over again. Why is it using "internal" instead of "external"? Why is it showing my 971 callerid and not the number I dialed? Where is the part of the logs that show all of the Rexex patterns being parsed to determine what context to use? freeswitch@ip-10-166-245-111> 2013-09-16 19:40:59.695046 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+1971238yyyy(a)67.231.1.115[1902fee8-f2d9-4678-b841-bf59948c4937] 2013-09-16 19:40:59.695046 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+1971238yyyy(a)67.231.1.115 [BREAK] 2013-09-16 19:40:59.695046 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+1971238yyyy(a)67.231.1.115 [BREAK] 2013-09-16 19:40:59.695046 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+1971238yyyy(a)67.231.1.115) Running State Change CS_NEW 2013-09-16 19:40:59.695046 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+1971238yyyy(a)67.231.1.115) State NEW 2013-09-16 19:40:59.715055 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected by acl "domains". Falling back to Digest auth. 2013-09-16 19:40:59.715055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+1971238yyyy(a)67.231.1.115 [BREAK] 2013-09-16 19:40:59.715055 [DEBUG] sofia.c:1787 detaching session 1902fee8-f2d9-4678-b841-bf59948c4937 2013-09-16 19:41:00.055048 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+1971238yyyy(a)67.231.9.59[3295c812-2f36-4d31-8cd4-ef9bbd5cdbdb] 2013-09-16 19:41:00.055048 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+1971238yyyy(a)67.231.9.59 [BREAK] 2013-09-16 19:41:00.055048 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+1971238yyyy(a)67.231.9.59 [BREAK] 2013-09-16 19:41:00.055048 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+1971238yyyy(a)67.231.9.59) Running State Change CS_NEW 2013-09-16 19:41:00.055048 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+1971238yyyy(a)67.231.9.59) State NEW 2013-09-16 19:41:00.075081 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected by acl "domains". Falling back to Digest auth. 2013-09-16 19:41:00.075081 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+1971238yyyy(a)67.231.9.59 [BREAK] 2013-09-16 19:41:00.075081 [DEBUG] sofia.c:1787 detaching session 3295c812-2f36-4d31-8cd4-ef9bbd5cdbdb I've poured through the dialplan docs and examples, but I really believe I could benefit from some explanation based on the problems I'm facing, which I'm sure is just me missing some key piece of insight. Oh I also just commented out the acl lines in both internal and external.xml, which removed the acl warnings, but things still loop. Thank you! James Mortensen

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Viproy VOIP pen testing metasploit plugin
by Mitch Capper 17 Sep '13

17 Sep '13

Hi Guys, Sadly every year ClueCon overlaps with my travel for defcon/blackhat but this year there was a fairly interesting tool released found at http://viproy.com/ for metasploit. It allows for a very large and robust set of VOIP attacks from DOS and fuzzing to some interesting manipulation support. It is not a point and go tool to get the most benefit but can certainly stress a variety of things on a VOIP server. I haven't had a chance to do too many fuzzing and depth attacks although on an old FS server I had it was vulnerable to an invite attack (most likely due to a configuration error on my part, although I didn't change my acl settings much on the updated server so I am not sure). If you use TLS only with certificate validation though you are secure against just about any sort of attack:) ~mitch

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OSX 10.8.5 update is fine with FreeSwitch
by Mario G 16 Sep '13

16 Sep '13

For interested parties: Built HEAD 1.5.6 from scratch on OSX 10.8.5 and Xcode 4.6.3 and all works fine. (except for fs-5375 and fs-5223 which I am still working on). Will update OS X wiki pages when wiki logon is fixed. Mario G

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DNS lookup on outbound proxy FQDN failing
by James Lotery 16 Sep '13

16 Sep '13

Hi FS user group, can anyone suggest any help with this problem. A provider with whom I am integrating with requires that we use their domain name in INVITEs on numbers they host on their end. Normally this is not a problem - I would just set that up in the gateway config with proxy="FQDN" etc however the domain name they are using does not resolve to an IP address I have added an entry in /etc/hosts for the FQDN and can ping it from my command line however FS outbound calls via the gateway are failing with nua_stack.c:271 nua_stack_event() nua(0x7fcc040023f0): event r_invite 503 DNS Error is there any way I can get around this without creating a fake DNS zone as I have seen some people suggest? I ask as I don't have a local DNS server. I am using the stable 1.2 build (Version 1.2.12) on Ubunutu 12.04 Thanks in advance. James

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Can't logon to wiki - says cookies disabled but they are not
by Mario G 16 Sep '13

16 Sep '13

Trying to update a couple of wiki pages but I can't logon anymore. I am using Safari. Any ideas? Mario G

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Re: [Freeswitch-users] OpenSIPs + Freeswitch + Registrations
by Chris B. Ware 16 Sep '13

16 Sep '13

Aleks, Of course I share registrations on DB, but it seems unuseful. I mean a call for a phone registered on an FS box, must be always sent to that box. I don't have NAT issues, since all FS boxes and phones are on the same LAN. What I'd like to do is "sharing the knowledge" between boxes, so that if the Invite goes to a box where the called phone is not registered, call is forwarded to the phone anyway, because there's a shared registrations DB! Many experts on IRC guided me to use paths or scripts to redirect the call to the right box, based on entries on table registrations. But the question is: why freeswitch doesn't do it automatically? Is there any configuration trick to do it? Thank you ------------------------------ Message: 3 Date: Mon, 9 Sep 2013 16:34:43 +0100 (BST) From: "Chris B. Ware" <chrisbware(a)yahoo.it> Subject: [Freeswitch-users] OpenSIPs + Freeswitch + Registrations To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Message-ID: <1378740883.57558.YahooMailNeo(a)web171802.mail.ir2.yahoo.com> Content-Type: text/plain; charset="iso-8859-1" Hi, I'm a bit confused (as usual) on call flow between FS boxes behind an OpenSIPs balancer. All servers are on the same LAN, so the phones. Registrations are stored on common DB. PhoneA sends a register and Opensips balancer forward it to FS1. PhoneB sends a register and Opensips balancer forward it to FS2. PhoneA send an INVITE for PhoneB and OpenSIPs forward it to FS1. FS1 answers "Originate Failed. ?Cause: USER_NOT_REGISTERED" An IRC friend says to use path (RFC 3327).? Is there a config solution or should I follow the his advice? Thanks, Chris B. Message: 4 Date: Mon, 9 Sep 2013 18:56:05 +0300 From: Aleksandr Chusov <chusov.alexsandr(a)gmail.com> Subject: Re: [Freeswitch-users] OpenSIPs + Freeswitch + Registrations To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Message-ID: <CABn9qfqw3e-FDPp2_gKuDZQ-8868B-fWf4-0pJQiLLrAmXC05Q(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 2013/9/9 Chris B. Ware <chrisbware(a)yahoo.it>: > USER_NOT_REGISTERED Hello, You need store the mod_sofia data in a shared database. Good Opensips configuration example: https://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeS…

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OpenSIPs + Freeswitch + Registrations
by Chris B. Ware 16 Sep '13

16 Sep '13

Hi, I'm a bit confused (as usual) on call flow between FS boxes behind an OpenSIPs balancer. All servers are on the same LAN, so the phones. Registrations are stored on common DB. PhoneA sends a register and Opensips balancer forward it to FS1. PhoneB sends a register and Opensips balancer forward it to FS2. PhoneA send an INVITE for PhoneB and OpenSIPs forward it to FS1. FS1 answers "Originate Failed. Cause: USER_NOT_REGISTERED" An IRC friend says to use path (RFC 3327). Is there a config solution or should I follow the his advice? Thanks, Chris B.

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Issue with JSSIP + Freeswitch
by Shahrzad A. 16 Sep '13

16 Sep '13

Hi everyone I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013') I'm using the default configuration and just uncommentated the ' <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound: [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client: 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/ 1000@10.0.14.16:5060 timer while HOT 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/ 1000@10.0.14.16:5060 Hot Hit 1 And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts: 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/ sip:1000@10.0.14.182:5065 Hot Hit 4 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/ sip:1000@10.0.14.182:5065 timer while HOT 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000! Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? Thanks in advanced! Sherry

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