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mod_avmd Not Detected Beep
by Timothy oladapo olawuyi 17 Apr '17

17 Apr '17

I was trying to implement avmd in lua script on mobile number that has voicemail enabled but AVMD was unable to detect the beep. Voice Mail was automatically activated mobile anytime a call was not answered. Below was the files and command use in implementing the AVMD. 1. AVMD configuration. https://pastebin.freeswitch.org/view/a25da7d8 2. My Lua Script https://pastebin.freeswitch.org/view/44403f72 Bgapi command was used to initiate the call. bgapi originate {sip_cid_type=pid,service_type=UKNOW,origination_caller_id_number=+234802356 5181,igonre_early_media=true}sofia/external/sip:+2347016922641@10.200.161.13 2 '&lua(/usr/local/freeswitch/scripts/obsession.lua)' Regards Olawuyi Timothy Oladapo | IT Operation | Information System Dept |Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 |Skype: daptims | <htttp://www.pethahiah.com/> htttp://www.pethahiah.com

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Bridge b-leg variables not present
by Daniel Finke 17 Apr '17

17 Apr '17

Hey, I am using the bridge command from mod_dptools as follows. I have a channel up on one FreeSWITCH server, let's call it FS1. I am trying to bridge to a conference running via mod_conference on another media server, FS2. This is the bridge command I am executing: bridge > {call_timeout=20,originate_timeout=20,ignore_early_media='false',hold_music=silence,sip_h_X-Conf-Flags-Moderator=false,custom_Test-Thing='some_value',local_var_clobber='true'}[sip_invite_to_uri=<sip:955@realm>,origination_caller_id_number='3001',origination_caller_id_name='Test > 1 > Phone',ignore_early_media='true',sip_auth_username='omitted',sip_auth_password='omitted']sofia/sipinterface_1/conference@FS2_IP > :11000 > This results in a new b-leg channel on FS2, but upon a "uuid_dump <b-leg-channel>" on FS2, I do not see the variables specified in the bridge command. Specifically, I am trying to get "custom-Test-Thing" to be present. I am using the following doc: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+bridge I have tried both the b-leg simple variables > (bridge > [variable_scope=endpoint-1]<target_endpoint>,[variable_scope=endpoint-2]<target_endpoint>) and b-leg thread global approaches > bridge > {variable_scope=thread_global}[variable_scope=endpoint-1]<target_endpoint>[,[variable_scope=endpoint-2]<target_endpoint>][|[variable_scope=endpoint-3]<target_endpoint>] > Hoping to hear that I am just doing something wrong and that it is simple! I can supply whichever logs are necessary.

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block udp for public profile
by Russell Treleaven 17 Apr '17

17 Apr '17

I have been thinking about only accepting tcp/tls on the public profile and blocking udp at the border. The peer will have to do the 3 way handshake which means we know they have a valid source ip:port before they get up the sip stack I'm sure this is not a new idea. Any thoughts? -- Sincerely, Russell Treleaven sip:rtreleaven@sip.bunnykick.ca;transport=tcp

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Random [CRIT] sofia_glue.c:1031 Error creating HANDLE! when using att_xfer
by Antonio Modesto 17 Apr '17

17 Apr '17

Hi folks, Some of my customers are complaining that sometimes when they are going to transfer a call, it drops. I noticed that when this happens, freeswitch reports the following in the console/logfile: aed6d1eb-c1fe-496f-b520-150ae23918a6 2017-04-04 12:58:13.900620 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/angela [aed6d1eb-c1fe-496f-b520-150ae 23918a6] aed6d1eb-c1fe-496f-b520-150ae23918a6 2017-04-04 12:58:13.900620 [CRIT] sofia_glue.c:1031 Error creating HANDLE! aed6d1eb-c1fe-496f-b520-150ae23918a6 url_str=[sip:angela@10.100.161.3:5060 ;transport=udp] aed6d1eb-c1fe-496f-b520-150ae23918a6 call_id=[N/A] aed6d1eb-c1fe-496f-b520-150ae23918a6 to_str=[<sip:angela@10.100.161.3:5060 ;transport=udp>] aed6d1eb-c1fe-496f-b520-150ae23918a6 from_str=["Outbound Call" < sip:keneja%sabia.pbx.isimples.com.br@10.50.0.1>] aed6d1eb-c1fe-496f-b520-150ae23918a6 invite_contact=[< sip:mod_sofia@10.50.0.1:5060>] aed6d1eb-c1fe-496f-b520-150ae23918a6 2017-04-04 12:58:13.900620 [NOTICE] mod_sofia.c:117 Hangup sofia/internal/angela [CS_INIT] [DESTINATION_OUT_OF_ORDER] Do you guys know what can be causing this? Thanks. -- Atenciosamente, Antônio Modesto

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a=fmtp:101 0-15
by Bilal Dar 16 Apr '17

16 Apr '17

We are having issues sending out DTMF, primarily when dial any conference bridge. My provider asked me to send 'a=fmtp:101 0-15' instead of 'a=fmtp:101 0-16'. They do not accept flash. Appreciate if anyone can let me know how we can fix that. Thanks Bilal

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communication probelm with Freeswitch through ESL
by Deepika Yadav 15 Apr '17

15 Apr '17

Hi, I am not bale to communicate with Freeswith over ESL: When I run the following program: import ESL con = ESL.ESLconnection('localhost', '8021', 'ClueCon') if con.connected: con.events('plain', 'ALL') while 1: e = con.recvEvent() if e: print e.serialize() I get output as : "Event-Name: SERVER_DISCONNECTED" similarly, If I run any command using "con.api("originate sofia/gateway/MySIP/XXXX &park")", the events do not get fired. Regards Deepika

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Allison Prompts Go Fund Me
by Brian West 15 Apr '17

15 Apr '17

FreeSWITCHers, I'm asking everyone to help us finish off the remainder so we can get the order placed before the end of the month and we can publish them along side our upcoming 1.8 release. https://www.gofundme.com/allison-prompts-for-freeswitch PRETTY PLEASE! ;) Thank You! -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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Control SDP IP used between internal and external bridge
by SIP Tech 14 Apr '17

14 Apr '17

Great work on the FS project guys! I have a specific use case on a seasoned freeswitch machine that seems to be giving me problems based on SDP, working NAT configs, and freeswitch. First off, NAT works and my machines are resolvable by internal IP within the network and publicly via static public IP. My issue is not a functional NAT issue but how freeswitch assigns a private or public IP in SDP. I would like to control IP used in SDP at bridge, so that I can use private IP or public IP on demand where I want to instead of letting freeswitch insert it based on config settings. My scenario is all controlled via configs within freeswitch. All machines have private and public IPs but specific traffic must pass over private IP and in other places calls must bridge to another FS machine via external profile and public IP. Internal Profile= UPD 5080 External Profile= UDP 5060 1. A private call comes into freeswitch A over an MPLS to my internal SIP profile on UDP 5080. 2. The call then routes from freeswitch A via bridge to freeswitch B via public IPs on the external profiles using UDP 5060. Freeswitch B is only accessible via its public IP address. 3. Freeswitch B routes calls to a 3rd party IVR that gives callers the option to transfer back to Freeswitch A. When the caller selects to be transferred within the IVR freeswitch B makes a bridge back to freeswitch A using public IPs. 4. Freeswitch A receives the incoming call from freeswitch B public IP to public IP. 5. Freeswitch A uses another dialplan to route the call back out over the MPLS using private IP to private IP. I am running into SDP IP issues and how freeswitch is handing SDP based on where the call came from and where it needs to go to. I have spent some time playing with configs in vars and the profiles around rtp ports and ext-rtp nat, autonat, etc. I have SDP working how I need it to up to the last step (4 and 5 above) but I cannot get freeswitch to send its internal IP out in that last SDP session on bridge back over the MPLS. The MPLS only talks private ip to private ip and FSA is sending its public IP in SDP. This is where I am looking for an explicit way to control which SDP IP (local machine private or public) is sent out on bridge. Is this possible? If not, Is there a better way or alternative way to approach this scenario in freeswitch? Thanks guys

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Using b-leg variables with execute_on_ring
by Iskren Hadzhinedev 14 Apr '17

14 Apr '17

Hi! I have the following extension at the beginning of my context: |<extension name="prepare call intercept" continue="true"> <condition> <action application="set" data="nolocal:execute_on_ring=db insert/call_intercept/${dialed_user}@{dialed_domain}/${originator}"/> </condition> </extension> | However, I get an error from the |db| application because these variables are expanded to empty strings. If I change the action’s data to |nolocal:execute_on_ring=info| I see the variables in the console. What am I missing here? Thank you! Kind regards, Iskren

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video conference
by Sugandha Joshi 12 Apr '17

12 Apr '17

Hi, We were trying to establish video conference by referring to https://freeswitch.org/confluence/display/FREESWITCH/mod_conference . In the mod_conference file they had mentioned one of users who tried how he configured his conference bridge i.e https://txlab.wordpress.com/2012/09/17/setting-up-a-conference-bridge-with-… . We followed the given steps but are unable to achieve it. Could you point us to a link which would help us set the video conference. Also, sometimes when i switch on my virtual machine on which the freeswitch server is running, i am not able to enter freeswitch console using the command fs_cli . It keeps displaying socket connection error. Could you help us solve it. Regards, Sugandha

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