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Number of ways for creating a dynamic ivr in freeswitch
by Aqs Younas 11 Nov '14

11 Nov '14

Hi, How can we create dynamic ivr in freeswitch. I have ivr data in json format from a web, i want this to be played in freeswitch as ivr. How can i do so.? In asterisk there are application availables like phpivr/SynIVR. How can this be done in Freeswitch. Any help would be much appreciated.

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Freeswitch webRTC - audio DTLS key err
by mbo 11 Nov '14

11 Nov '14

Hello, according to the instructions on https://freeswitch.org/confluence/display/FREESWITCH/WebRTC I have enabled webrtc on one Sip profile. I’ve added my own cerificate and installed in /usr/local/freeswitch/certs. I’m connecting to freeswitch using JsSip library. The signaling part works fine, I can see all SIP messages in the log, but then it fails to establish the audio/rtp connection: 2014-11-10 17:36:27.065688 [INFO] switch_core_media.c:5206 Skipping RTCP ICE (Same as RTP) 2014-11-10 17:36:27.065688 [INFO] switch_rtp.c:3065 Activate RTP/RTCP audio DTLS client 2014-11-10 17:36:27.065688 [ERR] switch_rtp.c:3117 audio DTLS key err [1] I’m a bit confused about this message, I thought all tls settings in a sip profile and the dtls-srtp.pem certificate is not relevant for webrtc / wss, or what does this error message want to tell me? Thanks Markus

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NAT: SDP with local IP in o-line unicast_address
by Peter Steinbach 11 Nov '14

11 Nov '14

Hell, we have the following problem: Our Freswitch is behind NAT. We are sending faxes to a SIP provider. Dependend on the destination number, the faxes are received or not. Faxes are always sent via the same SIP provider and the same dialplan, but I expect, they may be routed differently via other, subsequent SIP providers. Regarding the SDP I can see, that the c-line does contain the our external IP, but the o-Line does contain the local IP in the unicast_address field. We are routing the call via a defined gateway in the external profile, which has external_sip_ip set and external_rtp_ip set. Dialplan is: <action application="export" data="t38_passthru=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/gateway/QSC/06912345678(a)sip.qsc.de"/> Here is the SDP ==================================== v=0. o=FreeSWITCH 1037989557 1037989558 IN IP4 192.168.206.241. s=FreeSWITCH. c=IN IP4 212.xxx.xxx.106. t=0 0. m=audio 12056 RTP/AVP 0 8. a=rtpmap:0 PCMU/8000/1. a=rtpmap:8 PCMA/8000/1. a=maxptime:240. ==================================== I suspect, that the following SIP providers may have a problem with the o-line with the local IP. So - Is there any way to control this? E.g. via Dialplan variable? ACL also seems to be fine 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding 192.168.0.0/16 (allow) [] to list nat.auto I see the same behaviour also in https://freeswitch.org/jira/browse/FS-5909 "ext-xxx-ip ignored with proxy_media turned on" There is a link for a patch, which is no longer available. Did this go into the main release? Does anybody have this patch? -- With kind regards Peter

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UDP vs UDP-NAT
by Dion Phillips 11 Nov '14

11 Nov '14

Hi All I have a FS server in the cloud and have phones registering with FS from 2 different locations. From one location, the registration is UDP-NAT and just UDP from the second location. In the FS log, calls to the UDP connected phones work and they come up with an external IP address (sofia_glue.c:1232 sofia/internal/sip:1002@203.59.xxx.xxx:5060 sending invite) while the phones on the UDP-NAT site don't work because FS is trying to connect to them via the internal IP address (sofia_glue.c:1232 sofia/internal/sip:1000@192.168.0.111:5060 sending invite). The phones on both sites are behind a very standard modem with no port forwarding or NAT setting. What is wrong and how would I fix it? Thanks in advance. Dion.

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phpivr/SynIVR like application in freeswitch
by Aqs Younas 10 Nov '14

10 Nov '14

I have ivr data into json format. I want to execute that data in freeswitch.Is there any application available to do so.? Or how can i do so? Like we have phpivr/SynIVR in asterisk.

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Re: [Freeswitch-users] hash_remote <list>|<kill> [name]|<rescan>
by aademattia＠comcast.net 10 Nov '14

10 Nov '14

I could use this information too. Sent from my HTC ----- Reply message ----- From: "Andre Demattia" <andretodd(a)verizon.net> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] hash_remote <list>|<kill> [name]|<rescan> Date: Sun, Nov 9, 2014 7:39 PM Hi, What would the syntax look like to connect more than one FreeSwitch Limit hash_remote <list>|<kill> [name]|<rescan> Thanks Andre

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Confusing about FreeSWITCH callID variables
by Israel Fink 10 Nov '14

10 Nov '14

I'm a bit confusing about the channel variables that represent the call ID. I find that there are three variables, 1) Unique-ID 2) Channel-Call-UUID 3) variable_sip_call_id, what is the different between this. The value of Channel-Call-UUID seems to be stable during the entire call, the value of Unique-ID in the begin it is the same as Channel-Call-UUID but when bridging it changes it value to another ID, then it comes back to the beginning value, i.m not clear when and why. The variable variable_sip_call_id many time don't have any value, and also it changes the value when bridging and then comes back to the previous value. I have looked for an explanation in FreeSWITCH wiki, but don't find. In the mailing list a have found an explanation here: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-August/099035.h…, but still not enough. Can someone give a good explanation about this variables,what is their purpose, and so on. Israel Fink - Developer Machshevet team

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C Dialplan processing routines
by Abdul Hakeem 10 Nov '14

10 Nov '14

Hello, Could anyone give me an example on how to link a C code based routine to invoke FS API's to process not just Dialplan or Chatplan but other events such as Register, Reinvite etc ? Thank you in advance. Regards, Abdul Hakeem

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how to make ivr from json data
by Aqs Younas 10 Nov '14

10 Nov '14

I have ivr data in json format i want this data to be played in freeswitch like i used to do in asterisk with phpivr third party application. Does somebody know how to do so..or is there any application like phpivr in asterisk. https://code.google.com/p/phpivr/ Any help?

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Freeswitch port numbers assigned to phones
by Dion Phillips 10 Nov '14

10 Nov '14

Hi All I am looking for some insight into FS port numbers. The default internal port number is 5060 and when the IP phones register with FS, one of the phones gets assigned 5060 and the remaining phones get a random port number assigned (I am using this command to see the port numbers: "sofia status profile internal reg"). How is this random port number decided on? How can I tie a phone to a specific port number? Thanks Dion.

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