January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how can i do stream piping in freeswitch
by Aqs Younas 08 Jan '15

08 Jan '15

Currently i am playing a stream with mod_shout and this is my default xml. <context name="default"> <extension name="global-intercept"> <condition field="destination_number" expression="^14049002000$"> <action application="answer"/> <action application="playback" data="shout:// online.radiodifusion.net:8024/"/> </condition> </extension> </context> Every time a user makes a call for stream it opens a separate connection with the stream provider. If 100 users dials this number, there would be 100 connections with stream provider listening to same stream, means more rtp packets containing same data for different users. What i want is, if a user is listening to stream then other users must share the same listening connection that the first user is opened, instead of creating a separate connection with stream provider for same stream. Someone told me this is possible in asterisk, so there must be a way in freeswitch. How can i do this.? Any help would be much appreciated. Thanks

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Wrong ice candidate (internal ip) when using mod_verto
by Adam Ben-Ayoun 08 Jan '15

08 Jan '15

Hi, I am experimenting with mod_verto and using the demo to call to a conference on my server (logs on FreeSwitch seems ok generally), the issue is that I am not receiving/sending any audio (I would expect to hear moh at this point), I suspect this is because FreeSwitch sends an ice host candidate set to the internal ip instead of the external, here is the SDP: o=FreeSWITCH 1420641299 1420641300 IN IP4 10.245.43.195 s=FreeSWITCH c=IN IP4 10.245.43.195 t=0 0 a=msid-semantic: WMS Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br m=audio 26060 RTP/SAVPF 111 126 a=rtpmap:111 opus/48000/2 a=fmtp:111 maxplaybackrate=48000; minptime=10 a=rtpmap:126 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=fingerprint:sha-256 0D:BB:87:E0:B6:8A:AE:0D:32:3F:54:9F:36:FF:9B:89:28:AB:B4:17:BC:95:0D:2E:48:0C:33:30:78:52:5F:24 a=rtcp-mux a=rtcp:26060 IN IP4 10.245.43.195 a=ssrc:1435389439 cname:TfPkXSQ755IUEU1s a=ssrc:1435389439 msid:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br a0 a=ssrc:1435389439 mslabel:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br a=ssrc:1435389439 label:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Bra0 a=ice-ufrag:hQuEgraFWcUo2lGL a=ice-pwd:qmWe5Cb0xJL2tDkAjF4pW5UJ a=candidate:0460306999 1 udp 659136 10.245.43.195 26060 typ host generation 0 Using Wireshark I see that the client is trying to send packets to 10.245.43.195 which obviously fails.. How can I fix this? Thanks, Adam

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Register->From->Caller ID
by George F. Phelps 08 Jan '15

08 Jan '15

My current gateway REGISTER packet has a certain FROM header, as shown below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: <sip:+1404XXXXXXX@66.33.147.150>;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150> Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: <sip:gw+switch2voip.us@54.174.255.168:5080;transport=udp;gw=switch2voip.us> . . . What do I set/change in my gateway configuration to add a character string "1404XXXXXXX" to the FROM header? As shown in red below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: "1404XXXXXXX" <sip:+1404XXXXXXX@66.33.147.150>;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150> Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: <sip:gw+switch2voip.us@54.174.255.168:5080;transport=udp;gw=switch2voip.us> . . . I am already setting the following values (in my gateway definition): <param name="from-user" value="1404XXXXXXX"/> <param name="caller-id-in-from" value="true"/> Thanks, George

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Fwd: OverTime RAM usage keep inceasing
by indra sena 08 Jan '15

08 Jan '15

Hi Team, I have observed that overtime RAM utilization/usage is keep increasing maybe due to memory leaks ? when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 hours load test and oberserved now it is using almost 11GB/12GB starts swap also increasing now. Do you have config changes to fix or to take as work around ? How can we reduce RAM utilization overtime ? Could you please provide some solution for this ? Thanks & Regards, Indra.

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Hi Reg IVR and codec negotiation.
by indra sena 08 Jan '15

08 Jan '15

Hi , Happy new year to every body. I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks in advance. Thanks & Regards, Indra..

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BLF on Cisco 303/504/525
by Paul Nebb 08 Jan '15

08 Jan '15

Does anyone have templates or articles that would assist in setting up BLF on Cisco 303/504/525 Paul

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Multiple P-Asserted-Identity Headers
by George F. Phelps 07 Jan '15

07 Jan '15

How do I create two (2) unique, P-Asserted-Identity headers? As required by RFC 3325 <http://www.ietf.org/rfc/rfc3325.txt> : A P-Asserted-Identity header field value MUST consist of exactly one name-addr or addr-spec. There may be one or two P-Asserted-Identity values. If there is one value, it MUST be a sip, sips, or tel URI. If there are two values, one value MUST be a sip or sips URI and the other MUST be a tel URI. A valid, multiple P-Asserted-Identity headers example, taken from RFC 3325: INVITE sip:+14085551212@proxy.pstn.net SIP/2.0 Via: SIP/2.0/TCP useragent.cisco.com;branch=z9hG4bK-124 Via: SIP/2.0/TCP proxy.cisco.com;branch=z9hG4bK-abc To: <sip:+14085551212@cisco.com> From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9802748 Call-ID: 245780247857024504 CSeq: 2 INVITE Max-Forwards: 69 P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@cisco.com> P-Asserted-Identity: tel:+14085264000 Privacy: id When I use the following two (2) statements in a Freeswitch dialplan, the second "set," of course, overwrites the data stored by the first "set." <action application="set" data="sip_h_P-Asserted-ID=tel:+1404XXXXXXX"/> <action application="set" data="sip_h_P-Asserted-ID="George F Phelps" <sip:1404XXXXXXX@YY.YY.YYY.YYY>"/> Thanks, George

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Adding X-CID custom header for sipcapture integration
by Juan Antonio Ibañez Santorum 07 Jan '15

07 Jan '15

Hello, I am trying to add custom X-CID header to copy call id from a-leg to b-leg in order to make work call flow in sipcapture. I set channel variable sip_h_X-CID to add it to b-leg, I can see it is setted up but outgoing b-leg request does not contain the header. If I set up sip_h_B-CID it works but sip_h_X-CID not. Do you know why? Any other way to add the header? Regards

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Testing inbound calls from LOTS of origin points?
by Nathan Neulinger 07 Jan '15

07 Jan '15

Anyone know of any good way to test inbound calls from lots of different origins? Similar to what you do with things like 'global dns propagation checker'? I'd like to be able to periodically validate inbound calls from a bunch of different sources - ideally taking different paths inbound. (We've had a number of occasions lately where we suspect a routing problem external to our sip carrier only affecting certain inbound call sources.) I realize I can probably do this by hand, just by getting SIP trunks from a bunch of different providers, but is there any way that is a bit less manual/time consuming/third party to do the heavy lifting/etc? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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BLF status not always correct
by Raphael Lechner 07 Jan '15

07 Jan '15

Hi, Sometimes we have the issue that the status for the BLF linekeys don’t have the correct status. We have the same issue on different installations (different FreeSWITCH 1.4.X versions and different phone vendor/types, snom/yealink). Today we found the issue on a Installation with FreeSWITCH 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit). sip_subscriptions and sip_presence tables(at that moment the phone don’t have the right status) https://pastebin.freeswitch.org/23834 <https://pastebin.freeswitch.org/23834> Is that a known issue? How can I debug that and is there a workaround/solution for that? After rebooting the phones the status is working again and after some days the issue is back. On newer versions > 1.4.8 the problem is don’t occur so frequently. Thank you, Raphael

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