February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Adding P-Charge-Info SIP Header in a SIP Profile
by Ali Pey 10 Feb '15

10 Feb '15

Hello, Is there a way to add the P-Charge-Info header in a SIP Profile? So all outgoing Invites from that sip profile will always include the P-Charge_Info header. Thanks, Ali Pey

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DNS-based dialing?
by Thomas Auge 10 Feb '15

10 Feb '15

Maybe I'm missing the obvious, but I can't find a way to make sofia just contact the host part of a sip address (or use the corresponding NAPTR/SRV records) without creating a gateway record for each domain/server. A pointer in the right direction would be greatly appreciated. :-)

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pass_rfc2833 in 1.4.15
by Grant Bagdasarian 10 Feb '15

10 Feb '15

Hello, Have there been any know issue with the variable pass_rfc2833? We recently upgraded to 1.4.15 and are facing some DTMF issues. Before on 1.2.13 we also had issues with DTMF, so I added the pass_rfc2833 variable, and the problems went away. Now again with 1.4.15, the variable is still enabled, but the problems happen again. Any ideas? Grant

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Having issues accessing freeswitch.org?
by Brian West 10 Feb '15

10 Feb '15

Make sure your IP isn't on any spammers list. ;) Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Number of SIP concurrent calls based on first draft hardware spec
by Andrew Keil 09 Feb '15

09 Feb '15

To FreeSWITCH users, My client in the UK is asking how many concurrent SIP calls could be handled by the current production release of FreeSWITCH (based on the first draft of the hardware spec attached and the service usage shown below). The services will use the following and be written in Lua running inside FreeSWITCH: - G711 -A-law audio playback/record (since E1) - so no transcoding - DTMF detection - General string and number handling - which lua would do very well - TCP socket communication - No text to speech or speech recognition - Limited conference requirements (ie. Inbound call conferenced with an outbound call on the same server) Initially since they are a "windows-house" they intend to run the Windows 64-bit version of FreeSWITCH on Windows Server 2012 R2 Standard Edition. Eventually I hope to move them to Linux, however one step at a time. If stating how many potential concurrent SIP calls based on my hardware spec attached is too difficult, then the other way of stating your hardware spec (just the basics CPU/RAM/Hard disk subsystem) and how many concurrent SIP calls you can take would also help. I appreciate any assistance that anyone can give. Thanks in advance, Andrew Keil

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FreeSWITCH Week in Review (Master Branch) January 31st-February 6th
by Ken Rice 09 Feb '15

09 Feb '15

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1CMyefz FreeSWITCH Week in Review (Master Branch) January 31st-February 6th Hello, again. This past week in the FreeSWITCH master branch we had 8 commits. It was a quiet week with some small updates: updated freeswitch.spec for flite-2.0.0 and some minor updates to the mod_foo clean targets. New features that were added: FS-7149 Update freeswitch.spec for flite-2.0.0 FS-7245 Don’t rebuild core on mod_foo-clean targets The following bugs were squashed: FS-7235 Fix for call recording deleting recorded files in append mode if appended data is shorter than RECORD_MIN_SEC FS-7236 FS-7236 Added lock to prevent a race condition and segfault in mod_conference FS-7236 Fix code before declaration in mod_conference

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NAT traversal: replacing rttproxy with external RTP IP address in FS
by Ben Hood 09 Feb '15

09 Feb '15

Hi all, TL;DR: Does anybody know of a way to convince FS to insert an external RTP IP into the SDP even if the signalling is coming from a client behind the same NAT? I'm currently using kamailio and rtpproxy behind a NAT in conjunction with FS, but I'd like to eliminate rtpproxy from the equation and get FS to handle all of the RTP NAT. I've set the ext-rtp-ip in the SIP profile and FS seems to have accepted it, but FS still sends out SDPs with the local IP address in them. I'm assuming that this is happening because the client that is forwarding the INVITE is behind the same NAT and therefore FS is not inserting the external IP address into the SDP, but I might be wrong. I'm using version 1.4.15-1. Currently I've got this topology for signalling: CLIENT <-> PUBLIC INTERNET <-> NAT <-> KAMAILIO <-> FS And this topology for media: CLIENT <-> PUBLIC INTERNET <-> NAT <-> RTPPROXY <-> FS whereby rttproxy advertises itself on a public IP and the external firewall has static port forwards set up for ephemeral RTP ports. FS is a B2BUA in this setup. I'd like to replace the media topology with this: CLIENT <-> PUBLIC INTERNET <-> NAT <-> FS and port forward RTP to the FS instance in the firewall. As a proof of concept hack, I was able to get end to end media going between two different clients, by rewriting the local IP of the FS instance in the outgoing SDP on the outbound leg in kamailio. But I'd like to be able to set the correct external IP as part of the normal SDP patching process in FS. Thanks, Ben

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mod_vlc taking more time in playing a stream than mod_shout
by Aqs Younas 09 Feb '15

09 Feb '15

Hi, Users. I am facing an issue when trying to play a stream with mod_vlc i hear voice after 8 to 10 seconds after getting answer. But when i play same stream with mod_shout i get voice instantly. Why mod_vlc is taking much time in playing the stream? Is there any way to reduce this time? All thoughts are welcome. Thanks.

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Freeswitch behind a WS Proxy
by Christian Hoene 09 Feb '15

09 Feb '15

Hello, we have placed Freeswitch behind a Apache ws tunnel proxy. Now, Freeswitch give me an error message because the sip_via is not correct. nta.c:3146 agent_check_request_via() nta: Via check: invalid transport "SIP/2.0/WSS" from 127.0.0.1:39964 nta.c:2990 agent_recv_request() nta: REGISTER (1) has invalid Via How do I modify the sip via entry that Freeswitch is expecting? Where to set the sip_via_address/protocol/port variables? Thanks CH

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SWITCH_EVENT_NOTIFY will not send 'body' or 'content-length'
by Vik Killa 09 Feb '15

09 Feb '15

When firing a NOTIFY event using sendevent, mod_sofia does not accept 'body' or 'content-length' from the event; My goal is to be able to send custom 'dialog' events to send NOTIFYs to UAs I noticed FS will send the dialog event if I don't include the 'content-length' header and body. Here is an example event that works. mod_sofia will parse the event and fire the NOTIFY sendevent NOTIFY profile: internal event-string: dialog content-type: application/dialog-info+xml user: 1002 host: 3.local to-uri: 1002(a)3.local from-uri: 1002(a)3.local contact-uri: sip:1002@192.168.0.109:10536;fs_path=sip:192.168.207.156:5060 if I add content-length and a body, mod_sofia does not fire the NOTIFY: content-length: 493 <?xml version="1.0"?><dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="3" state="full" entity="sip:orbit+lot@3.local"><dialog id="lot" direction="initiator"><state>early</state><local><identity display="TEST">sip:lot@3.local;proto=orbit</identity><target uri="sip:lot@3.local;proto=orbit"><param pname="+sip.rendering" pvalue="no"/></target></local><remote><identity display="park">sip:lot</identity><target uri="sip:park+lot"/></remote></dialog></dialog-info> I noticed in mod_sofia.c, where it listens for SWITCH_EVENT_NOTIFY it does not parse the 'content-length' header but it does look for the body and even tries to send it in nua_notify() Is this the desired behavior? If SWITCH_EVENT_NOTIFY is trying to send the body of the event, shouldn't it look for a content-length header or try to create one based on the body? If it is not the desired behavior, could this be a bug? Thank you in advance.

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