- FreeSWITCH-users - FreeSWITCH Mailing Lists

capture/not sending DTMF to b-leg
by Dmitriy Shumaev 06 Jun '14

06 Jun '14

Hi. I use bind_meta_app and bind_digit_action: <action application="bind_meta_app" data="5 a b execute_extension::send_floor_notify_AT0 XML default"/> <action application="bind_meta_app" data="8 a b execute_extension::send_floor_notify_AT1 XML default"/> and <action application="bind_digit_action" data="default,1,exec:execute_extension,send_floor_notify_AT0 XML default"/> But I do not want to send *5 and *8 and 1 DTMF to b-leg. How can I do this? With best regards, Shumaev D.A., programmer @ KBR Ltd.

3 4

FreeSwitch as SIP route server
by Boris Kovalenko 06 Jun '14

06 Jun '14

Hello! Is it possible to use freeswitch as SIP route server? I have commercial SIP server which does not has LCR with it. Is uses external routing instead. It sends INVITE packet to external SIP server and waits for 3XX response. I tried to get FS working as RS. My DP is very simple, I just want to see I get a call. <extension name="del-group"> <condition field="destination_number" expression="^.*$"> <action application="log" data="ALERT Here I am"/> </condition> </extension> Console output: 2014-06-07 00:41:18.475135 [NOTICE] switch_channel.c:1053 New Channel sofia/lcr_standard/230069@127.0.0.1:5062 [bc7fe4e8-7496-48b6-9381-a5debd7c2423] 2014-06-07 00:41:18.475135 [NOTICE] sofia.c:7112 Hangup sofia/lcr_standard/230069@127.0.0.1:5062 [CS_HIBERNATE] [MANDATORY_IE_MISSING] 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/lcr_standard/230069@127.0.0.1:5062) Ended 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1636 Close Channel sofia/lcr_standard/230069@127.0.0.1:5062 [CS_DESTROY] So, if the FS can not be used as SIP route server may be You can suggest another software? -- Regards, Boris

2 2

CPU Affinity
by Andre 06 Jun '14

06 Jun '14

Hi, Can anyone give me some best performance advise how I should set my CPU Affinity for using Freeswitch in a high performance Dialer environment? Thanks

1 0

Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls?
by Mark Boots 06 Jun '14

06 Jun '14

We're using a major VoIP terminator and finding that on many outgoing calls, Freeswitch hangs up with INCOMPATIBLE_DESTINATION after trying to negotiate codecs. The carrier offers PCMA, and it looks like Freeswitch refuses to accept anything but PCMU: 2014-06-06 13:18:54.115197 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0011104555493618652! 2014-06-06 13:18:54.115197 [DEBUG] switch_channel.c:3277 (sofia/external/0011104555493618652) Callstate Change DOWN -> EARLY 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/0011104555391494140 [BREAK] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/0011104555391494140 [BREAK] 2014-06-06 13:18:54.175210 [DEBUG] sofia.c:5741 Channel sofia/external/0011104555391494140 entering state [proceeding][183] 2014-06-06 13:18:54.175210 [DEBUG] sofia.c:5750 Remote SDP: v=0 o=- 313783328916230866 1 IN IP4 81.52.176.165 s=- c=IN IP4 81.52.176.165 t=0 0 a=sendrecv m=audio 25080 RTP/AVP 8 13 101 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2585 Set comfort noise payload to 13 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [CN:13:8000:20:0]/[PCMU:0:8000:20:64000] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2824 Set 2833 dtmf send payload to 101 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/0011104555391494140 [BREAK] 2014-06-06 13:18:54.175210 [NOTICE] sofia.c:5868 Hangup sofia/external/0011104555391494140 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] However, our codec settings in vars.xml specify: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> The sip profile uses these settings. Any idea how to debug what codecs Freeswitch is willing to accept? Thanks! +Mark

2 1

Writing FS apps via mod event socket + node.js
by Alex Lake 06 Jun '14

06 Jun '14

I'm interested in using node.js to write scripts to control Freeswitch-based telephony apps. This is partly because some of the apps we've been writing seem to be stretching the dialplan/Lua way of doing things - or at least my comprehension of the model - beyond what is commercially viable. What I mean is that writing and extending our apps (particularly when working with multiple call legs with complex inter-related state) is too expensive. It seems from reading around, that externalising the choreography of sessions is the choice of the fs cognoscenti. However, there seems to be very little by the way of documentation and example non-trivial "cookbook" style apps around - or maybe I've just not been looking in the right places. I get the strong impression that there are plenty of people here who've been there, done that and have the t-shirt. Just wondering what the chances are of a bit of sharing (obviously I'm happy to invest time into publishing my findings and creating new examples, etc)

3 3

Can't make call from sip to skype.
by Russell Treleaven 06 Jun '14

06 Jun '14

I can do IM skype to sip and sip to skype I can call from skype to sip When I try to call from sip to skype it immediately goes to voicemailbox 1000(which does not exist). Actually about 1 in 20 times I can complete a call . dialplan <extension name="ext_skype_user"> <condition field="destination_number" expression="^skype/(.*)$" > <action application="log" data="DEBUG !in the skype extension"/> <action application="info"></action> <action application="bridge" data="skypopen/skype101/$1" ></action> </condition> </extension> log http://pastebin.freeswitch.org/22691

1 0

Grandstream GXP2000 living-dead registrations - NAT config problem???
by Alex Lake 06 Jun '14

06 Jun '14

Got a strange problem that has only begun to manifest itself since we started using a host running FreeSWITCH 1.2.24. We were previously using a host running 1.2.13. Both hosts are hosted at Hetzner. I'm not 100% sure that the relevant parts of configuration haven't changed, although I don't think they have. The phones are on a NATted LAN connected via ADSL to the Internet. The problem is that, despite showing up as registered (and continuing to re-register), the phones can get into a state where they don't respond to inbound INVITEs. Other brands of phone (eg. Linksys and Yealink) appear NOT to suffer from this problem. The thing that baffles me is that this appears to be repeatable, but differs between the hosts. What parts of FS config should I be looking at for differences?

1 2

Need some help with SIP Gateway
by deepak sujir 06 Jun '14

06 Jun '14

Hi, I am looking for some help with SIP gateway setup. I have a scenario where the SIP provider needs the *To-Header* and *Request-Line* to have their domain, but the request needs to be routed through a proxy. *Gateway configuration* <include> <gateway name="sip-domain.com"> <param name="realm" value="*sip-domain.com <http://sip-domain.com>*" /> <param name="proxy" value="*sip-proxy.com <http://sip-proxy.com>*"/> <param name="register" value="*false*"/> <param name="from-domain" value="sip-domain.com"/> </gateway> </include> *Dial plan * <action application="bridge" data="sofia/gateway/sip-domain.com/xxxx" /> Now when I send the call out through the gateway, the *To-Domain* and the *Request-Line *looks like *sip:xxxx@sip-proxy.com <sip%3Axxxx(a)sip-proxy.com>*, but i need it to be something like *sip:xxxx@sip-domain.com <sip%3Axxxx(a)sip-domain.com>* Is there a way where that I can set the To-domain? (may be something similar to param *from-domain* ) Thanks Deepak

1 0

Freeswitch Multiple early-media
by Marino 06 Jun '14

06 Jun '14

Hi to all, i'm trying to implement something like this with FS but with bad result :( -)From a client i make parallel forking using FS; -)All receivers replies with 183 w/SDP; -)I would like that all 183 will arrive to client that can stream it's video to all receivers -)The first receiver that pick-up call establish connection With a simple call 1 to 1 everything is ok..but with forking FS get all the 183s messages but it doesn't forward it to the caller. I don't know if RFC allows this kind of scenario, but i would only know if it possible or i need to modify FS sourcecode. Many thanks to all

1 0

5ess decoding
by Kasyanov Dmitry 06 Jun '14

06 Jun '14

Hi, Is there way to implement FS to decode binary 5ess data. I found some api to process 5ess and q931 data, but not quite sure how to use it. Sincerely Kasyanov Dmitry

1 0