- FreeSWITCH-users - FreeSWITCH Mailing Lists

Send A-leg response code based on B-leg's response code
by E. Schmidbauer 12 Jun '15

12 Jun '15

Hello, We are trying to send a specific response code back to A-leg based on B-leg's response code. We need to send a 503 to A-leg if B-leg returns a 480 We have the following setup in the A-leg dialplan before the bridge ... <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_ignore_remote_cause=true"/> <action application="set" data="proto_specific_hangup_cause=sip:503"/> <action application="bridge" data="sofia/internal/sip:blah@blahblah.net:5060 "/> .... The b-leg returns 480 and then FS returns a 480 back to A-leg We cannot use the respond app in this case because of how our dialplan is generated. >From what I read on confluence <action application="set" data="sip_ignore_remote_cause=true"/> and <action application="set" data="proto_specific_hangup_cause=sip:503"/> should accomplish what i need here, but it does not seem to work. Is this a bug or am I misunderstanding those channel variables? Is there anyway to send a response code using channel variables (not using respond app)? Thanks, Emmanuel

2 1

Low pass filter
by Paul Atreides 12 Jun '15

12 Jun '15

Hi, is there a simple way to apply a low pass filter to the bleg? The problem which I am having is that some people who are hard of hearing tend to speak much louder in the upper frequency band than usual which can be very annoying. Thanks

3 2

creating users on the fly
by ashish.nijai 12 Jun '15

12 Jun '15

Hi, Can anybody update me how to create users on the fly without adding XML files? Also to get similar results as above can we use regex in user name? Regards, Ashish Nijai -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/creating-users-on-the-fly-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

Problem with jssip and Freeswitch communication using direct websocket
by Ricardo Hernandez 12 Jun '15

12 Jun '15

Good day, we are trying to use jssip directly with the websocket functionality in Freeswitch, using wss in port 7443. We are using tryit.jssip.net to test the connection with jssip. When the connection is established it shows a message about "Bad Media Description" so we think it is maybe related to the SDP that Freeswitch and jssip exchange. We tried to use jssip with only audio and it seems to ask for video also in the SDP, also we tried sip.js and works good with only audio. What seems different in the SDP is that jssip use: a=group:BUNDLE audio video a=msid-semantic: WMS PawM1oe2pNKiSnlcrCjI4eaWfWkbHgWHqDn5 m=audio 40283 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 And sip.js use: a=group:BUNDLE audio a=msid-semantic: WMS sb3vos9wXefDV2LO3NQmNPJXww23Hh5zK0zn m=audio 35907 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 I am copying the whole INVITE-Response first from jssip and last from sip.js to Freeswitch: Thank you in advance. *INVITE sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com> SIP/2.0* *Via: SIP/2.0/WSS rp3mpnou5jnl.invalid;branch=z9hG4bK8006947* *Max-Forwards: 69* *To: <sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com>>* *From: <sip:1000@conference2.cibersys.com <sip%3A1000(a)conference2.cibersys.com>>;tag=eq4cu21u2r* *Call-ID: t3dk0o0dj41vcm1agqj2* *CSeq: 3025 INVITE* *Proxy-Authorization: Digest algorithm=MD5, username="1000", realm="conference2.cibersys.com <http://conference2.cibersys.com>", nonce="642839d2-6425-4248-a6cc-606f51d4506a", uri="sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com>", response="7f8537dfbfb3de1a34c151d2d2b41307", qop=auth, cnonce="9cr63aqqk5tg", nc=00000001* *X-Can-Renegotiate: true* *Contact: <sip:fhh33jaa@rp3mpnou5jnl.invalid;transport=ws;ob>* *Content-Type: application/sdp* *Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS* *Supported: ice,outbound* *User-Agent: JsSIP 0.6.30* *Content-Length: 3709* *v=0* *o=- 1430994607190998965 2 IN IP4 127.0.0.1* *s=-* *t=0 0* *a=group:BUNDLE audio video* *a=msid-semantic: WMS PawM1oe2pNKiSnlcrCjI4eaWfWkbHgWHqDn5* *m=audio 40283 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126* *c=IN IP4 200.82.236.209* *a=rtcp:47588 IN IP4 200.82.236.209* *a=candidate:2999745851 1 udp 2122194687 192.168.56.1 49826 typ host generation 0* *a=candidate:1013097898 1 udp 2122129151 10.0.1.8 49827 typ host generation 0* *a=candidate:2999745851 2 udp 2122194686 192.168.56.1 49828 typ host generation 0* *a=candidate:1013097898 2 udp 2122129150 10.0.1.8 49829 typ host generation 0* *a=candidate:50610563 1 udp 1685921535 200.82.236.209 40283 typ srflx raddr 10.0.1.8 rport 49827 generation 0* *a=candidate:50610563 2 udp 1685921534 200.82.236.209 47588 typ srflx raddr 10.0.1.8 rport 49829 generation 0* *a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0* *a=candidate:1927371098 1 tcp 1518149375 10.0.1.8 0 typ host tcptype active generation 0* *a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype active generation 0* *a=candidate:1927371098 2 tcp 1518149374 10.0.1.8 0 typ host tcptype active generation 0* *a=ice-ufrag:+BRKFdwXRfJPhO6x* *a=ice-pwd:aM60hJkwlM6FNqZUqp10mknm* *a=fingerprint:sha-256 34:95:A3:46:CF:B5:8C:3B:5C:E9:1C:80:39:54:67:7F:AA:FA:10:E3:2D:64:0F:00:43:EB:D2:FE:4A:58:0A:DC* *a=setup:actpass* *a=mid:audio* *a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level* *a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>* *a=sendrecv* *a=rtcp-mux* *a=rtpmap:111 opus/48000/2* *a=fmtp:111 minptime=10; useinbandfec=1* *a=rtpmap:103 ISAC/16000* *a=rtpmap:104 ISAC/32000* *a=rtpmap:9 G722/8000* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:106 CN/32000* *a=rtpmap:105 CN/16000* *a=rtpmap:13 CN/8000* *a=rtpmap:126 telephone-event/8000* *a=maxptime:60* *a=ssrc:550712101 cname:c9UmE3sQGsfoTBmx* *a=ssrc:550712101 msid:PawM1oe2pNKiSnlcrCjI4eaWfWkbHgWHqDn5 37864d75-0547-4bf1-aa91-d122004f0169* *a=ssrc:550712101 mslabel:PawM1oe2pNKiSnlcrCjI4eaWfWkbHgWHqDn5* *a=ssrc:550712101 label:37864d75-0547-4bf1-aa91-d122004f0169* *m=video 34574 RTP/SAVPF 100 116 117 96* *c=IN IP4 200.82.236.209* *a=rtcp:48136 IN IP4 200.82.236.209* *a=candidate:2999745851 1 udp 2122194687 192.168.56.1 49830 typ host generation 0* *a=candidate:1013097898 1 udp 2122129151 10.0.1.8 49831 typ host generation 0* *a=candidate:2999745851 2 udp 2122194686 192.168.56.1 49832 typ host generation 0* *a=candidate:1013097898 2 udp 2122129150 10.0.1.8 49833 typ host generation 0* *a=candidate:50610563 2 udp 1685921534 200.82.236.209 48136 typ srflx raddr 10.0.1.8 rport 49833 generation 0* *a=candidate:50610563 1 udp 1685921535 200.82.236.209 34574 typ srflx raddr 10.0.1.8 rport 49831 generation 0* *a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0* *a=candidate:1927371098 1 tcp 1518149375 10.0.1.8 0 typ host tcptype active generation 0* *a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype active generation 0* *a=candidate:1927371098 2 tcp 1518149374 10.0.1.8 0 typ host tcptype active generation 0* *a=ice-ufrag:+BRKFdwXRfJPhO6x* *a=ice-pwd:aM60hJkwlM6FNqZUqp10mknm* *a=fingerprint:sha-256 34:95:A3:46:CF:B5:8C:3B:5C:E9:1C:80:39:54:67:7F:AA:FA:10:E3:2D:64:0F:00:43:EB:D2:FE:4A:58:0A:DC* *a=setup:actpass* *a=mid:video* *a=extmap:2 urn:ietf:params:rtp-hdrext:toffset* *a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>* *a=extmap:4 urn:3gpp:video-orientation* *a=recvonly* *a=rtcp-mux* *a=rtpmap:100 VP8/90000* *a=rtcp-fb:100 ccm fir* *a=rtcp-fb:100 nack* *a=rtcp-fb:100 nack pli* *a=rtcp-fb:100 goog-remb* *a=rtpmap:116 red/90000* *a=rtpmap:117 ulpfec/90000* *a=rtpmap:96 rtx/90000* *a=fmtp:96 apt=100* *--------------------------------------------------------------* *SIP/2.0 200 OK* *Via: SIP/2.0/WSS rp3mpnou5jnl.invalid;branch=z9hG4bK8006947;received=200.82.236.209;rport=35929* *From: <sip:1000@conference2.cibersys.com <sip%3A1000(a)conference2.cibersys.com>>;tag=eq4cu21u2r* *To: <sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com>>;tag=3t8cX7KNSpgDD* *Call-ID: t3dk0o0dj41vcm1agqj2* *CSeq: 3025 INVITE* *Contact: <sip:9196@167.114.64.133:5060;transport=udp>* *User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20150609T042756Z~18063a4754~64bit* *Accept: application/sdp* *Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH,* *SUBSCRIBE* *Supported: timer, path, replaces* *Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer* *Session-Expires: 120;refresher=uas* *Content-Type: application/sdp* *Content-Disposition: session* *Content-Length: 1509* *Remote-Party-ID: "9196" <sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com>>;party=calling;privacy=off;screen=no* *v=0* *o=FreeSWITCH 1433962816 1433962817 IN IP4 167.114.64.133* *s=FreeSWITCH* *c=IN IP4 167.114.64.133* *t=0 0* *a=msid-semantic: WMS 56duKQZAnkHTimXh0kShnZHA9HK7K17F* *m=audio 18690 RTP/SAVPF 0 126* *a=rtpmap:0 PCMU/8000* *a=rtpmap:126 telephone-event/8000* *a=ptime:20* *a=fingerprint:sha-256 68:8F:C5:D4:DB:3B:F2:05:F6:D2:A0:3A:0F:3E:BF:A5:71:3B:36:EA:9B:CF:23:E4:B2:A9:7C:FC:FB:47:5B:9E* *a=setup:active* *a=rtcp-mux* *a=rtcp:18690 IN IP4 167.114.64.133* *a=ice-ufrag:NYH0OfrNM1KnTITH* *a=ice-pwd:KJ6spVngh1eQukf79NYOSfsv* *a=candidate:0344283282 1 udp 659136 167.114.64.133 18690 typ host generation 0* *a=ssrc:3313344033 cname:Yb0yRPZ40wO8ICgz* *a=ssrc:3313344033 msid:56duKQZAnkHTimXh0kShnZHA9HK7K17F a0* *a=ssrc:3313344033 mslabel:56duKQZAnkHTimXh0kShnZHA9HK7K17F* *a=ssrc:3313344033 label:56duKQZAnkHTimXh0kShnZHA9HK7K17Fa0* *m=video 25170 RTP/SAVPF 100* *b=AS:2048* *a=rtpmap:100 VP8/90000* *a=sendonly* *a=fingerprint:sha-256 68:8F:C5:D4:DB:3B:F2:05:F6:D2:A0:3A:0F:3E:BF:A5:71:3B:36:EA:9B:CF:23:E4:B2:A9:7C:FC:FB:47:5B:9E* *a=setup:active* *a=rtcp-mux* *a=rtcp:25170 IN IP4 167.114.64.133* *a=rtcp-fb:100 ccm fir* *a=rtcp-fb:100 nack* *a=rtcp-fb:100 nack pli* *a=ssrc:2596374144 cname:Yb0yRPZ40wO8ICgz* *a=ssrc:2596374144 msid:56duKQZAnkHTimXh0kShnZHA9HK7K17F v0* *a=ssrc:2596374144 mslabel:56duKQZAnkHTimXh0kShnZHA9HK7K17F* *a=ssrc:2596374144 label:56duKQZAnkHTimXh0kShnZHA9HK7K17Fv0* *a=ice-ufrag:SGyc7yGzJFAvvmIj* *a=ice-pwd:iaAqXYuZA3a57jtPQRSNgodP* *a=candidate:6626792977 1 udp 659136 167.114.64.133 25170 typ host generation 0* *-------------------------------------------* *INVITE sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com> SIP/2.0* *Via: SIP/2.0/WSS g4413q00qplk.invalid;branch=z9hG4bK3856295* *Max-Forwards: 70* *To: <sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com>>* *From: <sip:1000@conference2.cibersys.com <sip%3A1000(a)conference2.cibersys.com>>;tag=8i0i1tu4sa* *Call-ID: brrcaauhjvt62n23i72j* *CSeq: 2276 INVITE* *Contact: <sip:re2ehtpr@g4413q00qplk.invalid;transport=ws;ob>* *Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE* *Content-Type: application/sdp* *Support* *ed: outbound* *User-Agent: SIP.js/0.6.3-devel BB* *Content-Length: 2105* *v=0* *o=- 3214222955446291314 2 IN IP4 127.0.0.1* *s=-* *t=0 0* *a=group:BUNDLE audio* *a=msid-semantic: WMS sb3vos9wXefDV2LO3NQmNPJXww23Hh5zK0zn* *m=audio 35907 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126* *c=IN IP4 200.82.236.209* *a=rtcp:43823 IN IP4 200.82.236.209* *a=candidate:2999745851 1 udp 2122194687 192.168.56.1 64356 typ host generation 0* *a=candidate:1013097898 1 udp 2122129151 10.0.1.8 64357 typ host generation 0* *a=candidate:2999745851 2 udp 2122194686 192.168.56.1 64358 typ host generation 0* *a=candidate:1013097898 2 udp 2122129150 10.0.1.8 64359 typ host generation 0* *a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0* *a=candidate:1927371098 1 tcp 1518149375 10.0.1.8 0 typ host tcptype active generation 0* *a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype active generation 0* *a=candidate:1927371098 2 tcp 1518149374 10.0.1.8 0 typ host tcptype active generation 0* *a=candidate:50610563 1 udp 1685921535 200.82.236.209 35907 typ srflx raddr 10.0.1.8 rport 64357 generation 0* *a=candidate:50610563 2 udp 1685921534 200.82.236.209 43823 typ srflx raddr 10.0.1.8 rport 64359 generation 0* *a=ice-ufrag:qbn8dYtQ7XOYCeu8* *a=ice-pwd:qMkTalQhiMGIO5EdyBWGOP+i* *a=fingerprint:sha-256 6B:98:3C:1C:65:BC:C7:03:9C:6F:3E:3C:C4:DB:DF:64:42:2D:46:67:9D:71:1E:10:AB:3A:32:FD:3A:B5:32:82* *a=setup:actpass* *a=mid:audio* *a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level* *a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>* *a=sendrecv* *a=rtcp-mux* *a=rtpmap:111 opus/48000/2* *a=fmtp:111 minptime=10; useinbandfec=1* *a=rtpmap:103 ISAC/16000* *a=rtpmap:104 ISAC/32000* *a=rtpmap:9 G722/8000* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:106 CN/32000* *a=rtpmap:105 CN/16000* *a=rtpmap:13 CN/8000* *a=rtpmap:126 telephone-event/8000* *a=maxptime:60* *a=ssrc:1770879119 cname:Z4x7vzi6Pgjt5LSz* *a=ssrc:1770879119 msid:sb3vos9wXefDV2LO3NQmNPJXww23Hh5zK0zn 28966a00-fcda-4d5c-8b42-2d232033a248* *a=ssrc:1770879119 mslabel:sb3vos9wXefDV2LO3NQmNPJXww23Hh5zK0zn* *a=ssrc:1770879119 label:28966a00-fcda-4d5c-8b42-2d232033a248* *------------------------------------* *SIP/2.0 200 OK* *Via: SIP/2.0/WSS g4413q00qplk.invalid;branch=z9hG4bK4420536;received=200.82.236.209;rport=38294* *From: <sip:1000@conference2.cibersys.com <sip%3A1000(a)conference2.cibersys.com>>;tag=8i0i1tu4sa* *To: <sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com>>;tag=87685e86a3ZaQ* *Call-ID: brrcaauhjvt62n23i72j* *CSeq: 2277 INVITE* *Contact: <sip:9196@167.114.64.133:5060;transport=udp>* *User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20150609T042756Z~18063a4754~64bit* *Accept: application/sdp* *Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH,* *SUBSCRIBE* *Supported: timer, path, replaces* *Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer* *Session-Expires: 120;refresher=uas* *Content-Type: application/sdp* *Content-Disposition: session* *Content-Length: 823* *Remote-Party-ID: "9196" <sip:9196@conference2.cibersys.com <sip%3A9196(a)conference2.cibersys.com>>;party=calling;privacy=off;screen=no* *v=0* *o=FreeSWITCH 1433961875 1433961876 IN IP4 167.114.64.133* *s=FreeSWITCH* *c=IN IP4 167.114.64.133* *t=0 0* *a=msid-semantic: WMS avWcdKMQxwOgv3mzbNyZb72DIOJmABIA* *m=audio 19902 UDP/TLS/RTP/SAVPF 0 126* *a=rtpmap:0 PCMU/8000* *a=rtpmap:126 telephone-event/8000* *a=ptime:20* *a=fingerprint:sha-256 68:8F:C5:D4:DB:3B:F2:05:F6:D2:A0:3A:0F:3E:BF:A5:71:3B:36:EA:9B:CF:23:E4:B2:A9:7C:FC:FB:47:5B:9E* *a=setup:active* *a=rtcp-mux* *a=rtcp:19902 IN IP4 167.114.64.133* *a=ice-ufrag:oM6qJYUhCtt3V2uI* *a=ice-pwd:nXyi35Bvh0jmBxtQ9p7IeRQG* *a=candidate:2466874999 1 udp 659136 167.114.64.133 19902 typ host generation 0* *a=ssrc:3313344305 cname:WuD05fGsClzYfLng* *a=ssrc:3313344305 msid:avWcdKMQxwOgv3mzbNyZb72DIOJmABIA a0* *a=ssrc:3313344305 mslabel:avWcdKMQxwOgv3mzbNyZb72DIOJmABIA* *a=ssrc:3313344305 label:avWcdKMQxwOgv3mzbNyZb72DIOJmABIAa0*

2 3

TOO Frequent SIP Registration
by Sean Devoy 12 Jun '15

12 Jun '15

Hi everyone, Forgive my lack of knowledge here. I have a multi-tenant server setup and oddly everything is working correctly! Working correctly isn't odd, writing when everything is ok is odd. For one of my domains, all of the phones re-register about every 30 seconds. The log is filled with messages like: sofia_reg.c:1532 SIP auth challenge (REGISTER) on sofia profile 'external' for [43@<snip>] On the phones I changed the "Reg Min Expires" to 300 and they disappeared from registration!! Also, the phones are registered, I can call them and their status web page says next registration is xx (30 or less) seconds. So ... How often show registered phones re-register? How/Where do I set it? Why only one of my domains? (This domain may be the only one to get the latest firmware in the phones). For extra points does this relate to NAT Keepalive at all? I did google this but the only registration expiration info I can find is on gateways, not phones. Thanks, Sean

2 6

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 12 Jun '15

12 Jun '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

Setting up freeswitch server on Google cloud compute
by Pandurang Waghulde 12 Jun '15

12 Jun '15

Hi all, I am having trouble in setting up freeswitch server on Google cloud compute. Here are the details about issue that i am facing. http://stackoverflow.com/questions/30772637/setting-up-freeswitch-server-on… It would be great help if anyone can tell me how i can get rid of this issue? Thanks

7 9

FS on Intel NUC
by Nandy Dagondon 12 Jun '15

12 Jun '15

Hi folks! Anyone tried installing Freeswitch on an Intel NUC? Which board kit? Tks /Nandy

2 2

Call quality assurance
by Stanislav Sinyagin 11 Jun '15

11 Jun '15

I summarized some experience gained in a customer project, hopefully it will be useful: https://txlab.wordpress.com/2015/06/02/quality-assurance-for-voip-calls-2/ @Anthony, you suggested to use HPET timer for better precision, but it appears that it is affected by the CPU load: while the debian package build script was running, the quality of recorded calls has degraded -- even that the recording was done onto a TMPFS partition. With the default TSC timer, this was not the case.

4 8

Freetdm group
by Matt Broad 11 Jun '15

11 Jun '15

HI, I have a sangoma A102 card and have configured it using the Freetdm example (https://freeswitch.org/confluence/display/FREESWITCH/FreeTDM) I can make calls out using both spans, but I have a couple of questions: in the below CLI command, what does the "a" refer to? "originate freetdm/1/a/0123412345 &conference(123)" secondly, I will be using both ports to make outbound calls only and I am trying to add both spans to a group so that I can send calls out over the next available line. I have the following as my freetdm.conf *[span wanpipe wp1] trunk_type => e1 group=PSTN b-channel => 1:1-15 b-channel => 1:17-31 d-channel => 1:16 [span wanpipe wp2] trunk_type => e1 group=PSTN b-channel => 2:1-15 b-channel => 2:17-31 d-channel => 2:16 * When I try to make a call out using "originate freetdm/PSTN/a/0123412345 &conference(123)" I get the error "*[ERR] mod_freetdm.c:1390 Missing ftdm span or group: PSTN*" Any ideas why this would be? Do I need to declare the group elsewhere? I have tried configuring the card using "wancfg_fs" and doing it manually but both result in the error (though wancfg_fs requires the group to contain a number) thanks Matt

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