April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL Client Library Managed wrappers (Windows)
by Dave R. Kompel 04 May '13

04 May '13

I just spent the good part of this wekend trying to get them compilied, tested and working, it's been over 2 years since I had to build them, and it's a bitch. To save ppl time, I put the binaries up, and made them available if anyone needs them. You can grab them at: http://download.drknetworking.com/eslmanaged.zip --Dave

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Alarm, KISS OFF
by Miha 03 May '13

03 May '13

Hi, I have problem regarding alarm device. Alarm guys are saying that I am not detecting kiss off tone or something like that. Is is send on 1400Hz. I have been trying to detect this tone with mod spandsp but with no luck. has some manage to get this working or getting this to work on FS? Thanks! Miha

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Check if UA is still there?
by Peter Steinbach 03 May '13

03 May '13

Hello, is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs. Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface? An help is welcome. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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T38 Re-INVITE: FS responds 100 Trying, nothing further happens
by Örn Arnarson 03 May '13

03 May '13

Hello, I'm trying to get T38 gatewaying to work. I have a call coming in with FAX audio, and an endpoint that needs T38. The endpoint will send a T38 re-invite, to which FS responds 100 trying, and after 40 seconds the endpoint will give up (citing stale 100 trying). I set these variables before briding: <action application="set" data="fax_enable_t38=true"/> <action application="set" data="fax_enable_t38_request=true"/> And then I bridge like this (dialstring is sofia/profile/number@ip:port): <action application="bridge" data="{execute_on_image=t38_gateway self nocng}${dialstring}"/> I've tried all variants I can think of, and always have the same result, except for when fax_enable_t38 = false, in which case FS rejects the re-invite. I found some old discussions on the mailing list alluding to the same problem, but not much in the way of solutions. Can anybody give me a pointer as to what I'm doing wrong? I was running a git clone from 1.2.2, but I upgraded to latest stable (1.2.7) with the same results. Here's what happens in FS. At the very end you can see the Re-INVITE, and then nothing happens post that point. http://pastebin.freeswitch.org/20853 Best regards, Örn

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freetdm RBS configuration for multi-port cards
by bedgar＠vseinc.com 01 May '13

01 May '13

Have an issue for freetdm experts: 1. I have it working with one port but what is the proper syntax for multiple Sangoma RBS T1 ports in freetdm.conf and freetdm.conf.xml? The installation scripts wancfg_fs and wancfg_dahdi put them in the PRI type notation and cannot find how to rework for multiple RBS. Here is a working stripped down to only one port. This is after going through script and only setting one port as used. Current: freetdm.conf [span zt wp1] trunk_type => EM em-channel => 1-24 freetdm.conf.xml <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/>    </settings> <analog_em_spans> <span name="wp1"> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </analog_em_spans> </configuration> Thanks for looking and helping if you can! Brian C. Edgar, Jr.

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BT claims licensing on SIP
by Cal Leeming [Simplicity Media Ltd] 01 May '13

01 May '13

Hello, Thought I'd share this with you all... So there I was, quietly enjoying myself on theregister, when I stumbled upon what can only be described, as a complete and utter fail. http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_goog… Responses so far? <@SwK> lololol * foxx2 facepalm <coppice> BT come up with a regular stream of patents with no basis in reality <mcrane> they shouldn't be able to patent a standard protocol I'm fairly sure my immediate response would be to print out 50 copies of [1] and send it via recorded delivery. It would be interesting to hear from anyone who has already been contacted by BT about this, and how they plan to deal with it. I've had enough of the internet for one day ;/ Cal [1] http://25.media.tumblr.com/d0efdb79f57815eff2105f65da4f4df7/tumblr_mh0qugfw…

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Not Reachable
by Blake Priddy 30 Apr '13

30 Apr '13

I am still having the random no b-leg audio until 7 sec after call is answered issue and I have seen in wireshark that when someone says they have a delay the capture shows a destination unreachable about 2-4 times on wireshark. I have changed all my QoS settings for my vlans and I am still having an issue. Attached is a screenshot of the log -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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user-gateways: unregister gateway when user logs out?
by Michel Brabants 30 Apr '13

30 Apr '13

Hello, I'm user-specific gateways and they come up nicely when the user registers. However, they don't seem to be removed (which I desire) when the user unregisters (or there is timeout causing the unregister to happen). Is there a way to do this without resorting to esl? Thanks, Michel

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video call from X-lite(sip protocol) to Ekiga(h323 protocol) fails
by Navnath Sonavne 30 Apr '13

30 Apr '13

Hi Nikolay Kondratyev, I read your reply to opal to sip problem issue at http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066082… You said you use FS with mod_h323 ad sip<->h.323 gateway successfully. This what I want and I am hoping your help. Here is my issue: My video call is failing from X-Lite softphone(SIP protocol) to Ekiga softphone(h323 protocol). Ekiga softphone is answering the call but there is NO VIDEO transfer only audio transfer. My X-Lite softphone is at 192.168.9.165 and Ekiga softphone is at 192.168.8.27. Freeswitch server ip : 192.168.8.41 I have made changes in Local_Extension of default.xml such that if any registered extension has been dialed ,call will be forwarded to softphones at 192.168.8.27. <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.103"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension>. Here is my freeswitch console log for video call from X-Lite softphone(SIP protocol) to Ekiga softphone(h323 protocol): 2013-04-29 02:34:00.220733 [NOTICE] switch_channel.c:978 New Channel sofia/internal/1100(a)192.168.8.41 [c7188b62-81ca-4225-9c8e-ccdb32e97af2] 2013-04-29 02:34:00.260553 [INFO] mod_dialplan_xml.c:557 Processing navnath <1100>->1101 in context default 2013-04-29 02:34:00.479931 [NOTICE] switch_channel.c:978 New Channel h323/1101(a)192.168.8.27 [cf6f458c-c443-40ea-ab5a-6660853b5df8] 2013-04-29 02:34:00.479931 [WARNING] switch_core_session.c:635 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:34:00.479931 [WARNING] switch_core_session.c:648 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.711-ALaw-64k' format 'PCMA' 8 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.711-uLaw-64k' format 'PCMU' 0 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'GSM-06.10' format 'GSM' 3 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729A/B' format 'G729' 18 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729A' format 'G729' 18 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729B' format 'G729b' 18 2013-04-29 02:34:00.479931 [NOTICE] mod_h323.cpp:856 capability 'G.729' format 'G729' 18 (.navn 61 74 68 00 6c 05 81 31 31 30 31 70 05 81 31 31 ath.l..1101p..11 30 31 7e 03 06 05 20 b8 06 00 08 91 4a 00 06 01 01~... .....J... 01 80 44 34 28 c0 09 00 00 3d 14 46 72 65 65 53 ..D4(....=.FreeS 57 49 54 4 2013-04-29 02:34:00.600359 [NOTICE] mod_h323.cpp:1147 Ring-Ready h323/1101(a)192.168.8.27! 2013-04-29 02:34:00.600359 [INFO] h323ep.cxx:3048 Received alerting PDU. (.. 2013-04-29 02:34:00.620246 [NOTICE] mod_sofia.c:2035 Ring-Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:34:00.620246 [NOTICE] switch_ivr_originate.c:521 Ring Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:34:12.819989 [NOTICE] mod_h323.cpp:1012 Hangup h323/1101(a)192.168.8.27 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2013-04-29 02:34:12.839928 [INFO] mod_dptools.c:3088 Originate Failed. Cause: CALL_REJECTED 2013-04-29 02:34:12.839928 [NOTICE] mod_dptools.c:3208 Hangup sofia/internal/1100(a)192.168.8.41 [CS_EXECUTE] [CALL_REJECTED] 2013-04-29 02:34:12.839928 [INFO] h323.cxx:1273 Connection ip$localhost/18648 terminated. 2013-04-29 02:34:12.839928 [NOTICE] switch_core_session.c:1566 Session 1 (sofia/internal/1100(a)192.168.8.41) Ended 2013-04-29 02:34:12.839928 [NOTICE] switch_core_session.c:1570 Close Channel sofia/internal/1100(a)192.168.8.41 [CS_DESTROY] 2013-04-29 02:34:13.819657 [NOTICE] switch_core_session.c:1566 Session 2 (h323/1101(a)192.168.8.27) Ended 2013-04-29 02:34:13.819657 [NOTICE] switch_core_session.c:1570 Close Channel h323/1101(a)192.168.8.27 [CS_DESTROY] 2013-04-29 02:35:51.759989 [NOTICE] switch_channel.c:978 New Channel sofia/internal/1100(a)192.168.8.41 [23720e51-1818-4a08-aa36-958ebc19cf45] 2013-04-29 02:35:51.799989 [INFO] mod_dialplan_xml.c:557 Processing navnath <1100>->1101 in context default 2013-04-29 02:35:52.019984 [NOTICE] switch_channel.c:978 New Channel h323/1101(a)192.168.8.27 [09ee14e2-dffe-435d-b58f-6db0b8ac1314] 2013-04-29 02:35:52.019984 [WARNING] switch_core_session.c:635 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:35:52.019984 [WARNING] switch_core_session.c:648 h323/1101(a)192.168.8.27 does not support the proxy feature, disabling. 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.711-ALaw-64k' format 'PCMA' 8 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.711-uLaw-64k' format 'PCMU' 0 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'GSM-06.10' format 'GSM' 3 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729A/B' format 'G729' 18 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729A' format 'G729' 18 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729B' format 'G729b' 18 2013-04-29 02:35:52.019984 [NOTICE] mod_h323.cpp:856 capability 'G.729' format 'G729' 18 (.navn 61 74 68 00 6c 05 81 31 31 30 31 70 05 81 31 31 ath.l..1101p..11 30 31 7e 03 06 05 20 b8 06 00 08 91 4a 00 06 01 01~... .....J... 01 80 44 34 28 c0 09 00 00 3d 14 46 72 65 65 53 ..D4(....=.FreeS 57 49 54 4 2013-04-29 02:35:52.799941 [NOTICE] mod_h323.cpp:1147 Ring-Ready h323/1101(a)192.168.8.27! 2013-04-29 02:35:52.799941 [INFO] h323ep.cxx:3048 Received alerting PDU. (.. 2013-04-29 02:35:52.819889 [NOTICE] mod_sofia.c:2035 Ring-Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:35:52.819889 [NOTICE] switch_ivr_originate.c:521 Ring Ready sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:36:06.719989 [INFO] h323ep.cxx:3115 Received connect PDU. 2013-04-29 02:36:06.719989 [NOTICE] mod_h323.cpp:1258 Channel [h323/1101(a)192.168.8.27] has been answered 2013-04-29 02:36:06.739922 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1100(a)192.168.8.41! 2013-04-29 02:36:06.739922 [NOTICE] switch_ivr_originate.c:3411 Channel [sofia/internal/1100(a)192.168.8.41] has been answered 2013-04-29 02:36:12.959989 [NOTICE] mod_h323.cpp:1012 Hangup h323/1101(a)192.168.8.27 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-04-29 02:36:12.959989 [INFO] h323.cxx:1273 Connection ip$localhost/18649 terminated. 2013-04-29 02:36:12.979942 [NOTICE] switch_ivr_bridge.c:1516 Hangup sofia/internal/1100(a)192.168.8.41 [CS_EXECUTE] [NORMAL_CLEARING] 2013-04-29 02:36:12.979942 [NOTICE] switch_core_session.c:1566 Session 3 (sofia/internal/1100(a)192.168.8.41) Ended 2013-04-29 02:36:12.979942 [NOTICE] switch_core_session.c:1570 Close Channel sofia/internal/1100(a)192.168.8.41 [CS_DESTROY] 2013-04-29 02:36:13.979874 [NOTICE] switch_core_session.c:1566 Session 4 (h323/1101(a)192.168.8.27) Ended 2013-04-29 02:36:13.979874 [NOTICE] switch_core_session.c:1570 Close Channel h323/1101(a)192.168.8.27 [CS_DESTROY] -Navnath.

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mod_voicemail: skip instructions after recording
by Peter Steinbach 30 Apr '13

30 Apr '13

Hello, I am wondering how I can skip the instructions to the caller AFTER recoding his message. Skipping the instructions BEFORE recording works nicely, but I have found no information how to avoid the instructions being played AFTER the caller has spoken his message. I simply want mod_voicemail to hangup as soon as he has detected that the recording has finished. Anybody has a clue how to solve this? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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