August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

List of $$ variables
by Grant Bagdasarian 21 Aug '13

21 Aug '13

Hello, Is there a list available for variables starting with $$? Like $${local_ip_v4}? Does the following xml element support multiple values? <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto "/> Something like: <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.1.1,10.0.0.1 "/> #Notice the comma separating the bind IP's. Thanks, Grant

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Play sound over local_stream (Mix sounds)
by Sayyed Mohammad Emami Razavi 21 Aug '13

21 Aug '13

I want play sound over local_stream without stopping it! Now i use uuid_broadcast api with aleg argument but it stops MOH any solution?!

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nate-mode -nonat apply-nat-acl documentation
by Karl Schmidt 21 Aug '13

21 Aug '13

OK -nonat turns off NAT detection, but I don't know if this applies to external SIP clients behind NAT? =============================================================================================== There are clearly 2 separate cases where NAT detection may be needed and some way to enable each one independent of the other. =============================================================================================== The following explores my confusion: apply-nat-acl If value is rfc1918 enable 'nat-mode' if IP matches - OK but what exactly is nat-mode? And what is value="nat.auto" ? How does it relate to auto-nat? Does any of this do anything if I'm running with -nonat ? Is running with -nonat the opposite of nat-mode? nat-mode could mean several things and I can't find it defined anywhere. There is also -nonatmap which allows NAT detection but avoids UPnP/NAT-PMP In my situation, I have FS situated behind a firewall and have the proper ports DNATed. I'm running with -nonat and this works, and should not create any problems. BUT there could be situations where I am out of town and need to connect to the server from behind some NAT that I have no control over. Thus there are two separate situations that might want NAT detection - FS behind NAT and a SIP client behind NAT. ext-rtp-ip, ext-sip-ip, rtp-ip, and sip-ip Should tell FS what it needs to know about the server end - but I'm not finding clarity on what a sane setup would be for the possibility of an external SIP client behind NAT. ,.,. I think the best way to document this is to come up with a few scenarios and show a sane set up for each one. We can assume in all cases that phones on both sides of NAT need to register and if the server is on the LAN side of a NAT there could be need for a double transit of NAT. 1 - FS server on a LAN with ports DNATed by firewall - static IP - Phones on LAN and phones registering from Internet 2 - FS on public IP by phones on the other side of a NAT firewall need to register as well as phones on bare internet 3 - FS server on LAN with firewall on dynamic IP 4 - Just so we are on the same page: DNAT (Destination network address translation) DNAT is a technique for transparently changing the destination IP address of an en route packet and performing the inverse function for any replies. This use of DNAT is also called port forwarding. I'm trying to document this for myself and finding it similar to walking in a swamp each step is getting me bogged down. -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Inflation is taxation without legislation. Milton Friedman --------------------------------------------------------------------------------

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INCOMPATIBLE_DESTINATION
by Royce Mitchell III 20 Aug '13

20 Aug '13

Hi, My client has been running FreeSWITCH with low call volume and has been stable for several months now. This Tuesday morning we switched all call volume over to FreeSWITCH and everything was running fine until this morning. Out of the blue, *all* calls started rejecting with INCOMPATIBLE_DESTINATION. Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a trunk build from approximately 7/11/2013. Is this an issue that an update could fix? Could this be port starvation? Please assist, thanks. Here's a snippet from the freeswitch log where the first occurrence of that error: 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel sofia/external/******9333@192.168.1.212:5080[bfa3419b-0f79-4220-8807-773e3f680751] 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/******9333@192.168.1.212:5080 [BREAK] 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/******9333@192.168.1.212:5080 [BREAK] 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416 (sofia/external/******9333@192.168.1.212:5080) Running State Change CS_NEW 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434 (sofia/external/******9333@192.168.1.212:5080) State NEW 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel sofia/external/ ******9333@192.168.1.212:5080 entering state [received][100] 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP: v=0 o=- 1376662098 1 IN IP4 192.168.1.193 s=- c=IN IP4 192.168.1.193 t=0 0 m=audio 0 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup sofia/external/ ******9333@192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] Royce Mitchell, IT Consultant ITAS Solutions royce3(a)itas-solutions.com

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mod_sms endpoints
by Bob McCarthy 20 Aug '13

20 Aug '13

Is it possible to create a fictious endpoint to route sms messages to ? using mod_sms to send messages, I am intercepting the messages and disseminating them via the event socket. By sending them to nowhere I get a nuisance error message. Bob McCarthy

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different RTP port ranges per sip profile
by Łukasz Dobrodziej 20 Aug '13

20 Aug '13

Hello Is it possible to define different RTP port ranges for different sip profiles? For example the Internal sip profile would use ports 60100-60200 for RTP traffic and the External sip profile would use ports 1100-1200. I can set RTP port range in switch.conf.xml file using rtp-start-port and rtp-end-port params but it is global configuration. Moreover configuration from switch.conf.xml forces a continuity of RTP port range (one start and end). Is it possible to define disjunctive ranges? -- Lucas

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Inexplicable and silent FreeSWITCH crash
by Emrah 20 Aug '13

20 Aug '13

Guys, Very recently my FS instance has been crashing continuously and there is nothing in the logs to help… I am on log 9 and there is absolutely nothing out of the ordinary. FS just dies on me. This has started happening ever since I started playing with video and proxying media. How do I get more verbose and get to know what crashes FS? I tried both the Master branch and the V1.4.Beta. Thanks for any help I'm going nuts. Emrah

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Auto-detecting Local IP when no gateway set
by Siri MM 20 Aug '13

20 Aug '13

Hi All, I am using freeswitch without local_ip_v4 explicitly set, i.e, I am dependant on freeswitch auto-detecting the local IP address. >From the code, what I understand is, in order to determine the local IP address, freeswitch tries to ‘connect()’ using a DGRAM socket with BROADCAST option set. The problem that I am having is, my customer is working in a Local LAN, and is not very keen on setting the interface gateway. Without gateway set, connect() returns 'Network unreachable' error. Is there any way I could workaround this problem? (apart from setting local_ip_v4) Also, any other issues forseen if gateway isn't set ont he interface?

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Call not hanging up
by Peter 20 Aug '13

20 Aug '13

Running FS 1.2.12 in the following environment: FS Ext <---> FS SBC <---> FS Int I can successfully make calls in both directions. If the caller hangs up, the other end correctly hangs up. However, if the callee hangs up, the caller session remains active until the RTP timeout kicks in. Here's a snippet from the FS log where this occurs (1004 is the caller and 2010 is the callee) [NOTICE] sofia.c:716 Hangup sofia/internal/2010@10.1.1.206:5061[CS_HIBERNATE] [NORMAL_CLEARING] [NOTICE] switch_ivr_bridge.c:1109 Hangup sofia/internal/1004(a)10.1.1.204[CS_HIBERNATE] [NORMAL_CLEARING] [NOTICE] switch_core_session.c:1560 Session 4 (sofia/internal/ 2010@10.1.1.206:5061) Ended [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/ 2010@10.1.1.206:5061 [CS_DESTROY] [NOTICE] switch_core_session.c:1560 Session 3 (sofia/internal/ 1004(a)10.1.1.104) Ended [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/ 1004(a)10.1.1.104 [CS_DESTROY] I've tried setting hangup_after_bridge=true in the dialplan on FS Ext. I wasn't 100% sure whether it should go on FS Ext or FS SBC, so I tried both and it made no difference. What else should I be checking? Thanks Peter

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unable to subscrib to freeswitch-sec
by covici＠ccs.covici.com 20 Aug '13

20 Aug '13

Hi. I am unable to subscribe to the freeswitch-sec list. When I get the confirmation, the address it wants to send the reply to is tron.freeswitch.org which seems not to exist -- can't ping or anything. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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