October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_limit does not reset count when call gets to voicemail?
by Yehavi Bourvine 15 Oct '10

15 Oct '10

Hello, We are using mod_limit in order to track user's state and know when to play busy/call-wait tone. We encountered two problems (which are probably related): - When the call gets to the voicemail the extension is still though as busy until the caller hangs. - If a call is picked-up using the intercept application the original destination is still thought as busy. When looking in the limit_data table the record of the call is indeed there until a final call hangup. According to the wiki we tried setting the variable limit_ignore_transfer to false both with set command before calling the limit application and also in limit.conf.xml. However, this does not change this behaviour. Is this the intended behaviour? Can it work as I want? (i.e. once the call arrived to voicemail or picked-up the original destination is marked as free). We are using 1.0.6; since this is a production system it is quite hard for me to test the latest git. Thanks! __Yehavi:

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Java_ESL_Client - unhandled exception caught
by Vedran Zeljeznak 15 Oct '10

15 Oct '10

hello, i've found a bug in Java_ESL_Client and the info on the website (http://wiki.freeswitch.org/wiki/Java_ESL_Client#Issues) said it should be reported here so i'm sending it. It looks like sometimes FS sends malformed URL encoded values through Event Socket, probably because value is not properly terminated (probably on my version of FS only - FreeSWITCH Version 1.0.6 (hacked-20100420T130612Z) - official version). exception trace follows: #### TRACE / LOG info 14.10.2010. 11:16:29 org.jboss.netty.channel.SimpleChannelUpstreamHandler WARNING: EXCEPTION, please implement org.freeswitch.esl.client.inbound.InboundClientHandler.exceptionCaught() for proper handling. java.lang.IllegalArgumentException: URLDecoder: Illegal hex characters in escape (%) pattern - For input string: "\5" at java.net.URLDecoder.decode(URLDecoder.java:173) at org.freeswitch.esl.client.transport.event.EslEvent.parsePlainBody(EslEvent.java:187) at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:73) at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:62) at org.freeswitch.esl.client.internal.AbstractEslClientHandler.messageReceived(AbstractEslClientHandler.java:80) at org.freeswitch.esl.client.internal.debug.ChannelEventRunnable.run(ChannelEventRunnable.java:76) at org.jboss.netty.handler.execution.MemoryAwareThreadPoolExecutor$MemoryAwareRunnable.run(MemoryAwareThreadPoolExecutor.java:541) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:886) at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:908) at java.lang.Thread.run(Thread.java:619) #### TRACE / LOG info --- Vedran Zeljeznak

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tls/ssl gateway insists on registering SIP
by Jerre Cope 15 Oct '10

15 Oct '10

I'm attempting to implement the Hybrid model as shown on http://wiki.freeswitch.org/wiki/Tls#Hybrid_Encryption FS(Truck)=ssl/tls=>FS(Office) My idea was to create a dial plan so that when FS(Truck) dials 999, a secure connection could be made to the office. The pastebin shows my directory entry in FS(Office) and the external gateway in FS(truck) http://pastebin.freeswitch.org/14245 The gateway insists on connecting on 5060, calls made through the gateway appear in tshark to be using tls. I can block 5060 at the FS(office) firewall after registration and make calls through the gateway OK until FS(truck) tries to register again. The FS(truck) default gateway is voip.ms, the idea being non-secured calls would go out the voip.ms gateway so as to limit the traffic to FS(office). I'm operating way beyond my experience, so my attempt at this may be a goofy start. I'm asking for a little direction/perspective on how to implement. I'm using ssl because all my FS(trucks) will be dynamic IPs. Thanks for all the really cool programming/documentation so far. I'm having great fun!

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valet_park timeout and spot announcement
by Jeremy Stricker 15 Oct '10

15 Oct '10

Hello all, Is there anyway to set a call timeout when a call is valet parked so that it will ring back to the extension that parked it if it isn't answered by the intended recipient? We have an user who parks calls and then blindly announces the parking spot over an intercom. With this setup there is a great risk of a caller getting stuck in park. If there isn't a timeout method, can anyone recommend any alternate setup that would allow at least two parking spots with visual indication of calls parked on the handsets (Aastra 6730i and 6757i). Secondly, is there anyway to stop read-back of the parking spot on the parkee's side of the call? Currently, the parker performs an attended transfer, waits for and hears the spot read-back. The parkee hears MOH while the transfer is being performed, but then the spot announcement themselves as soon as the parker finishes the attended transfer by hanging up. Thanks for any help available. Jeremy -- Jeremy Stricker LightNex Communications e: jstricker(a)lightnex.com p: 877-342-3768 w: www.lightnex.com

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early media problem?
by Mario 15 Oct '10

15 Oct '10

I had an extension working fine that played a file, ringed, etc. to an inbound caller before we picked up. Trying to fix a problem the ITSP moved the account and now the caller only hears ringing, none of my stuff is heard even though it's shows as executed in trace. I looked on the wiki and tried adding <action application="ring_ready"/> at the top but that did not help. Anyone have any idea how to get all my work working again? Is there something I can change or the ITSP? Thanks. Mario

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reload caused freeswitch to hang
by Woody Dickson 15 Oct '10

15 Oct '10

Hi, I was able to use the reload API to reload any of my application, but after I upgraded to git version, I found that the RELOAD APP would cause the whole freeswitch to hang. Even "sofia profile internal restart" causes the entire freeswitch console to hang. Does anyone encounter the same issue I am having with the latest git? Thank, Woody

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"ab" logging in cdr_csv
by Christian Hiller 15 Oct '10

15 Oct '10

Hello, A-leg and B-leg have the same channel variables. When enabling "ab" logging in cdr_csv, how can i distinguish the variables of both legs ? Regards Chris Powered by yellox.com

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icall's carrier service returns 480 Temporarily Unavailable for outbound INVITE
by mdale111＠verizon.net 14 Oct '10

14 Oct '10

Hello all, I'm going on the fourth day with no response from icall's support on this issue, so I would greatly appreciate any guidance. Freeswitch successfully registers with icalls outbound server, sbc01-car.dal.us.icall.net, and incoming calls are handled properly. But when I try to call a land-line with pa call 1813968XXXX, I get a response of 480 Temporarily Unavailable. I've also registered my external ip address with icalls servers. Also, this works "originate sofia/internal/9996(a)conference.freeswitch.org 1000", but this doesn't "originate sofia/internal/1813968XXXX@sbc01-car.dal.us.icall.net:5060 1000." Windows XP SP3, 32-bit, 1.5gb FreeSWITCH Version 1.0.4 (14460) Here's the pa call sip trace from wireshark: INVITE sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060 SIP/2.0 Via: SIP/2.0/UDP 173.65.36.19;rport;branch=z9hG4bKg0DHp06gr9Utj Max-Forwards: 70 From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060> Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 INVITE Contact: <sip:gw+icall@173.65.36.19:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 316 Remote-Party-ID: "username" <sip:813995XXXX@sbc01-car.dal.us.icall.net:5060>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1283092013 1283092014 IN IP4 173.65.36.19 s=FreeSWITCH c=IN IP4 173.65.36.19 t=0 0 m=audio 23986 RTP/AVP 10 0 8 3 101 13 a=rtpmap:10 L16/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 SIP/2.0 100 Trying Via: SIP/2.0/UDP 173.65.36.19;rport=5060;branch=z9hG4bKg0DHp06gr9Utj From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060> Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 INVITE User-Agent: CoreSSP SBC Content-Length: 0 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 173.65.36.19;rport=5060;branch=z9hG4bKg0DHp06gr9Utj From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060>;tag=FQK8XBFmHggUg Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 INVITE User-Agent: CoreSSP SBC Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "1813968XXXX" <sip:1813968XXXX@8.19.97.6> ACK sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060 SIP/2.0 Via: SIP/2.0/UDP 173.65.36.19;rport;branch=z9hG4bKg0DHp06gr9Utj Max-Forwards: 70 From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060>;tag=FQK8XBFmHggUg Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 ACK Content-Length: 0 Thanks in advance.

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Configure anf libjpeg
by Kyle Haefner 14 Oct '10

14 Oct '10

Hi all, I've run into this a couple of times now, bootstrap and or configure need to test for libjpeg-devel, as mod_spandsp needs it. Creating mod_spandsp.la /usr/bin/ld: cannot find -ljpeg collect2: ld returned 1 exit status Kyle

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Speex VBR and VAD
by Dan Lane 14 Oct '10

14 Oct '10

Hi, Is there any way to tell Freeswitch to enable VBR and VAD on Speex channels as it appears to be disabled by default and I can't find any mention of it in the wiki or previous mailing list posts. The client is sending the following SDP: a=rtpmap:111 speex/16000/1 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000/1 a=fmtp:110 vbr=on a=rtpmap:0 GSM/22050/1 a=rtpmap:0 GSM/11025/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:113 iLBC/8000/1 a=fmtp:113 mode=30 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv And receiving back the following without vbr=on which disables it in the client: a=rtpmap:111 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Cheers, Dan

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