August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Snom and valet_parking
by soapee01.fs＠stubbornroses.com 19 Aug '13

19 Aug '13

Hi list: I have an issue with a new Snom 821. I've configured valet_parking for auto park mode (eg: my_lot auto in). After the attended transfer, I always receive a Temporarily Unavailable response on the Snom phone from FS. This does not hold true on my yealink/cisco/linksys/etc phones. I can do proper attended transfers from extension to extension on the Snom. I've tried this on a separate machine with latest stable, and latest git head. Dialplan: <extension name="valet_auto_park_in" > <condition field="destination_number" expression="^(7900)$" > <action application="valet_park" data="my_lot@${domain_name} auto in 7901 7999" /> </condition> </extension> <extension name="valet_auto_park_out" > <condition field="destination_number" expression="^(79\d\d)$" > <action application="answer" /> <action application="valet_park" data="my_lot@${domain_name} $1" /> </condition> </extension> screenshot from wireshark: http://tinypic.com/r/jtp5hc/5 Relevant Log: http://pastebin.freeswitch.org/21322 Snom Firmware Version: snom821-SIP 8.7.3.19 The call flow analysis on wireshark doesn't include any of the sip messages that follow the 480, so it doesn't really provide much there. Any pointers would be greatly appreciated. Duplication (for my own personal sanity) would also be appreciated. Could be I just got a bad phone... Regards, James PS: I know the timestamps on the wireshark don't match the log. They are from different tests, but the results are always the same.

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Congratulations and Wanted: Single Consultant
by Alex Lake 19 Aug '13

19 Aug '13

A little bird tells me that Avi Marcus is getting married (or "Bridged"?) which is lovely, but that means that there's a little project that we're looking for a Freeswitch guru to do for us.... It's essentially that we want a DTMF testing rig setup (on a "cloud-based" server of ours) that we can use as a model to determine why we're getting poor dtmf recognition from some phones/networks when they come to our current setup, but that Avi's test switch picked up just fine. So a consultant who's either young, free and single or one that's old, married and bored ;-) Email me if you're interested and available nowish (we'd like to get this done this week if possible) Alex

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Mod_shout using with icecast Big delay
by Jorge Núñez 19 Aug '13

19 Aug '13

Hi I am using mod_shout to send a conference call to icecast and hear it from a tag audio in html5 in realtime, but it has a big delay like 12 seconds. How can I reduce the latency of the audio sent from freeswitch or what can I do to improve this. Regards Jorge

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does fs have problems with a . in the password field?
by covici＠ccs.covici.com 19 Aug '13

19 Aug '13

Hi. I was having problems with one of my carriers -- they changed the password for my account without notice! The password they changed to had a . and they kept saying I had the wrong password or anyway an authentication error, till at their suggestion, I removed the on both sides, and bingo I was able to register. I am using FreeSWITCH Version 1.5.3b+git~20130709T135336Z~267ef728e1 (git 267ef72 2013-07-09 13:53:36Z) . Should a Jira be filed? -- 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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webrtc INCOMPATIBLE_DESTINATION
by Javier Menendez 19 Aug '13

19 Aug '13

Hi, I am trying to make an outbound call to a webrtc softphone using jssip, I initiate the call from an asterisk box : [Asterisk] -> [FS] ->[jssip] I always get an INCOMPATIBLE_DESTINATION error, looking at the trace logs I found out that the problem is the codec negotiation but I can not make it work, AFAIK the call should use the ALAW codec as it is compatible with all legs involved. this is the asterisk SDP 2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP: v=0 o=root 451068671 451068671 IN IP4 192.168.90.16 s=Asterisk PBX 1.8.16.0 c=IN IP4 192.168.90.16 t=0 0 m=audio 16126 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 And this is the SDP from FS wich is sent to the jssip client 2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP: v=0 o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP s=FreeSWITCH c=IN IP4 PUBLIC_IP t=0 0 a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps m=audio 21772 RTP/SAVPF 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=rtcp-mux a=rtcp:21772 IN IP4 PUBLIC_IP a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0 a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0 a=ice-ufrag:4BgP20qzvjuPqK7E a=ice-pwd:ADcc0S2Lqc2T5cB0 a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host generation 0 a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host generation 0 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7 a=ptime:20 a=sendrecv Shouldn't it include the PCMA codec? I hace tried to enable it in configuration but it doesn't work, also tried to set it with <action application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/> before bridge, but no luck. Funny thing is that it is working if i innitiate the call from the jssip client. [jssip] -> [FS] ->[Asterisk] Any clue?

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Ademco signaling with mod_spandsp
by Brian Foster 19 Aug '13

19 Aug '13

I've been tasked with securing a storage facility with an alarm system. Basically we'd use POTS or GSM on the site and a POTS line on the monitoring station. The alarm system is capable of using the ademco standard. I see there's some code in git in regards to the ademco standard. Is this usable inside Freeswitch? If so, how can I access it? I'd like to pay the wiki tax to doxument it if anyone can get me started. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster(a)davri.com Indianapolis, Indiana Sent from a mobile device.

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setting transfer context
by Babak Yakhchali 19 Aug '13

19 Aug '13

hi when a call is attended transferred, for example, to a number like 222, first bleg is hold and caller initiates a new channel with destination 222 which hits default context. if this new call is transferred to another context like public using transfer application with a different destination number like 333 after attended transfer completes the bleg is dropped. because it goes in routing state with destination number 333 in default dialplan. to solve this problem I'm using an extension in all contexts after unloop extension: <extension name="transfer_adjustment"> <condition field="${transfer_adjusted}" expression="^$"/> <condition field="${sip_refer_to}" expression="^<sip:(.+)@.*$"> <action application="log" data="DEBUG original tranfer was:$1"/> <action application="set" data="transfer_adjusted=true"/> <action application="transfer" data="$1 ${dialplan} ${context}"/> </condition> </extension> I check if transfer_adjusted is not set and the channel is transferred,first and second conditions, then I transfer it to the original number which caller called to transfer the call to correct destination. Is this the right way? or I'm missing something thanks

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Softphone not getting connected
by Ashish Mishra 18 Aug '13

18 Aug '13

I installed freeswitch on my ubuntu 12.04 machine...but when i am trying to connect the softphone installed on my windows 8 pc with ubuntu machine the softphone gives me an error that account not enabled...i have used a network cable to connect the two machines...also the firewall in both the machines is disabled... Kindly help.. Ashish Mishra

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Issue when receiving INVITE with both audio and image (T38)
by François 18 Aug '13

18 Aug '13

Hello list, I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending both audio+image (T38) in the initial INVITE. In a simple bridge scenario, FS will just ignore the T38 part and setup an audio connection only with b-leg. My issue is later on if/when we receive a re-INVITE (ex: SIP timers refresh) with the same SDP (audio+T38), FS now considers it as FAX and sends a re-INVITE to b-leg with T38 only! In case the destination is voice only, the call hangs up. --- Asterisk apparently had the same issue, they solved it by setting up both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL they just re-INVITE b-leg with T38. See: https://reviewboard.asterisk.org/r/208/ Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843 Is this the right way to go? (Also is it feasible in FreeSwitch)? I can add a JIRA if necessary. Thanks, François.

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Wait for ANSWER/HANGUP events in Mod lua
by Sami Montour 18 Aug '13

18 Aug '13

I am using Mod lua for call processing on the FreeSwitch. I would like to wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP in order to get the answer time and hangup time. Basically, something like Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP If (event_name = "CHANNEL_ANSWER") then answer_timestamp = getHeader("Caller-Channel-Answered-Time") If (event_name = "CHANNEL_HANGUP") then hangup_timestamp = getHeader("Event-Date-Timestamp") The session:getState() described in module lua gets the call state but not the channel state. For instance, 'CS_EXECUTE' is the channel state when it is executing a dialplan not when the call is answered. Is there a way to wait for channel events such as CHANNEL_ANSWER and CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call? Any input is very much appreciated. Thanks. Sam

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