- FreeSWITCH-users - FreeSWITCH Mailing Lists

Asynchronous PTIME
by Serge S. Yuriev 12 Jan '11

12 Jan '11

Hello, 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 20 to 60 I'm getting this warning and client hears chopped sound :( That is "Our end"? Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch All but MVTS under my control. I doesn't see any clue in logs and can't reproduce this with my testing via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices Which debug/logs I should take? Any ideas? Thanks a lot. btw how I can save debug into log not only console? -- wbr, Serge

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How to disable 5060-5080 with auto-nat at upnp?
by Alex Wang 12 Jan '11

12 Jan '11

Hi All Just don't want to open these port to the public, all my extensions are in my LAN. If I disable upnp, the outside sip register is not working, I mean no sound. How to disable the auto-nat publish 5060-5080 on the internet? Thanks Alex

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FS 502- Bad gateway to Patton SN4960
by Luis F Urrea 12 Jan '11

12 Jan '11

Hi guys, I have a call coming through a Patton GW in the same LAN as FS, call comes in and FS responds with 502 to GW. Here's a console + siptrace log http://pastebin.freeswitch.org/14978 I appreciate your input and advice on where to direct my troubleshooting efforts! TIA

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Try to get rid of Speex codec on invite
by Peter Steinbach 11 Jan '11

11 Jan '11

Hello, when our Freeswitch sends an INVITE to a phone, it only offers speex codec for voice: m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 When the phone then answers with ACK and OK and sends back it's own codecs during OK, then codecs are negociated to PCMA, which is fine. However our Aastra phones do not like Speex in the INVITE message, so they send back a "Bad request". We have set in vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,GSM,speex@8000h@20i,speex@16000h@20i"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU,G729,GSM,speex@8000h@20i,speex@16000h@20i"/> and the internal profile contains <param name="inbound-codec-prefs" value="PCMA,PCMU,speex,GSM,G729"/> <param name="outbound-codec-prefs" value="PCMA,PCMU,speex,GSM,G729"/> <param name="inbound-codec-negotiation" value="greedy"/> <param name="outbound-codec-negotiation" value="greedy"/> So Speex is not the preferred codec defined. Eeven 'scrooge' did not solve the problem. How can I get rid of this behaviour and force Freeswitch to invite with PCMA? -- With kind regards Peter Steinbach Telefaks Services GmbH Theo-Geisel-Strasse 25 D 61250 Usingen, Germany mailto:lists (att) telefaks.de Internet: www.telefaks.de

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(no subject)
by Madovsky 11 Jan '11

11 Jan '11

in mod_cepstral wiki page, to alter volume example is <prosody volume='15'>This is pretty softly spoken.</prosody>but where to put this xml in dialplan ?Thanks

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NDLB-force-rport safe
by Serge Yuriev 11 Jan '11

11 Jan '11

Hi Trying to use safe settings with no luck Profile starting w/o problem http://pastebin.freeswitch.org/14986 FS answers to wrong port http://pastebin.freeswitch.org/14988 Pls advice -- wbr, Serge

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bind_digit_action
by Rafqat . 11 Jan '11

11 Jan '11

Hi, Can someone please help here. I am trying to use bind_digit_action to hangup leg B only, if leg A presses ##, but it doesn't seem to be working. The INFO message is displayed but when I press ##, nothing happens. Any help will be much appreciated. <extension name="call-me-service"> <condition field="destination_number" expression="^9999$"> <action application="answer"/> ......... <action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/> <action application="set" data="bridge_pre_execute_bleg_data=START_LISTENING XML private"/> ....... </condition> </extension> <extension name="setup bind_digit_action_listening"> <condition field="destination_number" expression="^START_LISTENING$"> <action application="log" data="INFO Configuring bind_digit_action to listen..."/> <action application="bind_digit_action" data="start,##,exec:hangup,unknown"/> <action application="digit_action_set_realm" data="start"/> </condition> </extension> Cheers Raf

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Two meny qestions
by Mattias Hemmingsson 11 Jan '11

11 Jan '11

Hi So first realy like the freeswitchn config use to work with trixbox but i like to have everything in config /xml files. Bur i have stumbeld over two problems. And i think that they are realy easy to fix. i have a nice meny that greats the caller comming in. And it works when i choose to be transferd to user 1000 in the meny by pressing one. But when the is not online i want the user to be transferd to the users voicemail. I have voicemail working of i call the user from a nother externsion. But i would like it to wokr from the ivr meny as well. And second i whould like to if you press two you should be transferd to an ring group. I have an working ring group with extension 200. This is my ivr meny <include> <menu name="elino_ivr" greet-long="/opt/freeswitch/conf/elino/sounds/elino_ivr.wav" timeout="10000" inter-digit-timeout="2000" max-failures="1" max-timeouts="2" digit-len="1"> <entry action="menu-exec-app" digits="1" param="bridge sofia/internal/1000%10.0.1.6"/> <entry action="menu-exec-app" digits="2" param="bridge sofia/internal/1001%10.0.1.6"/> <entry action="menu-exec-app" digits="3" param="bridge sofia/internal/1000%10.0.1.6"/> </menu> </include> i have one doman called www.elino.se set upp as well. Regards Mattias

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cepstral problem
by Madovsky 11 Jan '11

11 Jan '11

I have one sentence like this in my dialplan <action application="speak" data="cepstral|david|want pizza"/> so when I start FS from a console directly in foreground the voice works. if I start FS from the init script in background, the voice becomes silence and nothing is running after this in the dialplan. if I want to shutdown manually FS, it makes a segmentation fault with core dump. Any idea ? Thanks

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Asterisk & FreeSwitch in recording view
by Leonid K 11 Jan '11

11 Jan '11

Hi, currently we're looking for voip recording solution - it must record all incoming/outgoing/internal/conference calls within the company. later on we going to develop applications that let us fing/analyze recordings. the main problem as I see at the moment is huge count of calls that is going through switch. what do u think about using Asterisk or/and FreeSwitch for this task? thanks in advance! -- Sincerely, Leonid Kryvoruchko Mobile +38 093 7609175

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