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mod_distributor and dial strings
by David Villasmil 12 Feb '11

12 Feb '11

hello all I'm playing with mod_distributor... it's nice.. but i'm having this problem: suppose i have 2 gateways i want to use in round-trip fashion.. that works fine... but suppose i must send to gw1 with a prefix and to gw2 with a different prefix... is this possible? thanks David

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hold_music var gets unset/lost by cancelled/failed att_xfer
by Fraser Redmond 11 Feb '11

11 Feb '11

I'm trying to do a second att_xfer on a call so that if the first attended transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that doesn't exist) then the call could be transferred to someone else. On the first att_xfer the person on hold hears the hold_music correctly. Once that transfer is cancelled or fails: -- On any subsequent att_xfer's the person on hold just hears silence. -- If they are put on hold they just hear silence. I tried setting hold_music again for each channel after the first att_xfer, but that didn't work, so it's probably not actually a problem with hold_music per se, but some other variable/setting that decides whether to use hold_music. I also tried doing a uuid_dump before and after each attempt, but didn't notice anything too different - unless it's a matter of unsetting one of the couple of changed/new vars like: variable_originate_disposition variable_current_application variable_playback_seconds I get the feeling other variables are probably also lost by the first failed transfer as the second att_xfer has some odd things happen if the third party does answer. Haven't been able to narrow it down as closely as the hold_music, but two things I've seen happen are: -- The party that initiated the transfer gets hung up automatically (after 30 sec) -- When the party that initiated the transfer hangs up it should connect the other two parties, but instead it hung up all three Cheers, Fraser

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Re: [Freeswitch-users] 503 Maximum Calls In Progress
by Gary Chen 11 Feb '11

11 Feb '11

After restart, everything back to normal. I can make call again. I will do more tests. I will let you know if it happen again. Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, February 11, 2011 12:54 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress If you stop freeswitch and restart it does it still say that or are you able to make some calls for a while before it throws that error? -MC On Fri, Feb 11, 2011 at 11:26 AM, Gary Chen <gchen00(a)insightbb.com> wrote: Yes. I did some test calls between two Cisco sip phones and they all work fine and then it just stop working. At that point, FS is still up and I can see the system information by press F keys. It just won’t accept any calls. I looked at log file and it just has that one line: 2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time. The register message is still working at that time. Gary

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Using an Empty FROM field for SIP Messages
by Matthew Fong 11 Feb '11

11 Feb '11

I am trying to setup a customer who wanted to utilize the airespring national did presence, where in they rewrite the Caller ID of outbound calls. In order to do this tho, they say I need to send a blank caller id. FreeSWITCH will send 0000000000 if the caller id is not specified. Airespring wants something formatted like INVITE sip:jungar@alpha-org.com SIP/2.0 Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr Max-Forwards: 70 To: 2556112121 <sip:jungar@alpha-org.com> From: <sip:jsmith@beta-org.com>;tag=13456 Call-ID: @zetamachine.beta-org.com with the From: field looking like any of the following From: <sip:jsmith@beta-org.com>;tag=13456 From: < @208.76.54.59>;tag=13456 From: @208.76.54.59;tag=13456 The closest I got was to make FreeSWITCH to the following, but airespring won't budge. Can anyone tell me how to make the From string as airespring wants for their national DID presence service. Thanks. From: "" <sip:0000000000@208.76.54.59>;tag=UyNBXjyUUay8m

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how to set up a DTMF callback function
by Frankie Yiu 11 Feb '11

11 Feb '11

If I want to get a DTMF callBack to a function when a caller presses a key, what function should I use to set it up? Thanks, Frankie ____________________________________________________________________________________ Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/

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Re: [Freeswitch-users] 503 Maximum Calls In Progress
by Gary Chen 11 Feb '11

11 Feb '11

Yes. I did some test calls between two Cisco sip phones and they all work fine and then it just stop working. At that point, FS is still up and I can see the system information by press F keys. It just won’t accept any calls. I looked at log file and it just has that one line: 2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time. The register message is still working at that time. Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Friday, February 11, 2011 12:13 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress Uhhh... which is the opposite of commenting it out? Gary, if you don't have that param in your file then it's not this. You say FS had been up an hour. Had it handled any successful calls in that time? Was FS shutting down at the time? -Steve 2011/2/11 David Ponzone <david.ponzone(a)ipeva.fr> So add it. David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 11/02/2011 à 17:01, Gary Chen a écrit : My switch.conf.xml does not have this param. Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Friday, February 11, 2011 10:39 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress Comment from anthm; edit conf/autoload_configs/switch.conf.xml comment out <param name="min-idle-cpu" value="25"/> It seems to not work well on 32 bit. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] 503 Maximum Calls In Progress
by Gary Chen 11 Feb '11

11 Feb '11

My switch.conf.xml does not have this param. Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Friday, February 11, 2011 10:39 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress Comment from anthm; edit conf/autoload_configs/switch.conf.xml comment out <param name="min-idle-cpu" value="25"/> It seems to not work well on 32 bit.

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Vestec Speech Engine: Large Vocabulary
by Kashif Kahn 11 Feb '11

11 Feb '11

Hello Everyone, Vestec recently launched a large vocabulary speech engine for "command-and-control" type IVR applications. The new engine - called Tier-2 - supports a vocabulary size of 2,500 keywords per recognition and complements our existing Tier-1 engine that has a 500 keywords vocabulary size. Tier-2 retails for $199 per port (ie. channel) while Tier-1 is priced at $99 per port. Acoustic models needs to be licensed separately at $9.99 per language. Our ASR is designed for keywords-based interaction and offers the best deal around for speech enabling a wide variety of IVR applications, including DTMF menus and multi-step business processes. The speech engine is standards based in terms of grammar writing format and platform integration protocols. We support SRGS grammar (ABNF & XML) as well as MRCP integration (v1 & v2). In addition, we offer a highly scalable architecture that is capable of supporting thousands of ports in an efficient manner. A starter kit - comprising a specially priced perpetual license for a full-function, full-feature standard ASR engine - is available for $25. Please visit Vestec webstore at: http://www.vestec.com/products Regards, -Kashif -- Kashif Kahn VP Business Development Vestec Inc Waterloo, ON Canada phone: +1 519 885-7615

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no voice for call in same lan network
by Sam 11 Feb '11

11 Feb '11

Here , i have bunch extension in the same lan network and when they talk with each other there is no voice , i have another bunch in other lan network the too do not get voice when talking with each other. Here the public ip for 2 bunches are different and when an extension from bunch A calls to extension of bunch B then they have voice. FS is on public IP :( >From the trace i could see that when making an internal call in the same LAN the RTP flows to private IP and when making and external call from bunch A to bunch B the RTP is send to public IP where the voice is heard, also while making and outbound call the RTP is send to public IP of the LAN and there is voice. Is this a known scenario to anybody ? Regards Sam

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Re: [Freeswitch-users] 503 Maximum Calls In Progress
by Gary Chen 11 Feb '11

11 Feb '11

My company blocked Git access. Can you tell me which config file need to be changed to fix the problem? Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Friday, February 11, 2011 10:12 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress Update to latest git. I think Tony did an update for this, it was in one of the config files. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Gary Chen Skickat: den 11 februari 2011 15:52 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] 503 Maximum Calls In Progress FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z) I have a freeswitch installed for testing. It just up for about an hour and there are two Cisco 7960 phones registered. After made several test calls between these two Cisco phones, FS stop working and give me these message on fs_cli: 2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time. In the SIP header it shows: 503 Maximum Calls In Progress What should I do to fix this? Here is the message by press F2, F3 and F4: F2: freeswitch@internal> status UP 0 years, 0 days, 1 hour, 12 minutes, 55 seconds, 590 milliseconds, 20 microseconds 39 session(s) since startup 0 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 F3: freeswitch@internal> show channels 0 total. F4: freeswitch@internal> show calls 0 total. Gary !DSPAM:4d554ec432767061315535!

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