July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to detect that a party is in a conversation?
by Eli Hayun 21 Jul '09

21 Jul '09

I want to know if the party that I am calling to is in a middle of a conversation. I did not get a busy line because he had more then one line defined. I tried to set max_calls=1 with no luck. any suggestions? Thanx Eli

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Re: [Freeswitch-users] call fails when using leg_timeout
by TTNC - Adnan Barakat 21 Jul '09

21 Jul '09

Hi, I seem to have come across a strange problem; Basically I'm trying to dial 3 destinations one after another, until the destination dialled is answered, and I only want the destination to ring for 20 seconds. If I do this from the console what I'm trying to achieve works fine; originate {leg_timeout=20,ignore_early_media=true}sofia/internal/123@1.2.3.4|sofia/internal/456@1.2.3.4|sofia/internal/789@1.2.3.4 &park() But if I do it from dialplan it doesn't; Doing this below dials the first destination for 20 seconds, then the second destination for 5 seconds or so, then the call terminates with ORIGINATOR_CANCEL <action application="bridge" data="{leg_timeout=20,ignore_early_media=true}sofia/internal/123@1.2.3.4|sofia/internal/456@1.2.3.4|sofia/internal/789@1.2.3.4" /> Any ideas what might be causing this and any solutions would be appreciated. Many thanks Adnan

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mod_fax problems
by Evgeniy Zolotov 21 Jul '09

21 Jul '09

Hello! I have some problems with receiving fax messages. My FreeSWITCH cann't decode some faxes. As I understood this problem has occurred with Panasonic and Panasonic-like fax-machines. I have recorded wav-file, which corresponds to the receiving fax (see fax.wav.tgz) and log-file of this session (fax.log.tgz). I tried decode this fax with the help of transferring them from one FreeSWITCH to another, but with no luck (I used for this SIP and openzap - result the same). Transferring from one FreeSWITCH to another with the help of 'txfax' works perfect. OS: Centos 5.2 FreeSWITCH: 1.0.2 (11053) Configuration file for mod_fax - original. So, please, help me to understand this situation. Thanks, Evgeniy. mailto:zolotov@altron.ua

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FreeSWITCH-Air, Another GUI?
by Seven Du 20 Jul '09

20 Jul '09

ALL, I know you guys more prefer a CLI version of softphone to a GUI version. But I still would like to share this: http://wiki.freeswitch.org/wiki/FsAir And feel free to give me feedbacks. I'v only played a few days of ActionScript, it's highly appreciated if someone can give me help on the following problems. 1) Bounce the icon on Mac on incoming call. 2) Show a small window on incoming call. 3) Is it possible to block read/write a socket? I want to implement sendRecv() in ActionScript like in the C version of ESL. Thanks.

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Re: [Freeswitch-users] mod_fax problems
by Evgeniy Zolotov 20 Jul '09

20 Jul '09

Hello, Brian. Ok, I understand. So I'll try to update to SVN trunk and then post log, if any problems still exists. Вы писали 20 июля 2009 г., 18:09:37: > If you have made changes to the included modules you'll need to report > your changes to jira http://jira.freeswitch.org > Thanks, > /b > On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: >> Hello, Brian. >> >> I understand that this is the first thing that I should make. >> But we have many our own changes in modules and after update of all >> project we'll need to transfer them too. But this is the great peace >> of job. >> So probably update of mod_fax and spandsp would be enough? > _______________________________________________ > 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 Thanks, Evgeniy. mailto:zolotov@altron.ua

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ClueCon 2009 - Last Chance To Book At The Wyndham
by Michael Collins 20 Jul '09

20 Jul '09

Greetings! We would like to let everyone know that the hotel where ClueCon 2009 is being held - The Wyndham Chicago - is completely booked up outside of the block of rooms that we have reserved. There are still rooms available. However, after noon CST tomorrow, July 21, those rooms will no longer be reserved for ClueCon attendees so you have until then to get your reservations in. After noon tomorrow we will no longer be able to guarantee that you have an chance to register with the hotel, so please act fast! See you in Chicago. Thanks for supporting ClueCon! -Michael http://www.cluecon.com 877.742.CLUE

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Creating a new User Agent
by velusamy velu 20 Jul '09

20 Jul '09

Dear All, I want to create a new User Agent like sip configurations in Asterisk. I checked default user agents 1000 to 1001. But I have bit confused the relationship between default user agents and sip_profiles. I need some help from you all for the following questions, How to create new user agent ? How to relate the new user agent with sip internal profile ?

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mod_conference: behavior when the conference is torn down before 31seconds
by Luis F Urrea 20 Jul '09

20 Jul '09

Hi all, I am experiencing a behavior that I cannot clearly understand. Basically I "autocall" a few phones into a conference with the sip_auto_answer set to true, as follows: <extension name="extension-intercom"> <condition field="destination_number" expression="^773$"> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="conference_set_auto_outcall" data="user/305"/> <action application="conference_set_auto_outcall" data="user/303"/> <action application="conference_set_auto_outcall" data="user/201"/> <action application="conference" data="412+flags{endconf|deaf}"/> <action application="conference" data="412 kick all"/> </condition> </extension> The conference establishes just fine and everyone can hear just fine. The "strange" behavior comes when the person calling to ext 773 hangs up before 31 seconds have passed, the rest of the phones stay up until they reach second 31 into the "conference". I am using snom phones and I see the BYE message arriving at the phones exactly at second 31 after the call establishes. The conference itself however does not exist after the person calling 773 hangs up (doing conference list on CLI shows NO active conferences). If the conference stays up more than 31 seconds, then when the person calling 773 hangs up, the rest of the phones hang up immediately as desired. Here's the log for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9773 Here's the log of the phone for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9774 Your inout is appreciated. Regards, Luis

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Re: [Freeswitch-users] caller_id 0000000000
by TTNC - Adnan Barakat 20 Jul '09

20 Jul '09

Hi We have a FreeSWITCH server receiving calls from a provider, we process the call (play greeting messages etc...) then pass the call out again to an end destination via the same provider. But if the caller is witholding their cli our provider send the call to us with the sip_from_user variable to 'nobody'. Is there anyway to remove the caller_id when we don't receive it rather than override it with 0000000000? or send the call in the same way we receive it ie. from 'nobody'? Thanks Adnan

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NAT'd FS / publice softphone problems
by Mark Campbell-Smith 19 Jul '09

19 Jul '09

Hi All, I know this question has come up before but I couldn't find the answer that I could understand! Sorry in advance. My setup is: Freeswtch NAT'd (192.168.x.x) -> Router -> Internet -> Softphone with public IP I can easily get the softphones to register, but when I try to call from the softphone to voicemail (for example), I don't get any audio. I checked out this page: http://wiki.freeswitch.org/wiki/External_profile (section Switch with External SoftPhone) but I am not clear how I can get this to work. I have played around with the rtp-ip and external-rtp-ip but without success. Is it possible for someone to help me configure this so softphones that are outside the nat'd lan get audio correctly? Help appreciated! Thanks

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