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Display update after consult transfer to bridge/loopback app
by Beeton, Carolyn (Carolyn) 25 Jan '11

25 Jan '11

I have an application defined in the dialplan, accessed through a "bridge" to a loopback socket application so that consult transfer to it works correctly. If I dial it directly on a Polycom set, the set displays "Outbound Call/<port>async full". I can overcome this by exporting the callee_id_name/number variables in the dialplan. However, if I transfer a set to this application via consult transfer, the set ends up with a "Outbound Call/app=socket:..." display and I can't figure out how to get it to show the name/number of the application. Any ideas? I'm running on today's git latest (6294bc1620c). Thanks, Carolyn

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Voiceglue support?
by Lukasz Karlowski 24 Jan '11

24 Jan '11

Dear Freeswitch Team, I am writting to you to ask if you are planning to add support for Voiceglue? It is immportant to us(me and my academic group) to get know about it. So we will be very greatefull for your anwser. Your sincerelly, Łukasz Karłowski with his acedmic group.

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email and reverse DNS
by Seven Du 24 Jan '11

24 Jan '11

Hi US-Team, We couldn't send email to some services like AOL. They need reverse DNS to work. So question: 1) How do we send email in the original eleutian system? If I'm right it directly send via google. So is there any limit on sending via google? how to set that? 2) We had reverse DNS on softlayer but as we are going to kill those servers, if 1) is not available we may need to ask reverse DNS from TCT. So, should we setup a centra mail server? 3) There current setup is to go directly from each server. Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Re: [Freeswitch-dev] [Freeswitch-users] Bounty Offered for A2Billing equivalent for FreeSWITCH
by EdPimentl 23 Jan '11

23 Jan '11

ASTPP already support FreeSwitch. Review the features missing you want to implement. If you decide ASTPP is a viable solution, I would contribute to the bounty. http://www.astpp.org/?q=node/165 -E

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Possible mod_smpp
by Michael Collins 19 Jan '11

19 Jan '11

Hello all, If you are interested in giving monetary support to have a professional software firm create mod_smpp then please contact me off list and I will give you more details. Thanks, Michael

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FreeSWITCH Conference Call Today
by Michael Collins 19 Jan '11

19 Jan '11

Hello all! The agenda for today's conference call is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_19 We have Mitch Capper scheduled to speak to us on his experiences with embedding FreeSWITCH in other applications. In his case he created a FS-based softphone for Windows. We look forward to hearing what he's got! -MC

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Before I submit a bug report, could someone take a quick look at this voicemail.conf?
by Mark Sobkow 18 Jan '11

18 Jan '11

Our last bug report on Erlang-Freeswitch integration turned out to be an XML error that wasn't reported by Freeswitch rather than a bug in the integration of the two. I was hoping someone could take a look at our response XML and confirm that we haven't got an XML error before I submit a bug report to JIRA. Right now it _appears_ that FS isn't processing the voicemail to email configuration response from Erlang, but as I said, we had a similar issue a few weeks ago that turned out to be an XML error. Thanks. <document type="freeswitch/xml"> <section name="configuration"> <configuration name="voicemail.conf" description="Voicemail Configuration"> <settings> </settings> <profiles> <profile name="default"> <param name="auto-playback-recordings" value="false" /> <param name="callback-context" value="" /> <param name="callback-dialplan" value="XML" /> <param name="change-pass-key" value="6" /> <param name="choose-greeting-key" value="2" /> <param name="config-menu-key" value="5" /> <param name="delete-file-key" value="7" /> <param name="digit-timeout" value="10000" /> <param name="email-key" value="4" /> <param name="ff-key" value="6" /> <param name="file-extension" value="wav" /> <param name="listen-file-key" value="1" /> <param name="login-keys" value="0" /> <param name="main-menu-key" value="0" /> <param name="max-login-attempts" value="3" /> <param name="max-record-len" value="300" /> <param name="max-retries" value="3" /> <param name="min-record-len" value="3" /> <param name="odvc-dsn" value="dsn:user:pass" /> <param name="operator-extension" value="operator XML default" /> <param name="pause-key" value="0" /> <param name="play-new-messages-key" value="1" /> <param name="play-saved-messages-key" value="2" /> <param name="record-coment" value="" /> <param name="record-copyright" value="" /> <param name="record-file-key" value="3" /> <param name="record-greeting-key" value="1" /> <param name="record-name-key" value="3" /> <param name="record-sample-rate" value="16000" /> <param name="record-silence-hits" value="2" /> <param name="record-silence-threshold" value="200" /> <param name="record-title" value="" /> <param name="restart-key" value="1" /> <param name="rew-key" value="4" /> <param name="save-file-key" value="2" /> <param name="skip-greet-key" value="#" /> <param name="storage-dir" value="/var/voicemail" /> <param name="terminator-key" value="#" /> <param name="tone-spec" value="%(1000, 0, 640)" /> <param name="undelete-file-key" value="8" /> <param name="vmail-key" value="*" /> <param name="vmain-extension" value="900" /> <param name="web-template-file" value="web-vm.tpl" /> <email> <param name="date-fmt" value="%A, %B %d %Y %I %M %p" /> <param name="email-from" value="${voicemail_account}@${voicemail_domain}" /> <param name="notify-template-file" value="notify-voicemail.tpl" /> <param name="template-file" value="voicemail.tpl" /> </email> </profile> </profiles> </configuration> </section> </document> <document type="freeswitch/xml"> <section name="directory"> <domain name="10.77.0.231"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <user id="5000"> <params> <param name="password" value="****"/> <parameter name="email_email-from" value="voicemail(a)marketel.com"/> <parameter name="vm-attach-file" value="true"/> <parameter name="vm-disk-quota" value="0"/> <parameter name="vm-email-all-messages" value="true"/> <parameter name="vm-keep-local-after-email" value="true"/> <parameter name="vm-mailto" value="user5000(a)marketel.com"/> <parameter name="vm-message-ext" value="wav"/> <parameter name="vm-notify-email-all-messages" value="true"/> <parameter name="vm-notify-mailto" value="user5000(a)marketel.com"/> <parameter name="vm-password" value="****"/> </params> <variables> <variable name="accountcode" value="5000"/> <variable name="callgroup" value="employee"/> <variable name="effective_caller_id_name" value="Extension 5000"/> <variable name="effective_caller_id_number" value="5000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="toll_allow" value="domestic,international,local"/> <variable name="user_context" value="default"/> </variables> </user> </domain> </section> </document> -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/

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relative_url_root in rails3
by Seven Du 18 Jan '11

18 Jan '11

our whole arch depending on relative url to work. It turns unefficient in rails2, now they changed http://stackoverflow.com/questions/3181746/what-is-the-replacement-for-acti… Please use scopy .. do in rails 3 ... we lost some flexibility between dev and production , but we could write a deploy script to substitute the routes named scope in the future. Anyway, hard code it for now. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Processing Live Audio during a call between 2 extensions
by Michel Habib 16 Jan '11

16 Jan '11

Dear Freeswitch Developers, i am relatively new to freeswitch and i am seeking some advice. I am creating an External Application that does some [live] processing on the Audio of the freeswitch call before sending it to the second Call Leg, and vice versa. Can you please direct me to the best way to do that and how to access/capture the live audio channel in both legs while it is streaming, and how to resubmit it again to the other leg? I read the book about freeswitch, but still couldnt figure the best approach to do that, any hint is appreciated. Best Regards, Michel Habib.

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FreeSWITCH 1.0.7
by Brian West 14 Jan '11

14 Jan '11

http://latest.freeswitch.org/ Enjoy! /b

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