April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

xml_curl response for voicemail_inject
by Frank Park 07 Apr '11

07 Apr '11

I had a quick question. I am trying to figure out what directory response is expected when a user wants to forward a voicemail to another extension. Looking at the post request of xml_curl, I see that it's invoking voicemail_inject. I've tried the same response as other voicemail function, which is similar to the authorization response, but that didn't seem to do it. Anybody care to share an example response when voicemail_inject is requested? Thanks, Frank -- ----=======================---- Frank Park Telonium Communications, LLC frank(a)telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----

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Job Opening at Barracuda Networks / CudaTel - Technical Escalation Engineer
by Anthony Minessale 07 Apr '11

07 Apr '11

Looking for someone well-versed in debugging voice and data using common tools like wireshark etc to take escalation incidents from tech support at CudaTel. Should be well versed in unix systems and internet tools and above average diagnostic skills. Preferred location or relocation to Ann Arbor MI but will entertain applicants from the bay area in CA as well. Please reply with resume to jobs(a)freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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FS does not relay BYE
by Eric Beard 07 Apr '11

07 Apr '11

Hello, I have just started using freeSwitch as a way to terminate calls from Microsoft Speech Server to voip gateways. I have almost everything working with a few exceptions. One of the problems I am having is that the final BYE issued by the terminator does not get relayed back to MSS, so MSS keeps the call open for an additional minute, then issues its own BYE, which freeSwitch can't match up to a call because it tore the call down already. The sequence: - MSS running on my machine originates a call, sends INVITE to freeSwitch running on a separate machine, with an internal and external NIC. - freeSwitch relays the INVITE to the gateway (in this case Affinity, but I get the same behavior with other gateways). - My cell phone rings, I pick it up, then hang up the call. - The gateway issues a BYE to freeSwitch, freeSwitch says OK and tears down the call without passing on the BYE. If I originate a call from my machine with a soft phone, it works fine. The only difference I can see is that the soft phone uses UDP, while MSS only talks SIP over TCP. I have pasted logs for the session at http://pastebin.freeswitch.org/16037. Thanks! ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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How to limit the max number of registration users
by fieldpeak 07 Apr '11

07 Apr '11

Could anyone help advise how to limit the max number of registration users? thanks if any advise. Regards, Charles

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Recording transfer audio
by Eric Beard 07 Apr '11

07 Apr '11

Hello, I asked this question yesterday over irc. My apologies if someone answered there already, my irc client kept crashing so I might have missed it. I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, except when the call gets transferred. When I do a transfer, I see two WAV files, one of which is the original session in which I can hear the call up until the transfer. The other file is all silence, which I assume is the session up to the point where the INVITE is answered. But I get no audio of the conversation after the transfer. My suspicion is that freeSwitch is not hairpinning the audio, so it does not have access to the RTP. I pasted everything from the console at http://pastebin.freeswitch.org/16027. Thanks! ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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Freeswitchsolutions.com?
by Alex Crow 07 Apr '11

07 Apr '11

Hi list, Anthony, We are investigating a possible move from a proprietary phone system to Freeswitch. Advertised on the FS site is the above address which appears to say it provides commercial support for FS. In the last two weeks both me and a colleague filled in the contact form asking about prices, SLAs etc but have had no response. Is the company real or is there a problem with the contact form? Thanks Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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What FreeSwitch version i am using after updating git pull?
by Shamun toha md 07 Apr '11

07 Apr '11

How do we know if i am using the latest Git version (nightly builids)? How to know if i am really using 1.0.7 or 1.0.6 oldest? So far tried as > git pull && make current > make current (showed some error) > make clean modwipe (completed successfully) > ./configure && make & make install > /usr/local/freeswitch/bin/freeswitch > > preesed F12 shows: > FreeSWITCH Version 1.0.head (git-828960a 2010-09-25 12-51-42 -0500) Let me know plz, is this the 1.0.7 ?

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Cannot compile freetdm
by Valery K 07 Apr '11

07 Apr '11

Enter: # cd /usr/local/freeswitch/libs/freetdm # ./configure --with-libisdn # make ... compile ... cc1: warnings being treated as errors src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34': src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen' src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span': src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of 'Q931SetLogCB' from incompatible pointer type make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1 Why? libisdn-0.0.1 installed -- View this message in context: http://old.nabble.com/Cannot-compile-freetdm-tp31340769p31340769.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Hylafax server emulation
by Steve Underwood 07 Apr '11

07 Apr '11

Hi, It has always been clear that a HylaFAX compatible FAX job submission server would add considerably to the value of the FAX facilities in Asterisk and Freeswitch, but somehow it hasn't happened until now. I recently found that in 2005 someone produced something fairly basic for Asterisk in Perl, but it doesn't seem to have been well publicised, and it looks like development stalled long ago. I now have the skeleton of HylaFAX compatible FAX job submission server, in C, working. It accepts FAX submissions from sendfax and a couple of the windows HylaFAX clients, though it needs a lot more polishing. Now I need to look at the best thing to do on the Freeswitch side. I aim to make the server maintain its own database of FAX jobs. It will attach to Freeswitch, by ESL; push the jobs through FS; deal with scheduling, retries, etc; and report the final result to the user, just as HylaFAX does. The thing I am rather unsure about is the best way to handle the accounts used to accept FAX jobs? Should I maintain a separate database of FAX accounts, or hook into an existing database? I would welcome suggestions for what would be the most useful approach. Steve

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softphone to outbound help
by Dave Bracken 06 Apr '11

06 Apr '11

Can anyone tell me what i need to do to be able to dial a 7 digit number on my softphone and have it dial out to a phone i have sitting on my desk? I guess what i really need to know is what all files do i need to set up and where, etc. I have never done anything like this, so i need your help. Thanks in advance, Dave

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