December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Compact Sip Headers
by Ira Tessler 20 Dec '11

20 Dec '11

Is Freeswitch compatible with compact sip headers? If so, is there a parameter I have to set on the external profile to get them to work?

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error compiling Freeswitchin mod_spandsp
by Cavalera Claudio Luigi 20 Dec '11

20 Dec '11

Hello, I'm facing a compilation issue regarding mod_spandsp which could be similar to this one: http://jira.freeswitch.org/browse/FS-3473?page=com.atlassian.jira.plugin.sy… or this one: http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-August/005159.html Here is the error on the pastebin: http://pastebin.freeswitch.org/18028 uname -ar output is Linux DBS_A6_A 2.6.9-89.ELsmp #1 SMP Mon Apr 20 10:33:05 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux and I have ./configure --enable-64 has someone any hints to narrow down the problem? Thanks, Claudio From: Michael Collins Post your question to the -users list and give us the details. -MC On Mon, Dec 19, 2011 at 1:47 AM, Cavalera Claudio Luigi <Claudio.Cavalera(a)italtel.it> wrote: Hello guys, I've read about your trouble on the mailing list http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-August/005159. html Unfortunately it does not come to a happy end :-\ Did you find a solution for this problem I'm also facing? :-) Kind Regards, Claudio PS: Please forgive the automatic disclaimer in the email. Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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FS-2828
by ib-freeswitch＠bzsolutions.it 19 Dec '11

19 Dec '11

Hi, I'm studing FS-2828, why freeswitch statemachine always rewrite a 180 with SDP in a 183 ? I already ported (it's a test and it must be cleaned from my debug logs) FS-2828 patch to current GIT but my patch didn't forward new SDP when 180 is received after first 183 http://pastebin.com/dqiBcYC8 Now I'm working also to forward same 18X to ALEG, i don't want to touch FS state machine i'm working exporting channels variables from BLEG to ALEG, i create a function: void sofia_glue_pass_18X(private_object_t *tech_pvt, int forward_18X, int forward_180_sdp); if (sofia_use_soa(tech_pvt)) { nua_respond(tech_pvt->nh, switch_true(switch_channel_get_variable(channel, "forward_180_sdp")) ? 180 : 183, switch_true(switch_channel_get_variable(channel, "forward_180_sdp")) ? "RINGING" : "PROGRESS", //SIP_183_PROGRESS Is it the right way? Thanks Igor-

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announce conf count (total callers minus 1)?
by Rodney 19 Dec '11

19 Dec '11

this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? conf/dialplan/default/01_Announce_Conf_Count.xml: <include>  <extension name="Announce Conf Count (Inline)"> <condition field="destination_number" expression="^ANNOUNCE_CONF_COUNT_PRIVATE$"> <action application="set" data="conf_count=${conference ${conference_name} list count}"/> <action application="log" data="INFO ${conf_count}"/> <action application="say" data="en number pronounced ${conf_count}"/> </condition> </extension> </include>

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PCI Compliance Over Telephone for Credit Cards- how?
by Avi Marcus 19 Dec '11

19 Dec '11

I'm planning on an IVR to accept credit card information for signing up and renewal of my services. Regarding fraud, I'm going to require at minimum a recording of name, who they are, or something or an actual live call. But for PCI compliance.. this says https://www.pcisecuritystandards.org/documents/protecting_telephone-based_p… on page 9: Call centers will need to ensure that transmission of cardholder data > across public networks is encrypted. > This is part of PCI DSS Requirement 4 and includes: > > - ... > > > - *Voice or data streams over Voice over IP (VoIP) telephone > systems, whenever sent over an open or public network. Note that only > those consumer or enterprise VoIP systems that provide strong > cryptography should be used. * > > > - Requiring agents to use analog telephone lines when a VoIP > telephone system does not provide strong cryptography. > > I'm doing dtmf, not voice, but I can't imagine that's LESS strict. I haven't really heard of any end-to-end encrypted origination lines. Is this guideline ignored? How do people deal with this? Does someone have T1 lines and offers encryption for origination...? -Avi Marcus

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Freeswitch ULIMIT on 64 bit
by Bob 19 Dec '11

19 Dec '11

Hi, Apologies if I'm double posting this, but I have not seen my original post to list appear in the archives, so I'm assuming it got lost in cyberspace ! Hi, As per the performance docs, I am trying to set the recommended ULIMIT of 8192 stacksize on a 64 bit machine. I am using the current version of Freeswitch (as pulled off GIT yesterday, Friday). I am running Scientific Linux 6.1 : Linux 2.6.32-131.21.1.el6.x86_64 #1 SMP Tue Nov 22 14:15:09 CST 2011 x86_64 x86_64 x86_64 GNU/Linux My present ulimit settings are as follows : $ ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited However when I start freeswitch "bin/freeswitch -nc -nonat -nonatmap", I get the following : Error: stacksize 8192 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... Surely this is incorrect, because I am running a 64 bit kernel, 240 is the recommended value for 32 bit ?

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Re: [Freeswitch-users] destination_number cleanup
by Elliott Vogel 19 Dec '11

19 Dec '11

Some custom softphone a call center we use has; they auto dial with it and I was able to make brea, x-lite and polycom phone (using uri dialing) send some messed up strings too. I was trying to see if breaks the sip specification so I could push back but I couldn't find anything what a uri couldn't use. It looks to me it can be almost anything - personally they are being lazy and not scrubbing the list first Thanks for the regex string and I will look into writing a script however I would prefer not too... From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 1:08 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] destination_number cleanup Yes, this is really broken input! I thought it was bad some do +1 or 001 1 or no prefix to dial America! To modify Ryan's code to make it more flexible... ^(?:\+1|001|1)?\D*(\d{3})\D*(\d{3})\D*(\d{4})$ $1, $2 and $3 should give you the full number, so bridge to 1$1$2$3. it will match anything that has xxx-xxx-xxxx with whatever separating non-digits is in between. But it can only handle (by discarding) a prefix of +1, 1, or 001. I'd suggest using xml_curl or a lua script and doing a regex replace of any non-digit characters, if you really need. But how in the world is this your input? What devices are sending this?? -Avi On Mon, Dec 19, 2011 at 4:12 AM, Brian West <brian(a)freeswitch.org<mailto:brian@freeswitch.org>> wrote: Why do you have broken devices sending you numbers in that screwed up format in the request URI in the first place? At this point you should be using XML_CURL then cleaning that up if this is what you're hitting. Or you tell your clients to fix their badly broken behavior. /b On Dec 18, 2011, at 8:08 PM, Ryan V wrote: On Mon, Dec 19, 2011 at 2:56 AM, Elliott Vogel <elliott(a)zoogmedia.com<mailto:elliott@zoogmedia.com>>wrote: I was wondering if anyone has a regex expression that works to return just digest? I have some clients sending requests formatted to +1 (555) 555-5555, 555-555-5555, 555.555.5555 which aren't be processed by our dial plan because we are expecting all numbers (5555555555) **** How about this? ^\+1\s+$(\d{3})$\s(\d{3})-(\d{4})$|^(\d{3})-(\d{3})-(\d{4})$|^(\d{3})\.(\d{3})\.(\d{4})$ $1, $2 and $3 should give you the full number. Regards, Ryan -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266<tel:%2B1%20%28918%29%20420-9266> Fax: +1 (918) 420-9267<tel:%2B1%20%28918%29%20420-9267> brian(a)freeswitch.org<mailto:brian@freeswitch.org> http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@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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FreeSWITCH Conference Call Today
by Michael Collins 19 Dec '11

19 Dec '11

Hello all, Today's conf call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_14 We are having Areski from the Newfies Dialer project come and talk to us! Please join us at 1PM EST, 10AM PST. Thanks! -Michael

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leave voicemail then drop to IVR
by Rodney 19 Dec '11

19 Dec '11

I am trying to send callers to a voicemail box using pagd and then after they leave the message get dropped back into IVR main_menu. what is the method for this? using the following dialplan. the caller gets hung up from the whole system. thank you condition destination_number ^757$ 1 action answer 3 action sleep 1000 4 action play_and_get_digits 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+ 12 action phrase spell,${vmb} 15 action voicemail default ${domain_name} ${vmb} 20

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N-tier hierarchy real time billing using FS and NibbleBill
by gaelmartin 19 Dec '11

19 Dec '11

Hi, I need to develop a project with FreeSwitch that can bill calls in realtime not just for the end user but for all its parent account at different rate too and if one of them run out of credit the call stops. The NibbleBill module only allow to rate 2 account and 2 rates at the same time if we bill the A and B leg, this would be fine but I sometime need to bill 3 or more tier but realistically this is unlikely to go beyond 5 tiers. I tried a proof of concept by looping the call to FreeSwitch and NibbleBill each loop using it's own account and rate until there's no parent account for a user and then go out to the PSTN gateways, I cannot use the loopback system because it would remove all the middle calls once the 2 ends are connected together, so I use a sofia bridge to the local IP, (I just used a different sofia interface than the one the end user normally register into and this looping interface is not public) This is working fine and does exactly what I want, so far I've tested it with 5 tiers and as soon as one tier run out of credit it will disconnect the entire chain of call and adjust each account the correct amount. I have not tested with multiple end user calls at the same time but I expect it would work the same way. My only concern is performance and any potential issue doing some many loopback in terms of quality or timing. Looping calls in telephony system are normally reserved only as a last chance attempt to sort a problem. So I was wondering if someone had done something similar (i.e. looping calls through the same FreeSwitch instance) with no issues, and if there were anything to watch for or any tweaks to the sofia interface to make sure this will not cause an issue when scaling this system. I'm fairly new to FreeSwitch so I dont know by heart all the different options yet. Thanks. PS: I'm happy to share my proof of concept if anyone's interested doing the same. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/N-tier-hierarchy-real-time-bi… Sent from the freeswitch-users mailing list archive at Nabble.com.

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