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Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds
by William Traenkle 16 Mar '10

16 Mar '10

Michael, Last question, what dictates when a call uses the public.xml or the default.xml dial plan? Thanks, ________________________________ From: William Traenkle <will.traenkle(a)yahoo.com> To: Michael Collins <msc(a)freeswitch.org>; freeswitch-users(a)lists.freeswitch.org Sent: Tue, March 16, 2010 1:03:32 AM Subject: Re: IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds Michael, Thanks for your help. I was able to get it work, however, I would like your opinion as how to do this correctly as I have a hunch I may have opened up some security risks or have gone against freeswitch best practices. If not, just let me know and I will leave it as is. I modified the public.xml file under /conf/dialplan/ I uncommented the following:  I added the following:  <extension name="send_to_voicemail"> <condition field="destination_number" expression="^\*99(\d{3,6})$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension>  <X-PRE-PROCESS cmd="include" data="default/*.xml"/> ________________________________ From: Michael Collins <msc(a)freeswitch.org> To: William Traenkle <will.traenkle(a)yahoo.com> Sent: Mon, March 15, 2010 5:11:57 PM Subject: Re: IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds On Mon, Mar 15, 2010 at 4:22 PM, William Traenkle <will.traenkle(a)yahoo.com> wrote: When a hunt group is accessed from the main IVR, the hunt group dials >an internal extension via correctly but when trying to dial an >external number such a cell phone via sip uri it is unable to connect >or ring the cell phone and it waits the correct timeout period before it >goes to voicemail. > >However, if you dial the hunt group directly without going through the main IVR, the external number such as cell phone that uses sip uri to connect to it work perfect. > > > >>There is only an issue when the hunt group is accessed from the main IVR when dialing an external number using sip uri. > >>1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR > >>2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3 In #2 your call to 7000 is hitting the public context. Most likely this isn't what you were anticipating. Try handling 7000 in the public context. Find the public_extensions dp entry and update your regex there: <condition field="destination_number" expression="^(10[01][0-9]|7000)$"> reloadxml and try again. If it works then consider creating a public context extension that specifically handles these calls, perhaps checking the IP address of the source of the call. -MC > >>3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3 > >>4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab > > >5) Dialplan default.xml: >http://pastebin.com/iAbNSW9E > >If you require additional information or have any other >questions, please let me know. > >>Thanks, > >>-Will > ________________________________ From: Michael Collins <msc(a)freeswitch.org> >To: freeswitch-users(a)lists.freeswitch.org >Sent: Tue, March 9, 2010 11:00:53 PM >Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds > > > > >On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle(a)yahoo.com> wrote: > >When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail. >> >>However, if you access the hunt group directly without going through the main IVR, all works perfectly. >> >>Any ideas where I should look and what line of code I might be missing? >> >> > >Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition. >> >-MC > > >

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voicemail options
by Madovsky 16 Mar '10

16 Mar '10

Hi all, is there any way to set the voicemail for leave a message without any option at the end ? like record or send the message by email once the caller hangs up or after N sec of silence. Thanks Franck

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Re: [Freeswitch-users] Set/Export channel variable [SOLVED]
by Ahmed Ajmal 16 Mar '10

16 Mar '10

So it seems that the FS console by default only displays the 'a' leg vars which is why I couldn't see it (and it makes sense too since gateway is on the b leg side). I have the ESL script now display both the inbound/outbound after which I can see the sip_gateway_name var. -ahmed From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ahmed Ajmal Sent: Tuesday, March 16, 2010 10:42 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Set/Export channel variable I am looking at the FS console after hangup is complete. I also have a ESL event hooked up for HANGUP_COMPLETE event. In both cases I am unable to see the sip_gateway_name variable. -ahmed From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, March 15, 2010 11:29 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Set/Export channel variable On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal(a)breezecom.ae> wrote: Hi I have now configured the gateways as suggested below but I am not able to see the sip_gateway_name variable when the call hangup event is complete. I am on Freeswitch 1.0.5. Where are you looking to find the variable? -MC

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Re: [Freeswitch-users] Set/Export channel variable
by Michael Collins 16 Mar '10

16 Mar '10

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal(a)breezecom.ae>wrote: > Hi > > > > I have now configured the gateways as suggested below but I am not able to > see the sip_gateway_name variable when the call hangup event is complete. I > am on Freeswitch 1.0.5. > > > Where are you looking to find the variable? -MC

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voicemail message "saved" error
by Madovsky 16 Mar '10

16 Mar '10

Hi, 2010-03-16 00:44:27.878289 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not match any patterns sh: line 1: 30667 Done /bin/cat /tmp/mail.126871466794cf 30668 Segmentation fault | sendmail -f infos(a)bibi.baba -t other(a)kiki.coco Thanks F

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Switch with External SoftPhone
by Vladimir Elizarov 16 Mar '10

16 Mar '10

Hello. What are the ways connect external users to the fs there? In such a scheme: fs with real ip <-> internet <-> nat <-> sip-phone I read the topic http://wiki.freeswitch.org/wiki/External_profile # Switch_with_External_SoftPhone I did not understand several things: - That should be in exteranl5090.xml? Whole profile external.xml c changes in paragraphs 2,3? - Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes in xxxx? External UA IP? Domain?" I have the same question. that x.x.x.x should be? But this seems not quite what I need. Must be external to the subscriber is practically no different from internal. -- Best regards, Vladimir Elizarov

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Re: [Freeswitch-users] voicemail options [SOLVED]
by Madovsky 16 Mar '10

16 Mar '10

Brian, I read the wiki 2 days ago already and din't help me ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 12:46 AM Subject: Re: [Freeswitch-users] voicemail options [SOLVED] as my sendmail is ultra secure I forgot that the FROM "email" MUST be registered in your mails server... but now there is another problem. I open a new thread Thanks for your time Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 12:33 AM Subject: Re: [Freeswitch-users] voicemail options there's a thread here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.… for exim with the same problem, but at the end he didn't say if it worked or not ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 12:20 AM Subject: Re: [Freeswitch-users] voicemail options So I checked t/mp WHILE the operator says "message saved" but nothing like /tmp/mail.xxxxxxx do you have an army of ideas ? ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 15, 2010 11:58 PM Subject: Re: [Freeswitch-users] voicemail options sendmail. ----- Original Message ----- From: Brian West To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 15, 2010 11:50 PM Subject: Re: [Freeswitch-users] voicemail options NO your mailer is crashing cuz its running out of stack space I suspect. What mailer are you using? /b On Mar 15, 2010, at 10:42 PM, Madovsky wrote: fingers ripped... no FS is ok, I updated to 16998, used the default voicemail config and it's the same /bin/cat: write error: Broken pipe sh: line 1: 15281 Done(1) /bin/cat /tmp/mail.12687097673f11 15282 Segmentation fault | sendmail -f gaga(a)gogo.kiki -t infos(a)bobo.baba I think it means that the cat doesn't find the file... so I checked the /tmp permission and it's ok (chmod 1777) also if I change in voicemail.conf.xml <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/> FS continues to use /tmp. Latest trunk, fedora 10 64bits ---------------------------------------------------------------------- _______________________________________________ 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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Attrafax
by Kristian Kielhofner 16 Mar '10

16 Mar '10

Disclaimers: - I haven't looked at it in detail (at all, really) - It is GPL2 (although perhaps Zoa and co can budge on this) http://www.venturevoip.com/news.php?rssid=2364 Any thoughts? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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PSTN connection
by Ken Gillett 16 Mar '10

16 Mar '10

If I am running FreeSwitch on a Mac, would I be able to use Apple's USB modem as a means to connect to the PSTN and utilise this line to place and receive calls? Ken G i l l e t t _/_/_/_/_/_/_/_/

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Re: [Freeswitch-users] voicemail options [SOLVED]
by Madovsky 16 Mar '10

16 Mar '10

as my sendmail is ultra secure I forgot that the FROM "email" MUST be registered in your mails server... but now there is another problem. I open a new thread Thanks for your time Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 12:33 AM Subject: Re: [Freeswitch-users] voicemail options there's a thread here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.… for exim with the same problem, but at the end he didn't say if it worked or not ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 12:20 AM Subject: Re: [Freeswitch-users] voicemail options So I checked t/mp WHILE the operator says "message saved" but nothing like /tmp/mail.xxxxxxx do you have an army of ideas ? ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 15, 2010 11:58 PM Subject: Re: [Freeswitch-users] voicemail options sendmail. ----- Original Message ----- From: Brian West To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 15, 2010 11:50 PM Subject: Re: [Freeswitch-users] voicemail options NO your mailer is crashing cuz its running out of stack space I suspect. What mailer are you using? /b On Mar 15, 2010, at 10:42 PM, Madovsky wrote: fingers ripped... no FS is ok, I updated to 16998, used the default voicemail config and it's the same /bin/cat: write error: Broken pipe sh: line 1: 15281 Done(1) /bin/cat /tmp/mail.12687097673f11 15282 Segmentation fault | sendmail -f gaga(a)gogo.kiki -t infos(a)bobo.baba I think it means that the cat doesn't find the file... so I checked the /tmp permission and it's ok (chmod 1777) also if I change in voicemail.conf.xml <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/> FS continues to use /tmp. Latest trunk, fedora 10 64bits ------------------------------------------------------------------------ _______________________________________________ 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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