suggestions for hardware.
First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!! I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions. Here is my plan, please let me know if it does not make sense, or if I am going about it System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server. System Design. Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ??? here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) #2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available? I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format? Thanks Orien
Hi, can someone pointme to where the valid dialing strings are specified ? I'm assuming that something, somewhere, tells FS that numbers are invalid before they get dialed ? regards Dave
On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" <stevendt@primrosebank.net> wrote:
Hi,
can someone pointme to where the valid dialing strings are specified ?
For SIP dialstrings check here: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings Also, if you send us examples of what you've tried we can help you figure out what's wrong.
I'm assuming that something, somewhere, tells FS that numbers are invalid before they get dialed ?
Pastebin some debug logs of what's happening. Check out this page which has lots of useful information on how to collect information: http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like it's just a matter of figuring out how to configure your specific setup. Please report back with more information and we'll be happy to help. -MC
Hi Michael, thanks for the reply. I think that I have got to the bottom of how to allow numbers to get to the VOIP gateway - at the moment, my dialplan just allows any. The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls to it from VOIP to the PSTN and I don't know why. I have posted a dump to the pastebin, hopefully, the messages in there will allow someone to see what the problem is and give me some pointers on how I might fix it regards Dave ----- Original Message ----- From: "Michael S Collins" <msc@freeswitch.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Saturday, November 07, 2009 6:49 PM Subject: Re: [Freeswitch-users] Valid Dial Strings
On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" <stevendt@primrosebank.net> wrote:
Hi,
can someone pointme to where the valid dialing strings are specified ?
For SIP dialstrings check here: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
Also, if you send us examples of what you've tried we can help you figure out what's wrong.
I'm assuming that something, somewhere, tells FS that numbers are invalid before they get dialed ?
Pastebin some debug logs of what's happening. Check out this page which has lots of useful information on how to collect information: http://wiki.freeswitch.org/wiki/Reporting_Bugs
It sounds like it's just a matter of figuring out how to configure your specific setup. Please report back with more information and we'll be happy to help.
-MC
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Hello, When you post something on pastebin, please post the link to your post so everyone can find it, what is the link to it? Have a nice day :) 2009/11/7 Dave Stevenson <stevendt@primrosebank.net>
Hi Michael,
thanks for the reply. I think that I have got to the bottom of how to allow numbers to get to the VOIP gateway - at the moment, my dialplan just allows any.
The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls to it from VOIP to the PSTN and I don't know why. I have posted a dump to the pastebin, hopefully, the messages in there will allow someone to see what the problem is and give me some pointers on how I might fix it
regards Dave
----- Original Message ----- From: "Michael S Collins" <msc@freeswitch.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Saturday, November 07, 2009 6:49 PM Subject: Re: [Freeswitch-users] Valid Dial Strings
On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" <stevendt@primrosebank.net> wrote:
Hi,
can someone pointme to where the valid dialing strings are specified ?
For SIP dialstrings check here: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
Also, if you send us examples of what you've tried we can help you figure out what's wrong.
I'm assuming that something, somewhere, tells FS that numbers are invalid before they get dialed ?
Pastebin some debug logs of what's happening. Check out this page which has lots of useful information on how to collect information: http://wiki.freeswitch.org/wiki/Reporting_Bugs
It sounds like it's just a matter of figuring out how to configure your specific setup. Please report back with more information and we'll be happy to help.
-MC
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Milena, thanks a lot for the reply - sorry, I'm new to this, but I'll remember that for next time. Actually, I found my way to the IRC site and the helpful chaps there got to the bottom of my problem. I had made an error copying the dialplan data from the "SPA3102 FreeSwitch HowTo" http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo I had read <action application="bridge" data="sofia/internal/${destination_number}@[IP address of SPA3102]:5061" /> as <action application="bridge" data="sofia/internal/$(destination_number)@[IP address of SPA3102]:5061" /> i.e., substituted "(...)" for "{...}" Being unfamiliar with the FreeSwitch dialplans, I'd NEVER have found the problem without help, regards Dave ----- Original Message ----- From: Milena To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 1:30 PM Subject: Re: [Freeswitch-users] Valid Dial Strings Hello, When you post something on pastebin, please post the link to your post so everyone can find it, what is the link to it? Have a nice day :) 2009/11/7 Dave Stevenson <stevendt@primrosebank.net> Hi Michael, thanks for the reply. I think that I have got to the bottom of how to allow numbers to get to the VOIP gateway - at the moment, my dialplan just allows any. The big problem is that the VOIP Gateway (Linksys 3102) rejects any calls to it from VOIP to the PSTN and I don't know why. I have posted a dump to the pastebin, hopefully, the messages in there will allow someone to see what the problem is and give me some pointers on how I might fix it regards Dave ----- Original Message ----- From: "Michael S Collins" <msc@freeswitch.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Saturday, November 07, 2009 6:49 PM Subject: Re: [Freeswitch-users] Valid Dial Strings > > On Nov 6, 2009, at 3:59 PM, "Dave Stevenson" > <stevendt@primrosebank.net> wrote: > >> Hi, >> >> can someone pointme to where the valid dialing strings are specified ? >> > For SIP dialstrings check here: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > > Also, if you send us examples of what you've tried we can help you > figure out what's wrong. > >> I'm assuming that something, somewhere, tells FS that numbers are >> invalid >> before they get dialed ? > > Pastebin some debug logs of what's happening. Check out this page > which has lots of useful information on how to collect information: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It sounds like it's just a matter of figuring out how to configure > your specific setup. Please report back with more information and > we'll be happy to help. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > 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 > _______________________________________________ FreeSWITCH-users mailing list 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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list 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
Hi, has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ? I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs. I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom. regards Dave
I have Siemens A58IP and Snom M3. Both work very well with pros and cons. Nonetheless, both lack HD .... JM On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
Hi,
has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ?
I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs.
I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom.
regards Dave
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Joao, thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ? regards Dave ----- Original Message ----- From: João Mesquita To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 6:50 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones I have Siemens A58IP and Snom M3. Both work very well with pros and cons. Nonetheless, both lack HD .... JM On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson <stevendt@primrosebank.net> wrote: Hi, has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ? I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs. I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom. regards Dave _______________________________________________ FreeSWITCH-users mailing list 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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list 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
On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote:
Joao,
thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN
Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ?
Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements.
regards Dave
Hi, thanks Brian, that's interesting. I had a comment "off list" which suggested the same thing. It did not quite fit with my aspiration for an all VOIP solution, but it sounds like the technology is not quite there yet for hands-free. That's great feedback before I spend some cash on a hands-free VOIP phone regards Dave ----- Original Message ----- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 8:05 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote: Joao, thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ? RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ? Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements. regards Dave ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list 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
I agree about the M3. I have the handset and it is not ergonomic at all. I also have a Siemens A580-IP. It does do G722 but has a few bugs related to G722 that I normally run it with G711 only. There is a quality difference between G722 and G711 when talking among the A580 handsets or the handset and my Polycom 450. Of the DECT handsets, I'd recommend the Siemens. It is also very inexpensive. Beware it has a 3 concurrent calls limit per base-station. On Mon, Nov 9, 2009 at 12:05 PM, Brian West <brian@freeswitch.org> wrote:
On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote:
Joao,
thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN
Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ?
Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements.
regards Dave
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-- -Rupa
Hi Rupa, thanks for the tip. I've had a look for the A580-PI - as you say, quite inexpensive and probably worth taking a chance on one. regards Dave ----- Original Message ----- From: "Rupa Schomaker" <rupa@rupa.com> To: <freeswitch-users@lists.freeswitch.org> Sent: Monday, November 09, 2009 8:23 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones I agree about the M3. I have the handset and it is not ergonomic at all. I also have a Siemens A580-IP. It does do G722 but has a few bugs related to G722 that I normally run it with G711 only. There is a quality difference between G722 and G711 when talking among the A580 handsets or the handset and my Polycom 450. Of the DECT handsets, I'd recommend the Siemens. It is also very inexpensive. Beware it has a 3 concurrent calls limit per base-station. On Mon, Nov 9, 2009 at 12:05 PM, Brian West <brian@freeswitch.org> wrote:
On Nov 9, 2009, at 1:38 PM, Dave Stevenson wrote:
Joao,
thanks for the note. The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN
Interesting about the HD, but do you notice the difference and find that you're disappointed with the quality of their sounds ?
Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements.
regards Dave
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-- -Rupa _______________________________________________ FreeSWITCH-users mailing list 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
On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote:
Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements.
That said, they are better than what else is around. I'd call them average. Nothing to write home about but you don't need to run away from them. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier
Hey Hadley, jump up on irc sometimes. Regards, Ognjen On Mon, Nov 9, 2009 at 9:26 PM, Hadley Rich <hads@nice.net.nz> wrote:
On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote:
Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements.
That said, they are better than what else is around.
I'd call them average. Nothing to write home about but you don't need to run away from them.
hads
-- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier
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The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad... I have Aastra 480i CT's which work well. jlc
asstra has one issue where if you look at them wrong they start telling the server that the media ip is 0.0.0.0 which we have never identified but they indeed seem to work better than snom m3 On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale <JCasale@activenetwerx.com>wrote:
The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad...
I have Aastra 480i CT's which work well.
jlc
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-- 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Beat me with a dead cat all you want but I rather the snom m3 than the Siemens A580IP.... Siemens has very low volume which makes its call quality suck despite of being ergonomic and all... That gigaset application sucks and the base station is slow as hell... Maybe I have a bad unit? The snom m3 has its downsides, but all and all, I am happy with the phone if you consider its price tag here in South America where a Polycom can easily cost over 200USD the cheapest unit. Regards, JM On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
asstra has one issue where if you look at them wrong they start telling the server that the media ip is 0.0.0.0 which we have never identified but they indeed seem to work better than snom m3
On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale < JCasale@activenetwerx.com> wrote:
The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad...
I have Aastra 480i CT's which work well.
jlc
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-- 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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2009/11/9 João Mesquita <jmesquita@freeswitch.org>:
Beat me with a dead cat all you want but I rather the snom m3 than the Siemens A580IP.... Siemens has very low volume which makes its call quality suck despite of being ergonomic and all...
Did you flip hte option in the base station that tells it to make the audio louder?
That gigaset application sucks and the base station is slow as hell... Maybe I have a bad unit?
I didn't play with any of the gigaset specific stuff, I've disabled any screen savers. Maybe if I had the more "fancy" handsets the apps would be more useful, but when using the base handset they are not. Oh, and the weather is in C rather than F -- good for the rest of the world but not for us in the US. The base station is very slow with firefox but when I use chrome isn't so bad. Dunno if it was a combination of extensions or what. Oh, and it looks like a new firmware came out for the Siemens today. Wonder what it fixes (and breaks). Hmm.. wonder where I can find a list of whats new.
The snom m3 has its downsides, but all and all, I am happy with the phone if you consider its price tag here in South America where a Polycom can easily cost over 200USD the cheapest unit.
Regards,
JM
On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
asstra has one issue where if you look at them wrong they start telling the server that the media ip is 0.0.0.0 which we have never identified but they indeed seem to work better than snom m3
On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale <JCasale@activenetwerx.com> wrote:
The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad...
I have Aastra 480i CT's which work well.
jlc
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-- 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
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-- -Rupa
On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker <rupa@rupa.com> wrote:
Oh, and it looks like a new firmware came out for the Siemens today. Wonder what it fixes (and breaks). Hmm.. wonder where I can find a list of whats new.
Well, it seems to totally break g722. I haven't had a chance to narrow it down further, but beware if g722 is important don't update... -- -Rupa
Rupa, I tried flipping, yes, but no, it works bad... Thanks for the heads up on G722! Regards, JM On Tue, Nov 10, 2009 at 1:09 PM, Rupa Schomaker <rupa@rupa.com> wrote:
On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker <rupa@rupa.com> wrote:
Oh, and it looks like a new firmware came out for the Siemens today. Wonder what it fixes (and breaks). Hmm.. wonder where I can find a list of whats new.
Well, it seems to totally break g722. I haven't had a chance to narrow it down further, but beware if g722 is important don't update...
-- -Rupa
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Thanks - pretty unambiguous reply ! I won't go down that route then :-) ----- Original Message ----- From: "Joseph L. Casale" <JCasale@activenetwerx.com> To: <freeswitch-users@lists.freeswitch.org> Sent: Monday, November 09, 2009 10:46 PM Subject: Re: [Freeswitch-users] Cordless VOIP Phones
The Snom M3 is one of the ones that I was looking at - I would be interested in the "Pro's & Cons" ?
Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad...
I have Aastra 480i CT's which work well.
jlc
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I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? regards Dave
Would something like this work for you? <extension name="some-extension"> <condition field="destination_number" expression="^100$"> <!-- do something --> </condition> </extension> <extension name="another-extension"> <condition field="destination_number" expression="^([1-9]\d{2})$"> <!-- do something else --> </condition> </extension> --Stephen On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
I **think** that the following will match any three character strings from 1xx to 399
I want to exclude 100 though, can anyone help me with the required RegEx please ?
^([1-3][0-9][0-9])$
I could (I think) do
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
But it does not "feel" elegant - is there a better way ?
regards Dave
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Hi Stephen, thanks for the reply. I'm not sure , does the code below handle all number from 101 to 399 ? It would rely on the 100 code being picked up by the dialplan before the other extensions were processed so the order of the code in the dialplan is significant. Is that how people normally write their code, i.e., the extension processing is position dependant in the file ? regards Dave ----- Original Message ----- From: Stephen Crosby To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 8:37 PM Subject: Re: [Freeswitch-users] RegEx Help Would something like this work for you?<extension name="some-extension"> <condition field="destination_number" expression="^100$"> <!-- do something --> </condition></extension><extension name="another-extension"> <condition field="destination_number" expression="^([1-9]\d{2})$"> <!-- do something else --> </condition></extension>--Stephen On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson <stevendt@primrosebank.net> wrote: I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? regards Dave _______________________________________________ FreeSWITCH-users mailing list 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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list 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
Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) <extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> </condition> </extension> That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox <action application="set" data="dialed_extension=$1"/> set the timeout for the call <action application="set" data="call_timeout=20"/> Added these lines - but don't know why - they are in the default extension code ????? <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> then go to voice mail on 100 <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> giving <extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> </condition> </extension> The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? regards Dave
See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
Hi Guys,
OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted.
I have created a group (100) which contains a number of phones - 101 to 105 at the moment.
When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code)
<extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="${group_call( primrose@${domain_name})}"/ <primrose@$%7Bdomain_name%7D%29%7D%22/>> </condition> </extension> That's fine and the nominated phones all ring.
I'm struggling to get it to do what I want when some doesn't pick up though.
All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in.
What I want to happen is for extension 100's voice mail to kick in after a time delay.
So, get the dialed exetension number so that I can point at the right mailbox
<action application="set" data="dialed_extension=$1"/> set the timeout for the call
<action application="set" data="call_timeout=20"/> Added these lines - but don't know why - they are in the default extension code ?????
<action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> then go to voice mail on 100
<action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/>
giving
<extension name="group_dial_primrose">
The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated.
<condition field="destination_number" expression="^100$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call( primrose@${domain_name})}"/ <primrose@$%7Bdomain_name%7D%29%7D%22/>> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="sleep" data="1000"/>
I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}"
<action application="voicemail" data="default ${domain_name}{dialed_extension}"/>
</condition> </extension> The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages.
Can someone help please ?
Where am i going wrong ?
Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC
Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! <condition field="destination_number" expression="^(100)$"> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson <stevendt@primrosebank.net> wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) <extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> </condition> </extension> That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox <action application="set" data="dialed_extension=$1"/> set the timeout for the call <action application="set" data="call_timeout=20"/> Added these lines - but don't know why - they are in the default extension code ????? <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> then go to voice mail on 100 <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> giving <extension name="group_dial_primrose"> The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. <condition field="destination_number" expression="^100$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="sleep" data="1000"/> I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> </condition> </extension> The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Well, I thought it was fixed - it is more or less working, with one more stumbling block. I have just posted a dump to the pastebin - from Dave (stevendt) The voice mail works - but too well. If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.# Is there something else wrong with the dialplan logic below ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 11:52 PM Subject: Re: [Freeswitch-users] DIalplan logic Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! <condition field="destination_number" expression="^(100)$"> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson <stevendt@primrosebank.net> wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) <extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> </condition> </extension> That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox <action application="set" data="dialed_extension=$1"/> set the timeout for the call <action application="set" data="call_timeout=20"/> Added these lines - but don't know why - they are in the default extension code ????? <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> then go to voice mail on 100 <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> giving <extension name="group_dial_primrose"> The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. <condition field="destination_number" expression="^100$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="sleep" data="1000"/> I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> </condition> </extension> The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list 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 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list 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
You set both hangup_after_bridge and continue_on_fail after you already called bridge. Try setting it *before* Seems to be a running theme here that things will be parsed in a linear fashion that you may want to take note of. On Mon, Nov 9, 2009 at 6:18 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
Well,
I thought it was fixed - it is more or less working, with one more stumbling block.
I have just posted a dump to the pastebin - from Dave (stevendt)
The voice mail works - but too well.
If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.#
Is there something else wrong with the dialplan logic below ?
regards Dave
----- Original Message ----- *From:* Dave Stevenson <stevendt@primrosebank.net> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Monday, November 09, 2009 11:52 PM *Subject:* Re: [Freeswitch-users] DIalplan logic
Michael,
thanks a lot - it's fixed......
you spotted exactly what the problem was !
<condition field="destination_number" expression="^(100)$">
<action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
regards Dave
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Monday, November 09, 2009 11:26 PM *Subject:* Re: [Freeswitch-users] DIalplan logic
See comment inline
On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
Hi Guys,
OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted.
I have created a group (100) which contains a number of phones - 101 to 105 at the moment.
When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code)
<extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="${group_call( primrose@${domain_name})}"/ <primrose@$%7Bdomain_name%7D%29%7D%22/>> </condition> </extension> That's fine and the nominated phones all ring.
I'm struggling to get it to do what I want when some doesn't pick up though.
All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in.
What I want to happen is for extension 100's voice mail to kick in after a time delay.
So, get the dialed exetension number so that I can point at the right mailbox
<action application="set" data="dialed_extension=$1"/> set the timeout for the call
<action application="set" data="call_timeout=20"/> Added these lines - but don't know why - they are in the default extension code ?????
<action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> then go to voice mail on 100
<action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/>
giving
<extension name="group_dial_primrose">
The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated.
<condition field="destination_number" expression="^100$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call( primrose@${domain_name})}"/ <primrose@$%7Bdomain_name%7D%29%7D%22/>> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="sleep" data="1000"/>
I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}"
<action application="voicemail" data="default ${domain_name}{dialed_extension}"/>
</condition> </extension> The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages.
Can someone help please ?
Where am i going wrong ?
Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC
------------------------------
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------------------------------
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-- 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks Anthony that did the trick ! Excuse my ignorance - this is all new to me . . . It would help if I knew what I was doing, as I commented below, I copied the <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> from the code for the default extensions (in the wrong order though obviously), without understanding what they were there for ! I've just stumbled across the Wiki page http://wiki.freeswitch.org/wiki/Extension_Status_Example - hopefully, I understand now regards Dave ----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 10, 2009 12:29 AM Subject: Re: [Freeswitch-users] DIalplan logic You set both hangup_after_bridge and continue_on_fail after you already called bridge. Try setting it *before* Seems to be a running theme here that things will be parsed in a linear fashion that you may want to take note of. On Mon, Nov 9, 2009 at 6:18 PM, Dave Stevenson <stevendt@primrosebank.net> wrote: Well, I thought it was fixed - it is more or less working, with one more stumbling block. I have just posted a dump to the pastebin - from Dave (stevendt) The voice mail works - but too well. If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.# Is there something else wrong with the dialplan logic below ? regards Dave ----- Original Message ----- From: Dave Stevenson To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 11:52 PM Subject: Re: [Freeswitch-users] DIalplan logic Michael, thanks a lot - it's fixed...... you spotted exactly what the problem was ! <condition field="destination_number" expression="^(100)$"> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 11:26 PM Subject: Re: [Freeswitch-users] DIalplan logic See comment inline On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson <stevendt@primrosebank.net> wrote: Hi Guys, OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted. I have created a group (100) which contains a number of phones - 101 to 105 at the moment. When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code) <extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> </condition> </extension> That's fine and the nominated phones all ring. I'm struggling to get it to do what I want when some doesn't pick up though. All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in. What I want to happen is for extension 100's voice mail to kick in after a time delay. So, get the dialed exetension number so that I can point at the right mailbox <action application="set" data="dialed_extension=$1"/> set the timeout for the call <action application="set" data="call_timeout=20"/> Added these lines - but don't know why - they are in the default extension code ????? <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> then go to voice mail on 100 <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> giving <extension name="group_dial_primrose"> The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated. <condition field="destination_number" expression="^100$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call(primrose@${domain_name})}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="sleep" data="1000"/> I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}" <action application="voicemail" data="default ${domain_name}{dialed_extension}"/> </condition> </extension> The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages. Can someone help please ? Where am i going wrong ? Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC ------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list 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 -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list 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 _______________________________________________ FreeSWITCH-users mailing list 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 -- 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list 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
Oops, you've got some lines that are in the wrong place: <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> Those lines need to come prior to the bridge call or they'll never be applied. :) -MC On Mon, Nov 9, 2009 at 4:18 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
Well,
I thought it was fixed - it is more or less working, with one more stumbling block.
I have just posted a dump to the pastebin - from Dave (stevendt)
The voice mail works - but too well.
If the call is answered by a someone at this end - everything is fine until the user hangs up, then the remote party gets the voicemail messages.#
Is there something else wrong with the dialplan logic below ?
regards Dave
----- Original Message ----- *From:* Dave Stevenson <stevendt@primrosebank.net> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Monday, November 09, 2009 11:52 PM *Subject:* Re: [Freeswitch-users] DIalplan logic
Michael,
thanks a lot - it's fixed......
you spotted exactly what the problem was !
<condition field="destination_number" expression="^(100)$">
<action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
regards Dave
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* freeswitch-users@lists.freeswitch.org *Sent:* Monday, November 09, 2009 11:26 PM *Subject:* Re: [Freeswitch-users] DIalplan logic
See comment inline
On Mon, Nov 9, 2009 at 2:56 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
Hi Guys,
OK, with the RegEx help that you gave me, I have separated out the processing of extension 100 from 101 to 399 as I wanted.
I have created a group (100) which contains a number of phones - 101 to 105 at the moment.
When the PSTN line rings, I want all the extensions in the group to ring - that's the easy bit (I think - it's a copy of extension 2000 code)
<extension name="group_dial_primrose"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="${group_call( primrose@${domain_name})}"/ <primrose@$%7Bdomain_name%7D%29%7D%22/>> </condition> </extension> That's fine and the nominated phones all ring.
I'm struggling to get it to do what I want when some doesn't pick up though.
All extensions ring as required, but their own dialplan entries (copies of the 1001 to 1005 code in the default dialplan) don't answer the call. That's fine, as you would not want every extension's voice mail to kick in.
What I want to happen is for extension 100's voice mail to kick in after a time delay.
So, get the dialed exetension number so that I can point at the right mailbox
<action application="set" data="dialed_extension=$1"/> set the timeout for the call
<action application="set" data="call_timeout=20"/> Added these lines - but don't know why - they are in the default extension code ?????
<action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> then go to voice mail on 100
<action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name}{dialed_extension}"/>
giving
<extension name="group_dial_primrose">
The following line needs to have 100 in parens like this: "^(100)$" because that's how you get $1 to be populated.
<condition field="destination_number" expression="^100$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call( primrose@${domain_name})}"/ <primrose@$%7Bdomain_name%7D%29%7D%22/>> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="sleep" data="1000"/>
I think this might be a typo? Shouldn't this next line be ... data="default ${domain_name} ${dialed_extension}"
<action application="voicemail" data="default ${domain_name}{dialed_extension}"/>
</condition> </extension> The voicemail kicks in, and prompts are correct (although the extension name is not spoken) but the wav file is saved in the 1001 directory not 100 and neither extension 100 or 1001 think that have any voice mail messages.
Can someone help please ?
Where am i going wrong ?
Make those changes, reloadxml, and then try again. Be sure to capture a debug log if it doesn't work and put that log in pastebin.freeswitch.org. -MC
------------------------------
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------------------------------
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On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
I **think** that the following will match any three character strings from 1xx to 399
I want to exclude 100 though, can anyone help me with the required RegEx please ?
^([1-3][0-9][0-9])$
I could (I think) do
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
But it does not "feel" elegant - is there a better way ?
expression="^[1-3]\d\d$" will match 100 to 399 -MC
Thanks Michael, but I want to exclude 100 ? regards Dave ----- Original Message ----- From: Michael Collins To: freeswitch-users@lists.freeswitch.org Sent: Monday, November 09, 2009 8:38 PM Subject: Re: [Freeswitch-users] RegEx Help On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson <stevendt@primrosebank.net> wrote: I **think** that the following will match any three character strings from 1xx to 399 I want to exclude 100 though, can anyone help me with the required RegEx please ? ^([1-3][0-9][0-9])$ I could (I think) do ^([1-3][1-9][0-9]|[2-3][0-9][0-9])$ But it does not "feel" elegant - is there a better way ? expression="^[1-3]\d\d$" will match 100 to 399 -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On Mon, Nov 09, Michael Collins wrote:
On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
I **think** that the following will match any three character strings from 1xx to 399
I want to exclude 100 though, can anyone help me with the required RegEx please ?
^([1-3][0-9][0-9])$
I could (I think) do
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
You mean (^1[0-9][1-9]$|^[2-3]\d\d$) --FC
Hi Frank Yup ! That's what I mean :-) thanks a lot, regards Dave ----- Original Message ----- From: "Frank Carmickle" <frank@carmickle.com> To: <freeswitch-users@lists.freeswitch.org> Sent: Monday, November 09, 2009 9:03 PM Subject: Re: [Freeswitch-users] RegEx Help
On Mon, Nov 09, Michael Collins wrote:
On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
I **think** that the following will match any three character strings from 1xx to 399
I want to exclude 100 though, can anyone help me with the required RegEx please ?
^([1-3][0-9][0-9])$
I could (I think) do
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
You mean
(^1[0-9][1-9]$|^[2-3]\d\d$)
--FC
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Dave, I think extensions are processed in order although I can't quickly find any documentation that says this, why don't you try it and see, it would take only a moment to find out for sure. --Stephen On Mon, Nov 9, 2009 at 1:03 PM, Frank Carmickle <frank@carmickle.com> wrote:
On Mon, Nov 09, Michael Collins wrote:
On Mon, Nov 9, 2009 at 12:22 PM, Dave Stevenson <stevendt@primrosebank.net>wrote:
I **think** that the following will match any three character strings from 1xx to 399
I want to exclude 100 though, can anyone help me with the required RegEx please ?
^([1-3][0-9][0-9])$
I could (I think) do
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
You mean
(^1[0-9][1-9]$|^[2-3]\d\d$)
--FC
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Dave Stevenson <stevendt@primrosebank.net> said:
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
Another possibility. ^(1(0[1-9]|[1-9]\d)|[2-3]\d{2}) -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events!
On Mon, Nov 9, 2009 at 1:10 PM, <Russell.Mosemann@cune.org> wrote:
Dave Stevenson <stevendt@primrosebank.net> said:
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
Another possibility.
^(1(0[1-9]|[1-9]\d)|[2-3]\d{2})
Yep this is the one. I'm sorry I didn't read the OP correctly the first time. Skipping 100 and matching 101 is the tricky part, obviously. This regex should fit the bill. -MC
If the global var "auto_hunt" is "true" the xml dialplan will try to find an extension where the "name" param matches the destination number. This is not the default The default is to try them in order from top to bottom. On Mon, Nov 9, 2009 at 3:20 PM, Michael Collins <msc@freeswitch.org> wrote:
On Mon, Nov 9, 2009 at 1:10 PM, <Russell.Mosemann@cune.org> wrote:
Dave Stevenson <stevendt@primrosebank.net> said:
^([1-3][1-9][0-9]|[2-3][0-9][0-9])$
Another possibility.
^(1(0[1-9]|[1-9]\d)|[2-3]\d{2})
Yep this is the one. I'm sorry I didn't read the OP correctly the first time. Skipping 100 and matching 101 is the tricky part, obviously. This regex should fit the bill. -MC
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-- 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
On Nov 6, 2009, at 3:21 PM, Orien Love <orien@tx.rr.com> wrote:
First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!!
I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions.
Here is my plan, please let me know if it does not make sense, or if I am going about it
System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server.
System Design.
Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ???
here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) #2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available?
I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format?
Thanks Orien
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On Nov 6, 2009, at 3:21 PM, Orien Love <orien@tx.rr.com> wrote:
First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!!
I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions.
Here is my plan, please let me know if it does not make sense, or if I am going about it
System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server.
System Design.
Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ???
here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010)
The FS devs don't endorse or recommend any specific hardware. However, many FreeSWITCH user are quite vocal about the hardware they prefer so we will let them speak. Just remember that YMMV depending on your specific setup. That being said, it does not sound like your HW reqs are very intense. Some of our members who've used the atom can probably give you specific feedback.
#2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available?
I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format?
Thanks Orien
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I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions.
4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server.
The last time I tried using spa3000's for PSTN connections, I had a lot of difficulty tuning the settings to get rid of some echo the users were hearing. I don't know what the local loop length was, and the wiring in that neighbourhood is pretty old - both probably had a strong influence on what I was seeing. My determination at that time (~3 years ago) was that they were good for home connections, but not suitable for business use. I ultimately went with a Sangoma card (A200 +HWEC module), it killed the echo and sounded great. YMMV - Firmware and hardware has probably changed quite a bit in 3 years, and your loop characteristics might be better. My inclination would be to buy one, and see how well it works at your site.
System Design.
Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ???
Depending on how you want your system to work - auto-attendant is a good way to go if you won't have DIDs for all or most individuals. Our receptionist's (and our client's) time was being chewed up by all calls being forced to go through her before we implemented an auto-attendant.
I have built some low-wattage systems before, and from what I have seen, you want to stay away from the low-wattage processors. On a price per performance per watt scale, the lowest wattage Core 2 Duo processors are the best bet. I have a logic-supply ITX system here at home that has a VIA processor in it and it is dirt slow. It cost about $350 - $400. It sucks about 5W max, according to my Watt meter but it takes forever to do anything. Compiling FreeSWITCH on it is an absolute nightmare (it takes hours). I have an alternate MicroATX system with a Core 2 Duo and it pulls about 15W all the time. If I compile FreeSWITCH on that system, it takes a few minutes. The Core 2 Duo system was bought about 6 months after the VIA system. The Core 2 Duo cost about $270 for everything, including 2 GB of RAM. The amount of money saved by running the VIA system at 5W is nowhere close to the inconvenience of waiting for it to do anything. Also, I had to go through about 5-6 different Linux distributions before I found one that would actually install on the VIA processor. I think Suse Linux was the one that finally worked on it. Tried CentOS, Slackware, Ubuntu, Debian, and a couple of others. Now, I have not tried the Atom processor, so it could be very different. However, I have read the Tom's Hardware review that also showed that the Core 2 Duo was several times better on the price per performance per watt scale than the Atom processor, and again, it was mainly because the Atom processor took so much longer to do anything. On Fri, Nov 6, 2009 at 6:21 PM, Orien Love <orien@tx.rr.com> wrote:
First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!!
I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions.
Here is my plan, please let me know if it does not make sense, or if I am going about it
System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server.
System Design.
Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ???
here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) #2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available?
I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format?
Thanks Orien
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-- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero
participants (17)
-
Anthony Minessale
-
Brian West
-
Dana Harding
-
Dave Stevenson
-
Eliot Gable
-
Frank Carmickle
-
Hadley Rich
-
Joseph L. Casale
-
João Mesquita
-
Michael Collins
-
Michael S Collins
-
Milena
-
Ognjen Seslija
-
Orien Love
-
Rupa Schomaker
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Russell.Mosemann@cune.org
-
Stephen Crosby