Re: [Freeswitch-users] How to initiate a call without dialing
Raymond Chandler wrote: Eli Hayun wrote: Is there is a way to initiate a call without making any dial manually? i think the api command originate is what you're looking for -Ray ___ Thanks, I figure that out, but now I have another problem. When I do that, the name display as FreeSwitch and the number is display as 000 I tried to set outbound_caller_name with no success. How should I solve that? Thanks Eli ___ 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
Re: [Freeswitch-users] How to initiate a call without dialing
Eli Hayun pisze: Raymond Chandler wrote: Eli Hayun wrote: Is there is a way to initiate a call without making any dial manually? i think the api command originate is what you're looking for -Ray ___ Thanks, I figure that out, but now I have another problem. When I do that, the name display as FreeSwitch and the number is display as 000 I tried to set outbound_caller_name with no success. How should I solve that? Thanks Eli Read wiki, it explains a lot. http://wiki.freeswitch.org/wiki/Mod_commands#originate use it like that: originate sofia/internal/1001%192.168.1.1 3001 XML default Name 1213232 OR originate sofia/internal/1001%192.168.1.1 conference(test) '' '' Name 1213232 ___ 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
Re: [Freeswitch-users] How to initiate a call without dialing
Szymon Olko wrote: Eli Hayun pisze: Raymond Chandler wrote: Eli Hayun wrote: Is there is a way to initiate a call without making any dial manually? i think the api command originate is what you're looking for -Ray ___ Thanks, I figure that out, but now I have another problem. When I do that, the name display as FreeSwitch and the number is display as 000 I tried to set outbound_caller_name with no success. How should I solve that? Thanks Eli Read wiki, it explains a lot. http://wiki.freeswitch.org/wiki/Mod_commands#originate use it like that: originate sofia/internal/1001%192.168.1.1 3001 XML default Name 1213232 OR originate sofia/internal/1001%192.168.1.1 conference(test) '' '' Name 1213232 ___ 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 Thanks alot. Its working now. Eli ___ 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
Re: [Freeswitch-users] Confusing handling of incoming calls
On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller p...@privateconnect.comwrote: My goal is: 0) figure out why the bandwidth gateway is being processed as internal (this is more of a security thing) they are probably terminating traffic on port 5060 rather than 5080. 5060 is internal, 5080 is external. 1) have both gateways enter at the same point in the dialplan (this seems to be the purpose of the Extension param) I'd drop the extension param and instead match on the destination_number (the DID used to reach you). 2) be able to identify which gateway the call came in on. I was hoping to set a param in the gateway configuration that would be passed through onto the channel, but have not found one. Worst case, I could have each gateway enter at a different extension in the dialplan, however, that doesn't seem to be working if the channel comes in the internal profile. Not sure here... gateways are an outbound thing. Inbound calls just hit your dialplan and you process from there. A sledgehammer approach would be to have a different sip_profile for each gateway. But that is just silly. Flowroute at least puts their name in the sdp Thanks for your help. I've provided INFO dumps from both gateways if they help... -pete -- -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
Re: [Freeswitch-users] Confusing handling of incoming calls
0) Rupa, you are absolutely right, I forgot that. ports was never an issue because previous gateways all REGISTERed. I will have to swap my ports around as bandwidth is not flexible.1) I thought of this, but I have hundreds of DID, (around 600 at the moment) and maintaining that mapping in the dialplan would be a mess. AFTER I know what gateway the call arrived on, I have a database for each gateway that helps me process from there.2) Yes, separate profiles would work, but does sound gross. I'm going to swap my ports around and see if that clears things up... -pete Original Message Subject: Re: [Freeswitch-users] Confusing handling of incoming calls From: Rupa Schomaker r...@rupa.com Date: Wed, July 22, 2009 2:12 am To: freeswitch-users@lists.freeswitch.org On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller p...@privateconnect.com wrote: My goal is:0) figure out why the bandwidth gateway is being processed as "internal" (this is more of a security thing) they are probably terminating traffic on port 5060 rather than 5080. 5060 is internal, 5080 is external. 1) have both gateways enter at the same point in the dialplan (this seems to be the purpose of the "Extension" param) I'd drop the extension param and instead match on the destination_number (the DID used to reach you). 2) be able to identify which gateway the call came in on. I was hoping to set a param in the gateway configuration that would be passed through onto the channel, but have not found one. Worst case, I could have each gateway enter at a different extension in the dialplan, however, that doesn't seem to be working if the channel comes in the "internal" profile. Not sure here... gateways are an outbound thing. Inbound calls just hit your dialplan and you process from there. A sledgehammer approach would be to have a different sip_profile for each gateway. But that is just silly. Flowroute at least puts their name in the sdp Thanks for your help. I've provided INFO dumps from both gateways if they help...-pete-- -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 ___ 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] Playing sound files in a conference
Hallo everybody! I would like to play soundfiles in a existing conference. The procedure is this: Someone calls the number of the conference. Then this person types the pin in to his phone. The next step is that he has to say the name for example John. This file is saved in a special folder. When he enters the conference room everybody in this existing conference should here: John has enterd the room. If he leaves then everybody should hear: John has left the room. Of course there are two soundfiles. John is what the caller has spoken into his phone and the second one a generated file from me. They should be played in succession. Is it possible to implement this with lua? If yes, how can I do that. Thanks for your help. Greetz ___ 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
Re: [Freeswitch-users] Confusing handling of incoming calls
On Jul 22, 2009, at 5:11 AM, Pete Mueller wrote: 0) Rupa, you are absolutely right, I forgot that. ports was never an issue because previous gateways all REGISTERed. I will have to swap my ports around as bandwidth is not flexible. What do you mean here? 1) I thought of this, but I have hundreds of DID, (around 600 at the moment) and maintaining that mapping in the dialplan would be a mess. AFTER I know what gateway the call arrived on, I have a database for each gateway that helps me process from there. XML_CURL? 2) Yes, separate profiles would work, but does sound gross. I'm going to swap my ports around and see if that clears things up... -pete ___ 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
Re: [Freeswitch-users] Dial up from confernece issue
The far end you're calling is sending a 302 can you check the sip traffic please. /b On Jul 22, 2009, at 12:45 AM, Łukasz Zwierko wrote: I'm not sure how this exactly works, but I suppose that it is a single leg call, which upon answer would be attached to the conference (?) somehow. But again, this call does not originate outside FS so what would be the cause for 302? ___ 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] Limit is not working when originate a call
Hi
I set the limit to 1 on the extension like that
action application=limit_hash data=${destination_number}
${destination_number} 1 /
When I am trying to make a call the that destination i transfered to
limit_exceeded dialplan, just like I want
The problem is, that when I am trying to make a call using originate I
am not getting the limitation.
Why is that?
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Re: [Freeswitch-users] Confusing handling of incoming calls
On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller p...@privateconnect.comwrote: 0) Rupa, you are absolutely right, I forgot that. ports was never an issue because previous gateways all REGISTERed. I will have to swap my ports around as bandwidth is not flexible. You can't tell bandwidth.com to use port 5080? 1) I thought of this, but I have hundreds of DID, (around 600 at the moment) and maintaining that mapping in the dialplan would be a mess. AFTER I know what gateway the call arrived on, I have a database for each gateway that helps me process from there. You have cases where the same DID maps differently for one gateway or another? If not, why is the gateway part of the database query? 2) Yes, separate profiles would work, but does sound gross. I'm going to swap my ports around and see if that clears things up... -pete Original Message Subject: Re: [Freeswitch-users] Confusing handling of incoming calls From: Rupa Schomaker r...@rupa.com Date: Wed, July 22, 2009 2:12 am To: freeswitch-users@lists.freeswitch.org On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller p...@privateconnect.comwrote: My goal is: 0) figure out why the bandwidth gateway is being processed as internal (this is more of a security thing) they are probably terminating traffic on port 5060 rather than 5080. 5060 is internal, 5080 is external. 1) have both gateways enter at the same point in the dialplan (this seems to be the purpose of the Extension param) I'd drop the extension param and instead match on the destination_number (the DID used to reach you). 2) be able to identify which gateway the call came in on. I was hoping to set a param in the gateway configuration that would be passed through onto the channel, but have not found one. Worst case, I could have each gateway enter at a different extension in the dialplan, however, that doesn't seem to be working if the channel comes in the internal profile. Not sure here... gateways are an outbound thing. Inbound calls just hit your dialplan and you process from there. A sledgehammer approach would be to have a different sip_profile for each gateway. But that is just silly. Flowroute at least puts their name in the sdp Thanks for your help. I've provided INFO dumps from both gateways if they help... -pete -- -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 ___ 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 -- -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
Re: [Freeswitch-users] Confusing handling of incoming calls
On Jul 22, 2009, at 8:04 AM, Rupa Schomaker wrote: On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller p...@privateconnect.com wrote: 0) Rupa, you are absolutely right, I forgot that. ports was never an issue because previous gateways all REGISTERed. I will have to swap my ports around as bandwidth is not flexible. You can't tell bandwidth.com to use port 5080? Yes you can... I do it all the time. 1) I thought of this, but I have hundreds of DID, (around 600 at the moment) and maintaining that mapping in the dialplan would be a mess. AFTER I know what gateway the call arrived on, I have a database for each gateway that helps me process from there. You have cases where the same DID maps differently for one gateway or another? If not, why is the gateway part of the database query? 2) Yes, separate profiles would work, but does sound gross. I'm going to swap my ports around and see if that clears things up... -pete ___ 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
Re: [Freeswitch-users] Playing sound files in a conference
you could hang on the event socket and catch the conference events, then play the sounds via the conference api commands -Ray Rudolf Denert wrote: Hallo everybody! I would like to play soundfiles in a existing conference. The procedure is this: Someone calls the number of the conference. Then this person types the pin in to his phone. The next step is that he has to say the name for example John. This file is saved in a special folder. When he enters the conference room everybody in this existing conference should here: John has enterd the room. If he leaves then everybody should hear: John has left the room. Of course there are two soundfiles. John is what the caller has spoken into his phone and the second one a generated file from me. They should be played in succession. Is it possible to implement this with lua? If yes, how can I do that. Thanks for your help. Greetz ___ 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
Re: [Freeswitch-users] Limit is not working when originate a call
because your not running limit at all when you are doing an originate
directly. You can use loopback to originate through a dialplan
extension.
Mike
On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote:
Hi
I set the limit to 1 on the extension like that
action application=limit_hash data=${destination_number}
${destination_number} 1 /
When I am trying to make a call the that destination i transfered to
limit_exceeded dialplan, just like I want
The problem is, that when I am trying to make a call using
originate I
am not getting the limitation.
Why is that?
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[Freeswitch-users] Good Information On How To Submit Bug Reports
FYI, Brian West called to my attention that one of our community members, John Wehle, has been very good at submitting useful bug reports, in many cases with patches. His style of reporting is worthy of imitation, so I've added a few links to the JIRA section of the Reporting Bugs wiki page: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Examples_Of_Well-Written_JIRA_Submissions Please feel free to check it out. If you have any questions on what a bug report should look like then definitely read some of John's submissions and emulate his style. By submitting useful bug reports you will save the FreeSWITCH developers countless hours and headaches, not to mention the warm fuzzies you'll feel inside. :) -Michael ___ 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
Re: [Freeswitch-users] Confusing handling of incoming calls
0. Probably, gonna have to go get them on the phone.1. No, but the information I need to process is stored in separate databases by gateway. There is no single table that has all of the DIDs and which gateway they belong to. I could create/maintain that table, but if I can determine the gateway before beginning my logic, I can avoid that. Original Message Subject: Re: [Freeswitch-users] Confusing handling of incoming calls From: Rupa Schomaker r...@rupa.com Date: Wed, July 22, 2009 6:04 am To: freeswitch-users@lists.freeswitch.org On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller p...@privateconnect.com wrote: 0) Rupa, you are absolutely right, I forgot that. ports was never an issue because previous gateways all REGISTERed. I will have to swap my ports around as bandwidth is not flexible. You can't tell bandwidth.com to use port 5080? 1) I thought of this, but I have hundreds of DID, (around 600 at the moment) and maintaining that mapping in the dialplan would be a mess. AFTER I know what gateway the call arrived on, I have a database for each gateway that helps me process from there. You have cases where the same DID maps differently for one gateway or another? If not, why is the gateway part of the database query? 2) Yes, separate profiles would work, but does sound gross. I'm going to swap my ports around and see if that clears things up... -pete Original Message Subject: Re: [Freeswitch-users] Confusing handling of incoming calls From: Rupa Schomaker r...@rupa.com Date: Wed, July 22, 2009 2:12 am To: freeswitch-users@lists.freeswitch.org On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller p...@privateconnect.com wrote: My goal is:0) figure out why the bandwidth gateway is being processed as "internal" (this is more of a security thing) they are probably terminating traffic on port 5060 rather than 5080. 5060 is internal, 5080 is external. 1) have both gateways enter at the same point in the dialplan (this seems to be the purpose of the "Extension" param) I'd drop the extension param and instead match on the destination_number (the DID used to reach you). 2) be able to identify which gateway the call came in on. I was hoping to set a param in the gateway configuration that would be passed through onto the channel, but have not found one. Worst case, I could have each gateway enter at a different extension in the dialplan, however, that doesn't seem to be working if the channel comes in the "internal" profile. Not sure here... gateways are an outbound thing. Inbound calls just hit your dialplan and you process from there. A sledgehammer approach would be to have a different sip_profile for each gateway. But that is just silly. Flowroute at least puts their name in the sdp Thanks for your help. I've provided INFO dumps from both gateways if they help...-pete-- -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 ___ 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 -- -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 ___ 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] Asterisk key during message hangs up call
Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars ___ 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
Re: [Freeswitch-users] Asterisk key during message hangs up call
I don't know if this may be related but in voicemail.conf.xml by default the two params that follow are defined: param name=operator-extension value=operator XML default/ param name=operator-key value=9/ And pressing 9 during the greeting does not send me to the operator. I am on trunk rev 14123M On Wed, Jul 22, 2009 at 2:13 PM, Lars Zeb larc...@yahoo.com wrote: Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don’t know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars ___ 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
Re: [Freeswitch-users] Asterisk key during message hangs up call
you are using a channel created with a script and you did not set js session.autoHangup(0) lua session:autoHangup(0) so when the * makes the call transfer the script kills the channel. On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb larc...@yahoo.com wrote: Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don’t know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars ___ 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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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
Re: [Freeswitch-users] Asterisk key during message hangs up call
Do you have anything on that extension? On Jul 22, 2009, at 7:21 PM, Luis F Urrea lfur...@gmail.com wrote: I don't know if this may be related but in voicemail.conf.xml by default the two params that follow are defined: param name=operator-extension value=operator XML default/ param name=operator-key value=9/ And pressing 9 during the greeting does not send me to the operator. I am on trunk rev 14123M On Wed, Jul 22, 2009 at 2:13 PM, Lars Zeb larc...@yahoo.com wrote: Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don’t know if my dialplan is causing the error or so mething in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars ___ 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
Re: [Freeswitch-users] Limit is not working when originate a call
Michael Jerris wrote:
because your not running limit at all when you are doing an originate
directly. You can use loopback to originate through a dialplan
extension.
Mike
On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote:
Hi
I set the limit to 1 on the extension like that
action application=limit_hash data=${destination_number}
${destination_number} 1 /
When I am trying to make a call the that destination i transfered to
limit_exceeded dialplan, just like I want
The problem is, that when I am trying to make a call using
originate I
am not getting the limitation.
Why is that?
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http://www.freeswitch.org
Thanks for answer.
I am calling Originate from JS. I tried to call limit_hash from JS
but with no success. I did it like that:
lmt = apiExecute(limit_hash, dialed_ext + + dialed_ext + 1);
I could't find any documentation on that.
can u help ?
Thanks
Eli
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