originating an automated call, default value for origination_caller_id_number
Hi, Thanks for great support so far. I have another two questions to you: 1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. My originate command looks like this: originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/606999888 6668 The '6668' extension is an httapi script that plays an audio file and collects some digits. The problem is that my httapi ivr starts the playback even before the mobile phone is answerred. As a result, the called person doesn't hear the beginning of the recording. Can I somehow make my httapi script wait for the mobile to be answerred? 2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send Remote-Party-ID: <sip:0000000000@sip.ipfon.pl> in INVITE request that goes to the voip gateway. Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options? Thanks RG
Hi, maybe it's an obvious 'discovery' for you, but i think i figured out how to wait for the call to be answerred when executing an outbound, mod_httapi-controlled call: simply use the 'wait_for_answer' application before starting the playback. Here's my current dial script: <document type="text/freeswitch-httapi"> <params> <stage>1</stage> </params> <variables /> <work> <execute application="wait_for_answer" /> <execute application="answer" /> <pause milliseconds="1000" /> <playback file="xyz.wav" name="user_answer" loops="2" digit-timeout="20000"> <bind>~\d</bind> </playback> </work> </document> Of course you need to enable 'wait_for_answer' application in mod_httapi config file. PS but if you know the answer to to my 2nd question please post it here because I'm not really happy having to hard-code the origination_caller_id in my call scripts Best regards RG On Fri, Jul 12, 2013 at 4:06 PM, Rafal Gwizdala <rafal.gwizdala@gmail.com>wrote:
Hi,
Thanks for great support so far. I have another two questions to you: 1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. My originate command looks like this:
originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/ 606999888 6668
The '6668' extension is an httapi script that plays an audio file and collects some digits. The problem is that my httapi ivr starts the playback even before the mobile phone is answerred. As a result, the called person doesn't hear the beginning of the recording. Can I somehow make my httapi script wait for the mobile to be answerred?
2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send Remote-Party-ID: <sip:0000000000@sip.ipfon.pl> in INVITE request that goes to the voip gateway. Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options?
Thanks RG
Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options?
You could try setting effective_caller_id_number through the gateway's outbound variables. http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Variables -Steve On 12 July 2013 15:06, Rafal Gwizdala <rafal.gwizdala@gmail.com> wrote:
Hi,
Thanks for great support so far. I have another two questions to you: 1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. My originate command looks like this:
originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/606999888 6668
The '6668' extension is an httapi script that plays an audio file and collects some digits. The problem is that my httapi ivr starts the playback even before the mobile phone is answerred. As a result, the called person doesn't hear the beginning of the recording. Can I somehow make my httapi script wait for the mobile to be answerred?
2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send Remote-Party-ID: <sip:0000000000@sip.ipfon.pl> in INVITE request that goes to the voip gateway. Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options?
Thanks RG
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Thanks, Steve, unfortunately setting gateway variables doesn't seem to work for me. R On Sat, Jul 13, 2013 at 4:22 PM, Steven Ayre <steveayre@gmail.com> wrote:
Is there a way to specify a default caller id for all outgoing connections
that go through a particular gateway, so I wouldn't have to specify it in originate options?
You could try setting effective_caller_id_number through the gateway's outbound variables. http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Variables
-Steve
On 12 July 2013 15:06, Rafal Gwizdala <rafal.gwizdala@gmail.com> wrote:
Hi,
Thanks for great support so far. I have another two questions to you: 1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. My originate command looks like this:
originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/ 606999888 6668
The '6668' extension is an httapi script that plays an audio file and collects some digits. The problem is that my httapi ivr starts the playback even before the mobile phone is answerred. As a result, the called person doesn't hear the beginning of the recording. Can I somehow make my httapi script wait for the mobile to be answerred?
2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send Remote-Party-ID: <sip:0000000000@sip.ipfon.pl> in INVITE request that goes to the voip gateway. Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options?
Thanks RG
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Rafal Gwizdala
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Steven Ayre