Hi,

Thanks for your suggestion, I have tried it and I am able to do a Inbound call via Nexmo now. But still I am not able to make any outbound call from my application.

I have checked the FreeSwitch log by siptrace enable and found that my call was terminated with a SIP message as "

IP/2.0 407 Proxy Authentication Required".

Please find the siptrace log for my call as an attachment. and let me know what changes or configuration I need to make for Proxy Authentication Header.

Thanks,

Regards,

Manish Talwar

From: freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Aviv Shaham <aviv@sent.com>
Sent: 06 November 2014 14:39
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP trunking with Nexmo

Hi Manish,

Nexmo expects your API KEY to be in the From header. To set the caller ID you will need to use "caller-id-name". Good timing btw, I just posted a reply to a similar question on Quora. Have a look: http://qr.ae/DEbk2 - also covers Plivo.

Aviv

On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote:

Hi,

I have make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below:

Name Type Data State

================================================================================================

external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0)

192.168.1.140 alias internal ALIASED

external profile sip:mod_sofia@192.168.1.140:5080 RUNNING (0)

external::example.com gateway sip:joeuser@example.com NOREG

external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4@sip.nexmo.com REGED

internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0)

internal profile sip:mod_sofia@192.168.1.140:5060 RUNNING (0)

================================================================================================

4 profiles 1 alias

But when I send the request to FreeSwitch by Dial command as:

<document type="xml/freeswitch-httapi"><params></params><work><execute application="set" data="sip_h_api_key=b9c280dd" /><execute application="set" data="sip_h_api_secret=7678b8c4" /><execute application="set" data="sip_h_to=919818753995" /><execute application="set" data="sip_h_from=18188535351 <sip:b9c280dd@sip.nexmo.com>" /><execute application="set" data="sip_h_answer_url=http://services.qpayi.com:8080/ivr/assets/NexmoTransfer.vxml" /><dial name="exten" action="http://localhost:8080/ivr/fsdialmenu/dialendresponse/" caller-id-name="HTTAPI Test"

caller-id-number="18188535351" context="default" Dialplan="XML" >919818753995</dial></work></document>

here, 18188535351 = Nexmo virtual number for connecting call.

919818753995 = mobile number where I am looking for making a call.

It will not connected to Nexmo and call will be terminated with message as:

2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED

Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) as an attachment.

Please let me know whether I am doing SIP trunking in correct way or need to change something.

Also, Please suggest me what will be my next step for making a call on mobile by this ways.

Thanks,

Regards,

Manish Talwar

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Email had 2 attachments:

FsCall.txt
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Nexmo.xml
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