[Freeswitch-users] SIP trunking with Nexmo
Manish Talwar
manish.talwar at nexxuspg.com
Fri Nov 7 11:05:34 MSK 2014
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 at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Aviv Shaham <aviv at sent.com>
Sent: 06 November 2014 14:39
To: freeswitch-users at 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 at 192.168.1.140:5080 RUNNING (0)
external::example.com gateway sip:joeuser at example.com NOREG
external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4 at 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 at 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 at 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
15k (text/plain)
* Nexmo.xml
3k (text/xml)
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