Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555@your-server.com and then have a dialplan such as: <extension name="IncomingNexmo"> <condition field="destination_number" expression="^nexmo_(\d+)$"> <action application="info"/> <action application="set" data="nexmo_forwarded_for=$1"/> <action application="lua" data="nexmo_handler.lua"/> </condition> </extension> The nexmo_forwarded_for session variable will now expose to you the dialed Nexmo phone number allowing your application or XML dialplan to use it. Let me know if you are having any other issues. Aviv On Fri, Nov 7, 2014, at 01:05 AM, Manish Talwar wrote:
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 applicati**on="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://loc**alhost: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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