[Freeswitch-users] SIP trunking with Nexmo

Fri Nov 7 19:18:23 MSK 2014

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 at 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 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
>> 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 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://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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>>
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>>
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>> http://www.freeswitch.org Email had 2 attachments:

>>  * FsCall.txt
>>
15k (text/plain)
>>  * Nexmo.xml
>>
3k (text/xml)
>
> ___________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
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