[Freeswitch-users] SIP trunking with Nexmo

Fri Nov 14 09:52:25 MSK 2014

Hi,

I have found that its connecting a outbound call fine but there is no ring tone coming by FreeSwitch via Nexmo even call is ringing on my mobile. Also, when I answered my connected call then there is no voice stream comes and I cant hear anything at both ends.

Please suggest me any configuration if any for getting the ring tone while calling a call and getting a voice at both ends after answering a call.

Thanks,

Regards,
Manish Talwar

________________________________
From: Manish Talwar
Sent: 11 November 2014 14:43
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP trunking with Nexmo

Hi,

Thanks a lot, I am able to make a outbound call by these settings. I have set "absolute_codec_string" as "PCMA,PCMU" and remove "nexmo_forwarded_for" from my dialplan.

I have tried this outbound call by "FsClient" and "Wizton" application. Call is running fine with "FsClient" but unable to received the call by "Wizton". In Wizton, Its tried to call on my mobile with similar logs in "FS_CLI" as calling from "FsClient" but after few seconds call was hangup with message as "Originate Failed.  Cause: ORIGINATOR_CANCEL".

I feel it might be some "codec" configuration problem only, Please find the "FsClient" and "Wizton" log file as an attachment. With, "FsClient" log file I am able to receive the call on my mobile (+919818753995).

Also, when its calling Outbound call by "FsClient" then there was no ring sound came on "FsClient" but call was coming on mobile.

Please suggest me, what I need to do for Ring sound while its calling a mobile and also Is there any other setting required while calling from "Wizton" or any other medium (like mobile, phone etc).

Thanks a lot.

Regards,

Manish Talwar

Also,

________________________________
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: 11 November 2014 00:47
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP trunking with Nexmo

Hi Manish,

First, no need to set nexmo_forwarded_for for outbound here, if you re-read my previous email you'll see that it was meant to be used for Nexmo DIDs you use to direct origination traffic into FS.

As for the error you are getting with this dialplan, you need to remove "@sip.nexmo.com:5080" from your origination string.

Hope it helps,

Aviv

On Fri, Nov 7, 2014, at 11:10 PM, Manish Talwar wrote:

Hi,

Thanks for your suggestion, I have make these changes and removed the L16 codec from request now. I have set "absolute_codec_string" and "nexmo_forwarded_for" and its not throwing any error message in SIP trace now.

But still, I am not able to make a call on my mobile number "1919818753995". Its show message on FreeSwitch log as "[RECOVERY_ON_TIMER_EXPIRE]" and hangup the freeswitch call. Also, there is no log created on Nexmo dashboard for this call's.

I am sending my call request to Nexmo from FreeSwitch by dialplan as.

<extension name="Dial through Nexmo">
         <condition field="destination_number" expression="^19(1\d{10})$">
                <action application="set" data="absolute_codec_string=PCMU,GSM"/>
                <action application="set" data="nexmo_forwarded_for=$1"/>
               <action application="bridge" data="{origination_caller_id_name='18188535351',ignore_early_media=true}sofia/gateway/nexmo/$1 at sip.nexmo.com:5080"/>
         </condition>
 </extension>

Please find the attached SipTrace file now and let me know what I need to update now.

In this log, values passed in "From" and "To" attribute as:

From: "18188535351" <sip:b9c280dd at sip.nexmo.com>;tag=D8g4a5NvH4emF
   To: <sip:19818753995 at sip.nexmo.com:5080>

I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also.

Thanks,

Regards,

Manish Talwar

________________________________

From: Aviv Shaham <aviv at sent.com>
Sent: 07 November 2014 21:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP trunking with Nexmo

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<mailto: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 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)

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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Email had 1 attachment:

  *   SipTrace.txt
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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