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/$1@sip.nexmo.com:5080"/>
</condition>
</extension>
Please find the attached SipTrace file now and let me know what I need to update now.
In this log, values passed in "From" and "To" attribute as:
From: "18188535351" <sip:b9c280dd@sip.nexmo.com>;tag=D8g4a5NvH4emFTo: <sip:19818753995@sip.nexmo.com:5080>
I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also.
Thanks,
Regards,
Manish Talwar
From: Aviv Shaham <aviv@sent.com>Sent: 07 November 2014 21:48To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP trunking with NexmoHi Manish,Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps.Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call.Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555@your-server.com and then have a dialplan such as:<extension name="IncomingNexmo"><condition field="destination_number" expression="^nexmo_(\d+)$"><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.AvivOn 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 14:39To: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] SIP trunking with NexmoHi 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.AvivOn 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 ALIASEDexternal profile sip:mod_sofia@192.168.1.140:5080 RUNNING (0)external::example.com gateway sip:joeuser@example.com NOREGexternal::sip.nexmo.com gateway sip:b9c280dd:7678b8c4@sip.nexmo.com REGEDinternal-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 aliasBut 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_UNSPECIFIEDPlease 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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