SIP trunking with Nexmo
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 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_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?
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
___________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments:
* FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml)
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 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_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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml)
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
___________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments:
* FsCall.txt
15k (text/plain)
* Nexmo.xml
3k (text/xml)
___________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment:
* SipTrace.txt 9k (text/plain)
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/$1@sip.nexmo.com:5080"/> </condition> </extension> Please find the attached SipTrace file now and let me know what I need to update now. In this log, values passed in "From" and "To" attribute as: From: "18188535351" <sip:b9c280dd@sip.nexmo.com>;tag=D8g4a5NvH4emF To: <sip:19818753995@sip.nexmo.com:5080> I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also. Thanks, Regards, Manish Talwar ________________________________ From: Aviv Shaham <aviv@sent.com> Sent: 07 November 2014 21:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555@your-server.com<mailto:nexmo_12121115555@your-server.com> and then have a dialplan such as: <extension name="IncomingNexmo"> <condition field="destination_number" expression="^nexmo_(\d+)$"> <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 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_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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 9k (text/plain)
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/$1@sip.nexmo.com:5080"/>
*
* </condition>* * </extension>*
Please find the attached SipTrace file now and let me know what I need to update now.
In this log, values passed in "From" and "To" attribute as: **
**
*From: "18188535351" <sip:b9c280dd@sip.nexmo.com>;tag=D8g4a5NvH4emF* * To: <sip:19818753995@sip.nexmo.com:5080>*
I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also.
Thanks,
Regards,
Manish Talwar
*From:* Aviv Shaham <aviv@sent.com> *Sent:* 07 November 2014 21:48 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] SIP trunking with Nexmo
Hi Manish,
Nexmo doesn't seem to handle it well if your first specified codec is L16.
Try to set absolute_codec_string to PCMU and see if that helps.
Also note that there is no need to
include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call.
Since you mentioned also needing inbound
- keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example:
nexmo_12121115555@your-server.com and then have a dialplan such as:
<extension name="IncomingNexmo"> <condition field="destination_number" expression="^nexmo_(\d+)$"> <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 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_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
___________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments:
* FsCall.txt
15k (text/plain)
* Nexmo.xml
3k (text/xml)
___________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment:
* SipTrace.txt
9k (text/plain)
___________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment:
* SipTrace.txt 16k (text/plain)
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@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Aviv Shaham <aviv@sent.com> Sent: 11 November 2014 00:47 To: freeswitch-users@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/$1@sip.nexmo.com:5080"/> </condition> </extension> Please find the attached SipTrace file now and let me know what I need to update now. In this log, values passed in "From" and "To" attribute as: From: "18188535351" <sip:b9c280dd@sip.nexmo.com>;tag=D8g4a5NvH4emF To: <sip:19818753995@sip.nexmo.com:5080> I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also. Thanks, Regards, Manish Talwar ________________________________ From: Aviv Shaham <aviv@sent.com> Sent: 07 November 2014 21:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555@your-server.com<mailto:nexmo_12121115555@your-server.com> and then have a dialplan such as: <extension name="IncomingNexmo"> <condition field="destination_number" expression="^nexmo_(\d+)$"> <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 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_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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 9k (text/plain) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 16k (text/plain)
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@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@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Aviv Shaham <aviv@sent.com> Sent: 11 November 2014 00:47 To: freeswitch-users@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/$1@sip.nexmo.com:5080"/> </condition> </extension> Please find the attached SipTrace file now and let me know what I need to update now. In this log, values passed in "From" and "To" attribute as: From: "18188535351" <sip:b9c280dd@sip.nexmo.com>;tag=D8g4a5NvH4emF To: <sip:19818753995@sip.nexmo.com:5080> I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also. Thanks, Regards, Manish Talwar ________________________________ From: Aviv Shaham <aviv@sent.com> Sent: 07 November 2014 21:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555@your-server.com<mailto:nexmo_12121115555@your-server.com> and then have a dialplan such as: <extension name="IncomingNexmo"> <condition field="destination_number" expression="^nexmo_(\d+)$"> <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 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_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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 9k (text/plain) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@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@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 16k (text/plain)
participants (2)
-
Aviv Shaham
-
Manish Talwar