I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider - "State=REGED". I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of "SIP/2.0 480 Temporarily Unavailable". Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks!
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com
wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
_________________________________________________________________________ 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
Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com
wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
_________________________________________________________________________ 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
_________________________________________________________________________ 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
Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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
http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com
wrote:
Chris Tunbridge,
1) I made the updates to my configuration, as suggested in the “ https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.
2) Extension x9161 is one of the default dialplan applications.
3) Call failure log posted at: http://pastebin.com/E4sqTLa4
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Saturday, December 27, 2014 2:30 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml
2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.
3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.
4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.
On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- DVG -- Imagination is more important than knowledge Albert Einstein
New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein
can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
New “pastebin” created:
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 4:04 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it."
On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge,
1) I made the updates to my configuration, as suggested in the “ https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.
2) Extension x9161 is one of the default dialplan applications.
3) Call failure log posted at: http://pastebin.com/E4sqTLa4
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Saturday, December 27, 2014 2:30 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml
2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.
3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.
4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.
On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
--
DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
-- DVG -- Imagination is more important than knowledge Albert Einstein
The full output from the “xml_locate dialplan” command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip.us"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension. Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein
Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: <extension name="o <http://switch2voip.us/>utside_calls"> <condition field="destination_number" expression="^(404.*)$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
The full output from the “xml_locate dialplan” command is already in the previously pasted logfile.
Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us:
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip.us">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension.
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 5:05 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
can you share your dialplan? It looks like you're dialing
"To: sip:4049392032@172.31.33.109"
but have no extension for that...
On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
New “pastebin” created:
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 4:04 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it."
On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge,
1) I made the updates to my configuration, as suggested in the “ https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.
2) Extension x9161 is one of the default dialplan applications.
3) Call failure log posted at: http://pastebin.com/E4sqTLa4
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Saturday, December 27, 2014 2:30 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml
2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.
3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.
4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.
On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
--
DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
--
DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
-- DVG -- Imagination is more important than knowledge Albert Einstein
George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. <http://switch2voip.us> On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea < david.villasmil@gmail.com> wrote:
Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like:
<extension name="o <http://switch2voip.us/>utside_calls">
<condition field="destination_number" expression="^(404.*)$">
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
The full output from the “xml_locate dialplan” command is already in the previously pasted logfile.
Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us:
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip.us">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension.
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 5:05 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
can you share your dialplan? It looks like you're dialing
"To: sip:4049392032@172.31.33.109"
but have no extension for that...
On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
New “pastebin” created:
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 4:04 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it."
On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge,
1) I made the updates to my configuration, as suggested in the “ https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.
2) Extension x9161 is one of the default dialplan applications.
3) Call failure log posted at: http://pastebin.com/E4sqTLa4
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Saturday, December 27, 2014 2:30 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml
2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.
3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.
4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.
On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
--
DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
--
DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
-- DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@50.160.141.159:13130> To: "George Phelps"<sip:17708410143@172.31.33.109> From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil@gmail.com> wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: <extension name="o <http://switch2voip.us/> utside_calls"> <condition field="destination_number" expression="^(404.*)$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: The full output from the “xml_locate dialplan” command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip.us"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension. Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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
Follow on… It appears, to me, that my outbound call is being processed in the “public” context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" <sip:1001@172.31.33.109>;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032@172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@50.160.141.159:13130> To: "George Phelps"<sip:17708410143@172.31.33.109> From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil@gmail.com> wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: <extension name="o <http://switch2voip.us/> utside_calls"> <condition field="destination_number" expression="^(404.*)$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: The full output from the “xml_locate dialplan” command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip.us"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension. Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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
Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps <GeorgePhelps@gfphelps.com
wrote:
Follow on…
It appears, to me, that my outbound call is being processed in the “public” context:
2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public
Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context.
(Trace segment below.)
Thanks,
George
send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.160.141.159:48815 ;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b
From: "George F Phelps" <sip:1001@172.31.33.109
;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U
To: sip:4049392032@172.31.33.109
Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ
CSeq: 10680 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit
Content-Length: 0
------------------------------------------------------------------------
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680)
nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying
nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called
nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public
2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan instruction, hanging up.
2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING]
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *George F. Phelps *Sent:* Monday, December 29, 2014 7:21 AM
*To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Chris Tunbridge,
I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension?
recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167:
------------------------------------------------------------------------
INVITE sip:17708410143@172.31.33.109 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:13130 ;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1000@50.160.141.159:13130>
To: "George Phelps"<sip:17708410143@172.31.33.109>
From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149
Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria 3 release 3.5.5 stamp 71238
Content-Length: 264
v=0
o=- 13064325826971649 1 IN IP4 192.168.1.100
s=Bria 3 release 3.5.5 stamp 71238
c=IN IP4 192.168.1.100
t=0 0
m=audio 50404 RTP/AVP 9 8 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
------------------------------------------------------------------------
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Chris Tunbridge *Sent:* Sunday, December 28, 2014 11:07 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
George, can you move your /usr/local/freeswitch/conf/dialplan/ switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder?
The main folder (not /default/) is used for context's, so it wouldn't get included in your default context.
On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea < david.villasmil@gmail.com> wrote:
Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like:
<extension name="o <http://switch2voip.us/>utside_calls">
<condition field="destination_number" expression="^(404.*)$">
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
The full output from the “xml_locate dialplan” command is already in the previously pasted logfile.
Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us:
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip.us">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension.
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 5:05 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
can you share your dialplan? It looks like you're dialing
"To: sip:4049392032@172.31.33.109"
but have no extension for that...
On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
New “pastebin” created:
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 4:04 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it."
On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge,
1) I made the updates to my configuration, as suggested in the “ https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.
2) Extension x9161 is one of the default dialplan applications.
3) Call failure log posted at: http://pastebin.com/E4sqTLa4
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Saturday, December 27, 2014 2:30 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml
2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.
3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.
4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.
On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ 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
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DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
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--
DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
--
DVG
-- Imagination is more important than knowledge Albert Einstein
_________________________________________________________________________ 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
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_________________________________________________________________________ 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
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Brian West, Okay, I’m sure there is an answer/solution there, but it’s over my head… Questions How do I check to see if I have inadvertently disabled “auth”? I am 99% sure that I have do changed it. I am 100% sure that I have not touched anything to do with “allow acl”. But how do I check? So are you saying that dialing an outside in the in the “public” context is correct? Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 10:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Follow on… It appears, to me, that my outbound call is being processed in the “public” context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" <sip:1001@172.31.33.109 <mailto:sip%3A1001@172.31.33.109> >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032@172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@50.160.141.159:13130> To: "George Phelps"<sip:17708410143@172.31.33.109> From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil@gmail.com> wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: <extension name="o <http://switch2voip.us/> utside_calls"> <condition field="destination_number" expression="^(404.*)$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: The full output from the “xml_locate dialplan” command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip.us"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension. Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- Brian West brian@freeswitch.org <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Brian West,
Okay, I’m sure there is an answer/solution there, but it’s over my head…
*Questions*
How do I check to see if I have inadvertently disabled “auth”? I am 99% sure that I have do changed it.
I am 100% sure that I have not touched anything to do with “allow acl”. But how do I check?
So are you saying that dialing an outside in the in the “public” context is correct?
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* Tuesday, December 30, 2014 10:32 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth.
On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Follow on…
It appears, to me, that my outbound call is being processed in the “public” context:
2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public
Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context.
(Trace segment below.)
Thanks,
George
send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.160.141.159:48815 ;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b
From: "George F Phelps" <sip:1001@172.31.33.109
;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U
To: sip:4049392032@172.31.33.109
Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ
CSeq: 10680 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit
Content-Length: 0
------------------------------------------------------------------------
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680)
nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying
nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called
nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public
2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan instruction, hanging up.
2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING]
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *George F. Phelps *Sent:* Monday, December 29, 2014 7:21 AM
*To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Chris Tunbridge,
I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension?
recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167:
------------------------------------------------------------------------
INVITE sip:17708410143@172.31.33.109 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:13130 ;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1000@50.160.141.159:13130>
To: "George Phelps"<sip:17708410143@172.31.33.109>
From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149
Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria 3 release 3.5.5 stamp 71238
Content-Length: 264
v=0
o=- 13064325826971649 1 IN IP4 192.168.1.100
s=Bria 3 release 3.5.5 stamp 71238
c=IN IP4 192.168.1.100
t=0 0
m=audio 50404 RTP/AVP 9 8 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
------------------------------------------------------------------------
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Chris Tunbridge *Sent:* Sunday, December 28, 2014 11:07 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
George, can you move your /usr/local/freeswitch/conf/dialplan/ switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder?
The main folder (not /default/) is used for context's, so it wouldn't get included in your default context.
On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea < david.villasmil@gmail.com> wrote:
Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like:
<extension name="o <http://switch2voip.us/>utside_calls">
<condition field="destination_number" expression="^(404.*)$">
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
The full output from the “xml_locate dialplan” command is already in the previously pasted logfile.
Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us:
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip.us">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension.
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 5:05 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
can you share your dialplan? It looks like you're dialing
"To: sip:4049392032@172.31.33.109"
but have no extension for that...
On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
New “pastebin” created:
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 4:04 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it."
On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge,
1) I made the updates to my configuration, as suggested in the “ https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.
2) Extension x9161 is one of the default dialplan applications.
3) Call failure log posted at: http://pastebin.com/E4sqTLa4
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Saturday, December 27, 2014 2:30 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml
2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.
3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.
4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.
On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
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DVG
-- Imagination is more important than knowledge Albert Einstein
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DVG
-- Imagination is more important than knowledge Albert Einstein
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DVG
-- Imagination is more important than knowledge Albert Einstein
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*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Brian West, I wiped out my configuration and started over. I am able to dial from one extension to another — with 2-way audio. However, I am still not able to dial an external phone number. I also cannot dial the example extensions, x9196, x5000, etc. I am able (via IPv6) to register with my VoIP gateway: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::switch2voip.us gateway sip:9221591504@66.33.147.150 REGED external profile sip:mod_sofia@54.174.255.168:5080 RUNNING (0) external::switch2voip.us gateway sip:9221591504@66.33.147.150 FAIL_WAIT 172.31.33.109 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@54.174.255.168:5060 RUNNING (0) ================================================================================================= 4 profiles 1 alias A failed dial attempt is shown below. Freeswitch is apparently still attempting to route the call to a non-existent local extension (sip:14049392032@172.31.33.109) instead of my VoIP provider. recv 880 bytes from udp/[50.160.141.159]:49334 at 15:57:15.095108: ------------------------------------------------------------------------ INVITE sip:14049392032@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 50.160.141.159:49334;rport;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032@172.31.33.109 Contact: "George F Phelps" <sip:1001@50.160.141.159:49334;ob> Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria Android 3.2.1 Content-Type: application/sdp Content-Length: 247 v=0 o=- 3629048232 3629048232 IN IP4 50.160.141.159 s=cpc_med c=IN IP4 50.160.141.159 t=0 0 m=audio 4000 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 406 bytes to udp/[50.160.141.159]:49334 at 15:57:15.095386: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032@172.31.33.109 Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Content-Length: 0 ------------------------------------------------------------------------ send 802 bytes to udp/[50.160.141.159]:49334 at 15:57:15.097601: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: <sip:14049392032@172.31.33.109>;tag=ctKjQv8vKyZtB Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "14049392032" <sip:14049392032@172.31.33.109>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ My outbound dialplan “/usr/local/freeswitch/conf/dialplan/default/00_outbound_calls.xml”: <include> <extension name="outbound_calls"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> </include> My gateway “/usr/local/freeswitch/conf/directory/default/example.com.xml”: <include> <user id="$${default_provider}"> <gateways> <gateway name="$${default_provider}"> <param name="username" value="$${default_provider_username}"/> <param name="password" value="$${default_provider_password}"/> <param name="from-user" value="$${default_provider_username}"/> <param name="from-domain" value="$${default_provider_from_domain}"/> <param name="realm" value="$${default_provider_from_domain}"/> <param name="proxy" value="66.33.147.150"/> <param name="expire-seconds" value="600"/> <param name="register" value="$${default_provider_register}"/> <param name="retry-seconds" value="30"/> <param name="extension" value="$${default_provider_contact}"/> <!--<param name="contact-params" value="domain_name=$${domain}"/>--> <param name="context" value="public"/> </gateway> </gateways> <params> <param name="password" value="$${default_provider_password}"/> </params> </user> </include> And in “/usr/local/freeswitch/conf/vars.xml”: <X-PRE-PROCESS cmd="set" data="default_provider=switch2voip.us"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=9221591504"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=XXXXXXXX"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=switch2voip.us"/> <!-- true or false --> <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider (66.33.147.150). I did make the additional configuration updates as documented here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 1:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Brian West, Okay, I’m sure there is an answer/solution there, but it’s over my head… Questions How do I check to see if I have inadvertently disabled “auth”? I am 99% sure that I have do changed it. I am 100% sure that I have not touched anything to do with “allow acl”. But how do I check? So are you saying that dialing an outside in the in the “public” context is correct? Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 10:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Follow on… It appears, to me, that my outbound call is being processed in the “public” context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" <sip:1001@172.31.33.109 <mailto:sip%3A1001@172.31.33.109> >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032@172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@50.160.141.159:13130> To: "George Phelps"<sip:17708410143@172.31.33.109> From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil@gmail.com> wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: <extension name="o <http://switch2voip.us/> utside_calls"> <condition field="destination_number" expression="^(404.*)$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: The full output from the “xml_locate dialplan” command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip.us"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension. Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- Brian West brian@freeswitch.org <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ 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 -- Brian West brian@freeswitch.org <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
I added a debug log message to my dialplan: <include> <extension name="outbound_calls"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <action application="log" data="DIALING via switch2voip.us"/> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> </include> The log message is not showing up when I dial, as an example, “14049392032”. So I assume that some other section of the dialplan is handling my outbound call? How can I determine what section is handling the call, without manually adding multiple, additional log messages? I pasted my current “xml_locate dialplan” output here: http://pastebin.com/BRB5Lpmg Extracted “condition field="destination_number"” lines from the pastebin: <action application="deflect" data="${destination_number}"></action> <condition field="destination_number" expression="^886$"> <condition field="destination_number" expression="^\*8$"> <condition field="destination_number" expression="^\*\*(\d+)$"> <condition field="destination_number" expression="^(redial|870)$"> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></action> <condition field="destination_number" expression="^9001$"> <condition field="destination_number" expression="^9000$"> <condition field="destination_number" expression="^88(\d{4})$|^\*0(.*)$"> <condition field="destination_number" expression="^779$"> <condition field="destination_number" expression="^\*69$|^869$|^lcr$"> <condition field="destination_number" expression="^80(\d{2})$"> <condition field="destination_number" expression="^81(\d{2})$"> <condition field="destination_number" expression="^82(\d{2})$"> <condition field="destination_number" expression="^83(\d{2})$"> <condition field="destination_number" expression="^8(10[01][0-9])$"> <condition field="destination_number" expression="^(10[01][0-9])$"> <condition field="destination_number" expression="^(11[01][0-9])$"> <action application="bridge" data="skinny/internal/${destination_number}"></action> <condition field="destination_number" expression="^2000$"> <condition field="destination_number" expression="^2001$"> <condition field="destination_number" expression="^2002$"> <condition field="destination_number" expression="^(operator|0)$"> <condition field="destination_number" expression="^vmain$|^4000$|^\*98$"> <condition field="destination_number" expression="^sip:(.*)$"> <condition field="destination_number" expression="^(30\d{2})$"> <condition field="destination_number" expression="^(31\d{2})$"> <condition field="destination_number" expression="^(32\d{2})$"> <condition field="destination_number" expression="^(33\d{2})$"> <condition field="destination_number" expression="^9(888|8888|1616|3232)$"> <condition field="destination_number" expression="^0911$"> <condition field="destination_number" expression="^0912$"> <condition field="destination_number" expression="^0913$"> <condition field="destination_number" expression="^5000$"> <condition field="destination_number" expression="^5001$"> <condition field="destination_number" expression="^pagegroup$|^7243$"> <condition field="destination_number" expression="^5900$"> <condition field="destination_number" expression="^5901$"> <condition field="destination_number" expression="^(6000)$"> <condition field="destination_number" expression="^((?!6000)60\d{2})$"> <condition field="destination_number" expression="park\+(\d+)"> <condition field="destination_number" expression="^parking$"></condition> <condition field="destination_number" expression="callpark"></condition> <condition field="destination_number" expression="pickup"></condition> <condition field="destination_number" expression="^wait$"> <condition field="destination_number" expression="^9178$"> <condition field="destination_number" expression="^9179$"> <condition field="destination_number" expression="^9180$"> <condition field="destination_number" expression="^9181$"> <condition field="destination_number" expression="^9182$"> <condition field="destination_number" expression="^9183$"> <condition field="destination_number" expression="^9184$"> <condition field="destination_number" expression="^9191$"> <condition field="destination_number" expression="^9192$"> <condition field="destination_number" expression="^9193$"> <condition field="destination_number" expression="^9194$"> <condition field="destination_number" expression="^9195$"> <condition field="destination_number" expression="^9196$"> <condition field="destination_number" expression="^9197$"> <condition field="destination_number" expression="^9198$"> <condition field="destination_number" expression="^9787$"> <condition field="destination_number" expression="^9664$"></condition> <condition field="destination_number" expression="^9386$"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <condition field="destination_number" expression="^9170$"> <condition field="destination_number" expression="^9171$"> <condition field="destination_number" expression="^9172$"> <condition field="destination_number" expression="^(\d{7})$"> <condition field="destination_number" expression="^(\d{11})$"> <condition field="destination_number" expression="^(011\d+)$"> <condition field="destination_number" expression="^101$"> <condition field="destination_number" expression="^(pizza|74992)$"></condition> <condition field="destination_number" expression="^(.*)$"> <condition field="destination_number" expression="^dx$"> <condition field="destination_number" expression="^att_xfer$"> <condition field="destination_number" expression="^is_transfer$"></condition> <condition field="destination_number" expression="^cf$"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="deflect" data="${destination_number}"></action> <condition field="destination_number" expression="^(10[01][0-9])$"> <condition field="destination_number" expression="^(5551212)$"> <condition field="destination_number" expression="^(9[0-9]{3})$"> <condition field="destination_number" expression="^(10[01][0-9])$"> <condition field="destination_number" expression="^(11[01][0-9])$"> <condition field="destination_number" expression="^(0[0-9]{10,})$"> <condition field="destination_number" expression="^(.{14,})$"> Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Wednesday, December 31, 2014 4:37 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Brian West, I wiped out my configuration and started over. I am able to dial from one extension to another — with 2-way audio. However, I am still not able to dial an external phone number. I also cannot dial the example extensions, x9196, x5000, etc. I am able (via IPv6) to register with my VoIP gateway: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::switch2voip.us gateway sip:9221591504@66.33.147.150 REGED external profile sip:mod_sofia@54.174.255.168:5080 RUNNING (0) external::switch2voip.us gateway sip:9221591504@66.33.147.150 FAIL_WAIT 172.31.33.109 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@54.174.255.168:5060 RUNNING (0) ================================================================================================= 4 profiles 1 alias A failed dial attempt is shown below. Freeswitch is apparently still attempting to route the call to a non-existent local extension (sip:14049392032@172.31.33.109) instead of my VoIP provider. recv 880 bytes from udp/[50.160.141.159]:49334 at 15:57:15.095108: ------------------------------------------------------------------------ INVITE sip:14049392032@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 50.160.141.159:49334;rport;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032@172.31.33.109 Contact: "George F Phelps" <sip:1001@50.160.141.159:49334;ob> Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria Android 3.2.1 Content-Type: application/sdp Content-Length: 247 v=0 o=- 3629048232 3629048232 IN IP4 50.160.141.159 s=cpc_med c=IN IP4 50.160.141.159 t=0 0 m=audio 4000 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 406 bytes to udp/[50.160.141.159]:49334 at 15:57:15.095386: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032@172.31.33.109 Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Content-Length: 0 ------------------------------------------------------------------------ send 802 bytes to udp/[50.160.141.159]:49334 at 15:57:15.097601: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: <sip:14049392032@172.31.33.109>;tag=ctKjQv8vKyZtB Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "14049392032" <sip:14049392032@172.31.33.109>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ My outbound dialplan “/usr/local/freeswitch/conf/dialplan/default/00_outbound_calls.xml”: <include> <extension name="outbound_calls"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> </include> My gateway “/usr/local/freeswitch/conf/directory/default/example.com.xml”: <include> <user id="$${default_provider}"> <gateways> <gateway name="$${default_provider}"> <param name="username" value="$${default_provider_username}"/> <param name="password" value="$${default_provider_password}"/> <param name="from-user" value="$${default_provider_username}"/> <param name="from-domain" value="$${default_provider_from_domain}"/> <param name="realm" value="$${default_provider_from_domain}"/> <param name="proxy" value="66.33.147.150"/> <param name="expire-seconds" value="600"/> <param name="register" value="$${default_provider_register}"/> <param name="retry-seconds" value="30"/> <param name="extension" value="$${default_provider_contact}"/> <!--<param name="contact-params" value="domain_name=$${domain}"/>--> <param name="context" value="public"/> </gateway> </gateways> <params> <param name="password" value="$${default_provider_password}"/> </params> </user> </include> And in “/usr/local/freeswitch/conf/vars.xml”: <X-PRE-PROCESS cmd="set" data="default_provider=switch2voip.us"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=9221591504"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=XXXXXXXX"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=switch2voip.us"/> <!-- true or false --> <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider (66.33.147.150). I did make the additional configuration updates as documented here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 1:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Brian West, Okay, I’m sure there is an answer/solution there, but it’s over my head… Questions How do I check to see if I have inadvertently disabled “auth”? I am 99% sure that I have do changed it. I am 100% sure that I have not touched anything to do with “allow acl”. But how do I check? So are you saying that dialing an outside in the in the “public” context is correct? Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 10:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Follow on… It appears, to me, that my outbound call is being processed in the “public” context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" <sip:1001@172.31.33.109 <mailto:sip%3A1001@172.31.33.109> >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032@172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@50.160.141.159:13130> To: "George Phelps"<sip:17708410143@172.31.33.109> From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil@gmail.com> wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: <extension name="o <http://switch2voip.us/> utside_calls"> <condition field="destination_number" expression="^(404.*)$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: The full output from the “xml_locate dialplan” command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip.us"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension. Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- Brian West brian@freeswitch.org <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ 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 -- Brian West 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Additional “freeswitch.log” debug output is included below, for a failed external call. According to the debug output, the call attempt is indeed being made in the “public” context. My outbound dialplan is located in the “default” context, so apparently no “action” match is found. Questions 1) Should my dialplan be located in the “public” or “default” context? 2) Should my call attempt be taking place in the “public” or “default” context? 3) What (how?) do I need to change to have both the dialplan and the dial attempt in the same (correct) context? For reference, I pasted my current “xml_locate dialplan” output here: http://pastebin.com/BRB5Lpmg And this is my Freeswitch version information: FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit) Thanks, George ---- 2015-01-02 09:44:19.406789 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->14049392032 in context public Dialplan: sofia/external/1001@172.31.33.109 parsing [public->unloop] continue=false Dialplan: sofia/external/1001@172.31.33.109 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1001@172.31.33.109 parsing [public->outside_call] continue=true Dialplan: sofia/external/1001@172.31.33.109 Absolute Condition [outside_call] Dialplan: sofia/external/1001@172.31.33.109 Action set(outside_call=true) Dialplan: sofia/external/1001@172.31.33.109 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/1001@172.31.33.109 parsing [public->call_debug] continue=true Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/1001@172.31.33.109 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [public_extensions] destination_number(14049392032) =~ /^(10[01][0-9])$/ break=\ Dialplan: sofia/external/1001@172.31.33.109 parsing [public->public_did] continue=false Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [public_did] destination_number(14049392032) =~ /^(5551212)$/ break=on-false 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:216 (sofia/external/1001@172.31.33.109) State Change CS_ROUTING -> CS_EXE\ 2015-01-02 09:44:19.406789 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1001@172.31.33.109 [BREAK] 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1001@172.31.33.109) State ROUTING going to sleep 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1001@172.31.33.109) Running State Change CS_EXECUTE 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1001@172.31.33.109) State EXECUTE 2015-01-02 09:44:19.406789 [DEBUG] mod_sofia.c:178 sofia/external/1001@172.31.33.109 SOFIA EXECUTE 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:258 sofia/external/1001@172.31.33.109 Standard EXECUTE EXECUTE sofia/external/1001@172.31.33.109 set(outside_call=true) 2015-01-02 09:44:19.406789 [DEBUG] mod_dptools.c:1435 sofia/external/1001@172.31.33.109 SET [outside_call]=[true] EXECUTE sofia/external/1001@172.31.33.109 export(RFC2822_DATE=Fri, 02 Jan 2015 09:44:19 -0500) 2015-01-02 09:44:19.406789 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 02 Jan 2015 09:44:19 -0500] 2015-01-02 09:44:19.406789 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan ins\ 2015-01-02 09:44:19.406789 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARI\ 2015-01-02 09:44:19.406789 [DEBUG] switch_channel.c:3222 Send signal sofia/external/1001@172.31.33.109 [KILL] 2015-01-02 09:44:19.406789 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1001@172.31.33.109 [BREAK] 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1001@172.31.33.109) State EXECUTE going to sleep ---- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 01, 2015 8:13 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable I added a debug log message to my dialplan: <include> <extension name="outbound_calls"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <action application="log" data="DIALING via switch2voip.us"/> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> </include> The log message is not showing up when I dial, as an example, “14049392032”. So I assume that some other section of the dialplan is handling my outbound call? How can I determine what section is handling the call, without manually adding multiple, additional log messages? I pasted my current “xml_locate dialplan” output here: http://pastebin.com/BRB5Lpmg Extracted “condition field="destination_number"” lines from the pastebin: <action application="deflect" data="${destination_number}"></action> <condition field="destination_number" expression="^886$"> <condition field="destination_number" expression="^\*8$"> <condition field="destination_number" expression="^\*\*(\d+)$"> <condition field="destination_number" expression="^(redial|870)$"> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></action> <condition field="destination_number" expression="^9001$"> <condition field="destination_number" expression="^9000$"> <condition field="destination_number" expression="^88(\d{4})$|^\*0(.*)$"> <condition field="destination_number" expression="^779$"> <condition field="destination_number" expression="^\*69$|^869$|^lcr$"> <condition field="destination_number" expression="^80(\d{2})$"> <condition field="destination_number" expression="^81(\d{2})$"> <condition field="destination_number" expression="^82(\d{2})$"> <condition field="destination_number" expression="^83(\d{2})$"> <condition field="destination_number" expression="^8(10[01][0-9])$"> <condition field="destination_number" expression="^(10[01][0-9])$"> <condition field="destination_number" expression="^(11[01][0-9])$"> <action application="bridge" data="skinny/internal/${destination_number}"></action> <condition field="destination_number" expression="^2000$"> <condition field="destination_number" expression="^2001$"> <condition field="destination_number" expression="^2002$"> <condition field="destination_number" expression="^(operator|0)$"> <condition field="destination_number" expression="^vmain$|^4000$|^\*98$"> <condition field="destination_number" expression="^sip:(.*)$"> <condition field="destination_number" expression="^(30\d{2})$"> <condition field="destination_number" expression="^(31\d{2})$"> <condition field="destination_number" expression="^(32\d{2})$"> <condition field="destination_number" expression="^(33\d{2})$"> <condition field="destination_number" expression="^9(888|8888|1616|3232)$"> <condition field="destination_number" expression="^0911$"> <condition field="destination_number" expression="^0912$"> <condition field="destination_number" expression="^0913$"> <condition field="destination_number" expression="^5000$"> <condition field="destination_number" expression="^5001$"> <condition field="destination_number" expression="^pagegroup$|^7243$"> <condition field="destination_number" expression="^5900$"> <condition field="destination_number" expression="^5901$"> <condition field="destination_number" expression="^(6000)$"> <condition field="destination_number" expression="^((?!6000)60\d{2})$"> <condition field="destination_number" expression="park\+(\d+)"> <condition field="destination_number" expression="^parking$"></condition> <condition field="destination_number" expression="callpark"></condition> <condition field="destination_number" expression="pickup"></condition> <condition field="destination_number" expression="^wait$"> <condition field="destination_number" expression="^9178$"> <condition field="destination_number" expression="^9179$"> <condition field="destination_number" expression="^9180$"> <condition field="destination_number" expression="^9181$"> <condition field="destination_number" expression="^9182$"> <condition field="destination_number" expression="^9183$"> <condition field="destination_number" expression="^9184$"> <condition field="destination_number" expression="^9191$"> <condition field="destination_number" expression="^9192$"> <condition field="destination_number" expression="^9193$"> <condition field="destination_number" expression="^9194$"> <condition field="destination_number" expression="^9195$"> <condition field="destination_number" expression="^9196$"> <condition field="destination_number" expression="^9197$"> <condition field="destination_number" expression="^9198$"> <condition field="destination_number" expression="^9787$"> <condition field="destination_number" expression="^9664$"></condition> <condition field="destination_number" expression="^9386$"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <condition field="destination_number" expression="^9170$"> <condition field="destination_number" expression="^9171$"> <condition field="destination_number" expression="^9172$"> <condition field="destination_number" expression="^(\d{7})$"> <condition field="destination_number" expression="^(\d{11})$"> <condition field="destination_number" expression="^(011\d+)$"> <condition field="destination_number" expression="^101$"> <condition field="destination_number" expression="^(pizza|74992)$"></condition> <condition field="destination_number" expression="^(.*)$"> <condition field="destination_number" expression="^dx$"> <condition field="destination_number" expression="^att_xfer$"> <condition field="destination_number" expression="^is_transfer$"></condition> <condition field="destination_number" expression="^cf$"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="deflect" data="${destination_number}"></action> <condition field="destination_number" expression="^(10[01][0-9])$"> <condition field="destination_number" expression="^(5551212)$"> <condition field="destination_number" expression="^(9[0-9]{3})$"> <condition field="destination_number" expression="^(10[01][0-9])$"> <condition field="destination_number" expression="^(11[01][0-9])$"> <condition field="destination_number" expression="^(0[0-9]{10,})$"> <condition field="destination_number" expression="^(.{14,})$"> Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Wednesday, December 31, 2014 4:37 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Brian West, I wiped out my configuration and started over. I am able to dial from one extension to another — with 2-way audio. However, I am still not able to dial an external phone number. I also cannot dial the example extensions, x9196, x5000, etc. I am able (via IPv6) to register with my VoIP gateway: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::switch2voip.us gateway sip:9221591504@66.33.147.150 REGED external profile sip:mod_sofia@54.174.255.168:5080 RUNNING (0) external::switch2voip.us gateway sip:9221591504@66.33.147.150 FAIL_WAIT 172.31.33.109 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@54.174.255.168:5060 RUNNING (0) ================================================================================================= 4 profiles 1 alias A failed dial attempt is shown below. Freeswitch is apparently still attempting to route the call to a non-existent local extension (sip:14049392032@172.31.33.109) instead of my VoIP provider. recv 880 bytes from udp/[50.160.141.159]:49334 at 15:57:15.095108: ------------------------------------------------------------------------ INVITE sip:14049392032@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 50.160.141.159:49334;rport;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032@172.31.33.109 Contact: "George F Phelps" <sip:1001@50.160.141.159:49334;ob> Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria Android 3.2.1 Content-Type: application/sdp Content-Length: 247 v=0 o=- 3629048232 3629048232 IN IP4 50.160.141.159 s=cpc_med c=IN IP4 50.160.141.159 t=0 0 m=audio 4000 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 406 bytes to udp/[50.160.141.159]:49334 at 15:57:15.095386: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032@172.31.33.109 Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Content-Length: 0 ------------------------------------------------------------------------ send 802 bytes to udp/[50.160.141.159]:49334 at 15:57:15.097601: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: <sip:14049392032@172.31.33.109>;tag=ctKjQv8vKyZtB Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "14049392032" <sip:14049392032@172.31.33.109>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ My outbound dialplan “/usr/local/freeswitch/conf/dialplan/default/00_outbound_calls.xml”: <include> <extension name="outbound_calls"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> </include> My gateway “/usr/local/freeswitch/conf/directory/default/example.com.xml”: <include> <user id="$${default_provider}"> <gateways> <gateway name="$${default_provider}"> <param name="username" value="$${default_provider_username}"/> <param name="password" value="$${default_provider_password}"/> <param name="from-user" value="$${default_provider_username}"/> <param name="from-domain" value="$${default_provider_from_domain}"/> <param name="realm" value="$${default_provider_from_domain}"/> <param name="proxy" value="66.33.147.150"/> <param name="expire-seconds" value="600"/> <param name="register" value="$${default_provider_register}"/> <param name="retry-seconds" value="30"/> <param name="extension" value="$${default_provider_contact}"/> <!--<param name="contact-params" value="domain_name=$${domain}"/>--> <param name="context" value="public"/> </gateway> </gateways> <params> <param name="password" value="$${default_provider_password}"/> </params> </user> </include> And in “/usr/local/freeswitch/conf/vars.xml”: <X-PRE-PROCESS cmd="set" data="default_provider=switch2voip.us"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=9221591504"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=XXXXXXXX"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=switch2voip.us"/> <!-- true or false --> <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider (66.33.147.150). I did make the additional configuration updates as documented here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 1:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Brian West, Okay, I’m sure there is an answer/solution there, but it’s over my head… Questions How do I check to see if I have inadvertently disabled “auth”? I am 99% sure that I have do changed it. I am 100% sure that I have not touched anything to do with “allow acl”. But how do I check? So are you saying that dialing an outside in the in the “public” context is correct? Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 10:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Follow on… It appears, to me, that my outbound call is being processed in the “public” context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" <sip:1001@172.31.33.109 <mailto:sip%3A1001@172.31.33.109> >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032@172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143@172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@50.160.141.159:13130> To: "George Phelps"<sip:17708410143@172.31.33.109> From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil@gmail.com> wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: <extension name="o <http://switch2voip.us/> utside_calls"> <condition field="destination_number" expression="^(404.*)$"> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: The full output from the “xml_locate dialplan” command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip.us"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> My suspicion is that some other dialplan, other than my “switch2voip.us” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension. Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032@172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: New “pastebin” created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable <http://pastebin.com/E4sqTLa4> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. <?xml version="1.0" encoding="utf-8"?> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="switch2voip"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/> </condition> </extension> 4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me… Thanks, George From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com> wrote: I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider — “State=REGED”. I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks! _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- Brian West brian@freeswitch.org <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ 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 -- Brian West brian@freeswitch.org <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
For the default config the public context listens on port 5080, the default (internal) one on 5060. Which context you hit depends on which port you contact. The ports are defined in the sip_profiles xmls. On 02.01.2015 12:49, George F. Phelps wrote:
Additional “freeswitch.log” debug output is included below, for a failed external call.
According to the debug output, the call attempt is indeed being made in the “public” context. My outbound dialplan is located in the “default” context, so apparently no “action” match is found.
*Questions***
1) Should my dialplan be located in the “public” or “default” context?
2) Should my call attempt be taking place in the “public” or “default” context?
3) What (how?) do I need to change to have both the dialplan and the dial attempt in the same (correct) context?
For reference, I pasted my current “xml_locate dialplan” output here:
*http://pastebin.com/BRB5Lpmg*
And this is my Freeswitch version information:
FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)
Thanks,
George
----
2015-01-02 09:44:19.406789 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->14049392032 in context public
Dialplan: sofia/external/1001@172.31.33.109 parsing [public->unloop] continue=false
Dialplan: sofia/external/1001@172.31.33.109 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/1001@172.31.33.109 parsing [public->outside_call] continue=true
Dialplan: sofia/external/1001@172.31.33.109 Absolute Condition [outside_call]
Dialplan: sofia/external/1001@172.31.33.109 Action set(outside_call=true)
Dialplan: sofia/external/1001@172.31.33.109 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/external/1001@172.31.33.109 parsing [public->call_debug] continue=true
Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/1001@172.31.33.109 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [public_extensions] destination_number(14049392032) =~ /^(10[01][0-9])$/ break=\
Dialplan: sofia/external/1001@172.31.33.109 parsing [public->public_did] continue=false
Dialplan: sofia/external/1001@172.31.33.109 Regex (FAIL) [public_did] destination_number(14049392032) =~ /^(5551212)$/ break=on-false
2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:216 (sofia/external/1001@172.31.33.109) State Change CS_ROUTING -> CS_EXE\
2015-01-02 09:44:19.406789 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1001@172.31.33.109 [BREAK]
2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1001@172.31.33.109) State ROUTING going to sleep
2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1001@172.31.33.109) Running State Change CS_EXECUTE
2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1001@172.31.33.109) State EXECUTE
2015-01-02 09:44:19.406789 [DEBUG] mod_sofia.c:178 sofia/external/1001@172.31.33.109 SOFIA EXECUTE
2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:258 sofia/external/1001@172.31.33.109 Standard EXECUTE
EXECUTE sofia/external/1001@172.31.33.109 set(outside_call=true)
2015-01-02 09:44:19.406789 [DEBUG] mod_dptools.c:1435 sofia/external/1001@172.31.33.109 SET [outside_call]=[true]
EXECUTE sofia/external/1001@172.31.33.109 export(RFC2822_DATE=Fri, 02 Jan 2015 09:44:19 -0500)
2015-01-02 09:44:19.406789 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 02 Jan 2015 09:44:19 -0500]
2015-01-02 09:44:19.406789 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 has executed the last dialplan ins\
2015-01-02 09:44:19.406789 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARI\
2015-01-02 09:44:19.406789 [DEBUG] switch_channel.c:3222 Send signal sofia/external/1001@172.31.33.109 [KILL]
2015-01-02 09:44:19.406789 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1001@172.31.33.109 [BREAK]
2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1001@172.31.33.109) State EXECUTE going to sleep
----
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *George F. Phelps *Sent:* Thursday, January 01, 2015 8:13 AM *To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
I added a debug log message to my dialplan:
<include>
<extension name="outbound_calls">
<condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
<action application="log" data="DIALING via switch2voip.us"/>
<action application="bridge" data="sofia/gateway/switch2voip.us/$1"/>
</condition>
</extension>
</include>
The log message is not showing up when I dial, as an example, “14049392032”. So I assume that some other section of the dialplan is handling my outbound call? How can I determine what section is handling the call, without manually adding multiple, additional log messages?
I pasted my current “xml_locate dialplan” output here:
Extracted “condition field="destination_number"”lines from the pastebin:
<action application="deflect" data="${destination_number}"></action>
<condition field="destination_number" expression="^886$">
<condition field="destination_number" expression="^\*8$">
<condition field="destination_number" expression="^\*\*(\d+)$">
<condition field="destination_number" expression="^(redial|870)$">
<action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></action>
<condition field="destination_number" expression="^9001$">
<condition field="destination_number" expression="^9000$">
<condition field="destination_number" expression="^88(\d{4})$|^\*0(.*)$">
<condition field="destination_number" expression="^779$">
<condition field="destination_number" expression="^\*69$|^869$|^lcr$">
<condition field="destination_number" expression="^80(\d{2})$">
<condition field="destination_number" expression="^81(\d{2})$">
<condition field="destination_number" expression="^82(\d{2})$">
<condition field="destination_number" expression="^83(\d{2})$">
<condition field="destination_number" expression="^8(10[01][0-9])$">
<condition field="destination_number" expression="^(10[01][0-9])$">
<condition field="destination_number" expression="^(11[01][0-9])$">
<action application="bridge" data="skinny/internal/${destination_number}"></action>
<condition field="destination_number" expression="^2000$">
<condition field="destination_number" expression="^2001$">
<condition field="destination_number" expression="^2002$">
<condition field="destination_number" expression="^(operator|0)$">
<condition field="destination_number" expression="^vmain$|^4000$|^\*98$">
<condition field="destination_number" expression="^sip:(.*)$">
<condition field="destination_number" expression="^(30\d{2})$">
<condition field="destination_number" expression="^(31\d{2})$">
<condition field="destination_number" expression="^(32\d{2})$">
<condition field="destination_number" expression="^(33\d{2})$">
<condition field="destination_number" expression="^9(888|8888|1616|3232)$">
<condition field="destination_number" expression="^0911$">
<condition field="destination_number" expression="^0912$">
<condition field="destination_number" expression="^0913$">
<condition field="destination_number" expression="^5000$">
<condition field="destination_number" expression="^5001$">
<condition field="destination_number" expression="^pagegroup$|^7243$">
<condition field="destination_number" expression="^5900$">
<condition field="destination_number" expression="^5901$">
<condition field="destination_number" expression="^(6000)$">
<condition field="destination_number" expression="^((?!6000)60\d{2})$">
<condition field="destination_number" expression="park\+(\d+)">
<condition field="destination_number" expression="^parking$"></condition>
<condition field="destination_number" expression="callpark"></condition>
<condition field="destination_number" expression="pickup"></condition>
<condition field="destination_number" expression="^wait$">
<condition field="destination_number" expression="^9178$">
<condition field="destination_number" expression="^9179$">
<condition field="destination_number" expression="^9180$">
<condition field="destination_number" expression="^9181$">
<condition field="destination_number" expression="^9182$">
<condition field="destination_number" expression="^9183$">
<condition field="destination_number" expression="^9184$">
<condition field="destination_number" expression="^9191$">
<condition field="destination_number" expression="^9192$">
<condition field="destination_number" expression="^9193$">
<condition field="destination_number" expression="^9194$">
<condition field="destination_number" expression="^9195$">
<condition field="destination_number" expression="^9196$">
<condition field="destination_number" expression="^9197$">
<condition field="destination_number" expression="^9198$">
<condition field="destination_number" expression="^9787$">
<condition field="destination_number" expression="^9664$"></condition>
<condition field="destination_number" expression="^9386$">
<condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
<condition field="destination_number" expression="^9170$">
<condition field="destination_number" expression="^9171$">
<condition field="destination_number" expression="^9172$">
<condition field="destination_number" expression="^(\d{7})$">
<condition field="destination_number" expression="^(\d{11})$">
<condition field="destination_number" expression="^(011\d+)$">
<condition field="destination_number" expression="^101$">
<condition field="destination_number" expression="^(pizza|74992)$"></condition>
<condition field="destination_number" expression="^(.*)$">
<condition field="destination_number" expression="^dx$">
<condition field="destination_number" expression="^att_xfer$">
<condition field="destination_number" expression="^is_transfer$"></condition>
<condition field="destination_number" expression="^cf$">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="deflect" data="${destination_number}"></action>
<condition field="destination_number" expression="^(10[01][0-9])$">
<condition field="destination_number" expression="^(5551212)$">
<condition field="destination_number" expression="^(9[0-9]{3})$">
<condition field="destination_number" expression="^(10[01][0-9])$">
<condition field="destination_number" expression="^(11[01][0-9])$">
<condition field="destination_number" expression="^(0[0-9]{10,})$">
<condition field="destination_number" expression="^(.{14,})$">
Thanks,
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *George F. Phelps *Sent:* Wednesday, December 31, 2014 4:37 PM *To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Brian West,
I wiped out my configuration and started over. I am able to dial from one extension to another — with 2-way audio. However, I am still not able to dial an external phone number. I also cannot dial the example extensions, x9196, x5000, etc.
I am able (via IPv6) to register with my VoIP gateway:
Name Type Data State
=================================================================================================
external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0)
external-ipv6::switch2voip.us gateway sip:9221591504@66.33.147.150 REGED
external profile sip:mod_sofia@54.174.255.168:5080 RUNNING (0)
external::switch2voip.us gateway sip:9221591504@66.33.147.150 FAIL_WAIT
172.31.33.109 alias internal ALIASED
internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)
internal profile sip:mod_sofia@54.174.255.168:5060 RUNNING (0)
=================================================================================================
4 profiles 1 alias
A failed dial attempt is shown below. Freeswitch is apparently still attempting to route the call to a non-existent local extension (sip:14049392032@172.31.33.109) instead of my VoIP provider.
recv 880 bytes from udp/[50.160.141.159]:49334 at 15:57:15.095108:
------------------------------------------------------------------------
INVITE sip:14049392032@172.31.33.109 SIP/2.0
Via: SIP/2.0/UDP 50.160.141.159:49334;rport;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo
Max-Forwards: 70
From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr
To: sip:14049392032@172.31.33.109
Contact: "George F Phelps" <sip:1001@50.160.141.159:49334;ob>
Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT
CSeq: 30128 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Bria Android 3.2.1
Content-Type: application/sdp
Content-Length: 247
v=0
o=- 3629048232 3629048232 IN IP4 50.160.141.159
s=cpc_med
c=IN IP4 50.160.141.159
t=0 0
m=audio 4000 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 406 bytes to udp/[50.160.141.159]:49334 at 15:57:15.095386:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo
From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr
To: sip:14049392032@172.31.33.109
Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT
CSeq: 30128 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit
Content-Length: 0
------------------------------------------------------------------------
send 802 bytes to udp/[50.160.141.159]:49334 at 15:57:15.097601:
------------------------------------------------------------------------
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo
Max-Forwards: 70
From: "George F Phelps" <sip:1001@172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr
To: <sip:14049392032@172.31.33.109>;tag=ctKjQv8vKyZtB
Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT
CSeq: 30128 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Remote-Party-ID: "14049392032" <sip:14049392032@172.31.33.109>;party=calling;privacy=off;screen=no
------------------------------------------------------------------------
My outbound dialplan “/usr/local/freeswitch/conf/dialplan/default/00_outbound_calls.xml”:
<include>
<extension name="outbound_calls">
<condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
<action application="bridge" data="sofia/gateway/switch2voip.us/$1"/>
</condition>
</extension>
</include>
My gateway “/usr/local/freeswitch/conf/directory/default/example.com.xml”:
<include>
<user id="$${default_provider}">
<gateways>
<gateway name="$${default_provider}">
<param name="username" value="$${default_provider_username}"/>
<param name="password" value="$${default_provider_password}"/>
<param name="from-user" value="$${default_provider_username}"/>
<param name="from-domain" value="$${default_provider_from_domain}"/>
<param name="realm" value="$${default_provider_from_domain}"/>
<param name="proxy" value="66.33.147.150"/>
<param name="expire-seconds" value="600"/>
<param name="register" value="$${default_provider_register}"/>
<param name="retry-seconds" value="30"/>
<param name="extension" value="$${default_provider_contact}"/>
<!--<param name="contact-params" value="domain_name=$${domain}"/>-->
<param name="context" value="public"/>
</gateway>
</gateways>
<params>
<param name="password" value="$${default_provider_password}"/>
</params>
</user>
</include>
And in “/usr/local/freeswitch/conf/vars.xml”:
<X-PRE-PROCESS cmd="set" data="default_provider=switch2voip.us"/>
<X-PRE-PROCESS cmd="set" data="default_provider_username=9221591504"/>
<X-PRE-PROCESS cmd="set" data="default_provider_password=XXXXXXXX"/>
<X-PRE-PROCESS cmd="set" data="default_provider_from_domain=switch2voip.us"/>
<!-- true or false -->
<X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
<X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider (66.33.147.150).
I did make the additional configuration updates as documented here:
https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2”
Thanks,
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* Tuesday, December 30, 2014 1:24 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact.
On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps <GeorgePhelps@gfphelps.com <mailto:GeorgePhelps@gfphelps.com>> wrote:
Brian West,
Okay, I’m sure there is an answer/solution there, but it’s over my head…
*Questions*
How do I check to see if I have inadvertently disabled “auth”? I am 99% sure that I have do changed it.
I am 100% sure that I have not touched anything to do with “allow acl”. But how do I check?
So are you saying that dialing an outside in the in the “public” context is correct?
Thanks,
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Brian West *Sent:* Tuesday, December 30, 2014 10:32 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth.
On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps <GeorgePhelps@gfphelps.com <mailto:GeorgePhelps@gfphelps.com>> wrote:
Follow on…
It appears, to me, that my outbound call is being processed in the “public” context:
2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 <tel:4049392032> in context public
Don’t I want it be processed in my “default” context? My local extensions are in the “default” context. My dialplan (for my gateway) is in the “default” context.
(Trace segment below.)
Thanks,
George
send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b
From: "George F Phelps" <sip:1001@172.31.33.109 <mailto:sip%3A1001@172.31.33.109>>;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U
To: sip:4049392032 <tel:4049392032>@172.31.33.109 <http://172.31.33.109>
Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ
CSeq: 10680 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit
Content-Length: 0
------------------------------------------------------------------------
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680)
nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying
nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called
nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001@172.31.33.109 <mailto:1001@172.31.33.109> [3a7e8146-7167-4276-a90c-70955ed5c250]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 <tel:4049392032> in context public
2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001@172.31.33.109 <mailto:1001@172.31.33.109> has executed the last dialplan instruction, hanging up.
2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001@172.31.33.109 <mailto:1001@172.31.33.109> [CS_EXECUTE] [NORMAL_CLEARING]
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *George F. Phelps *Sent:* Monday, December 29, 2014 7:21 AM
*To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Chris Tunbridge,
I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension?
recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167:
------------------------------------------------------------------------
INVITE sip:17708410143@172.31.33.109 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1000@50.160.141.159:13130>
To: "George Phelps"<sip:17708410143@172.31.33.109>
From: "George F Phelps"<sip:1000@172.31.33.109>;tag=13860149
Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria 3 release 3.5.5 stamp 71238
Content-Length: 264
v=0
o=- 13064325826971649 1 IN IP4 192.168.1.100
s=Bria 3 release 3.5.5 stamp 71238
c=IN IP4 192.168.1.100
t=0 0
m=audio 50404 RTP/AVP 9 8 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
------------------------------------------------------------------------
Thanks,
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Sunday, December 28, 2014 11:07 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us <http://switch2voip.us> into the /usr/local/freeswitch/conf/dialplan/default/ folder?
The main folder (not /default/) is used for context's, so it wouldn't get included in your default context.
On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil@gmail.com <mailto:david.villasmil@gmail.com>> wrote:
Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like:
<extension name="o <http://switch2voip.us/>utside_calls">
<condition field="destination_number" expression="^(404.*)$">
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 <http://switch2voip.us/$1>"/>
</condition>
</extension>
On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps@gfphelps.com <mailto:GeorgePhelps@gfphelps.com>> wrote:
The full output from the “xml_locate dialplan” command is already in the previously pasted logfile.
Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us <http://switch2voip.us>:
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip.us <http://switch2voip.us>">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 <http://switch2voip.us/$1>"/>
</condition>
</extension>
My suspicion is that some other dialplan, other than my “switch2voip.us <http://switch2voip.us>” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension.
Thanks,
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 5:05 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
can you share your dialplan? It looks like you're dialing
"To: sip:4049392032 <tel:4049392032>@172.31.33.109 <http://172.31.33.109>"
but have no extension for that...
On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <GeorgePhelps@gfphelps.com <mailto:GeorgePhelps@gfphelps.com>> wrote:
New “pastebin” created:
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *David Villasmil Govea *Sent:* Sunday, December 28, 2014 4:04 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it."
On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <GeorgePhelps@gfphelps.com <mailto:GeorgePhelps@gfphelps.com>> wrote:
Chris Tunbridge,
1) I made the updates to my configuration, as suggested in the “https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.
2) Extension x9161 is one of the default dialplan applications.
3) Call failure log posted at: http://pastebin.com/E4sqTLa4
Thanks,
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Chris Tunbridge *Sent:* Saturday, December 27, 2014 2:30 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml
2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.
3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.
4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.
On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <GeorgePhelps@gfphelps.com <mailto:GeorgePhelps@gfphelps.com>> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us <http://switch2voip.us>) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 <http://switch2voip.us/$1>"/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <GeorgePhelps@gfphelps.com <mailto:GeorgePhelps@gfphelps.com>> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
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-- Imagination is more important than knowledge Albert Einstein
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*/Twitter: @FreeSWITCH , @briankwest/* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 <tel:%2B19184209001> | *F:*+19184209002 <tel:%2B19184209002> | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
It's a configuration problem. You'd configured it to load, but not installed it. Although logged as a critical error FreeSWITCH would have started anyway so it wouldn't matter unless you were trying to run Javascript scripts. More of a warning, but logged with a higher log level so it's more likely to get noticed. There isn't a default configuration, although there are some example ones provided (vanilla etc) On 26 December 2014 at 14:57, George F. Phelps <GeorgePhelps@gfphelps.com> wrote:
Chris Tunbridge, et al.,
1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.
2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.
3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway.
<?xml version="1.0" encoding="utf-8"?>
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="switch2voip">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" /-->
<action application="bridge" data="sofia/gateway/switch2voip.us/$1 "/>
</condition>
</extension>
4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…
Thanks,
George
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Chris Tunbridge *Sent:* Thursday, December 25, 2014 9:25 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration
2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)
3) Do you have an outbound route configured that matches your dial string?
4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end
On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps < GeorgePhelps@gfphelps.com> wrote:
I am debugging a new/initial Freeswitch configuration.
I believe that I have successfully registered with my VoIP provider — “State=REGED”.
I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.
When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?
And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.
I am getting this critical error on startup. Problem #4.
2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so
**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**
Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?
Version info:
FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)
Thanks!
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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participants (6)
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Brian West
-
Chris Tunbridge
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David Villasmil Govea
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George F. Phelps
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Steven Ayre
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Thomas Auge