No outbounds calls
Ok guys, I give up. So I must ask help. :-( In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please? This is my gateway conf: <include> <gateway name="outbound-my_provider"> <param name="username" value="123456789"/> <!-- param name="auth-username" value="123456789@ims.my_provider.net"/ --> <!-- param name="password" value="my_password"/ --> <param name="realm" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="123.ims.my_provider.net:5060"/> <param name="register" value="false"/> <!-- param name="register-transport" value="udp"/ --> <param name="context" value="public"/> <param name="extension" value="123456789"/> <param name="extension-in-contact" value="true"/> <param name="expire-seconds" value="3600"/> <!--param name="cid-type" value="rpid"/--> <!-- param name="contact-params" value="domain_name=$${domain}"/ --> </gateway> <gateway name="123456789"> <param name="username" value="123456789@ims.my_provider.net"/> <param name="password" value="my_password"/> <param name="extension" value="123456789"/> <param name="proxy" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="register-proxy" value="123.ims.my_provider.net:5060"/> <param name="expire-seconds" value="1800"/> <param name="retry-seconds" value="120"/> <param name="register" value="true"/> <param name="dtmf-type" value="rfc2833"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> This is my dialplan: <!-- /etc/freeswitch/dialplan/public/my_gateway.xml --> <include> <extension name="outbound_calls"> <condition field="destination_number" expression="(^\d{5,14}$)"> <action application="set" data="effective_caller_id_name=123456789"/> <action application="set" data="effective_caller_id_number=123456789"/> <!-- action application="set" data="sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net"/ --> <action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/> </condition> </extension> </include> I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209. This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95 Thank you in advance. Best regards -- Distinti saluti /Paolo Visnoviz/
Hi Paolo, SIP/2.0 404 Not Found Via: SIP/2.0/UDP my_public_ip:5080;rport=5080;branch=z9hG4bK9X12v734KmrZe To: <sip:number_to_call@123.ims.my_provider.net>;tag=ztesipsuDgPzVN*2-4-20481*giag.2 From: "?"<sip:123456789@ims.my_provider.net>;tag=8g41KmQBgB0pm Call-ID: 9b55b86c-0972-1238-0aa8-080027755653 CSeq: 5714138 INVITE X-ZTE-Cause: "CSCF-1.3154123179.miicscf1.ims.my_provider.net" Content-Length: 0 Thanks & Regards On Sat, Jun 15, 2019 at 12:50 AM paolo.visnoviz@vipcomputers.it <paolo.visnoviz@vipcomputers.it> wrote:
Ok guys, I give up. So I must ask help. :-(
In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please?
This is my gateway conf:
<include> <gateway name="outbound-my_provider"> <param name="username" value="123456789"/> <!-- param name="auth-username" value="123456789@ims.my_provider.net"/ --> <!-- param name="password" value="my_password"/ --> <param name="realm" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="123.ims.my_provider.net:5060"/> <param name="register" value="false"/> <!-- param name="register-transport" value="udp"/ --> <param name="context" value="public"/> <param name="extension" value="123456789"/> <param name="extension-in-contact" value="true"/> <param name="expire-seconds" value="3600"/> <!--param name="cid-type" value="rpid"/--> <!-- param name="contact-params" value="domain_name=$${domain}"/ --> </gateway> <gateway name="123456789"> <param name="username" value="123456789@ims.my_provider.net"/> <param name="password" value="my_password"/> <param name="extension" value="123456789"/> <param name="proxy" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="register-proxy" value="123.ims.my_provider.net:5060"/> <param name="expire-seconds" value="1800"/> <param name="retry-seconds" value="120"/> <param name="register" value="true"/> <param name="dtmf-type" value="rfc2833"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway>
This is my dialplan:
<!-- /etc/freeswitch/dialplan/public/my_gateway.xml --> <include> <extension name="outbound_calls"> <condition field="destination_number" expression="(^\d{5,14}$)"> <action application="set" data="effective_caller_id_name=123456789"/> <action application="set" data="effective_caller_id_number=123456789"/> <!-- action application="set" data="sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net"/ --> <action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/> </condition> </extension> </include>
I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209.
This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95 Thank you in advance. Best regards
-- Distinti saluti Paolo Visnoviz
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Yes, and unallocated number too, but I can't figure out why... :-( Il 15/06/19 08:35, Brian : ha scritto:
Hi Paolo,
SIP/2.0 404 Not Found Via: SIP/2.0/UDP my_public_ip:5080;rport=5080;branch=z9hG4bK9X12v734KmrZe To: <sip:number_to_call@123.ims.my_provider.net>;tag=ztesipsuDgPzVN*2-4-20481*giag.2 From: "?"<sip:123456789@ims.my_provider.net>;tag=8g41KmQBgB0pm Call-ID: 9b55b86c-0972-1238-0aa8-080027755653 CSeq: 5714138 INVITE X-ZTE-Cause: "CSCF-1.3154123179.miicscf1.ims.my_provider.net" Content-Length: 0
Thanks & Regards
On Sat, Jun 15, 2019 at 12:50 AM paolo.visnoviz@vipcomputers.it <paolo.visnoviz@vipcomputers.it> wrote:
Ok guys, I give up. So I must ask help. :-(
In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please?
This is my gateway conf:
<include> <gateway name="outbound-my_provider"> <param name="username" value="123456789"/> <!-- param name="auth-username" value="123456789@ims.my_provider.net"/ --> <!-- param name="password" value="my_password"/ --> <param name="realm" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="123.ims.my_provider.net:5060"/> <param name="register" value="false"/> <!-- param name="register-transport" value="udp"/ --> <param name="context" value="public"/> <param name="extension" value="123456789"/> <param name="extension-in-contact" value="true"/> <param name="expire-seconds" value="3600"/> <!--param name="cid-type" value="rpid"/--> <!-- param name="contact-params" value="domain_name=$${domain}"/ --> </gateway> <gateway name="123456789"> <param name="username" value="123456789@ims.my_provider.net"/> <param name="password" value="my_password"/> <param name="extension" value="123456789"/> <param name="proxy" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="register-proxy" value="123.ims.my_provider.net:5060"/> <param name="expire-seconds" value="1800"/> <param name="retry-seconds" value="120"/> <param name="register" value="true"/> <param name="dtmf-type" value="rfc2833"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway>
This is my dialplan:
<!-- /etc/freeswitch/dialplan/public/my_gateway.xml --> <include> <extension name="outbound_calls"> <condition field="destination_number" expression="(^\d{5,14}$)"> <action application="set" data="effective_caller_id_name=123456789"/> <action application="set" data="effective_caller_id_number=123456789"/> <!-- action application="set" data="sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net"/ --> <action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/> </condition> </extension> </include>
I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209.
This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95 Thank you in advance. Best regards
-- Distinti saluti Paolo Visnoviz
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com -- Distinti saluti /Paolo Visnoviz/
Hi Paolo, I had a similar problem with one provider in the US. It may be unrelated, but I thought I would mention it. The issue was in the outgoing number. In my case the provider required “+” then 1 and my area code and number. Are you sure you have the outbound number syntax correct to meet their requirements? Regards, Sean From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> On Behalf Of paolo.visnoviz@vipcomputers.it Sent: Saturday, June 15, 2019 5:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] No outbounds calls Yes, and unallocated number too, but I can't figure out why... :-( Il 15/06/19 08:35, Brian : ha scritto: Hi Paolo, SIP/2.0 404 Not Found Via: SIP/2.0/UDP my_public_ip:5080;rport=5080;branch=z9hG4bK9X12v734KmrZe To: <sip:number_to_call@123.ims.my_provider.net><mailto:sip:number_to_call@123.ims.my_provider.net>;tag=ztesipsuDgPzVN*2-4-20481*giag.2 From: "?"<sip:123456789@ims.my_provider.net><mailto:sip:123456789@ims.my_provider.net>;tag=8g41KmQBgB0pm Call-ID: 9b55b86c-0972-1238-0aa8-080027755653 CSeq: 5714138 INVITE X-ZTE-Cause: "CSCF-1.3154123179.miicscf1.ims.my_provider.net" Content-Length: 0 Thanks & Regards On Sat, Jun 15, 2019 at 12:50 AM paolo.visnoviz@vipcomputers.it<mailto:paolo.visnoviz@vipcomputers.it> <paolo.visnoviz@vipcomputers.it><mailto:paolo.visnoviz@vipcomputers.it> wrote: Ok guys, I give up. So I must ask help. :-( In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please? This is my gateway conf: <include> <gateway name="outbound-my_provider"> <param name="username" value="123456789"/> <!-- param name="auth-username" value="123456789@ims.my_provider.net"<mailto:123456789@ims.my_provider.net>/ --> <!-- param name="password" value="my_password"/ --> <param name="realm" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="123.ims.my_provider.net:5060"/> <param name="register" value="false"/> <!-- param name="register-transport" value="udp"/ --> <param name="context" value="public"/> <param name="extension" value="123456789"/> <param name="extension-in-contact" value="true"/> <param name="expire-seconds" value="3600"/> <!--param name="cid-type" value="rpid"/--> <!-- param name="contact-params" value="domain_name=$${domain}"/ --> </gateway> <gateway name="123456789"> <param name="username" value="123456789@ims.my_provider.net"<mailto:123456789@ims.my_provider.net>/> <param name="password" value="my_password"/> <param name="extension" value="123456789"/> <param name="proxy" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="register-proxy" value="123.ims.my_provider.net:5060"/> <param name="expire-seconds" value="1800"/> <param name="retry-seconds" value="120"/> <param name="register" value="true"/> <param name="dtmf-type" value="rfc2833"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> This is my dialplan: <!-- /etc/freeswitch/dialplan/public/my_gateway.xml --> <include> <extension name="outbound_calls"> <condition field="destination_number" expression="(^\d{5,14}$)"> <action application="set" data="effective_caller_id_name=123456789"/> <action application="set" data="effective_caller_id_number=123456789"/> <!-- action application="set" data="sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net"<mailto:sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net>/ --> <action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/> </condition> </extension> </include> I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209. This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95 Thank you in advance. Best regards -- Distinti saluti Paolo Visnoviz _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Distinti saluti Paolo Visnoviz
Maybe stupid question, but You’re sending To: <sip:number_to_call@123.ims.my_provider.net> Number_to_call As the “to” user, did you remove the actual number or are sending that? On Mon, 17 Jun 2019 at 11:25, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi Paolo,
I had a similar problem with one provider in the US. It may be unrelated, but I thought I would mention it. The issue was in the outgoing number. In my case the provider required “+” then 1 and my area code and number. Are you sure you have the outbound number syntax correct to meet their requirements?
Regards,
Sean
*From:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> *On Behalf Of *paolo.visnoviz@vipcomputers.it *Sent:* Saturday, June 15, 2019 5:16 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] No outbounds calls
Yes, and unallocated number too, but I can't figure out why... :-(
Il 15/06/19 08:35, Brian : ha scritto:
Hi Paolo,
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP my_public_ip:5080;rport=5080;branch=z9hG4bK9X12v734KmrZe
To: <sip:number_to_call@123.ims.my_provider.net> <sip:number_to_call@123.ims.my_provider.net>;tag=ztesipsuDgPzVN*2-4-20481*giag.2
From: "?"<sip:123456789@ims.my_provider.net> <sip:123456789@ims.my_provider.net>;tag=8g41KmQBgB0pm
Call-ID: 9b55b86c-0972-1238-0aa8-080027755653
CSeq: 5714138 INVITE
X-ZTE-Cause: "CSCF-1.3154123179.miicscf1.ims.my_provider.net"
Content-Length: 0
Thanks & Regards
On Sat, Jun 15, 2019 at 12:50 AM paolo.visnoviz@vipcomputers.it
<paolo.visnoviz@vipcomputers.it> <paolo.visnoviz@vipcomputers.it> wrote:
Ok guys, I give up. So I must ask help. :-(
In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please?
This is my gateway conf:
<include>
<gateway name="outbound-my_provider">
<param name="username" value="123456789"/>
<!-- param name="auth-username" value="123456789@ims.my_provider.net" <123456789@ims.my_provider.net>/ -->
<!-- param name="password" value="my_password"/ -->
<param name="realm" value="ims.my_provider.net"/>
<param name="from-user" value="123456789"/>
<param name="from-domain" value="ims.my_provider.net"/>
<param name="caller-id-in-from" value="true"/>
<param name="proxy" value="123.ims.my_provider.net:5060"/>
<param name="register" value="false"/>
<!-- param name="register-transport" value="udp"/ -->
<param name="context" value="public"/>
<param name="extension" value="123456789"/>
<param name="extension-in-contact" value="true"/>
<param name="expire-seconds" value="3600"/>
<!--param name="cid-type" value="rpid"/-->
<!-- param name="contact-params" value="domain_name=$${domain}"/ -->
</gateway>
<gateway name="123456789">
<param name="username" value="123456789@ims.my_provider.net" <123456789@ims.my_provider.net>/>
<param name="password" value="my_password"/>
<param name="extension" value="123456789"/>
<param name="proxy" value="ims.my_provider.net"/>
<param name="from-user" value="123456789"/>
<param name="from-domain" value="ims.my_provider.net"/>
<param name="register-proxy" value="123.ims.my_provider.net:5060"/>
<param name="expire-seconds" value="1800"/>
<param name="retry-seconds" value="120"/>
<param name="register" value="true"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="register-transport" value="udp"/>
<param name="context" value="public"/>
</gateway>
This is my dialplan:
<!-- /etc/freeswitch/dialplan/public/my_gateway.xml -->
<include>
<extension name="outbound_calls">
<condition field="destination_number" expression="(^\d{5,14}$)">
<action application="set" data="effective_caller_id_name=123456789"/>
<action application="set" data="effective_caller_id_number=123456789"/>
<!-- action application="set" data="sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net" <sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net>/ -->
<action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/>
</condition>
</extension>
</include>
I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209.
This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95
Thank you in advance. Best regards
--
Distinti saluti
Paolo Visnoviz
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
-- Distinti saluti *Paolo Visnoviz* _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
Sean, yes I tried with every kind of number form: with/without + , 00 and so on. Same result. David, sorry but I can't understand your question. "number_to_call" is the number that I try to call, here removed for obvious privacy reasons. In the coming days, as soon as I can, I will add more details. The problems steams because in Italy, until recently, the majority of providers give us an adsl or fttc or ftth with a preconfigured router, usually with two pstn ports. We cant' use (without some tricks) others router than one that our providers give us. And was very difficult to use directly the voip. Now, like in other european country, the law require to our provider to release the connection parameters, voip included. But often they are very criptical in doing so. And they don't give us no clarification, not even if you call paid assistance. And this is the parameters released to me: PPPoE Fisso USER-VLAN 8/35 in 802.1Q user ID PPP (Internet) username ppp Password PPP (Internet) pwd ppp Username SIP (Voce) mynumber@ims.tiscali.net Password SIP (Voce) my_sip_pwd URI 1 mynumber outbound proxy/proxy server • srv: srvmi.p.ims.tiscali.net • fqdn: core2.p.ims.tiscali.net • IP: 94.32.130.112 • Protocollo UDP: porta 5060 Protocollo Voip SIP RFC 3261 Dominio/Registrar ims.tiscali.net Codec list g711alaw; g729 DTMF rfc2833 payload type 97 symmetric implementation Fax g 711 pass-through (T38 disabled) Session refresh Update method PRACK Supported 100rel MWI notify unsolicited ( subscribe disable) CLIP PAI-FROM Well, with this values I tried in first to configure my existing Asterisk, without success. This because our telco provider give us a trunk that require 100rel and TEL RFC 3966 (not only RFC 3261, but they omit to tell). Now Asterisk 16 with PJSIP support 100Rel, but not RFC3966 (so it is impossible to receive any call: protocol error), and they don't actually works to implement it. Asterisk chan_sip, instead, support RFC3966, but not 100Rel and chan_sip is not more actively developed on by Digium or Sangoma. So I search for an alternative. I found Freeswitch, and I was impressed for its flexibility and power. If I will find an italian language for Ivr, i will leave Asterisk. With freeswitch I was able to register my trunk and receive calls, but I can't made any outbound calls. I analyzed with Wireshark the sip traffic of provider router given us and, when as soon as I can, I will post it. There are some difference with my freeswitch outbound calls configuration, of course. But, maybe someone could ask, why don't simply use their router provided to us? Because I need an 8 ip pool too. Well, where is the problem? No problems at all, in fact my provider gives them to me. But, and here the things becomes ridiculous, you can't configure them because their router is armored, and you can access only to a limited sets of functions. If you call assistance and ask them to set the router, they treat you like an idiot and recommended to ask for system admin. Idiot, I'm a system admin - i was thinking during assistance call - but if I can't change the router modality from natted to routed or, better, routed+natted (and many other things), I couldn't do that even if I were Richard Stallman. Anyway I also need a native voip connection, not only two pstn archaic ports. Sorry if I was so verbose, but it was only to give you a picture of the situation. In the next days I will post the sip traces of outbound calls coming from router provider and freeswitch. Maybe someone will find were I'm wrong. For the moment, thank you all! Il 18/06/19 01:14, David Villasmil ha scritto:
Maybe stupid question, but You’re sending
To: <sip:number_to_call@123.ims.my_provider.net <mailto:sip%3Anumber_to_call@123.ims.my_provider.net>>
Number_to_call
As the “to” user, did you remove the actual number or are sending that?
On Mon, 17 Jun 2019 at 11:25, Sean Devoy <sdevoy@bizfocused.com <mailto:sdevoy@bizfocused.com>> wrote:
Hi Paolo,
I had a similar problem with one provider in the US. It may be unrelated, but I thought I would mention it. The issue was in the outgoing number. In my case the provider required “+” then 1 and my area code and number. Are you sure you have the outbound number syntax correct to meet their requirements?
Regards,
Sean
*From:*FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>> *On Behalf Of *paolo.visnoviz@vipcomputers.it <mailto:paolo.visnoviz@vipcomputers.it> *Sent:* Saturday, June 15, 2019 5:16 AM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] No outbounds calls
Yes, and unallocated number too, but I can't figure out why... :-(
Il 15/06/19 08:35, Brian : ha scritto:
Hi Paolo,
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP my_public_ip:5080;rport=5080;branch=z9hG4bK9X12v734KmrZe
To:<sip:number_to_call@123.ims.my_provider.net> <mailto:sip:number_to_call@123.ims.my_provider.net>;tag=ztesipsuDgPzVN*2-4-20481*giag.2
From: "?"<sip:123456789@ims.my_provider.net> <mailto:sip:123456789@ims.my_provider.net>;tag=8g41KmQBgB0pm
Call-ID: 9b55b86c-0972-1238-0aa8-080027755653
CSeq: 5714138 INVITE
X-ZTE-Cause: "CSCF-1.3154123179.miicscf1.ims.my_provider.net <http://CSCF-1.3154123179.miicscf1.ims.my_provider.net>"
Content-Length: 0
Thanks & Regards
On Sat, Jun 15, 2019 at 12:50 AMpaolo.visnoviz@vipcomputers.it <mailto:paolo.visnoviz@vipcomputers.it>
<paolo.visnoviz@vipcomputers.it> <mailto:paolo.visnoviz@vipcomputers.it> wrote:
Ok guys, I give up. So I must ask help. :-(
In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please?
This is my gateway conf:
<include>
<gateway name="outbound-my_provider">
<param name="username" value="123456789"/>
<!-- param name="auth-username" value="123456789@ims.my_provider.net" <mailto:123456789@ims.my_provider.net>/ -->
<!-- param name="password" value="my_password"/ -->
<param name="realm" value="ims.my_provider.net <http://ims.my_provider.net>"/>
<param name="from-user" value="123456789"/>
<param name="from-domain" value="ims.my_provider.net <http://ims.my_provider.net>"/>
<param name="caller-id-in-from" value="true"/>
<param name="proxy" value="123.ims.my_provider.net:5060 <http://123.ims.my_provider.net:5060>"/>
<param name="register" value="false"/>
<!-- param name="register-transport" value="udp"/ -->
<param name="context" value="public"/>
<param name="extension" value="123456789"/>
<param name="extension-in-contact" value="true"/>
<param name="expire-seconds" value="3600"/>
<!--param name="cid-type" value="rpid"/-->
<!-- param name="contact-params" value="domain_name=$${domain}"/ -->
</gateway>
<gateway name="123456789">
<param name="username" value="123456789@ims.my_provider.net" <mailto:123456789@ims.my_provider.net>/>
<param name="password" value="my_password"/>
<param name="extension" value="123456789"/>
<param name="proxy" value="ims.my_provider.net <http://ims.my_provider.net>"/>
<param name="from-user" value="123456789"/>
<param name="from-domain" value="ims.my_provider.net <http://ims.my_provider.net>"/>
<param name="register-proxy" value="123.ims.my_provider.net:5060 <http://123.ims.my_provider.net:5060>"/>
<param name="expire-seconds" value="1800"/>
<param name="retry-seconds" value="120"/>
<param name="register" value="true"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="register-transport" value="udp"/>
<param name="context" value="public"/>
</gateway>
This is my dialplan:
<!-- /etc/freeswitch/dialplan/public/my_gateway.xml -->
<include>
<extension name="outbound_calls">
<condition field="destination_number" expression="(^\d{5,14}$)">
<action application="set" data="effective_caller_id_name=123456789"/>
<action application="set" data="effective_caller_id_number=123456789"/>
<!-- action application="set" data="sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net" <mailto:sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net>/ -->
<action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/>
</condition>
</extension>
</include>
I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209.
This is my pastebin of my external call attempt:https://pastebin.freeswitch.org/view/05befe95
Thank you in advance. Best regards
--
Distinti saluti
Paolo Visnoviz
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWirehttps://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real timehttps://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com <mailto:sales@freeswitch.com>
Official FreeSWITCH Sites
https://freeswitch.org/confluence
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWirehttps://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real timehttps://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com <mailto:sales@freeswitch.com>
Official FreeSWITCH Sites
https://freeswitch.org/confluence
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
-- Distinti saluti /Paolo Visnoviz/
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com <mailto:sales@freeswitch.com> https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com <mailto:david.villasmil.work@gmail.com> phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Can you provide a trace? Are you sending the PAI header? On Fri, 21 Jun 2019 at 10:47, Paolo Visnoviz - VI.P. Computers < paolo.visnoviz@vipcomputers.it> wrote:
Sean, yes I tried with every kind of number form: with/without + , 00 and so on. Same result.
David, sorry but I can't understand your question. "number_to_call" is the number that I try to call, here removed for obvious privacy reasons.
In the coming days, as soon as I can, I will add more details. The problems steams because in Italy, until recently, the majority of providers give us an adsl or fttc or ftth with a preconfigured router, usually with two pstn ports. We cant' use (without some tricks) others router than one that our providers give us. And was very difficult to use directly the voip. Now, like in other european country, the law require to our provider to release the connection parameters, voip included. But often they are very criptical in doing so. And they don't give us no clarification, not even if you call paid assistance.
And this is the parameters released to me:
PPPoE Fisso USER-VLAN 8/35 in 802.1Q user ID PPP (Internet) username ppp Password PPP (Internet) pwd ppp Username SIP (Voce) mynumber@ims.tiscali.net Password SIP (Voce) my_sip_pwd URI 1 mynumber outbound proxy/proxy server
• srv: srvmi.p.ims.tiscali.net • fqdn: core2.p.ims.tiscali.net • IP: 94.32.130.112 • Protocollo UDP: porta 5060
Protocollo Voip SIP RFC 3261 Dominio/Registrar ims.tiscali.net Codec list g711alaw; g729 DTMF rfc2833 payload type 97 symmetric implementation Fax g 711 pass-through (T38 disabled) Session refresh Update method PRACK Supported 100rel MWI notify unsolicited ( subscribe disable) CLIP PAI-FROM
Well, with this values I tried in first to configure my existing Asterisk, without success. This because our telco provider give us a trunk that require 100rel and TEL RFC 3966 (not only RFC 3261, but they omit to tell). Now Asterisk 16 with PJSIP support 100Rel, but not RFC3966 (so it is impossible to receive any call: protocol error), and they don't actually works to implement it. Asterisk chan_sip, instead, support RFC3966, but not 100Rel and chan_sip is not more actively developed on by Digium or Sangoma.
So I search for an alternative. I found Freeswitch, and I was impressed for its flexibility and power. If I will find an italian language for Ivr, i will leave Asterisk.
With freeswitch I was able to register my trunk and receive calls, but I can't made any outbound calls.
I analyzed with Wireshark the sip traffic of provider router given us and, when as soon as I can, I will post it. There are some difference with my freeswitch outbound calls configuration, of course. But, maybe someone could ask, why don't simply use their router provided to us? Because I need an 8 ip pool too. Well, where is the problem? No problems at all, in fact my provider gives them to me. But, and here the things becomes ridiculous, you can't configure them because their router is armored, and you can access only to a limited sets of functions. If you call assistance and ask them to set the router, they treat you like an idiot and recommended to ask for system admin. Idiot, I'm a system admin - i was thinking during assistance call - but if I can't change the router modality from natted to routed or, better, routed+natted (and many other things), I couldn't do that even if I were Richard Stallman.
Anyway I also need a native voip connection, not only two pstn archaic ports.
Sorry if I was so verbose, but it was only to give you a picture of the situation. In the next days I will post the sip traces of outbound calls coming from router provider and freeswitch. Maybe someone will find were I'm wrong.
For the moment, thank you all!
Il 18/06/19 01:14, David Villasmil ha scritto:
Maybe stupid question, but You’re sending
To: <sip:number_to_call@123.ims.my_provider.net>
Number_to_call
As the “to” user, did you remove the actual number or are sending that?
On Mon, 17 Jun 2019 at 11:25, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi Paolo,
I had a similar problem with one provider in the US. It may be unrelated, but I thought I would mention it. The issue was in the outgoing number. In my case the provider required “+” then 1 and my area code and number. Are you sure you have the outbound number syntax correct to meet their requirements?
Regards,
Sean
*From:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> *On Behalf Of *paolo.visnoviz@vipcomputers.it *Sent:* Saturday, June 15, 2019 5:16 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] No outbounds calls
Yes, and unallocated number too, but I can't figure out why... :-(
Il 15/06/19 08:35, Brian : ha scritto:
Hi Paolo,
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP my_public_ip:5080;rport=5080;branch=z9hG4bK9X12v734KmrZe
To: <sip:number_to_call@123.ims.my_provider.net> <sip:number_to_call@123.ims.my_provider.net>;tag=ztesipsuDgPzVN*2-4-20481*giag.2
From: "?"<sip:123456789@ims.my_provider.net> <sip:123456789@ims.my_provider.net>;tag=8g41KmQBgB0pm
Call-ID: 9b55b86c-0972-1238-0aa8-080027755653
CSeq: 5714138 INVITE
X-ZTE-Cause: "CSCF-1.3154123179.miicscf1.ims.my_provider.net"
Content-Length: 0
Thanks & Regards
On Sat, Jun 15, 2019 at 12:50 AM paolo.visnoviz@vipcomputers.it
<paolo.visnoviz@vipcomputers.it> <paolo.visnoviz@vipcomputers.it> wrote:
Ok guys, I give up. So I must ask help. :-(
In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please?
This is my gateway conf:
<include>
<gateway name="outbound-my_provider">
<param name="username" value="123456789"/>
<!-- param name="auth-username" value="123456789@ims.my_provider.net" <123456789@ims.my_provider.net>/ -->
<!-- param name="password" value="my_password"/ -->
<param name="realm" value="ims.my_provider.net"/>
<param name="from-user" value="123456789"/>
<param name="from-domain" value="ims.my_provider.net"/>
<param name="caller-id-in-from" value="true"/>
<param name="proxy" value="123.ims.my_provider.net:5060"/>
<param name="register" value="false"/>
<!-- param name="register-transport" value="udp"/ -->
<param name="context" value="public"/>
<param name="extension" value="123456789"/>
<param name="extension-in-contact" value="true"/>
<param name="expire-seconds" value="3600"/>
<!--param name="cid-type" value="rpid"/-->
<!-- param name="contact-params" value="domain_name=$${domain}"/ -->
</gateway>
<gateway name="123456789">
<param name="username" value="123456789@ims.my_provider.net" <123456789@ims.my_provider.net>/>
<param name="password" value="my_password"/>
<param name="extension" value="123456789"/>
<param name="proxy" value="ims.my_provider.net"/>
<param name="from-user" value="123456789"/>
<param name="from-domain" value="ims.my_provider.net"/>
<param name="register-proxy" value="123.ims.my_provider.net:5060"/>
<param name="expire-seconds" value="1800"/>
<param name="retry-seconds" value="120"/>
<param name="register" value="true"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="register-transport" value="udp"/>
<param name="context" value="public"/>
</gateway>
This is my dialplan:
<!-- /etc/freeswitch/dialplan/public/my_gateway.xml -->
<include>
<extension name="outbound_calls">
<condition field="destination_number" expression="(^\d{5,14}$)">
<action application="set" data="effective_caller_id_name=123456789"/>
<action application="set" data="effective_caller_id_number=123456789"/>
<!-- action application="set" data="sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net" <sip_h_P-Preferred-Identity=sip:123456789@ims.my_provider.net>/ -->
<action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/>
</condition>
</extension>
</include>
I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209.
This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95
Thank you in advance. Best regards
--
Distinti saluti
Paolo Visnoviz
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services
sales@freeswitch.com
Official FreeSWITCH Sites
https://freeswitch.org/confluence
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
-- Distinti saluti *Paolo Visnoviz* _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Servicessales@freeswitch.comhttps://freeswitch.com
Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.c...
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswit...
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
participants (5)
-
Brian :
-
David Villasmil
-
Paolo Visnoviz - VI.P. Computers
-
paolo.visnoviz＠vipcomputers.it
-
Sean Devoy