[Freeswitch-users] No outbounds calls

David Villasmil david.villasmil.work at gmail.com
Fri Jun 21 16:59:08 UTC 2019 

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 at 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 at 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 at 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 at 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 at lists.freeswitch.org> *On
>> Behalf Of *paolo.visnoviz at vipcomputers.it
>> *Sent:* Saturday, June 15, 2019 5:16 AM
>> *To:* freeswitch-users at 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 at 123.ims.my_provider.net> <sip:number_to_call at 123.ims.my_provider.net>;tag=ztesipsuDgPzVN*2-4-20481*giag.2
>>
>>    From: "?"<sip:123456789 at ims.my_provider.net> <sip:123456789 at 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 at vipcomputers.it
>>
>> <paolo.visnoviz at vipcomputers.it> <paolo.visnoviz at 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 at ims.my_provider.net" <123456789 at 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 at ims.my_provider.net" <123456789 at 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 at ims.my_provider.net" <sip_h_P-Preferred-Identity=sip:123456789 at 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 at 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 at 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 at 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 at 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*
>> _________________________________________________________________________
>>
>> 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 at 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 at 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 at 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 at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://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 at 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 at 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 at gmail.com
phone: +34669448337
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