[Freeswitch-users] No outbounds calls
Paolo Visnoviz - VI.P. Computers
paolo.visnoviz at vipcomputers.it
Tue Jun 18 23:36:56 UTC 2019
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
> <mailto:sip%3Anumber_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
> <mailto: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
> <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *On Behalf
> Of *paolo.visnoviz at vipcomputers.it
> <mailto:paolo.visnoviz at vipcomputers.it>
> *Sent:* Saturday, June 15, 2019 5:16 AM
> *To:* freeswitch-users at lists.freeswitch.org
> <mailto: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> <mailto: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> <mailto: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 <http://CSCF-1.3154123179.miicscf1.ims.my_provider.net>"
>
> Content-Length: 0
>
>
>
> Thanks & Regards
>
>
>
> On Sat, Jun 15, 2019 at 12:50 AMpaolo.visnoviz at vipcomputers.it <mailto:paolo.visnoviz at vipcomputers.it>
>
> <paolo.visnoviz at vipcomputers.it> <mailto: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" <mailto:123456789 at 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 at ims.my_provider.net" <mailto: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 <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 at ims.my_provider.net" <mailto: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 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 at freeswitch.com <mailto: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 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
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>
> https://freeswitch.com
>
>
>
> _________________________________________________________________________
>
>
>
> 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 at freeswitch.com <mailto: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 <mailto: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 <mailto: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
> <mailto: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
> <mailto: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 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
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