Hi David

when change the
the Register message send out with wrong format : from sip:071234@x.x.x.x
the right format : sip:071234@sip.yeah.com

is there any way to fix it

Thank you

Từ: David Ponzone
Đến: freeswitch-users@lists.freeswitch.org
Gửi ngày: 15:50:00, Thứ Tư, 31 tháng 3 2010
Chủ đề: Re: [Freeswitch-users] Về: Về: need help on call to gateway

I guess you need to put the IP of the proxy also in:

David Ponzone Direction Technique

email: david.ponzone@ipeva.fr

tel: 01 74 03 18 97

gsm: 06 66 98 76 34

Service Client IPeva

tel: 0811 46 26 26

www.ipeva.fr - www.ipeva-studio.com

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

Le 31/03/2010 à 10:22, false a écrit :

Hi David

the domain sip.yeah.com cannot resolve in the FS host
the thing here is FW support for Registration to remote SIP server
so the domain no need to resolve
how to do the same thing when call out, no need to resolve the domain name: sip.yeah.com,

Thank you

Từ: David Ponzone <david.ponzone@gmail.com>
Đến: freeswitch-users@lists.freeswitch.org
Gửi ngày: 15:12:50, Thứ Tư, 31 tháng 3 2010
Chủ đề: Re: [Freeswitch-users] Về: need help on call to gateway

This looks like a DNS error.
Are you sure you can resolve sip.yeah.com from the FS host ?

David Ponzone Direction Technique

email: david.ponzone@ipeva.fr

tel: 01 74 03 18 97

gsm: 06 66 98 76 34

Service Client IPeva

tel: 0811 46 26 26

www.ipeva.fr - www.ipeva-studio.com

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

Le 31/03/2010 à 09:51, false a écrit :

Hi David and Brian

sorry to bother you again
the domain name : sip.yeah.com is not real, How to fix the issue

FreeSwitch register to sip server with user 071234 successull
FreeSwitch receive incomming from sip server and then transfer to 1000 extension successfull
the extension make call out using the sip server is fail
and the error :
sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
nta: for "sip.yeah.com" query "sip.yeah.com" A (cached)
nua(0x9473108): call state changed: init -> calling, sent offer
soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c], [(nil)]) called
nua(0x9473108): event i_state INVITE sent
nua(0x9473108): event r_invite 503 DNS Error
nua(0x9473108): call state changed: calling -> init
nua(0x9473108): event i_state 503 DNS Error
nua(0x9473108): event i_terminated 503 DNS Error

i create 3 file:
    1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for register to sip server
    1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/default for outbound call
    1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for incomming call

the content of 071234.xml :

the content of 00_inbound_did_fpt.xml:






the content of 01_fpt.net.xml:





Từ: David Ponzone <david.ponzone@gmail.com>
Đến: freeswitch-users@lists.freeswitch.org
Gửi ngày: 14:26:13, Thứ Ba, 30 tháng 3 2010
Chủ đề: Re: [Freeswitch-users] need help on call to gateway

False,

I think Brian is sleeping so I give you a quick answer.

For OUTBOUND:

Following my previous mail, you have to check if your user 1000 has the default context configured.

If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.

(You may also remove from this path the remaining .xml files from the default install)

In 01_whatever.xml, put:







So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.

For INBOUND:

In conf/dialplan/public/, add a file named whatever.xml with the following content:







Here, YOUR_DID is the DID number your provider allocated to you.

That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.

Of course, after doing all this, reloadxml.

It will perhaps not fit in your config exactly, but you should get the idea to get started.

David Ponzone Direction Technique

email: david.ponzone@ipeva.fr

tel: 01 74 03 18 97

gsm: 06 66 98 76 34

Service Client IPeva

tel: 0811 46 26 26

www.ipeva.fr - www.ipeva-studio.com

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

Le 30/03/2010 à 09:01, false a écrit :

Hi Brian

could you guide me more to config on freeswitch

i want xilte with user 1000 can make call out to sip server
and the prefix when make call out to sip server : 0
and receive incomming call from SIP server

what file name i should create

Thank you

--- Ngày Thứ 3, 30/03/10, Brian West <brian@freeswitch.org> đã viết:

Từ: Brian West <brian@freeswitch.org>
Chủ đề: Re: [Freeswitch-users] need help on call to gateway
Đến: freeswitch-users@lists.freeswitch.org
Ngày: Thứ Ba, 30 tháng 3, 2010, 4:37

its looking in context default... not public.

/b

On Mar 29, 2010, at 11:33 PM, false wrote:

Hi all

i use freeswitch to register to sip server
the domain is not real: sip.yeah.com
the outbound proxy : 118.69.145.5

i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder

    sip.yeah.com">


        sip.yeah.com"/>







so i use xlite register to freeswitch with user 1000
i start freeswitch , freeswitch start ok,
freeswitch register to sip server successfull with username 071234
xlite register to freeswitch successfull

so i edit the public.xml
for incomming call from sip server, i will forward to extension 1000
    sip.yeah.com">





but when i use xlite make call out from freeswitch to sip server and get no routing

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