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 : </include> <gateway name="071234"> <param name="username" value="071234"/> <param name="from-user" value="071234"/> <param name="proxy" value="sip.yeah.com"/> <param name="username" value="071234"/> <param name="password" value="hadn"/> <param name="register-proxy" value="x.x.x.x"/> <param name="register" value="true"/> <param name="retry-seconds" value="10"/> </gateway> </include>
the content of 00_inbound_did_fpt.xml: <include> <extension name="public_did_fpt"> <condition field="destination_number" expression="071234"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include>
the content of 01_fpt.net.xml: <include> <extension name="sip.yeah.com"> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="sofia/gateway/071234/$1"/> </condition> </extension> </include>
Từ: David Ponzone <david.ponzone@gmail.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 14:26:13, Thứ Ba, 30 tháng 3 2010 Chủ đề: Re: [Freeswitch-users] need help on call to gateway
False,
I think Brian is sleeping so I give you a quick answer.
For OUTBOUND:
Following my previous mail, you have to check if your user 1000 has the default context configured. If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/. (You may also remove from this path the remaining .xml files from the default install) In 01_whatever.xml, put:
<include> <extension name="Route_to_sip.yeah.com"> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="sofia/gateway/ sip.yeah.com/0$1"/> </condition> </extension> <include>
So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.
For INBOUND:
In conf/dialplan/public/, add a file named whatever.xml with the following content:
<include> <extension name="Inbound_from_sip.yeah.com"> <condition field="destination_number" expression="^YOUR_DID$"> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include>
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 </include> <gateway name="sip.yeah.com"> <param name="username" value="071234"/> <param name="from-user" value="071234"/> <param name="proxy" value="sip.yeah.com"/> <param name="username" value="071234"/> <param name="password" value="1234"/> <param name="register-proxy" value="118.69.145.5"/> <param name="register" value="true"/> <param name="retry-seconds" value="10"/> </gateway> </include>
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 <extension name="sip.yeah.com"> <condition field="destination_number" expression="071234"> <action application="transfer" data="1000 XML default"/> </condition> </extension>
but when i use xlite make call out from freeswitch to sip server and get no routing
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