need help on call to gateway
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 nta: received INVITE sip:071234@192.168.1.90;user=phone SIP/2.0 (CSeq 3621) nta: INVITE (3621) going to a default leg nua(0xb6e20f90): adding session usage nta: sent 100 Trying for INVITE (3621) nua(0xb6e20f90): event i_invite 100 Trying nua(0xb6e20f90): call state changed: init -> received, received offer nua(0xb6e20f90): event i_state 100 Trying 432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000@192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006] 446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default 455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000@192.168.1.90 to enum[071234@default] 566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting 566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000@192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION] nua(0xb6e20f90): recv signal r_respond 404 Not Found i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch please help, Thank you Hãy cùng chat với bạn bè trên Yahoo! Pingbox mới toanh! Tạo một không gian riêng để chat trên blog là chuyện nhỏ. http://vn.messenger.yahoo.com/pingbox/
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
nta: received INVITE sip:071234@192.168.1.90;user=phone SIP/2.0 (CSeq 3621) nta: INVITE (3621) going to a default leg nua(0xb6e20f90): adding session usage nta: sent 100 Trying for INVITE (3621) nua(0xb6e20f90): event i_invite 100 Trying nua(0xb6e20f90): call state changed: init -> received, received offer nua(0xb6e20f90): event i_state 100 Trying 432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000@192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006] 446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default 455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000@192.168.1.90 to enum[071234@default] 566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting 566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000@192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION] nua(0xb6e20f90): recv signal r_respond 404 Not Found
i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch
please help,
Thank you
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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 ____________________________________________________________ Tại sao đàn ông lại thích nhìn phụ nữ có mái tóc dài hơn? Khám phá tại Yahoo! Hỏi & Đáp http://vn.answers.yahoo.com
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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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
________________________________
Tốt hơn, thoáng gọn hơn, nhanh hơn - Trải nghiệm Yahoo! Mail mới hôm nay! _______________________________________________
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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
Tốt hơn, thoáng gọn hơn, nhanh hơn - Trải nghiệm Yahoo! Mail mới hôm nay!_______________________________________________ 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 http://www.freeswitch.org
Thiết kế ngay một Pingbox độc đáo cho riêng bạn! Tạo một nơi để chat trên blog là chuyện nhỏ._______________________________________________ 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 http://www.freeswitch.org
Hi David the domain sip.yeah.com cannot resolve in the FS host the thing here is FW support<param name="register-proxy" value="x.x.x.x"/> 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 : </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
________________________________
Tốt hơn, thoáng gọn hơn, nhanh hơn - Trải nghiệm Yahoo! Mail mới hôm nay!_______________________________________________
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 http://www.freeswitch.org
________________________________ Thiết kế ngay một Pingbox độc đáo cho riêng bạn! Tạo một nơi để chat trên blog là chuyện nhỏ._______________________________________________ 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 http://www.freeswitch.org
Lướt web nhanh hơn. Internet Explorer 8 tối ưu hóa cho Yahoo!, tự động khởi động 2 trang bạn thích mỗi lần mở trình duyệt. Tải IE8 tại đây! http://downloads.yahoo.com/vn/internetexplorer/
I guess you need to put the IP of the proxy also in: <param name="proxy" value="x.x.x.x"/> 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 <param name="register-proxy" value="x.x.x.x"/> 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 : </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
Tốt hơn, thoáng gọn hơn, nhanh hơn - Trải nghiệm Yahoo! Mail mới hôm nay!_______________________________________________ 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 http://www.freeswitch.org
Thiết kế ngay một Pingbox độc đáo cho riêng bạn! Tạo một nơi để chat trên blog là chuyện nhỏ._______________________________________________ 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 http://www.freeswitch.org
Yahoo! Mail nay NHANH HƠN - Thử ngay! _______________________________________________ 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 http://www.freeswitch.org
Hi David when change the <param name="proxy" value="x.x.x.x"/> 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 <david.ponzone@gmail.com> 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: <param name="proxy" value="x.x.x.x"/> 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 <param name="register-proxy" value="x.x.x.x"/> 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 : </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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Then either provide DNS to that system so that it can resolve the address, either create a static binding in /etc/hosts file on that host for sip.yeah.com On Wed, 31 Mar 2010 20:36:37 +0800 (SGT), false wrote: 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 NGY: 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 [1] 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 ĐẾN: freeswitch-users@lists.freeswitch.org [3] GỬI NGY: 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 [4] 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 ĐẾN: freeswitch-users@lists.freeswitch.org [6] GỬI NGY: 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 [7] 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 __ đã viết: Từ: Brian West Chủ đề: Re: [Freeswitch-users] need help on call to gateway Đến: freeswitch-users@lists.freeswitch.org [9] 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 [10] 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 ------------------------- Tốt hơn, thoáng gọn hơn, nhanh hơn - Trải nghiệm Yahoo! Mail mới hôm nay! [11]_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [12] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [13] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [14] http://www.freeswitch.org [15] ------------------------- Thiết kế ngay một Pingbox độc đáo cho riêng bạn! [16] Tạo một nơi để chat trên blog là chuyện nhỏ._______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [17] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [18] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [19] http://www.freeswitch.org [20] ------------------------- Yahoo! Mail nay NHANH HƠN - Thử ngay! [21]_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [22] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [23] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [24] http://www.freeswitch.org [25] ------------------------- Yahoo! Mail nay nhanh và nhiều không gian thoáng hơn. Hãy trải nghiệm ngay hôm nay! [26] Links: ------ [1] mailto:david.ponzone@ipeva.fr [2] mailto:david.ponzone@gmail.com [3] mailto:freeswitch-users@lists.freeswitch.org [4] mailto:david.ponzone@ipeva.fr [5] mailto:david.ponzone@gmail.com [6] mailto:freeswitch-users@lists.freeswitch.org [7] mailto:david.ponzone@ipeva.fr [8] mailto:brian@freeswitch.org [9] mailto:freeswitch-users@lists.freeswitch.org [10] http://sip.yeah.com [11] http://vn.mail.yahoo.com [12] mailto:FreeSWITCH-users@lists.freeswitch.org [13] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [14] http://lists.freeswitch.org/mailman/options/freeswitch-users [15] http://www.freeswitch.org [16] http://sg.rd.yahoo.com/vn/messenger/pingbox/mailtagline/*http://vn.messenger... [17] mailto:FreeSWITCH-users@lists.freeswitch.org [18] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] http://lists.freeswitch.org/mailman/options/freeswitch-users [20] http://www.freeswitch.org [21] http://vn.mail.yahoo.com [22] mailto:FreeSWITCH-users@lists.freeswitch.org [23] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] http://lists.freeswitch.org/mailman/options/freeswitch-users [25] http://www.freeswitch.org [26] http://vn.mail.yahoo.com
Hi Francios Thank you for you recommend from your recommendation, i add the host name : sip.yeah.com into the /etc/hosts but i am still getting error on DNS Failed with status DNS Error [503] it seems that FS needs resolve the sip.yeah.com from the DNS server. the FS support register with domain - domain is not real and the parameter = register-proxy. Is there the same paramater for make call out?? i do ping sip.yeah.com successfull but when i do nslookup and the DNS server response cannot find the domain name [root@localhost external]# nslookup sip.yeah.com Server: 8.8.8.8 Address: 8.8.8.8#53 ** server can't find sip.yeah.com: NXDOMAIN [root@localhost external]# ping sip.yeah.com PING sip.yeah.com (118.69.239.250) 56(84) bytes of data. 64 bytes from sip.yeah.com (118.69.239.250): icmp_seq=2 ttl=250 time=6.84 ms Thank you Ha` ________________________________ Từ: François Legal <devel@thom.fr.eu.org> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 20:51:17, Thứ Tư, 31 tháng 3 2010 Chủ đề: Re: [Freeswitch-users] Về: Về: Về: need help on call to gateway Then either provide DNS to that system so that it can resolve the address, either create a static binding in /etc/hosts file on that host for sip.yeah.com On Wed, 31 Mar 2010 20:36:37 +0800 (SGT), false <cucku.cucku@yahoo.com.vn> wrote: 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
________________________________ Tốt hơn, thoáng gọn hơn, nhanh hơn - Trải nghiệm Yahoo! Mail mới hôm nay!_______________________________________________ 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 http://www.freeswitch.org
________________________________ Thiết kế ngay một Pingbox độc đáo cho riêng bạn! Tạo một nơi để chat trên blog là chuyện nhỏ._______________________________________________ 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 http://www.freeswitch.org
Yahoo! Mail nay NHANH HƠN - Thử ngay!_______________________________________________ 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 http://www.freeswitch.org
Yahoo! Mail nay nhanh và nhiều không gian thoáng hơn.Hãy trải nghiệm ngay hôm nay!
Xem hình có âm mới với ông Bụt và cô Tấm…! Tải phiên bản Yahoo! Messenger mới nhất bằng tiếng Việt tại đây. http://vn.messenger.yahoo.com
Why not get a provider that actually does this stuff correctly? My blood boils ever time I hear of this stupid setup. /b On Apr 1, 2010, at 9:17 AM, false wrote:
Hi Francios
Thank you for you recommend
from your recommendation, i add the host name : sip.yeah.com into the /etc/hosts
but i am still getting error on DNS Failed with status DNS Error [503]
it seems that FS needs resolve the sip.yeah.com from the DNS server. the FS support register with domain - domain is not real and the parameter = register-proxy. Is there the same paramater for make call out??
i do ping sip.yeah.com successfull but when i do nslookup and the DNS server response cannot find the domain name
[root@localhost external]# nslookup sip.yeah.com Server: 8.8.8.8 Address: 8.8.8.8#53
** server can't find sip.yeah.com: NXDOMAIN
[root@localhost external]# ping sip.yeah.com PING sip.yeah.com (118.69.239.250) 56(84) bytes of data. 64 bytes from sip.yeah.com (118.69.239.250): icmp_seq=2 ttl=250 time=6.84 ms
Thank you Ha`
Le Thu, 1 Apr 2010 22:17:48 +0800 (SGT), false <cucku.cucku@yahoo.com.vn> a écrit :
Thank you for you recommend
from your recommendation, i add the host name : sip.yeah.com into the /etc/hosts
but i am still getting error on DNS Failed with status DNS Error [503]
This is because /etc/host.conf isn't configured to first pick a name into /etc/hosts before a DNS resolution occurs (man host.conf) But it is usually better to create a specific DNS zone, such as: ; Zone for *.fusionpbx.set $TTL 1D $ORIGIN fusionpbx.set. ; @ IN SOA fusionpbx.set. hostmaster.fusionpbx.set. ( 2010030401 ; serial 8H ; refresh 4H ; retry 4W ; expire 1D); ; DNS names IN NS ns1.fusionpbx.set. IN NS ns2.fusionpbx.set. ; MX IN MX 10 mail.fusionpbx.set. ; Virtual Machines & Hosts IN A 192.168.1.25 ns1 IN A 192.168.1.1 ns2 IN A 192.168.1.2 mail IN A 192.168.1.50 *.fusionpbx.set. IN A 192.168.1.25 This way, you can address any subdomain (but may be that's not what you're looking for?) -- "Take that, you hostile sons-of-bitches!" -- James Coburn, in the finale of _The_President's_Analyst_
Hi Jean yeah, your way solves my issue Thank you ________________________________ Từ: Jean-Yves F. Barbier <12ukwn@gmail.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 21:50:05, Thứ Năm, 1 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] : need help on call to gateway Le Thu, 1 Apr 2010 22:17:48 +0800 (SGT), false <cucku.cucku@yahoo.com.vn> a écrit :
Thank you for you recommend
from your recommendation, i add the host name : sip.yeah.com into the /etc/hosts
but i am still getting error on DNS Failed with status DNS Error [503]
This is because /etc/host.conf isn't configured to first pick a name into /etc/hosts before a DNS resolution occurs (man host.conf) But it is usually better to create a specific DNS zone, such as: ; Zone for *.fusionpbx.set $TTL 1D $ORIGIN fusionpbx.set. ; @ IN SOA fusionpbx.set. hostmaster.fusionpbx.set. ( 2010030401 ; serial 8H ; refresh 4H ; retry 4W ; expire 1D); ; DNS names IN NS ns1.fusionpbx.set. IN NS ns2.fusionpbx.set. ; MX IN MX 10 mail.fusionpbx.set. ; Virtual Machines & Hosts IN A 192.168.1.25 ns1 IN A 192.168.1.1 ns2 IN A 192.168.1.2 mail IN A 192.168.1.50 *.fusionpbx.set. IN A 192.168.1.25 This way, you can address any subdomain (but may be that's not what you're looking for?) -- "Take that, you hostile sons-of-bitches!" -- James Coburn, in the finale of _The_President's_Analyst_ _______________________________________________ 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 http://www.freeswitch.org Yahoo! Mail nay NHANH HƠN - Thử ngay! http://vn.mail.yahoo.com
False, The logic in FS is that a call will go through the dialplan context attached to the SIP profile that the call is coming into. So if you have the SIP Profile internal bound to 10.0.0.1:5080 for instance, and then you receive an INVITE on 10.0.0.1:5080, FS will parse the dialplan context configured in internal.xml (by default, public). For registered users though, you can alter this behavior by specifying a context in the user variables: <variable name="user_context" value="your_own_context_for_this_user"/> If you don't, the context configured in the SIP profile that the user is sending INVITEs to will be used. 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 06:33, false a écrit :
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
nta: received INVITE sip:071234@192.168.1.90;user=phone SIP/2.0 (CSeq 3621) nta: INVITE (3621) going to a default leg nua(0xb6e20f90): adding session usage nta: sent 100 Trying for INVITE (3621) nua(0xb6e20f90): event i_invite 100 Trying nua(0xb6e20f90): call state changed: init -> received, received offer nua(0xb6e20f90): event i_state 100 Trying 432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000@192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006] 446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default 455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000@192.168.1.90 to enum[071234@default] 566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting 566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000@192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION] nua(0xb6e20f90): recv signal r_respond 404 Not Found
i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch
please help,
Thank you
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participants (5)
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Brian West
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David Ponzone
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false
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François Legal
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Jean-Yves F. Barbier