Re: [Asterisk-Users] SIP URLs
On 08-Dec-2004, Alex Barnes wrote: The reason its probably not working is because your Xlite is sending the request to the Asterisk. The Asterisk isn't a SIP proxy hence all it does is see if it recognises the addressee. This isn't strictly true. A SIP proxy is one solution to this demand, but there's no reason asterisk can't be made to route URI-based calls just as easily as it routes number-based calls. The mechanics for asterisk to route a call to 555-1212 are no different than the mechanics involved in asterisk routing a call to sip:[EMAIL PROTECTED]. I have published my dialplan which allows asterisk to route both URI-target and number-target calls. http://slacker.com/~nugget/asterisk7.php -- David McNett [EMAIL PROTECTED] http://slacker.com/~nugget/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP URLs
-Original Message- From: Dan Goscomb [mailto:[EMAIL PROTECTED] Sent: 07 December 2004 15:38 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP URLs I have set up an asterisk server and can successfully call between extensions using SIP. i wish to be able to call other sip users using URLs such as sip:[EMAIL PROTECTED] and have no idea how this works... every time i try it (using X-Lite soft phone), i just get a 404: not found error. The reason its probably not working is because your Xlite is sending the request to the Asterisk. The Asterisk isn't a SIP proxy hence all it does is see if it recognises the addressee. You either need a proxy in the middle of your SIP UA's and the Asterisk or more simply (if u have only a few UA's) do not set an outbound proxy address. The support for this differs greatly from SIP UA to SIP UA, for instance some require it. Also some phones have dial plans that can be setup to make life of the user much eaiser. e.g. 6XXX always gets sent to IP of * or unless a domain part has been explicitly entered. If your phones / UA's can't do this then you will be stuck dialing the full asterisk IP / DNS name every time you call. If a SIP proxy sounds like your best bet then SER gets banded around the mailing list very often tho I cannot atest to it personally. Cheers Alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP URLs
Cheers i realised this last night and now have SER set up i can call between phones on SER, and to extensions handled on asterisk (for example voicemail). However... if i dial an extension which used to be assigned to a SIP phone, it tells me the user is on the phone... is there any way to get this sorted so i may call UA - SER - asterisk - SER - UA so i can dial UAs on SER with their asterisk extension number? Cheers Dan On Wed, 2004-12-08 at 10:23 +, Alex Barnes wrote: -Original Message- From: Dan Goscomb [mailto:[EMAIL PROTECTED] Sent: 07 December 2004 15:38 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP URLs I have set up an asterisk server and can successfully call between extensions using SIP. i wish to be able to call other sip users using URLs such as sip:[EMAIL PROTECTED] and have no idea how this works... every time i try it (using X-Lite soft phone), i just get a 404: not found error. The reason its probably not working is because your Xlite is sending the request to the Asterisk. The Asterisk isn't a SIP proxy hence all it does is see if it recognises the addressee. You either need a proxy in the middle of your SIP UA's and the Asterisk or more simply (if u have only a few UA's) do not set an outbound proxy address. The support for this differs greatly from SIP UA to SIP UA, for instance some require it. Also some phones have dial plans that can be setup to make life of the user much eaiser. e.g. 6XXX always gets sent to IP of * or unless a domain part has been explicitly entered. If your phones / UA's can't do this then you will be stuck dialing the full asterisk IP / DNS name every time you call. If a SIP proxy sounds like your best bet then SER gets banded around the mailing list very often tho I cannot atest to it personally. Cheers Alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dan Goscomb [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
