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📄 serfaq.sgml

📁 用来作为linux中SIP SERVER,完成VOIP网络电话中服务器的功能
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		</question>		<answer>		    <simpara>			FIXME.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Is it possible to get access to iptel's working copy of ser.cfg?		    </simpara>		</question>		<answer>		    <simpara>			We no longer disclose our operational policy to the public audience. The			configuration file is only available to our customers under <acronym>NDA</acronym>.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Can you send me detail for me to sign up for the commercial program. I would			also like detail on your levels of support you offer (ie...paid support)?		    </simpara>		</question>		<answer>		    <simpara>			See <ulink			    url="http://iptel.org/support/">http://iptel.org/support/</ulink>. Don't			    hesitate to approach us if you have any further questions.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			I need a method to determine if a called &uri; has an account on the system.			I found a function in the groups module,			<function>is_user_in</function>. but haven't yet figured out how to use it.		    </simpara>		</question>		<answer>		    <simpara>			The uri module's <function>does_uri_exist</function> is what you are seeking.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			We have a question regarding usrloc: where are the active sessions			being stored? We were not able to find any entries in MySQL.		    </simpara>		</question>		<answer>		    <simpara>			&ser; has no notion of active session. &ser; is a transaction-stateful proxy,			that means it knows about transactions (INVITE transaction, BYE			transaction), but it keeps no state if there is no transaction active.		    </simpara>		    <simpara>			That means, it keeps state when an INVITE comes and until a final response			to the INVITE is sent. The same for BYE.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Has the timeout for re-registering at the &ua; to be the same as in the tm			module mentioned in <filename>ser.cfg</filename> ?		    </simpara>		</question>		<answer>		    <simpara>			No.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			How do we deal with aliases? If for example uid=mic is authenticated, he is			available with sip address sip:mic@comnect.at. If I would want to be			addressable with michael.dosser@comnect.at how is this accomplished with ser			?		    </simpara>		</question>		<answer>		    <simpara>			Aliases are tied to user location. It uses a special table which has the			same structure as user location table and includes an entry for each			alias. You do not have to do something special for that. You have to create			this table and then you can use serctl to add aliases.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			I would like to know if &ser; supports <acronym>IPv6</acronym>. If not, do			you have any plans for it ? When ?		    </simpara>		</question>		<answer>		    <simpara>			Yes, &ser; supports <acronym>IPv6</acronym>.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Is there any way to change the default log file to a special SER.log file ?		    </simpara>		</question>		<answer>		    <simpara>			Try logging to stderr and redirecting it to a file:		    </simpara>		    <screen format="linespecific">ser -E     2>/tmp/ser.log</screen>		    <simpara>			(by default ser logs to syslog)		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Does anybody know anything about the P-Hint Header Field added from &ser;?		    </simpara>		</question>		<answer>		    <simpara>			<ulink url="http://www.iptel.org/ser/doc/seruser-html/c638.html#AEN729">			    http://www.iptel.org/ser/doc/seruser-html/c638.html#AEN729			</ulink>		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			When using the Jabber Gateway, some users get the following error from time to			time:		    </simpara>		    <simpara>			ERROR: Connection to Jabber server lost. You have to login to Jabber server			again (join again the conferences that you were participating, too).		    </simpara>		    <simpara>			sip_to_jabber_gateway says:		    </simpara>		    <simpara>			INFO: Your are now offline in Jabber network. Thank you for using &sip;-Jabber			gateway.		    </simpara>		    <simpara>			Do you know the reason why these messages appear and if it is possible to			avoid them?		    </simpara>		</question>		<answer>		    <simpara>			that usually occurs because Jabber server crashes or, for some unknown			reasons, the <acronym>TCP</acronym> connection with jabber server is			down. The second message is to inform the users that they are no more			connected to Jabber network.		    </simpara>		    <simpara>			There is no way to disable sending these messages. I may introduce a new			parameter to enable/disable them. But I am not sure it is a good idea (users			must be informed about the changes of the status).		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			I would like know if &ser; support also the transmission protocol			<acronym>TCP</acronym>, or <acronym>TLS</acronym>.		    </simpara>		</question>		<answer>		    <simpara>			Yes, &ser; supports <acronym>TCP</acronym>. <acronym>TLS</acronym> is work			in progress.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			I'm trying to rewrite the to domain, as in:		    </simpara>		    <simpara>			To: &lt;sip:19723236598@augustvoice.net&gt; ;user=phone.		    </simpara>		    <simpara>			I can't find any rewrite* function to rewrite the			to domain.		    </simpara>		</question>		<answer>		    <simpara>			The proxy is not supposed to touch To or From &uri;s. Only Request-&uri; can			be changed.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			When the messenger sends a message for the jabber gateway the following			error occurs:		    </simpara>		    <simpara>			<quote>ERROR: Your message was not sent. You do not have permissions to use the			    gateway.</quote>		    </simpara>		    <simpara>			What could be the problem here?		    </simpara>		</question>		<answer>		    <simpara>			You have to create the database for SIMPLE2Jabber gateway and after that you			have to associate &sip; users with Jabber IDs.		    </simpara>		    <simpara>			<ulink			    url="http://www.iptel.org/ser/doc/jabgw/xjab-manual.html#5._Admins_guide">			    http://www.iptel.org/ser/doc/jabgw/xjab-manual.html#5._Admins_guide			</ulink>		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Authentication doesn't work!!!  Where am I wrong?		    </simpara>		</question>		<answer>		    <simpara>			Windows Messenger needs same string both realm and SIP_DOMAIN, and it wants			to reach this address oh your network.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			I was wondering whether there is a ser module for			<acronym>SIP-CGI</acronym>. Or whether there are attempts at creating one?		    </simpara>		</question>		<answer>		    <simpara>			There is no <acronym>SIP-CGI</acronym> module for &ser;. The most similar,			though much simpler, functionality is provided by the exec module.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			How could I integrate &sip; and H323 together?		    </simpara>		</question>		<answer>		    <simpara>			You will need a &sip; to H.323 gateway. &ser; is a &sip; proxy only, it			cannot act as H.323 gateway.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			How can I contribute code ?		    </simpara>		</question>		<answer>		    <simpara>			FIXME.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			What, if anything, should the &ser; server do with a OPTIONS method?		    </simpara>		</question>		<answer>		    <simpara>			FIXME.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			What should I do to see the detailed debugs?		    </simpara>		</question>		<answer>		    <simpara>			Set debug=9 in your config script.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Is it possible for &ser; to forward a call to another phone if the recipient			does not answer?		    </simpara>		</question>		<answer>		    <simpara>			This could be achieved with sequential forking (by trying contacts in			decreasing q order), but sequential forking is not yet supported in &ser;.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Is it possible to configure &ser; so that it only has the latest			registration?		    </simpara>		</question>		<answer>		    <simpara>			FIXME.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			mkdir: cannot create directory `/usr/local/etc/ser': Permission denied			make: *** [/usr/local/etc/ser] Error 1		    </simpara>		    <simpara>			what may be the error ?		    </simpara>		</question>		<answer>		    <simpara>			You have no write permissions to the directory, try it again as root.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Can a proxy terminate a call ?		    </simpara>		</question>		<answer>		    <simpara>			No, proxy cannot do that because it is transaction stateful only.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			i want to know what accounting support is available using MySql. what			settings need to be done? are any records generated and placed in the			database? how do i access these records?		    </simpara>		</question>		<answer>		    <simpara>			FIXME.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			It seems that &ser; supports only strict routing. Please tell me if it			supports loose routing. if yes, do i have to enable it somehow?		    </simpara>		</question>		<answer>		    <simpara>			Yes, it does support loose routing as of 0.8.11, you don't have to enable it.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			I need to know if currently generated call records can be put into MySql			database. if yes, is there any tool available to view these records?		    </simpara>		</question>		<answer>		    <simpara>			Serweb can do it.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			If I want to use &ser; commercially do I need to purchase any license for			the same, as long as I'm not going to charge the customer for the &sip;			service but only for the termination of calls.		    </simpara>		</question>		<answer>		    <simpara>			&ser; is distributed under the &gpl; so you don't need to purchase any			license, you can use it freely.		    </simpara>		</answer>	    </qandaentry>	    <qandaentry>		<question>		    <simpara>			Is there a support for accounting ?		    </simpara>		</question>		<answer>		    <simpara>

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