<DIV>I've created many aliases. I've found that if a single alias points to two different sip addresses, that it will only ring through to one of them.</DIV>
<DIV> </DIV>
<DIV>i.e. a call come in from the PSTN for phone number 6600 [my PSTN service is configured to send only 4 digits]. This number (6600) is an alias for sip:info@mydomain.com. </DIV>
<DIV> </DIV>
<DIV>If I then apply the following</DIV>
<DIV> </DIV>
<DIV>root # serctl alias add 6600 sip:userb@mydomain.com</DIV>
<DIV> </DIV>
<DIV>I can add a second address for that alias, but the call is still redirected to sip:info@mydomain.com. </DIV>
<DIV> </DIV>
<DIV>Are you suggesting that I remove the 6600 alias and recreate it so it points to sip:userb@mydomain.com?</DIV>
<DIV> </DIV>
<DIV>What if I actually want it to ring both sip:info@mydomain.com and sip:userb@mydomain.com?</DIV>
<DIV> </DIV>
<DIV>Can I use the add contact feature in serweb for this? If I log into serweb as "info" and then add sip:userb@mydomain.com as a contact, the call still doesn't get forked to userb.<BR></DIV>
<DIV>Any ideas?</DIV>
<DIV> </DIV>
<DIV>Thanks,</DIV>
<DIV>G.</DIV>
<DIV><BR><B><I>Jan Janak <jan@iptel.org></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">
<P>Hello,<BR><BR>On 18-09 18:26, Gavin Bensom wrote:<BR>> Hello All,<BR>> <BR>> I'm interested in knowing if there is a way to do the equivalent of append_branch or rewriteuri/rewriteuser without changing the ser.cfg file and restarting ser.<BR>> <BR>> The scenario is this. I have a main line that is typically answered by one individual. Lets say he goes to lunch and the line is going to be covered by another individual. How could the individuals involved set this up by themselves using serweb or another method? I don't want to have to modify ser.cfg and restart the server every time calls need to be directed differently.<BR><BR>You can use serctl utility and create an alias. The alias will point<BR>to the person to which the calls should be redirected</P>
<P><BR>> I tried adding a contact in serweb. From user A's account I tried adding user B's username or alias. Neither seemed to work. When I dial user A, user B's line does not ring. <BR>> <BR><BR>If you add a contact in serweb then the call will be forked--that<BR>means both phones will ring.<BR><BR>> Any ideas? I think all I need is a simple redirect function that will automatically redirect calls from one user to another without needed to change the config file and restart the server.<BR><BR>Right now you can use serctl. We are working on some more advanced<BR>support for call-redirection, but there is no time line yet.<BR><BR>> Also, is there anything special I need to do to get serweb to display accounting and missed call information? Serweb is running but is not displaying this info. I believe I complied with accounting turned on but I can't remember. Is there any way to check what modules are enabled or what config switches were set during complie?<BR><BR>You must
load acc module and set flag for transactions that should be<BR>accounted, see acc module documentation for more details.<BR><BR>Jan.</P></BLOCKQUOTE><p><hr SIZE=1>
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