Yeah, I've tried modifying the RPID header... It looks like that's working for me. I had to make up a hard coded display name...
This is slightly off topic but in my current T1's I need to add in an Asterisk 'wait' command for the facility IE with the Calling name to be sent through. ie. --> http://www.voipinfo.org/wiki/view/CallerID Is there something similar for OpenSIPS? - Julian On Wed, Mar 18, 2009 at 9:49 AM, Brett Nemeroff <[email protected]> wrote: > It's not really standards compliant, but you can do it. I'm not sure why you > want to take stuff out of the contact header and stick it in from. From > shouldn't ever be changed, if you can help it. If your really trying to > change the resultant display name (ie: caller id) then go for manipulating / > adding RPID headers instead. > http://www.opensips.org/html/docs/modules/1.4.x/auth.html#append-rpid-hf-no-params > > I've had to rewrite "From" headers because non-compliant carriers have > insisted on using data in the from header for E911 call routing.. idiots.. > :P > If you insist on changing the From header; this may help.. you won't be able > to do it the way you are presently trying.. > take a look here: > http://www.opensips.org/index.php?n=Resources.DocsTipsFaqs > and here: > http://www.opensips.org/html/docs/modules/1.4.x/uac.html#id227417 > > On Wed, Mar 18, 2009 at 2:37 PM, Julian Yap <[email protected]> wrote: >> >> I just tested and this does not work: >> remove_hf("From"); >> append_hf("From: $ct;$ft\r\n"); >> >> So basically I want to rewrite the From header by using the details >> from the Contact header. >> >> Any suggestions? >> >> - Julian >> >> On Wed, Mar 18, 2009 at 1:15 AM, Julian Yap <[email protected]> wrote: >> > I have a scenario where the PSTN to SIP gateway (AudioCodes) I am >> > using sets the From header to 'anonymous' when it does not receive a >> > Calling Name from the PSTN side. >> > >> > The modified INVITE from the gateway then looks like this (changed >> > some numbers and IP's): >> > From: "anonymous" <sip:[email protected]>;tag=1c49690767. >> > To: <sip:[email protected];user=phone>. >> > CSeq: 1 INVITE. >> > Contact: <sip:[email protected]>. >> > >> > When the gateway does receive the Calling Name from the PSTN, it looks >> > like this: >> > From: "HONOLULU HI" >> > <sip:[email protected]>;tag=1c1248847826. >> > To: <sip:[email protected];user=phone>. >> > CSeq: 1 INVITE. >> > Contact: <sip:[email protected]>. >> > >> > In the first instance, I want to re-write the From header because I do >> > in fact have the calling number from the Contact header. In theory, >> > the PSTN gateway should sort this out for me and not send me the >> > 'anonymous' From header but I've searched the manuals and it doesn't. >> > Anyone else encountered this? >> > >> > This is the IF statement that satisfies the criteria: >> > if($fu=='sip:[email protected]' && >> > $ct=~"^<sip:[2-9][0-9]{2}[2-9][0-9]{6}@") >> > { >> > xlog("L_INFO", "fix anonymous\n"); >> > } >> > >> > The final From would be: >> > From: $ct;$ft >> > --> That is Contact header variable;From tag >> > >> > I had a look at the UAC module but using the function >> > uac_replace_from(), I don't know how to strip the '<' and '>' of the >> > Contact header to satisfy the arguments of the function. Is it >> > possible? I'm thinking that using the UAC is preferable to what I'm >> > proposing below. >> > >> > Does this method sound sane? Or is it dangerous?: >> > remove_hf("From"); >> > append_hf("From: $ct;$ft\r\n"); >> > >> > Thanks, >> > Julian >> > >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
