1. diff chan_sip.c.bak chan_sip.c

From: "Andrei Petrescu" 
To: <support@voip-info.org>
Sent: Tuesday, April 04, 2006 6:51 PM
Subject: UPDATE: caller id name , asterisk and d-link dvg1120m


after a few late nights with much midnight oil burnt,
i found out that:

Asterisk cannot transmit CallerIDName to the DVG1120s

And i started to investigate:
1) x-lite (using the dvg as a "proxy") transmits it fine, all phones display it ok
2) same x-lite, same settings, but going through asterisk - not

3) started sniffing both xlite and asterisk at work:
The only difference was in the From: SIP header
x-lite says
From: test22 <sip:3333@172.18.174.175>;tag=263456542

//ip addresses and other personal data altered - kept same format

asterisk says something like:
From: "test22" <sip:22@172.18.174.173>;tag=ax3y3z4e2b

it seems that the DVG reacts badly to the quotes in the SIP headers
BUT on the bright side, everyone else accepts the header WITHOUT the quotes just fine

so - get rid of them

Unfortunately, the sip header protocol is not configurable in this respect in asterisk,
so i had to hack very deep. (Limbo rock in the background - "how low-level can you go ?")

Finally i came up with:

1) cd ....../asterisk-1.2.4/channels    ;#  in the source tree for asterisk
2) rm chan_sip.o chan_sip.so

make the following adjustment to chan_sip.c:
--- chan_sip.c-patched  2006-04-04 23:51:48.000000000 -0400
+++ chan_sip.c-out-of-the-box     2006-01-28 08:52:15.000000000 -0500
@@ -28,7 +28,6 @@
   * \todo Better support of forking
   */

-int my1,my2;

  #include <stdio.h>
  #include <ctype.h>
@@ -3686,17 +3685,6 @@
                                 var = aliases[x].shortname;
         }

-/* VVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVVV
- MY PATCH FOR CallerID Name from Asterisk to D-Link DVG-1120S  */
-if (!strcmp("From",var)) { /* the From header needs quotes stripped */
-       if (value[0]==34) { /*opening quotes */
-       for (my1=0,my2=1;value[my1+1] ;my1++,my2++) {
-                       if (value[my2]==34) {my2++;}  /* jump over closing quotes */
-                       value[my1]=value[my2] ; }
-       value[my1]=0;
-       }
-}
-/* ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ */
         snprintf(req->header[req->headers], sizeof(req->data) - req->len - 4, "%s: %s\r\n", var, value);
         req->len += strlen(req->header[req->headers]);
         req->headers++;



  cd ..
  make all
  mv /usr/lib/asterisk/modules/chan_sip.so /usr/lib/asterisk/modules/chan_sip.so-out-of-the-box
  cp ....../asterisk-1.2.4/channels/chan_sip.so /usr/lib/asterisk/modules/chan_sip.so
  asterisk -rx "restart now"



========================================

Also the manual should be updated - there are some errorneous statements in that
page about the FXO port:

one can use the ACR feature to route some calls to the FXO port even when the SIP connection
is on

Also incoming calls will be received on that port (useful if one still has a Bell line
or a condo-style intercom)

Unfortunately all calls unresolved by the sip server are also dumped to the FXO port.


Another usage of it would be to keep it disconnected and make ACR route all calls to it
except a list of permitted valid numbers which will go through VoIP (e.g. only local
and free and emergency - all toll calls dumped on the empty fxo socket and getting dead air)
Not user friendly, but efficient in managing toll calls.