I discovered that if pjsip puts a call on hold, it does not trigger
music-on-hold using the Digium Switchvox pbx.
This appears to be because the version number in the sdp that pjsip generates is
overflowing a 32 bit int used in the Asterisk version used by switchvox.
The patch below (against 1.10) solves the issue, and I believe it should be
harmless to anyone else.
-m@
Index: callbridge/pjproject-1.10/pjmedia/src/pjmedia/endpoint.c
===================================================================
--- callbridge/pjproject-1.10/pjmedia/src/pjmedia/endpoint.c (revision 33933)
+++ callbridge/pjproject-1.10/pjmedia/src/pjmedia/endpoint.c (revision 33936)
@@ -343,7 +343,11 @@
pj_gettimeofday(&tv);
sdp->origin.user = pj_str("-");
- sdp->origin.version = sdp->origin.id = tv.sec + 2208988800UL;
+ // this used to add a big value that pushed the number past 32 bits
+ // and broke a few things in switchvox's asterisk, notably music-on-hold.
+ // This number is just a time in the not-too-distant past to get the
+ // numbers into a reasonable range.
+ sdp->origin.version = sdp->origin.id = tv.sec - 1304628710UL;
sdp->origin.net_type = STR_IN;
if (addr0->addr.sa_family == pj_AF_INET()) {
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