<PRE>I just upgraded my two asterisk boxes to 1.0.5 stable and I've noticed
that callerid is not functioning properly. My setup looks like this:
SIP Phone <--> SER <--> Asterisk <--> Asterisk <---> PSTN
No iax is being used at this time.
The problem can be best described by the following scenarios:
1.) SIP to PSTN call: When a SIP phone calls a PSTN bound number, the
callerid displayed on the PSTN phone is the number of the PSTN phone
instead of the SIP phone's number.
2.) PSTN to SIP call: When a PSTN phone calls a SIP Phone number, the
callerid displayed on the SIP phone is the number of the SIP phone instead
of the PSTN phone's number.
</PRE><PRE>For both scenarios - ${CALLERID}, ${EXTEN}, and ${CALLERIDNUM} all have the number of the called phone for ZAP to SIP, SIP to ZAP, and SIP to SIP. I have noticed that explicitly declaring SetCallerID(${CALLERID}) before my dial seems to fixe this issue for only the ZAP to SIP piece. In the next Asterisk where a SIP to SIP relay is occurring ${CALLERID} ends up matchign ${EXTEN} again.
This is causing some havoc with users calling cell phone from SIP phones.
Some users are being dumped into certain company's cell phone voicemail
because the callerid is keyed to the called phone's number.
Has anyone else experienced this problem with 1.0.5 stable? I checked the
bugs.digum.com page and found a similar bug with regard to the call being
delivered to the manager API. Also, I searched the configs and I did not
see any new settings related to callerid. If this is a simple
configuration change introduced into version 1.0.5, any info would be
greatly appreciated.
</PRE>