Greetings,
I have spent a considerable amount of time the past few days trying to figure this one out. But after giving up on Google searches I am wondering if anyone here has any ideas.
I am passing calls from an incoming SS7 trunk (libss7) outgoing on a SIP connection to a VOIP provider. Normal calls work fine, however, when I try to pass any sort of additional parameters such as “Original Called Number” or “LRN” or “Number Dipped” I am having all sorts of troubles.
I found the SIP standard for including a dipped number and LRN here (voip-info.org/wiki/view/SIP+ … Parameters)
Example:
INVITE sip:8085551212;rn=8087321000;npdi=yes@192.168.100.100:5060 SIP/2.0
I’ve tried several different things like setting the ${SIP_URI_OPTIONS} variable, ${VXML_URI} variable.
Here is what I have so far, let me know if I’m completely off on this.
exten => _X.,1,Set(SIP_URI_OPTIONS=rn=1234560000)
exten => _X.,1,Set(SIP_URI_OPTIONS=npdi=yes)
exten => _X.,n,Dial(${SIPTRUNK}/1${EXTEN})
exten => _X.,n,Hangup()
When I capture the SIP messaging with wireshark I do not see any of the additional SIP options that I add. It is always the same. I also tried using SipAddHeader , however, I’m not able to modify the To: header only add an additional header which isn’t how the SIP RFC 4694 specify.
Asterisk version: 1.6.0.1
Any help is appreciated.
Thanks,