I am stuck in channel PJSIP trying to see the real flow of SIP
messages, what in regular sip
we used to type sip set debug on
Also, is there an automated way to convert sip.conf options to pjsip.conf?
Philip
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I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)},
so if I use PJSIP, for scalability, how do I read what the signalling
IP where the inbound call is coming from and what is the inbound
codec?
You would think that the new channel would set those variables up, isn't it?
Philip
CDR wrote:
I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)},
so if I use PJSIP, for scalability, how do I read what the signalling
IP where the inbound call is coming from and what is the inbound
codec?
You would think that the new channel would set those variables up, isn't
On Mon, Sep 23, 2013 at 5:20 AM, CDR vene...@gmail.com wrote:
I am stuck in channel PJSIP trying to see the real flow of SIP
messages, what in regular sip
we used to type sip set debug on
Also, is there an automated way to convert sip.conf options to pjsip.conf?
First, please keep in mind
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Doug
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OK list. Just in case anyone cares. I did figure out (sort of) why I was
not receiving any IAX packets.
http://lists.digium.com/pipermail/asterisk-users/2013-September/280590.html
My modules.conf file was the same one I had previously used in 1.8 and
it specifically loaded necessary modules
smime.p7s
Description: S/MIME Cryptographic Signature
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On Sun, Sep 22, 2013 at 1:22 PM, Digium's Asterisk Development Team
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On Monday, September 23rd, 2013, all Asterisk mailing lists may have
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