We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection_permanent=no How do I achieve the same with chan_s..
Author : Benoit Panizzon
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Hi Im attempting to use ICE to be able to present all possible RTP transports to peers. 16.28.0~dfsg-0+deb11u2 (I know its old, but unfortunately Asterisk was removed from debian stable and the version in sid is just broken (opus + voicemail dont w..
Hey! I just upgraded our machines from Ubuntu focal to jammy. A separate package asterisk-opus does not exist any more. https://launchpad.net/ubuntu/+source/asterisk-opus/+changelog It looks like this is now included in the default packages. Requi..
Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communiti..
Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for ..
My last post did not make it back or to the archive… testing…
Mit freundlichen Grüssen
-Benoît Panizzon-
—
I m p r o W a r e A G-Leiter Comme..
Gang I noticed, that when I enable multiple codecs and rtp encrypting (generating a large SDP) invites with credentials do not get through anymore. So sniffed the connection and found that the IP packets have the dont fragment bit set, causing a V..
I can reproduce Asterisk 16.16.1 segfaulting in this situation: Asterisk configured with Application ReceiveFax. Incoming call with SDP: v=0 o=prt-cbl-sbc1 1418830458 1418830459 IN IP4 157.161.X.X s=sip call c=IN IP4 157.161.X.X t=0 0 m=audio 11828 RTP/..
Just ran into another weird issue… In Swiss Telephone Interconnection, ptime=20 is a requirement. So on our SBC we enforce the presence of ptime=20 to avoid issues. I have an asterisk with chan_sip in the LAB which behaves weirdly… Inbound SDP au..
I have come over a codec negotiation issue. A (asterisk) is sending in INVITE containing * opus (type 107) * g722 * alaw (type 8) B answers with 183 containing SDP * alaw a=sendrecv B then answer the call with 200 and NO SDP I suppose that result..