I use both confbridge to bring several devices into a receive only or listen mode, then allow the one person on the phone to speak live over those devices. Works great.
However - now I would like to be able to "play a tone" into the conference before the person speaks.
How might that be accomplished?
HI LIST CAN U HELP ME
If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name
firstname.lastname@example.org/line1 email@example.com/line2 [line1] type=peer username=myaccount1 host=sip.myitsp.com [line2] type=peer username=myaccount2 host=sip.myitsp.com
If sip.myitsp.com directs a call to asterisk with a request line of:
INVITE line1@mybindaddr SIP/2.0
then it is matched to the line2 peer whereas it would probably be better matched to the line1 peer
Given these occassional errors on my Asterisk CLI:
[Jul 2 10:23:36] WARNING: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission firstname.lastname@example.org:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 2 10:23:49] WARNING: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission email@example.com:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response
E. g. I have the transmission numbers
on which packets timed out.
Is there any way I can determine the IP addresses where those packets referred to by these errors timed out on?
What is Sequence 102?
192.168.11.31 is my Asterisk…
I built an LXC container with an "image" of asterisk 11.18 precompiled and installed. It runs fine on the dev platform, which is a Dell R320 running Ubuntu 14.04LTS. I shutdown the container, tarred it up, and untarred on a Dell PE1850, also running Ubuntu 14.04LTS. The container itself is Ubuntu 14.04LTS. Both platforms as far as I know are amd64.
The container boots fine on the 1850, but trying to run asterisk segfaults. The source tree was still in the container, so I just did a make clean; make; make install. It now runs fine.
Is there some compile flag I…
I want to know if it is somehow possible for asterisk to consider new registration attempts instead of matching them with old nonce
Correct auth, but based on stale nonce received from '"test" < sip:firstname.lastname@example.org>;tagya401979bffd0d9o0'
I see messages like the one above, I understand it is because of existing expire value but would like the previous expire timer to reset and issue a new registration instead
Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked?
Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.
I see in my log file this:
Jun 30 21:44:26] NOTICE[C-000002f3] chan_sip.c: Call from '' (
18.104.22.168:5076) to extension '011972592675431' rejected because extension not found in context 'default'.
which is great its rejected - however in my sip.conf file I have
deny=0.0.0.0 permit=x.y.z.z/255.255.255.255 permit=a.b.c.d/255.255.255.255
So I'm expecting to deny everything and only allow the two addresses I have listed of which the 22.214.171.124 is not one of?
What is wrong with my permit/deny ?
At first should take a look to cable pinout (RAD documents) as pin 1,2, Transmit (output) and 4, 5 Receive (input) for Digium card you should use a straight cable (try to test with new cable one too). Second check Dahdi configuration parameters, use dahdi commands as; dahdi show status, service dahdi restart and check result, (could a mistake on parameter value on system.conf). Mc GRATH Ricardo
We have used AIS for disturbed Device State in the past, BLF and MWI, We are in the process of an update on one of our clustered systems, We are looking at XMPP and I found a few discussions on a Corosync with has OpenAIS built in. My question is which should I be looking at to replace my current AIS option I currently have. XMPP or Corosync? It looks like the Corosync is just the AIS option more nicely packaged. Is XMPP a better solution as I grow my network? Are there down sides to XMPP that AIS/Corosync does…