search for: cc_monitor_policy

Displaying 8 results from an estimated 8 matches for "cc_monitor_policy".

2011 May 02
1
sip busy detect
...not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage li...
2013 Apr 30
2
Asterisk QSIG doesnt send the calling name to Nortel CS1000
...el to Asterisk, I have the display of the Snom "Marco <2002>" and the display of Nortel "<1001>" This is my / etc / asterisk / chan_dahdi.conf [channels] cc_offer_timer=20 ccbs_available_timer=4800 ccnr_available_timer=7200 cc_recall_timer=20 cc_agent_policy=native cc_monitor_policy=native pridialplan=private prilocaldialplan=private context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgai...
2011 May 20
5
Restart asterisk destroy all registered SIP peers
Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks S -------------- next part -------------- An HTML attachment was scrubbed... URL:
2014 May 12
1
new install: no re-invite and unwanted transcoding
...yes sendrpid=pai type=friend nat=no port=5060 qualify=yes qualifyfreq=60 transport=udp avpf=no icesupport=no encryption=no callgroup= pickupgroup= allow=g722 allow=ulaw allow=alaw dial=SIP/1001 mailbox=1001 at device permit=0.0.0.0/0.0.0.0 callerid=John Doe <1001> callcounter=yes faxdetect=no cc_monitor_policy=generic The dial command produced by FreePBX also looks reasonable: -- Executing [s at macro-dial-one:43] Dial("SIP/1002-0000007e", "SIP/1001,,rI") in new stack A second issue is that on outbound PSTN calls, Asterisk is accepting the phone's first-preference codec (g7...
2015 Aug 12
2
Busy level in Asterisk 11
Hi I need to set the number of incoming calls to one, but the outgoing calls should be unlimited. I think the busylevel parameter is for it(incoming calls), but not works. My config is: cat sip.conf [general] [template](!) qualify=yes cc_agent_policy=generic cc_monitor_policy=generic call-limit=2 busylevel=1 callcounter=yes subscribecontext = hint allowsubscribe=yes [100](template) type=friend context=default host=dynamic secret=*** [101](template) type=friend context=default host=dynamic secret=*** [102](template) type=friend context=default host=dynamic secret=***...
2015 Mar 03
0
TLS, SRTP, Asterisk11 and Snom870s
...transport=tls,udp,tcp > avpf=no > force_avp=no > icesupport=no > encryption=yes > callgroup= > pickupgroup= > dial=SIP/41712 > mailbox=41712 at device > permit=192.168.6.0/255.255.255.0 > callerid=James B Byrne <41712> > callcounter=yes > faxdetect=no > cc_monitor_policy=generic > > If I change the transport setting to TLS then I get this reported: > > [2015-03-03 11:10:08] ERROR[22244]: tcptls.c:875 > ast_tcptls_client_start: Unable to connect SIP socket to > 192.168.6.112:5060: Connection refused > > I cannot seem to configure the Snom870...
2015 Mar 03
6
TLS, SRTP, Asterisk11 and Snom870s
...ndrpid=no type=friend nat=no port=5060 qualify=yes qualifyfreq=60 transport=tls,udp,tcp avpf=no force_avp=no icesupport=no encryption=yes callgroup= pickupgroup= dial=SIP/41712 mailbox=41712 at device permit=192.168.6.0/255.255.255.0 callerid=James B Byrne <41712> callcounter=yes faxdetect=no cc_monitor_policy=generic If I change the transport setting to TLS then I get this reported: [2015-03-03 11:10:08] ERROR[22244]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.6.112:5060: Connection refused I cannot seem to configure the Snom870 to listen for TCP on 5060. There is...
2013 May 01
0
asterisk-users Digest, Vol 105, Issue 39
...Marco <2002>" and the display of > Nortel "<1001>" > > This is my / etc / asterisk / chan_dahdi.conf > > [channels] > cc_offer_timer=20 > ccbs_available_timer=4800 > ccnr_available_timer=7200 > cc_recall_timer=20 > cc_agent_policy=native > cc_monitor_policy=native > pridialplan=private > prilocaldialplan=private > > context=default > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > cancallforward=yes > callre...