vicidial.org

hello ,,

I installd vicibox server,

i only configurend the siptrunk , sip user and the dialplan.

it works but not perfekt.. now i had following probs:;

6 WARNING[5149]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb7011f78', 9 retries!

some times ,, this one

Auto-congesting SIP/SIPtrunk-081faf80
>> -- SIP/SIPtrunk-081faf80 is circuit-busy


or



Executing DeadAGI("SIP/cc101-b7011d50", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----7") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --20-----7 completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("SIP/cc101-b7011d50", "agi://127.0.0.1:4577/call_log") in new stack
Jun 12 14:29:46 WARNING[5093]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb7017290', 9 retries!
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-b7011d50", "SIP/004369981*****@SIPtrunk|25|tTo") in new stack
-- Called 004369981*****@SIPtrunk
-- SIP/SIPtrunk-0820b618 is making progress passing it to SIP/cc101-b7011d50
-- SIP/SIPtrunk-0820b618 is ringing
== Spawn extension (default, 70043699*******, 2) exited non-zero on 'SIP/cc101-b7011d50'
-- Executing DeadAGI("SIP/cc101-b7011d50", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/cc101-b7011d50", "agi://127.0.0.1:4577/call_log") in new stack
Jun 12 14:29:55 WARNING[5093]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb7017290', 9 retries!
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-b7011d50", "SIP/*************@SIPtrunk|25|tTo") in new stack
-- Called 00436*********@SIPtrunk
-- SIP/SIPtrunk-0820b618 is making progress passing it to SIP/cc101-b7011d50
-- SIP/SIPtrunk-0820b618 is ringing
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 70043************, 2) exited non-zero on 'SIP/cc101-b7011d50'
-- Executing DeadAGI("SIP/cc101-b7011d50", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
Vicidial-test*CLI>

sometimes it works.. and sometimes it dials .. but no ring on test mobile..

only the message: nobody picked up in 25 seconds,, but the mobile never ringing.. ???


i dont know ,, did any one know these Problems ?

thx

Have you tried calling a land-line?

Hello Matt,

If you mean with landline in my country ?

then yess.

or if you mean something other ?

thx

What country are you in?

hello matt..

the country is Austria.!


now i tried set up a new vicidial now.. but i had the same problem as
vicibox..

Executing AGI("Local/7004369981178753@default-11fd,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/70043699*******@default-11fd,2", "SIP/004369981178753@SIPtrunk|25|to") in new stack
-- Called 00436998*******@SIPtrunk
Jun 12 14:19:43 NOTICE[2943]: chan_sip.c:2035 auto_congest: Auto-congesting SIP/SIPtrunk-0901c770
-- SIP/SIPtrunk-0901c770 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/7004369981178753@default-11fd,2", "") in new stack
== Spawn extension (default, 70043699*******, 3) exited non-zero on 'Local/7004369981178753@default-11fd,2'
-- Executing DeadAGI("Local/70043699*******@default-11fd,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------") in new stack

new fresch system ,,

dialplan

; Austria SIP nemox
exten => _70043.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _70043.,2,Dial(SIP/${EXTEN:1}@SIPtrunk,25,tTo)
exten => _70043.,3,Hangup

thx for help

That's a SIP congestion issue, not much we can do about that one.

Have you tried another carrier?

auto-congest? is that a lag(timeout) congestion instead of an actual call rejection? what are your settings for this sip provider?

do you have qualify=###?

can you register to this sip provider successfully?

[SIPtrunk]
type=friend
username=XXXXXXX
secret=XXXXXXX
disallow=all
allow=alaw
host=sip.nemox.net
dtmfmode=rfc2833
qualify=yes


sip.nemox.net:5060 XXXXXX 45 Registered


64 bytes from sip.nemox.net (83.137.41.34): icmp_seq=1 ttl=54 time=34.5 ms
64 bytes from sip.nemox.net (83.137.41.34): icmp_seq=2 ttl=54 time=34.5 ms
64 bytes from sip.nemox.net (83.137.41.34): icmp_seq=3 ttl=54 time=34.3 ms
64 bytes from sip.nemox.net (83.137.41.34): icmp_seq=4 ttl=54 time=35.8 ms
64 bytes from sip.nemox.net (83.137.41.34): icmp_seq=5 ttl=54 time=35.2 ms


at the moment i didnt try another carrier because i had no no one other..

thx

auto-congest is being generated by your lag time (qualify=yes). qualify will track the "lag time" of your "peer" and if the lag time is too high asterisk will refuse to connect the call (as a result of your request to do so via setting qualify). so your system thinks the lag-time is too high, and it's method of rejecting the call is "congestion".

try "sip show peers" to find the lag time and take steps to reduce it (better network connection?)

or

remove qualify=yes from your context. the call may succeed, but you may notice a serious quality issue (or serious lag time).

i generally use qualify=500, i'm not sure what "yes" will convert to as a number (never needed to know it, cuz i always specify a number).

Hello matt, & william,

i try so much settings .. but every time the sam problem.

**That's a SIP congestion issue, not much we can do about that one.
Have you tried another carrier?**

yes i have .. but the same problem every time auto congestion
sometimes one call is ok..

lag times : i had to my dialer 22-28 ms .. from dialer to sip terminierung 2-4 ms.

but cames auto congestion also wehn the number not exist is ?

thx speed

now ,, il take another provider !

and it runs ... and run ,, and run..

this auto congestions is only when you had to much lag times from
server to siptrunk.

or when you dial a number whos not in system..

i think now it´s good ,,

i hope ,, my nervs ...

thx for help !