search for: congest

I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten => 9XXX976XXXX,1,Congestion exten => 91XXX976XXXX,1,Congestion

I've not seen an answer to this in any forum. I make a call through Asterisk, with a VOIP phone, doesn't matter which. The call gets made, I leave a voicemail, or complete the call in some manner, and the other side hangs up. I hear a busy signal on the phone on my end. If I have an extension that looks like this, after the hangup() is executed, my phone gives busy signals until I

...34,1,Playback(abandon-all-hope) include => extentions exten => i,1,Playback(vm-goodbye) exten => i,2,Hangup() exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() [outbound] ignorepat => 9 exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9XXXXXXXXXX,2,Congestion() exten => _9XXXXXXXXXX,102,Congestion() exten => _91800NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _91800NXXXXXX,2,Congestion() exten => _91800NXXXXXX,102,Congestion() exten => _91888NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _91888NXXXXXX,2,Congestion() ext...

We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These "Ringing Congestions" start to pile up, which eventually crashes Asterisk. H323 Gateway -> Asterisk (chan_h323) -> Tor2/PRI -> PSTN Has anyone ran into this problem or kno...

Hi, I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system, up to recently everything was fine but we are starting to experience the call limitation of the line (15). So as to warn user of the problem i attached a vocal notification to the CONGESTION status after a Dial(), but it looks like it also catch other congestion case (maybe on the receiver side). Should i / Could i use the ChanIsAvail() on a Dahdi channel ? will it work better to detect a full group ? regards

...s Asterisk server and the settings in the Linksys PAP2T box but could not get it to work. I removed the Linksys PAP2T and replaced it with an HT503 because I read a lot of good recommendations for this device. It seems to have almost the same problem. I say almost because when the Linksys would get congestion I would hear the Asterisk recording tell me "All circuits are busy now, good-bye" but the HT503 only gets a busy tone. All the SIP phones can call out no problem but these two ATA boxes that I am trying to use the FXS ports to connect old analog POTS phones to are not working. I have...

...etection is returning an error saying that no fax extension is present when I have defined one. The console returns this error: Aug 26 10:58:41 NOTICE[1112745536]: chan_zap.c:3989 zt_read: Fax detected, but no fax extension extensions.conf has: [default] exten => fax,1,Hangup exten => fax,2,Congestion exten => fax,102,Congestion exten => f,1,Hangup exten => f,2,Congestion exten => f,102,Congestion Any ideas? Simon Brown

On my wild learning curve, I encountered numerous occasions when a channel remained in "Congestion" state after a Congestion() step without going to the next step, which is Hangup(). I couldn't find a definite pattern but it seems to happen when a channel is hung up by the other party or by some other action. Any recommendation about preventing such? Yuan Liu

...> 0x2afec424c430 -- Probation passed - setting RTP source address to 192.168.1.212:57592 > 0xc5922b0 -- Probation passed - setting RTP source address to 217.195.xx.xxx:29674 -- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s at macro-dialout-trunk:23] NoOp("SIP/306-000000b8", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack -- Executing [s at macro-dialout-trunk:24] GotoIf("SIP/306-000000b8", &quo...

In the bufferbloat age, anything that can make for a kinder, gentler bulk transfer protocol seems desirable. This add support for user and server selectable congestion control algorithms. As examples: --congestion-alg=lp # For the tcp-lp algorithm on the command line Or in a subsection: [mystuff] congestion alg = westwood # for a wireless connection And diffserv support: --diffserv=8 for setting the CS1 bit This could be improved by being able to specify...

...wrong.. I am still a newbie and have only had a working system for about a week.. Thanks.. ===== extensions.conf - start ===== [general] static=yes writeprotect=no [globals] [default] ;incoming calls exten => s,1,Dial(Zap/2) [extensions] exten => 1234,1,Dial(Zap/2,,t) exten => 1234,2,Congestion exten => 1235,1,Dial(SIP/1235,,t) exten => 1235,2,Congestion exten => 1236,1,Dial(SIP/1236,,t) exten => 1236,2,Congestion include => parkedcalls include => dialout [dialout] ignorepat => 9 exten => _9[01][1-9].,1,Dial(Zap/1/${EXTEN:1}) exten => _9[01][1-9].,2,Congest...

Is there a recommended way to find out the cause of DIALSTATUS = CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS variable and I also count the active ISDN channels for the ISDN trunk in question. Counting the active ISDN channels seems somewhat clumsy as the mapping to a specific trunk must be done by hand (or write even more code). I have...

...ming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL ,CONGESTION,ANSWER) exten => s-CHANUNAVAIL,1,NoOp("CHANUNAVAIL") exten => s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account: ${ACCOUNTCODE}^${CALLERIDNUM}) exten => s-CONGESTION,1,NoOp("CONGESTION") exten => s-CONGESTION,2,UserEvent(Congestion|Account: ${ACCOUNTCODE}^${C...

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004", "Modem/g1:xxxxxxxxxx|50|Ttr") in new stack -- Called g1:0342426530 -- Asked to indicate '3' (Dialing) condition on channel SCCP/michi...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can the Dial() command tell the difference between busy and congestion? At the moment it seems to be treating them both the same on my server. I want to route the calls out via a SIP gateway unless that is congested, in which case dial out through my POTS line (using an X100P). It seems a bit pointless to try dialling the POTS line when the SIP dial is busy i...

...uot;, "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("Zap/4-1", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing Dial("Zap/4-1", "ZAP/g0/0827751492") in new stack == Everyone is busy/congested at this time -- Executing Macro("Zap/4-1", "outisbusy") in new stack -- Executing Playback("Zap/4-1", "allison7/all-circuits-busy-now") in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Pla...

Here's a segment of my dialplan, I'm working on the freenum/ISN functionality: same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) same => n,GotoIf($["${isnresult}" != ""]?:fn-CONGESTION,1) ; set up our outgoing call state same => n,Set(SIPFROMUSER=${CALLERID(num)}) same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" == ""]?dial:) same => n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) same => n(dial),Dial(SIP/${isnresult},40) same => n,Goto(fn-${DIALS...

I am trying to implement a rollover of extensions. exten => 3000,1,GotoIf($[${line1} = Congestion]?3:2) exten => 3000,2,Dial(${line1},15,rt) exten => 3000,3,GotoIf($[${line2} = Congestion]?5:4) exten => 3000,4,Dial(${line2},15,rt) exten => 3000,5,GotoIf($[${line3} = Congestion]?7:6) exten => 3000,6,Dial(${line3},15,rt) exten => 3000,7,GotoIf($[${line4} = Congestion]?1:8) ex...

Hi all, Does anybody know a package to control the bandwidth using "TCP Congestion Control" method? Best Regards Hoomaan Naimi Afranet Network Administrator

...- Probation passed - setting RTP source address to > 192.168.1.212:57592 > > 0xc5922b0 -- Probation passed - setting RTP source address to > 217.195.xx.xxx:29674 > -- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060 > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing [s at macro-dialout-trunk:23] NoOp("SIP/306-000000b8", "Dial > failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") > in new stack > -- Executing [s at macro-dialout-trunk:24] GotoIf("SIP/306-...