search for: congested

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 =>

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

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

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

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

...0-5 at 192.168.0.100 CSeq: 41 INVITE User-Agent: FPBX-2.4.0(1.4.19.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:301 at 192.168.0.1> Content-Length: 0 <------------> -- SIP/301-0822de30 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s at macro-dial:8] Set("SIP/302-08221a38", "DIALSTATUS=CONGESTION") in new stack -- Executing [s at macro-exten-vm:10] Set("SIP/302-08221a38", "SV_DIALSTATUS=CONGESTION") in new stack -- Executing [s at macro-e...

I have found that fax detection 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 =>

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", "...

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

Hi, How does * read and process the extension.conf file?? The reason I ask is that I think it probably has a very large impact on how the calls are routed and processed by the system especially when it comes to least cost routing.. Let me explain...with an example.. I am using the * Devkit to get to grips with the system, so I have and X100P (Zap/1) and and S100U (Zap/2).. Below is my

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 a setup where outgoing calls

...answer Outbound calls are made using Manager originate interface from a meetme room channel Local/4000/n where 4000 is an extension which accesses the meetme room. ITSP is terminating outbound calls to me via IAX2. I need to be able to see the CAUSE CODE status of the call if it is answered, CONGESTED or BUSY. my ITSP is in Australia - as am I. the IAX2 debug clearly indicates a zero CAUSE CODE on most call cases. Any idea what I might be able to do to make the CAUSE CODE a little more meaningful? Also, does ${DIALSTATUS} or ${HANGUPCAUSE} work better on PRI? Cheers, Mark. -- regards,...

...") in new stack -- Called g1:0342426530 -- Asked to indicate '3' (Dialing) condition on channel SCCP/michiel-00000004 -- Current tone (36) is equiv to wanted tone (36). Ignoring. -- Modem[i4l]/ttyI3 is busy -- Hungup 'Modem[i4l]/ttyI3' == Everyone is busy/congested at this time -- Sending tone 127 -- Executing Congestion("SCCP/michiel-00000004", "") in new stack I only have this with the bank. Is it possible there is some PBX at the bank that messes up normal call progress in * ??? This is a Dutch bank, maybe ppl in Holland using...

-----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 instead of congested. (I expected Dial() to treat congestion like other network error conditions such as a timeout) - Steve Jabber: steve@ne...

...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 Playb...

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)}" ==

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)

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/306-00...