search for: congestioned

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

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

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

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

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

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

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",

-----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

Greetings *`s, I am manually creating call files and dropping them into /var/spool/asterisk/outgoing to be picked up by *. Presently, when I use local/internal parameters using SIP it works..ie I make an internal call from device to device. However, when I try dial an outside number which I have set up in a custom conf file, it bombs out with the following message :

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

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit <salah.elharit200 at gmail.com> wrote: > hello list, > > i have asterisk 11.15.0 and i have some trunks sip from my provider > > we have some ip phone astra 6731i > > each Ip-phone is configured with trunk and we call > > no ihave configured another trunk from the same provider in my asterisk > > i can call