similar to: Causes of auto-congestion on SIP?

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

Hello, sometimes I'm experiencing autocongest error due "slow response", anyone knows, what this means? Second or third attempt after that happens pass successfully... this happens ever in fastethernet lan, so no problem with lag in wan environment, I'm using idefisk 1.32 on client side (winxp or linux)... PJ -- Executing Dial("IAX2/bill-7",

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 running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and try to make a call I get the following error message: ======================================================================================== -- Executing [6781948 at default:1] Dial("IAX2/iaxy-7477", "DAHDI/g1/96781948") in new stack [Apr 25 13:00:10] WARNING[3772]: app_dial.c:1806

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

On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: > I'm using asterisk-16.30.1 > > When I try to call another asterisk server over IAX I get a busy signal, > > chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow > response >     -- IAX2/192.168.143.1:4569-656 is circuit-busy > > Asterisk-16.16 is working normally, no congestion error. There is not

I'm using asterisk-16.30.1 When I try to call another asterisk server over IAX I get a busy signal, chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.143.1:4569-656 is circuit-busy Asterisk-16.16 is working normally, no congestion error. -- Thelma

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote: > > On 1/2/24 15:13, asterisk at phreaknet.org wrote: >>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: >>> I'm using asterisk-16.30.1 >>> >>> When I try to call another asterisk server over IAX I get a busy signal, >>> >>> chan_iax2.c:4739 __auto_congest:

Thanks James, What i need is to make the fax machines connected to the audiocodes mediant 1000 be able to send and receive fax throught Asterisk (connected to a pri) I know it's not reliable, but it should work at leaste, what should i do on Asterisk and Mediant to make this work? Im quite confuse with all these fax issues :S Thanks in advance > > Message: 11 > Date: Fri, 9 Apr

Alright I've been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn't. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these == Using SIP RTP

Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James

Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx "memory show summary" .... 1780466242 bytes (1780181594 cache) in 2352909 allocations in file frame.c ... Seems like a ridiculous cache. On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna <jlamanna at gmail.com> wrote: > cat /proc/cpuinfo lists 4 cores. >

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

Hello, I'm using Asterisk with an ISDN30e PRI line (only 16 channels active). Every now and then I get a CONGESTION error even-though there are only 2 channels in use out of the 16. When this happens, the user just needs to re-dial and the call goes through OK. [2008-10-14 15:41:40] -- Executing [s at macro-to-isdn:1] Dial("SIP/216-bc0aab90", "Zap/g1/0123456789") in

Hello, I'm using Asterisk with an ISDN30e PRI line (only 16 channels active). Every now and then I get a CONGESTION error even-though there are only 1 or 2 channels in use out of the 16 at that time. When this happens, the user just needs to re-dial and the call goes through OK. On a SNOM phone when the problem occurs, a "Service Unavailable 907" error is shown. [2008-10-14

> > Its up to 5.8G of resident memory with 28321 calls processed. > The OOM killer is going to kill this soon at this rate (8GB RAM machine). > This seems like a pretty serious problem. > It looks like I'll need to restart asterisk every night.... Hi the number of cpu cores that you see with top times 512Mbyte is the level of ram that's needed e.g. a hp-gen8 with 2 octo

Hi, Does anyone have some good examples of a Kamalio or OpenSips configuration that integrates with Asterisk? Essentially I want to use the SIP router as the UA, but still run all the calls through Asterisk (for dialplan, etc..) I've looked for examples on the project web sites, but I haven't found anything decent yet. Thanks. -- James

We are planning a very large Asterisk deployment, using Wifi SIP phones. We've done installs using Spectralink and the SVP to manage congestion at the access points, but we have a client that doesn't want Spectralinks. Anyone have experience with an alternative congestion management (AP association management ?) technology with Asterisk? Anything open source that I'm not aware of?

Hy all! Your Asterisk server is return this log : *CLI> -- Executing Dial("Khomp/B0C0", "IAX2/*.*.*.*/9834|30|r") in new stack -- Called *.*.*.*/9834 Mar 14 15:35:40 NOTICE[4212]: chan_iax2.c:2836 auto_congest: Auto-congesting call due to slow response -- IAX2/*.*.*.*:4569-1 is circuit-busy -- Hungup 'IAX2/*.*.*.*:4569-1' == Everyone is