similar to: How does chan_sip match an ACK?

I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that is behind a network device to which I don't have ready access, which is performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 system on a public IP. My question is very specific, and I don't need right now to discuss the ins and outs of the above setup. What I am seeing is that when I have set up a

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

On 2016-08-09 10:06, Faheem Muhammad wrote: > trip time and Call Setup time of SIP Requests. > In case of GSM Network with high delay you need to set the T1 timer a > higher value like 1000ms (500 ms default). Similarly you can reduce the > Call setup time by configuring 'T2' upto you choice as per you telephony > network. Configure t1min, timert1 and timerb according to

Hello List, I tried to change the following parameters in sip.conf file, but looks like it cannot be changed, Defaut values: ;t1min=100 ;timert1=500 ;timerb=32000 I have changed to: ;t1min=100 timert1=100 timerb=6400 Sometime I can see too many retransmission of BYE to some of the UAs if UA is unreachable. Is there a way that I can reduce the number of retransmission of BYE message?

Hi, I'm trying to diagnose a particularly elusive problem, and am wondering if anyone else here has seen anything similar and can offer any ideas. I have a conference bridge running Asterisk 1.2.32 (with slight mods), in a colo talking via a LAN to an ITSP using SIP/RTP. It is dedicated to a single customer. On several occasions over the last few months, the customer has reported instances

I have some more investigation to do on this, but I wanted to see if anyone here had any insight into the issue I've run into. The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one of several systems that have been running without issue since 2010/2011. They have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen 5 card), libpri 1.2.8 and

In article <CAHZ_z=w5DMg93gShtC93kuC+fnmraPgV46BS956U5BQXVgyhxg at mail.gmail.com>, Matt Fredrickson <creslin at digium.com> wrote: > On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield <tony at softins.co.uk> wrote: > > I have some more investigation to do on this, but I wanted to see if anyone > > here had any insight into the issue I've run into. > > >

I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one : Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> ) exten =>

In article <7025a0a8-1471-530d-dad0-3770e902ca31 at uw.edu>, John H Nyhuis <jnyhuis at uw.edu> wrote: > The mtx binary requires my tape library to be assigned a sg# driver, but > the kernel periodically renumbers the sg devices. Normally, we would > write a udev rule to manually assign a persistent name, but it looks > like things have changed as I can't seem to get

One of our customers has an Asterisk conference bridge connected to a SIP trunk from an ITSP. Yesterday, they had two inbound calls that didn't get hung up properly. From the tcpdump SIP trace that we have running continuously, I can see that no BYE was received by the bridge, and when some hours later the hangup was forced from the bridge end, the bridge sent a BYE to which it received a 481

In article <n009u2$85v$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > Apologies, this is slightly off-topic being to do with an EPEL package, > although it's running on CentOS6, so I thought others here might have come > across this issue. > > I have five CentOS 6 systems running fail2ban from EPEL, and this > package was updated

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

In article <ml1jnh$afr$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > When I started using CentOS 6 instead of CentOS 5, I discovered that > "less" no longer understood \< and \>, which I had been used to using > since almost forever. > > Eventually research revealed that in the Fedora version on which > RHEL 6 was

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

In article <CAG2kNCyjsQZ2qW_8BBLp8BH_20=JgxoEYpn9BSwZhXg7_rHBbg at mail.gmail.com>, Gianluca Cecchi <gianluca.cecchi at gmail.com> wrote: > On Mon, Nov 21, 2016 at 12:49 PM, Tony Mountifield <tony at softins.co.uk> > wrote: > > > I am just applying the latest C6 updates to a couple of KVM Linodes. > > It appears that the latest update of util-linux-ng has

In article <301a2e78-d490-3805-e30f-41b668aac5c1 at sysnux.pf>, Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > > Hi Tony, > > Le 20/07/2019 à 06:29, Tony Mountifield a écrit : > > Are there any other languages/libraries I should be considering? > > Same here, after years of AGI / AMI, I recently made my first project > using ARI on Asterisk-16. I love

I want to use fail2ban on CentOS 6 to monitor Apache with the standard default logfile format ("combined"). Has anyone here succeeded in doing so? The format has the IP at the start of the line, followed by two dashes (if no authentication) and THEN the timestamp. What I've read on the fail2ban wiki seems to say that the timestamp must ALWAYS be at the start of the line, followed by

On 06/09/2015 12:48 AM, Nicolas Thierry-Mieg wrote: > On 06/09/2015 12:33 AM, tony at softins.co.uk (Tony Mountifield) wrote: >> In article <ml1jnh$afr$1 at softins.softins.co.uk>, >> Tony Mountifield <tony at softins.co.uk> wrote: >>> When I started using CentOS 6 instead of CentOS 5, I discovered that >>> "less" no longer understood \<