similar to: Detected digit 'f'

Hi, I'm trying to receive faxes using hylafax / iaxmodem but I just can't get it to work. We're using Sangoma E1 cards and have calls coming in over PSTN. I've tried turning hardware echo cancellation off but it makes no difference. This is what I get in /var/spool/hylafax/log: [root at faxhost log]# cat c000000003 Jan 14 12:44:43.39: [ 3403]: SESSION BEGIN 000000003 18005551212

Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful.

HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call is always a ring group called '600', my problem is that after press 1 (but this problem is present also with press 2) before that the inbound call is transfer to extension pass 10/11 seconds ! In attach log file about incoming call. I use Trixbox with Asterisk-1.6.0.10. Thanks. ------ Salvatore.

Hi, We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. I've found an issue when Asterisk tries to make a SIP call out using auth, but has the wrong credentials and keeps getting returned a SIP 407, in this example to an OpenSIPs server requiring user auth. Basically this happens: 1. Asterisk sends plain INVITE to OpenSIPs 2. OpenSIPs responds with SIP 407 auth required with a

On Wed, 2020-10-28 at 14:40 -0300, Joshua C. Colp wrote: > This is not yet fixed, but is being worked on. I have it as a > security issue currently out of caution (although I don't think we'll > treat it as one after further investigation). Right OK, thanks. Do you have any idea of the sort of timescale, and whether it'll be available as a patch that we can apply to our

Hi, if we were to use shred to delete a file on a gluster volume, will the correct blocks be overwritten on the bricks? (still using Gluster 3.6.3 as have been too cautious to upgrade a mission critical live system). Cheers, Kingsley.

Hi, is this a rare problem? Cheers, Kingsley. On Tue, 2023-03-14 at 19:31 +0000, Kingsley Tart wrote: > Hi, > > using Gluster 9.2 on debian 11 I'm trying to set up geo replication. > I am following this guide: > > https://docs.gluster.org/en/main/Administrator-Guide/Geo-Replication/#password-less-ssh > > I have a volume called "ansible" which is only a

Hi, We've recently hit an issue with Asterisk 18.8.0 where a call comes in via SIP (using pjsip) but it can take 5 seconds before starting to execute the dialplan. This was intermittent, but frequent (eg approx half of the calls). We have verbose logging on, but I didn't see any errors. Running asterisk -r -vvvv and then watching SIP traffic in another window showed the INVITE coming

Hi, I'm not sure whether this is possible but if it is, I'm sure someone on here might know ... Is it possible to use Monitor() to record a conversation[1], but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on for a long time? We already have this happening if the callers press a specific key sequence (which we've defined in features.conf)

Hi, using Gluster 9.2 on debian 11 I'm trying to set up geo replication. I am following this guide: https://docs.gluster.org/en/main/Administrator-Guide/Geo-Replication/#password-less-ssh I have a volume called "ansible" which is only a small volume and seemed like an ideal test case. Firstly, for a bit of feedback (this isn't my issue as I worked around it) I had this

A few weeks ago, I lost my Trixbox that was all set up with Hylafax and IAXmodem. I am trying to set it up for email > procmail > faxmail > iaxmodem > asterisk >sipext > ATA (with attached fax). I have followed all the instructions on creating the IAX extension and configuring the IAXmodem config file. I can see the connection in Asterisk: iax2 show peers Name/Username

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

Hi, Most people using iaxmodem + HylaFAX or spandsp from an Asterisk application get satisfactory results for both transmitting and receiving FAXes. However, quite a few people find FAX receive is reliable, but FAX transmit can be flaky. Its box dependent. The cause of this has been known since the early days of spandsp. Asterisk + zaptel/dahdi is unable to produce a consistent audio stream

On 11/08/2017 11:36 PM, Kingsley Tart wrote: > Hi, > > if we were to use shred to delete a file on a gluster volume, will the > correct blocks be overwritten on the bricks? > > (still using Gluster 3.6.3 as have been too cautious to upgrade a > mission critical live system). When I strace `shred filename`, it just seems to write + fsync random values into the file based on

Hello, I have anolog lines coming throug Dahdi to Asterisk Server, one of the anolog lines is used for fax line. I received fax fine without any problems using Iaxmodem with Hylafax Server. Outgoint fax is the problem, when IAXMODEM dial out using Dahdi channel, dahdi answers and start to dial the outside number however Iaxmodem thinks that dahdi is the remote fax machine and starts sending fax

Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Our setup uses a digium b410p card with asterisk 1.6, latest libpri and dahdi, hylafax with iaxmodem, and all this on 1

I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk and

I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to

I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on the same machine. After rebooting the iaxmodems don't register to asterisk. Stoping and starting the relevant services gets it working, but what is the point of using init scripts if it does not work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it

Can anybody point me in the right direction please? I'm having some issues getting iaxmodem and hylafax to talk to each other. I have no doubt that someone has had this type of issue before but I can't find anything useful in the archives or on Google. Under RH9, with chan_capi 7.1, Asterisk 1.2.24, with an AVI Fritz ISDN2e BRI card all working perfectly... I've downloaded and