similar to: Delay in IVR

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, Does anyone know what it means when I've got an incoming fax routed through to iaxmodem+hylafax and then I see this in the asterisk log: DEBUG[18902] chan_dahdi.c: Detected digit 'f' This happens just after the initial fax negotiation has started and seems to correspond with the sending fax machine giving up. Googling hasn't helped me here :( -- Cheers, Kingsley.

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, sorry for my insistence but this is a big problem for me..my steps for remove card are ok ? Thanks. ------ Salvatore. ----- Original Message ----- From: "Sasa" <sasa at shoponweb.it> To: <asterisk-users at lists.digium.com> Sent: Monday, November 26, 2007 4:25 PM Subject: [asterisk-users] Remove a TDM Card > Hi, I would like remove a Digium TDM2400P from

Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) & Portech MV-370, my problem is that when arrived an external call I don't view (on my internal phone) the phone number but I have the number extension that is configured on MV-370. The MV-370 configuration is: Mobile to Lan Table : 0 * 192.168.1.1 Lan to Mobile Table: 0 * # SIP Setting: Display Name: Portech User Name: 1005

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

Hi, For a 20 users prospective customer, I'm wondering if any GUI would allow and end user to edit an Asterisk IVR tree ? For instance, I'm looking for something allowing to edit interactions like : "wait up to 20 seconds and say this "to reach sales department, type 1, to reach tech support type 2" message" Regards -------------- next part -------------- An HTML

Dear ALL I want to use IVR on my asterisk auto attendent call feature so basically how do i configure IVR and how do i tell to my asterisk which call is local or which call comming from outside world --------------------------------- Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. -------------- next part -------------- An

Hello Team, IVR selection of QUEUEMETRICS As we know queuemetrics had an IVR selection functionality where it can get the IVR keypress of a caller. We saw this link http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0 and upon checking, its only determined the Queue, I want to get is the per IVR of a caller. Can you help me guys regarding this? I want to implement this with

Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018

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

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, 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

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, 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

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 All, I ' m using Cisco 7970 IP Phone and Asterisk 1.6.0.10-FONCORE-r40 (Tirxbox). My problem is that I upgrade my phone to SIP image but now this phone is not registering. The error likes this : SIP/2.0 403 Forbidden (Bad auth) The phone and Trixbox are in the same network. There arenot any NAT rules. Can you help me please?

Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: <loadInformation>SIP11.8-0-4SR1S</loadInformation> ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local

Hi I want to setup an IVR on Trixbox and use it to send calls to agents, and i want to integrate this with sugar CRM that comes with tixbox. can some one please help me Adi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060620/66d8121f/attachment.htm