similar to: dtmf / misdn

Hello, I have some extensions that are using variables loaded by an AGI program. Everything works fine and I am able to use NoOp to see the value of my variables when using IAX, but the same variables don't work when using SIP. I can provide further details, but right off of the bat does is there something I need to know about the use of user defined variables in with SIP channels vs. IAX

Hello, We're about to deploy an Asterisk system. So far we have the following (below) configuration but before we start anything we would like to hear some suggestions on it, specification or recommendation too... Thank you! miniITX board dual core CPU 2 LAN 1x IDE (or more) 2x SATA (or more) 2x PCI slots (must work with Linux correctly in regards to IRQs) Optional: On board hardware RAID

Hi all, Perhaps someone has dealt with it before. I want to activate a bunch of my own scripts after someone has registred om my asterisk, or when his cient has de-registerded. have been skimming through AGI and AMI, and seen a lot of nice features, but not the (de-)registering events. Kind regards, Hans

Just curious, I noticed that with SetLanguage() you can change it into a lot of other languages. Yes one can record them easily enough with "record", but don't like to re-invent the wheel.. Browsed through a lot of google-pages but failed to find any other languages (except for FR and ES on the digium site) Any pointers for german, dutch, greek, italian, .... prompts? Hans --

Morning! This may be the "day of the dead" in some regions, but we expect the usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This Sunday, I believe the USA falls back to Standard time. Future VUC are still at 12 Noon EST. Info site: http://voipusersconference.org PSTN (724) 444-7444 enter 22622# 1#

Hello For customers who need a small IP PBX to handle up to four ISDN lines (in France, so I guess that means EuroISDN) instead of a PC + Asterisk and an ISDN gateway box, has someone already played with the Atcom IP-4B? www.atcom.cn/IP-BRIM.html Any feedback appreciated.

My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part

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Hi, I do hope this hasn't been covered. I googled and looked in the archives but didn't see anything related, so hopefully I am not covering old ground. I recently upgraded from 1.0.10 to 1.1rc5. I am seeing this error from clients: May 25 11:01:54 finity dovecot: Panic: IMAP(xxx at here.org): file mailbox-tree.c: line 171 (mailbox_tree_iterate_set_next_node): assertion failed:

Hi List, i have to put an * between two other SIP gateways and due to some circumstances, i have to use sip over tcp. With 1.6.2.6 this is working fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B (ocs) and that's about it. In the other direction however (ocs -> me -> deverto4) the call setup is complete but there is no audio. I can see the audio in the form of

I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can

On Saturday, November 2, 2019 10:40:45 PM EET Stephan Bosch <stephan at rename-it.nl> wrote: > I think we already know about this issue (tracked as DOP-942). It > apparently happens because qmail masks the SIGCHLD signal while Dovecot > doesn't unmask it before waiting for children. You're right. Isn't this a simple fix - a call to sigprocmask ? This should not be a

Hello, I tried the above setups and am having large problems sending and receiving faxes, even in very low speeds (9600 and far below). Faxes are either shifted (i.e. missing lines) or corrupt at all, some faxes do break with communication errors (although CRC-alike algorithms are turned off) or in alot of cases only the upper quarter or 1/8 is transferred. Iam seeking for any idea on how to

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn

Hello I fixed my problem. I changed user and group in /etc/mISDN.conf: <devnode user="asterisk" group="asterisk" mode="644">mISDN</devnode> now it works again! Thanx 4 help! ;) Ingrid -- "Bonnie & Clyde der Postmaster-Szene!" approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org --------------

Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060) iend(0x8fd5060) and running on Asterisk 1.4.21.2: pbx*CLI> misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:0

I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga -------------- next part -------------- An HTML attachment was scrubbed... URL:

Put faxdetect = none in the misdn.conf and you'll be fine. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 12:40 -->> To: asterisk-users at lists.digium.com -->> Subject: Re: [asterisk-users]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, We have a BRI installation who is experiencing constant echo with the Digium b410p and mISDN. I have adjusted gains, and originally thought that the desktop phones were at fault, although we experience the same problem when a call is bridged out to an external number. I've tried it with 128 taps (the default and I believe the supported

Hi on the list, does anyone of you have experience with asterisk 1.6 and mISDN, pri primarily? Thanks in advance & Regards, Christophorus