similar to: ISDN BRI voice one way only

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a general question about SIP access for nonregistered users. I would like to make it possible for basically anybody to make a SIP call to my asterisk without having to have a user account, but in a specific context. So that e.g. somebody could make a SIP call to SIP/stefan at my.asterix.pbx and it would go like this: [incoming_guest]

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got

Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply

I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary, when she is out, she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100 at

Hello! I'm curently using * with two old Teles PCI card (wich, btw, were hard to install and make good use of) with ISDN4Linux. The sound quality is simply perfect. However both dialing in and out through the ISDN line, there seems to be a _little_ bit of echo that eventually gets on your nerves ! Also the echo seems to get a _little_ bigger after a minute or so into the conversation. Now,

Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello! After LOTS of research on this list and internet in general I managed to get an old Teles PCI card working with Asterisk throught ISDN4Linux. No echos, no delays, simply perfect -- electronic poetry ! :) eheheheheh I just didn't get it to work with CAPI and "chan_capi" but, since isdn4linux is doing such a good job, I'll

Hi, i'm traying to compile asterisk on my pc, a laptop whit SUSE 9.1 and a desktop with SUSE 8.2, with a teles S0 16/3 PnP. With Kernel 2.4 (Desktop) Asterisk run but it's umpossible to compile the driver ISDN-utils for Teles. With kernel 2.6 I can't compile zaptel (not necessary with my laptop) and asterisk, in both cases I receve errors during make or make linux26 (I saw the notes

Hello, folks! This is my first post here. I installed Asterisk from scratch and after reading a lot of information on voip-info and this mailing list I was able to get started. I can make sip-to-sip calls (just on a basic extentions structure, let's say for beginners) but now I'm trying to make this system works with my Teles ISDN BRI PCI card. I can make and receive calls through X-Ten

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help?

Hi all, I'm struggling some time now with this problem. Googling and searching on this topic did not deliver the answer yet, so my last hope is this list. Analogue to the things which are possible with modem.conf, where I can configure the MSN's to act on, I would like to have similar functionality. This is the idea: I have 1 ISDN line, it can be reached by 4 different MSN's. I have

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.

Hi I'm using a Teles ISDN passive card configured in modem.conf. when i make call from my sip client (xtex x-pro) to the external world i have more than 1 second of delay and echo very. There is some tuning to do? The performance is better with an active ISDN card or CAPI compatible driver? thanks mark balester

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?