similar to: Chatterbug

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks [CC] app_cepstral.c -> app_cepstral.o In file included from /usr/src/asterisk/include/asterisk/linkedlists.h:23, from

Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? /rg

I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.com

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

I downloaded Cepstral to my Asterisk Box. I did the install and let it install to /opt/swift. I brought down a new CVS-HEAD as of today 10/1. I added APPS+=app_cepstral.so into the Makefile in /usr/src/asterisk/apps/Makefile Like: # Obsolete things... # #APPS+=app_sql_postgres.so #APPS+=app_sql_odbc.so APPS+=app_cepstral.so # I did this piece but wasn't sure exactly what part of the

Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

Does anyone know how to suppress the "Missed Calls" indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg

Anyone know how to bypass the CONFIRMATION of the user announcement recording in MeetMe? While I like the "please say your name" to announce a user into a conference, I find it confusing and time consuming to make the user to press 1 to accept a recording they haven't even previewed. I'm not a coder, but I'd be happy to comment out the confirmation loop if someone

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabled by default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory

We recently received a bug report about a compiler issue when using GNUTLS: http://bugs.gentoo.org/show_bug.cgi?id=67628. I can confirm the same bug using 0.99.11. Help? Thanks, g2boojum -- Grant Goodyear Gentoo Developer g2boojum at gentoo.org http://www.gentoo.org/~g2boojum GPG Fingerprint: D706 9802 1663 DEF5 81B0 9573 A6DC 7152 E0F6 5B76 -------------- next part -------------- A non-text

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it

hello there. i do have debian 6 in alix 2d13 that runs asterisk 11.7. i do need to interface an ISDN line to asterisk and i bought an external isdn card usb. it is a dlink du 128ta. i have seen that i need misdn and lcr. in a clean install of debian 6.0.8 i installed through git the misdn and misdn user. unofortunatelly i am stuck with the lcr. i have installed misdn, misdnuser and LCR through git

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.