similar to: FOP2 in Digium repository?

Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -------------- next part -------------- An HTML attachment was

Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

Dears; I need to order Digium card and not able to know which one is the best quality? Is it that of AEX with the end E or EF or P or B? I saw those card that its slot is small (I think those that end by EF), are they the best card? Really I am caring to have a card that has echo cancelation and the voice volume is high enough (because previously I faced the problem that the voice volume was

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hi, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ....) for cases when a hardware ech canceller is present or not. I read in TEXXX manual that when setting echocancel=yes in chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo cancellation was enabled. 1. I'm correct thinking that it is then

Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. P.S. Are both options part of the configs of Asterisk or need modules to be selected and installed before doing the

Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!!!!!!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e99afa31/attachment.htm>

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at

Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000 at mydomain to call 9000 at my domain(which is the call center)

Dear list, I would like to do some calculation using different grouping variables. My 'df' looks like this: # Some data set.seed(345) id <- seq(200,400, by=10) ids <- sample(substr(id,1,1)) group1 <- rep(1:3, each=7) group2 <- rep(1:2, c(10,11)) group3 <- rep(1:4, c(5,5,5,6)) df <- data.frame(id, ids, group1, group2, group3) df <- rbind(df, df, df) df$time <-

Hi list. I have Asterisk installed on a Debian 1.8 6 64-bit. What happens is the following, some channels are not being hangup properly. They run the hangup in dialplan, but the output of the command "core show channels" shows several channels with status "rsrvd." Checking the server's memory, the "top" command shows multiple processes and stopped using the

So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a "make menuselect" in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) res_smdi gets built fine, dahdi is

Hello List, I work in an environment where I have to request IPTABLES changes rather than doing them myself. Is there a way to utilize my SSH (port 22) to get a functional (and with good sound) Asterisk installation with multiple channels up without requesting the 5060(SIP) 5061 (TLS) and UDP/RTP (usually 10001-20000) IPTABLES allowances? Thanks Danny Nicholas

Hello, I am using asterisk realtime with a postgresql database on the same server. In res_pgsql.conf I have specified [general] dbhost=localhost dbport=5432 dbname=asteriskdb dbuser=psql dbsock=/tmp/.s.PGSQL.5432 Since both asterisk and db are on same server, I would like asterisk to connect to db using the local unix socket. However asterisk is not using the local unix socket to connect to