similar to: Multiple Outgoing Lines: extensions.conf

Hi All am using trixbox with call queues..I've set setinterfacevars=yes in queues.conf and following is dial plan : [test] exten => s,1,Answer() exten => s,2,Set(FILE_NAME=${CALLERID(num)}-${MEMBERINTERFACE}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten => s,3,Monitor(wav,${FILE_NAME},m) exten => s,4,queue(55365) exten => s,5,Hangup() but MEMBERINTERFACE is always empty - i

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

hello, I would like to know if it's possible to change the codec of a call in the middle of the call. I have an asterisk without g729 codecs and I recieve an incoming call. The codec is negotiated in ulaw althought who is calling have a g729 codec. My * plays and announcements and call and extension to pass the call to it. The extension have a g729 codec too. It is possible to change the

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

I'm curious as to what versions of everything you are using. Reason being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing it to SIP/5000-00000000". It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to SIP/801-0000000c" [1-1 being the span and channel

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -------------------------- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed Details:

Hello again dear list. Could you please help with this? Thank you for all support, you are great, and i am now at a late stage in the setup and tweaking this server, So I hope you can help me again. I Can't make include the context nighttime. Just to demonstrate if it works, I have a playback function there. But CLI reports: CLI [Jun 18 14:20:22] WARNING[2287]: pbx.c:9542

I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my searches have failed me, I hope someone can help. (By the way, I added an asterisk index for quick navigation on the blog http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.) Here is the snag and I am hoping for a little help from the collective. Inbound I have 2 different numbers. I can call in on both

Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be

Hello, I'm trying to monitor asterisk with snmp. I'm using asterisk 1.4.4 compiled with res_snmp on a debian stable: *CLI> module show like snmp Module Description Use Count res_snmp.so SNMP [Sub]Agent for Asterisk 0 I've configured asterisk in res_snmp.conf: [general] subagent = yes enabled = yes and when asterisk start print

Loan, Thanks for your help in this matter. Having never used astdb before, can you point me to an example on this?? Thanks hugely, Clara >> Hi Clara, You could put some data into astdb and query for the outgoing line and callerid based on internal callerid (extension). something like user/201/outline 89859715 user/201/outcallerid 89859715 and so on... By the way:

Hi all, I have a problem with my asterisk box and an X100P FXO card. I am able to place outgoing calls from my SIP phone (Cisco 7940) to any external number using my PSTN line, but when I call my PSTN line from my cell phone, the Cisco doesn't ring (and no message appears in the Asterisk CLI). Here are my config files: zaptel.conf fxsks=1 loadzone = be defaultzone = be

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>

Hey all. We have a DS3 circuit with GBLX split off into 7 systems with a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P cards in the other 5. GBLX numbers their spans from 0 to 3 instead of 1-4 and we have a NFAS configuration with the d-channel on chan 96. All of our systems are running 1.0.7 for stability reasons (and no good time for maintaince, the entire platform