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Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at

Hello, Just got a core dump, and thanks to somebody named Matt, I ran a gdb command, and this is the cusp of it: Reading symbols from /lib/libgcc_s.so.1...done. Loaded symbols for /lib/libgcc_s.so.1 Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 ast_senddigit_end (chan=0x0, digit=49 '1', duration=100) at

Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream =>

Hi, I'm planning on migrating someone who uses a very mature system. They would be logging in either as AgentLogin() or AQM. The main requirement however, is: The supervisor will have a control panel, where he will see how many of his agents are on call. If they are, he can "right-click" on the agent and get the options Call Monitor (where the super just listens in on the call,

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Help needed, please. From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Mark Hamilton Sent: March 13, 2008 2:16 PM To: asterisk-dev at lists.digium.com Subject: [asterisk-dev] Call failed, reason 0 explanation. Hello, I understand that Asterisk interprets SIP error codes as 'reason'. We use AMI origination for

Hello, Didn't get much help on asterisk-dev, so here it is. Please help. Thanks. From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Mark Hamilton Sent: March 13, 2008 2:10 PM To: asterisk-dev at lists.digium.com Subject: [asterisk-dev] Hardware and CentOS tweaks. Hello, We're working on using Asterisk as an

Hello, Asterisk Server A makes an outbound call, and upon connect: exten =>1,n,RetryDial(/var/lib/asterisk/sounds/connecting,0,3,SIP/${connectto},,tT ) (${connectto} most of the time happens to be 12345 at 66.xx.xx.66 or 54321 {IP masqueraded ofcourse}) ..transfers it to * Server B (i.e 66.xx.xx.66) via SIP. (Background info, Server B registers on Server A as 1000, and Server A

Hello, I have two sites. Both the sites will require queues for their own reason, own campaigns, etc. Like site1 would handle product1, product2, product3, while site2 can do product4, but can also do customer support for product1 and if anything, can transfer to site1's product1 queue (usually via SIP) Also, a few of these queues will require AgentLogin() type logging in, and some

Hello, Have any of you had the chance to see what TeleVantage call monitor is? It basically shows queues, other extensions, calls in waiting, voicemail, etc. Does anyone have if there is something out there like it, so each and every agent can have one of those? And not just a manager? I'll be putting in a screenshot if someone wants to see. Thank you, Mark --------------

Hi, I have a website where customers enter their phone numbers to be called. I'd like them to have to put in information and 'schedule' a call. 1) Call Immediately 2) Call in the next _ minutes 3) Call me tomorrow, same time. So, Asterisk will pull two variables from this php websites, $phonenumber and $timetocall. $timetocall will need to be calculated as