similar to: Can't dial out through AMI

Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

Hi there, I?ve been looking to do an outbound dialer for systems alerting, etc. and have in large part followed the recipe here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out That and the associated pages at voip-info give a basic set of recipes for callfiles, but nowhere there or in my copy of the O?Reilly book by Meggelen, Madsen, & Smith can I find a detailed

Hello, CDR fans! I'm looking at some issues brought forward over time: 12726/10668: someone wants me to revert the changes I made via bug 10668, last Sept; (that's they are messing him up. And I didn't do the change suggested in ForkCDR, for fear of lousing up folks depending on current behavior. Which probably sparked: 11721 :

We have a Shortel system at out main site. We're putting Asterisk servers at several smaller remote sites. I know I'll be able to get the Asterisk servers to talk to each other via IAX, but can they talk to the Shoretel server? Basically, I'd like to be able to, from the main site with Shoretel, dial an extension, and reach that phone at a remote site, and vice-versa. Thank

Dear members of the list; I am writing in the hope to get some help with a very peculiar problem with my new asterisk 1.6.0.1 installation. The same code runs on version 1.2 without problems, but it seems the behaviour has changed (also on 1.4.7, which I tried). Please consider the following extension: exten => 1213,1,Answer exten => 1213,n,Set(counter=X) exten =>

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

Okay, so I'm a noob. Asterisk looks very promising, so I say "thanks" and "good job" to all who contribute. My * test box is up and running with soft phones using IAX and SIP, so now I'm on to testing hard phones. I borrowed a couple ShoreTel 210 phones from somebody who had them on hand but they only support MGCP. I see that there's an mgcp.conf in

Gang, I'm having this error pop up when I do a ForkCDR, and I'm not sure how to get around it. Here are a few log lines: Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing ForkCDR("Zap/49-1", "") in new stack Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a CDR The scenario occurs like this: I use a .call file to generate a call on

Hello, I'm looking for a way to restore old behaviour (before Arkadia patch #0010668) of ForkCDR application in 1.4.18 I've done some research directly in the code (cdr.c & forkcdr.c), but can't find any flag. I am just f*c*ed or do you have something to suggest ? :) Thank you for help. Mathieu

Hi, I had Asterisk threads consuming 100% CPU at times since last week. Of course, last week an extra card was installed (we had a 1PRI, a 4PRI was added) so search concentrated on that, but to no avail. Today, I installed DDD on the machine and quickly found out that it was looping because cdr->next->next == cdr in ast_cdr_setapp(). I patched this up with some simple code in

Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is

Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to

I would appreciate hearing from anyone who has figured this one out. Here's the scenario: I have a context wherein I give the called party the option to dial the digit 9. If he does so, he is transferred a la this extension entry: exten => 9,1,Playback(pls-hold-while-try) exten => 9,n,Noop(Attempting to bridge to ${agentext}) exten =>

Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the "answer" field is always empty, "billsec" field is 0 and "disposition" field is

Hi Gang I have not yet managed to find a solution to correctly generate CDRs for this situation: Alice calls Bob. Bob has call forwarding delayed 20s to Charlie. Charlie picks up immediately. exten => bob,1,DBget(cfwdly=CFDLY/${exten}); $cfwdly contains charlie same => n,Set(CDR(src)=${CALLERID(number)}) ; src 'alice' same => n,Set(CDR(dst)=${exten}) same =>

Hello, In my 1.6.1.18-powered system, I've got the following dialplan (in extensions.ael) : Dial(SIP/foo,15); if (${DIALSTATUS}="NOANSWER") Dial(SIP/bar,15); When SIP/baz dials peer SIP/foo which do not answer, I've got a single CDR entry like this: SIP/baz SIP/bar time_when_foo_started_to_ring time_when_bar_ended_talking ANSWERED How can I get two CDR entries :

Hi Guys, The ForkCDR is doing pretty well, except for the call disposition. The time stamps are great, but the disposition always reads answered, even when the call is busy or not answered at all. It should read BUSY and NO ANSWER. I can send a snippet of the CDR record with forked calls for an example if needed. I think the record is not forking properly, but is maintaining the disposition

Hey, all. While ADA can still be downloaded, that's about all that I see. No development, no recent mention, and -- perhaps worst of all -- it appears not to work properly under 64-bit systems. So, assuming Digium's abandoned it, are there any suggestions of alternatives? Right now, I'm replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly fat client they

Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch