similar to: dialplan for monitoring outbound calls

This is my nth post regarding this matter. Where in the dialplan should I put AgentMonitorOutgoing? Can somebody show me how to use it? in extensions.conf: exten => x,1,AgentMonitorOutgoing(c) in agents.conf I set updatecdr to yes. It supposed to put agent/agent id in the "channel" column in the CDR, but instead it puts SIP/<extension number>. This link seems to be the

Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks???

Recently installed 1.4.5, and still unfortunate to find out that the 'c' option in AgentMonitorOutgoing() still doesn't work ('c' - change the CDR so that the source of the call is 'Agent/agent_id'). It wont change the source channel column in the CDR to 'Agent/agent_id' like it used to in 1.4.2 and other releases older than that. I hope this gets fixed in the

Hello list, I don't know if anybody faced this issue, but I finally found a workaround. I am using an external program with an AMI connection to originate outbound calls to Local/ channels, and on the dialplan context I dial outside with the corresponding trunk according to the prefix of the dialed number. The problem here was CDR, more precisely the uniqueid. Reading these mail lists I

I would like to use chanspy and restrict it to Agent channels. Then setup it on an extension for the supervisor to monitor people taking calls from the queue and making outbound calls. Is there a way to use the agent channel with outbound calls? I don't want to record the calls(which AgentMonitorOutgoing seems to do), just kinda flag them so that they can be spyed upon. Any kind odd

Hello everyone, I have a problem with getting name of the recorded file of agent calls. As I've googled I found that the name of the recording file should be inserted in userfield of CDR table. To do this I set createlink=yes in agents.conf but still userfield of cdr is empty but the recrding file is created. my agents.conf file is like this(That part related to recording options): ; Enable

We have the following scenario: Incoming call to a queue, Agent "A" answers. Agent "A" determines after about 20 seconds that agent "B" needs to deal with this call. A puts call on hold, calls and speaks to B, and then transfers the call to B. B speaks to the incomming caller for 5 minutes. That's all fine. However, the CDR records the call as incomming to

List members, Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but

thanks for your response i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue the server asterisk and the ip-phone where the number is configured are in the same network 192.168.1.X Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording

Summary: Is it possible to append to a user-defined text file from the dialplan without using AGI ? Background: We are using our Asterisk server to place outbound calls alone. We've 8 E1s that enables us to place about 240 calls concurrently. The server is a bit old, and we are not able to reach our maximum capacity. The dialplan makes use of MySQL to record call status information to our

Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI> -- Starting simple switch on 'Zap/69-1' -- Executing Wait("Zap/69-1", ".1") in new

I intend to add to my asterisk system a feature similar to cisco call manager's extension mobility so that agents can log in to any phone in the office and keep their profile (ex. the agent's specific directory number). But before doing that, I need to confirm that asterisk doesn't have a native solution for that (ex. application/addon), and that nobody has come up with their own

Installed Asterisk 1.4.4 hoping that this bug (http://bugs.digium.com/view.php?id=8680)<http://bugs.digium.com/view.php?id=8680> is fixed. It was fixed, however a new problem came up. It seems that AgentMonitorOutgoing(c) is no longer able to change the 'channel' value in the CDR into Agent/<AgentNumber> format. This feature is very useful to us because we can identify calls

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

I have a little call recording script that I am running and it works fine, but I have one problem. I get CDR when a user calls into the extension, but I do not get CDR for the call that it makes outbound on # 17. Any idea why? Here is the extensions info: [default] exten => 2211,1,Answer exten => 2211,2,Wait(1) exten => 2211,3,Playback(/etc/asterisk/recording/getshop) exten =>

Hi I'm using asterisk 1.8.23.1 but I've seen this same issue in previous versions of 1.8. I have created some work arounds but the behaviour is incorrect. This is the scenario: Call comes in and goes to appropriate dialplan In the dialplan the call is forwarded to another number using a Local channel (and using /n ) e.g. Dial(Local/<my-number>@outbound-context/n,60) The number is

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was