[Asterisk-Users] how to record all agent calls
I want to record incoming calls that are queued when the call is connected to an agent. I added the following lines to agents.conf before the list of agents: ; Enable recording calls addressed to agents. It's turned off by default. recordagentcalls=yes ; ;The format to be used to record the calls ;wav, gsm, wav49. ; By default its wav. recordformat=gsm ; ; Insert into CDR userfield a name of the the created recording ; By default it's turned off. createlink=no ; ; The text to be added to the name of the recording. Allows forming a url link. ;urlprefix=http://host.domain/calls/ ; ; The optional directory to save the conversations in. The default is ; /var/spool/asterisk/monitor ;savecallsin=/var/calls and added to the queues.conf file: ; monitor-format = gsm|wav|wav49 monitor-format = gsm ...and then issued the reload command in the Asterisk CLI console. I even created the /var/log/asterisk/monitor directory because it did not exist. Is there something else that needs to happen to record calls between agents and callers so you can hear both sides of the conversation? Thanks in advance. --- Jeff Crews Eastern Oregon Net, Inc. La Grande Oregon Email [EMAIL PROTECTED] Voice 541-963-2625 or 800-785-7873, extension 11 personal efax 503-907-6704, standard company fax 541-962-7818 web http://home.eoni.com
[Asterisk-Users] Agent Cleanup Time?
Previously there was discussion about people seeking the ability to have an delay between calls to the agents so that the agent could clean-up or wrap up the documentation on the call that just hung up...before the next call is connected to the agent. Was that option added to Asterisk? And if so...what is it officially called and how do we enable it? Jeff
Re: [Asterisk-Users] Non working 800 numbers
I just dialed all those numbers you gave that failed for you from my Cisco 7960 speaking SIP to my Asterisk box that is connected via PRI (to my CLEC switch) which is connected to the PSTN. When you have explicitly sent the caller id (ANI) try calling another phone from your Asterisk PBX (like on your desk, your cell phone, or some other that is connected to something on PSTN that is hopefully connected to another carrier network) that displays caller ID digits to see if at least the correct caller ID number comes through. This will help to confirm the ANI is getting through Asterisk and on to the PSTN. The name may not show...but at least the number should be correct. It is possible you are sending the ANI but it is getting removed some place in your local carrier's switch or someone other switch before it gets to these toll-free numbers that fail. I have found if I fail to send caller ID digits (ANI) some calls are not completed when I dial them. Just as a test I commented out this line in my sip.conf for my Cisco 7960 phone: callerid=Jeff Crews (541) 624-2611) and now I cannot call any of numbers I just said I could dial...I get this recording your call did not go through please try your call again 0 9 3 T with no failure logged on the Asterisk console. I restored my callerid line from above, told Asterisk to reload the config...and I hit the redial button on the call that would not complete...and it works now. So...I know that the caller ID can have an impact. I hope this helps. Jeff At 01:18 PM 4/6/2004, Matthew Branton wrote: Hey guys, I am having a strange problem with certain 800 numbers not working, specifically American Airlines 800-882-8880 800- 843-3000 800- 237-7976 and UPS 800-742-5877 I can't seem to figure out what is causing them not to pick up. Prior to using asterisk on our outbound PRI lines there was no problem. I tried explicitly setting callerid/ani etc on outbound calls, but so far no dice. Has anyone else had a similiar problem? What was the solution? Thanks, Matt --- Jeff Crews Eastern Oregon Net, Inc. La Grande Oregon Email [EMAIL PROTECTED] Voice 541-963-2625 or 800-785-7873, extension 11 personal efax 503-907-6704 standard company fax 541-962-7818 web http://www.eoni.com
[Asterisk-Users] Agents and delay before and after they handle a call
Is there a way for Agents logging in with AgentLogin to have the the agent hear the beep and then have the option to press # or some button to indicate they are ready to take the next call?Sometimes an agent is taking a drink of water or coughing...and logging off and logging back seem lengthy to do. I have tried to use AgentCallbackLogin but it seems to require that each Agent has their own DID phone number so that that the application can call them back at that specific number. We do not have DID to each agent implemented yet...as we are using Asterisk with our old phone system. Thanks. --- Jeff Crews Eastern Oregon Net, Inc. La Grande Oregon Email [EMAIL PROTECTED] Voice 541-963-2625 or 800-785-7873, extension 11 personal efax 503-907-6704 standard company fax 541-962-7818 web http://www.eoni.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Controlling queue size and queue options
I see that in queues.conf there is a maxlen variable to control the maximum size of the queue. So...if you set the queue to a maxlen = 3...my test caller gets dead air if they are queued to a queue with 3 calls already in the queue. I thought I could increment a variable each time a call is queued and decrement a variable when the call is connected to the agent...however I do not know how to build such a structure in extensions.conf to make this work. It also *seems* like when an agent releases/hangs up/finishes a call...that the incoming caller is disconnected in such a way that additional steps in the dialing plan in extensions.conf are not processed. Does anyone have a sample extensions.conf I can see that does something like this? I thought I would try to give call center managers the ability to dial an extension, be authenticated, and then enter a number of their choice to allow them to set how many calls can be in a given queue so that if there are more agents available...the queue can take more calls...and when fewer agents are available...callers might hear a greeting indicating delays and be given the option to leave voice mail.Does that sound like a reasonable idea? I thought when I feel really crafty I would make a web interface in ColdFusion ( I do not speak PHP yet) and have Asterisk copy config files generated by my ColdFusion application from a cronjob to update the running Asterisk config. Thanks in advance for any help...this list is great Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Easy access to visual busy status and call transfer buttons
I want to say thanks for the great posts to this list...I learn something know about every day reading this list. Anyway...I have been using * in a test environment for 10 months and really like it. I have PRI to the PSTN and SIP to 2 Snoms and 1 Cisco 7960. I have frequently used ATT/Lucent/Avaya phone systems such as Definity or Partner that provide the ability to assign LEDs on individual phones that allow you to visually see the status of specific extensions to determine if the extension is on a call, do not disturb, or idle. If I use * to speak SIP to the phones...such as the Cisco 7960...how do you provide users with this easy visual way to see the status of an extension? Further...using a button associated with these busy status indicators makes transferring calls fast. I see some people use software on a PC to get this functionality. It still seems that there should be a way to do this on a SIP phone. Am I the only person that thinks these status LEDs are valuable? Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Expire old voice mail messages, et al
I have Asterisk deliver all voice mail to users as email attachments.
I found by accident that there is a limit of 99 messages in your INBOX in
Asterisk.
The 100th attempt to record a voice mail causes the system to play your
greeting and then never record the 100th message and silently disconnect
the caller.
So...is it safe to simply use the UNIX find command to delete any files in
the INBOX directory that are older than X days old?
I did not know if Asterisk would lose track of which message number was
next...or otherwise screw up the mail box by doing this.
If my use of a daily cron like this:
/usr/bin/find /var/spool/asterisk/vm/33/INBOX/* -mtime +15 -exec rm {} \;
is a bad idea...perhaps having a message retention period defined in
voicemail.conf on a global or per user basis.
Any thought of having maximum number of messages be defined globally in
voicemail.conf or on a per user basis?
Also, does anyone feel a need to have the voicemail system speak the date
and time the voice mail message arrived for those that access messages by
phone instead of the usual email?
Finally...am I the only person who does not have a need for separate busy
and no answer outgoing messages? When I change my greeting...I change the
not available...and have a cron job copy the unavailable to the busy file
so the messages are the same.
Thanks.
---
Jeff Crews
Eastern Oregon Net, Inc.
La Grande Oregon
Email [EMAIL PROTECTED]
Voice 541-963-2625 or 800-785-7873, extension 11
personal efax 503-907-6704
standard company fax 541-962-7818
web http://www.eoni.com
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