jlumby
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Posts posted by jlumby
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I am running 3.0.0.2992 with cisco phones. I have found that if someone does an attended transfer to an extension that has 2 or more phones registered to it, the initial screening process of the transfer works fine, and you can talk with the operator that is transferring the call to you, however when the transfer is completed (transfer button pressed a second time by the operator), and the caller is put through, it rings to the other registered extension, and not the extension that the screened part of the transfer was being done on.
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Okay, just another idea. When you call from your cell to an auto attendant, and that number is in your extension, then the PBX gives you the option to call out. In that case IMHO it would use the extension caller-ID. Maybe that solves the problem?
It exibits the same problem as the calling card, where it sends my cell phone number as caller ID. I feel like it may be a bug in your implimentation of ANI with respect to the calling card service, since that is the method I am using to send my caller ID, and I doubt many other people are using both ani, and calling card together.
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Well the first time someone enters the queue should be only half of the waiting time. But I get the point. Maybe we should extend the syntax for the waiting time and say the first number if for all prompts and the second (if present) specially for the first prompt. Default would still be the duration divided by 2.
Gap between announcements (s): 20 5
Would mean: Usually wait 20 seconds, and for the first prompt wait 5 seconds. What do you think?
Sounds like a great solution
Another Idea I had (which only partially fixed the problem) was to have the trunk send to the extension, and set the extension to busy call forward to the agent group. The 2 prblems this brings up is that there are not service flag fields in the extension, so it rings endlessly after hours, and the extension cannot have voicemail.
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The problem with not recording 0 is that then when someone does hit the queue because the agents are busy, then callers just hear music with no explination of what is going on with their call for the first 20 seconds or so. If you decrease the timer, then they are constantly hearing the recordings, and not the background music.
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I was wondering 2 things about caller ID name.
The first is is it possible to create a local database of caller ID names, so when the PBX sees a number, it inserts the name out of the local database, or can overwrite the caller ID name that comes from the phone company if the name exists locally. The reason I ask is we have some customers that receive a lot of calls from DID numbers at other companies where it will just show the caller ID name of the entire company, and not specific person who has the DID.
Secondly in the case where the local provider only provides caller id number, it there a public database you can query to get the name? I would think something similar to reverse DNS would exist. If such a thing does exist, how could I get the PBX to query it?
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I wish it was taking my extension number, and sending it. It is sending my cell phone number.
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Is there a way to set the agent group so that if there are agents available, and their recovery timer has expired that the caller does not hear the initial announcement for the agent group, and they go direclty into ringing one of the available agents?
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Thanks for the help, I edited the config file, and it stopped the emails. As for the loop, I must have described it improperly. What I was saying is that the email account that you setup so the PBX can send emails, you would assume is just a service account to authenticate with to send. Most people would assume that it would never receive anything. The actuality is that the PBX sends emails to that account as well. You would also assume that all of the logging to email features would be listed as options in the GUI. You wouls also assume that if any logging to email was being done, that there would be a space within the GUI for an email logging recipient (similar to the daily call log recipient). But the end result is that it is fixed now.
It also made me look into why I was getting so many emails. Turns out that some phones I have, the value for when the registration expires is set in seconds, while others are measured in minutes. The end result is I had a bunch of extensions re-registering every minute. This has since been cut way back.
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I noticed that the service email account that is utilized to authenticate, and send daily email call summaries was sending emails to itself. It is doing this everytime anything is written to the log file. This means it is clogging it's own email box with emails that only show one line like the following:
Source address for sip:208@mydomain.com has changed
to udp:192.168.3.141:52868
I stumbled upon this when looking at mail server stats, and it had more than 200,000 emails in its box. I cannot figure out how to turn this off. My logging is set down at level 5. Any hints at how to disable this are appreciated. I am running 3.0.0.2980, and based on the number of messages, I would guess it has been doing this for several versions now.
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I have noticed that if you have numbers filled in the User Input field of the Direct Destinations section of the auto attendant, despite having no playback set, a few seconds after your recording is played, the pre-recorded voice in the PBX will begin saying the digits that you have put in the user input field.
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Ok, to eliminate variables, and gain more insight than I would have on the phone's interface, I ran the same test from Internet explorer. When I enter the following:
http://ip_of_Pbxnsip/tftp/services.xml
It successfully returns the document to internet explorer.
When I enter:
http://ip_of_Pbxnsip/tftp/services.xml?loc...SEP00070E16446E
It returns:
File services.xml?locale=English_United_States&name=SEP00070E16446E not found
Please ask your system administrator to check the log file.
It is taking the ? and everything after as the file name.
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In the phone's config file, I am actually specifying the complete URL which is http://pbxnsip ip address/tftp/services.xml I am not sure why it does not show the tftp in the packet capture.
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Just ran into a second situation I did not think about. What if you need to do it to an ITSP that requires registration? Can you some how include the username/password in the static registration, or can you tell it to use the trunk's?
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After presenting to a customer how his remote sites can be networked together with multiple PBXnSIP systems, I came to realize that the PAC will only monitor one of these systems at a time. Is it possible to set it up in a way similar to outlook where you can have it check several accounts at once (or in this case phone systems). It would also be nice if the PAC would remenber it's previous window size, and position on the desktop.
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This works without the authentication problem that I was having when using the HTML folder, however some phones send additional info in the url that causes your web server to return a 404 error. IIS ignores the additional info. The following is the get command from a packet capture.
GET /services.xml?locale=English_United_States&name=SEP00070E16446E HTTP/1.1\r\n
Is there any way to allow your built in web server to allow this, and ignore everything after the ? since I am not utilizing the different localles
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Thanks, it works perfectly!
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If a call is forked to a cell phone, is it possible to use features such as call forward via star codes?
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Is there any way to allow the extensions that are ringing via a huntgroup to still call fork to a cell phone?
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With the latest fix to caller ID with respect to attended transfers, is it possible to get an option to set the call forking timer for the extension to 2 seconds. This is a good compromise for people who would like immediately, however will be getting some of their calls via attended transfer. This 2 second timer would allow the transfer to be completed in just enough time to show the correct caller ID on the cellphone.
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Switching the caller ID to Block on the calling card account had no effect.
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Is there a way where the moderator can mute select participants in the conference? We have had occasions where someone within the conference is very noisy, or puts the conference on hold, and we are forced to stop, and restart the entire conference to get rid of the problem caller.
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I am attempting to use a calling card as a way to hide my cell phone caller ID, and send my office phone's caller ID. I am running 3.0.0.2976 (Win32) and am using your ANI fields to send caller ID to my ITSP. I find that nomatter what is filled in the trunk's account field, or in the ANI field, the calling card displays my cellphone's caller ID to whomever I am calling.
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I am running 3.0.0.2976 (Win32) and if you select any music on hold other than the domain default, you only get silence
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You were correct, it was in the dialplan. The problem was the number that outlook was showing me matched the dialplan, however if I looked at the packet capture, it was sending a 1 infront of the number that outlook was displaying.
Thanks for the help
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The version of 3.0.0.2997 posted on the PBXnSIP web site for download to the CS410 is not in english. Is there a setting I am missing?