jlumby
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Posts posted by jlumby
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I just got hit by the friendly scanner again, this time the source IP was 92.61.60.3 Unfortunately since Version 4 with DoS protection is still under development, it took the server down, until I could block it at the firewall. THe packet capture looks identical to the one I posted above
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I have not turned compression on. When the server was originally built, compression was not an option. I will try it to see if it helps, as well as i think your increase to 16 meg will help as well.
THANKS!!!
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I did a search of the entire PBX directory for files larger than 4mb Other than PBXCTRL.EXE I came back with hundreds of call recordings, most that averaged around 10 meg, and a few that were up to 50 meg, in addition to that, there were 2 non call recording files that were about 6 meg each. One was a voicemail where the person hung up, however the voicemail system kept recording silence for 1 hour, and the other was a name recording where there was nothing but noise in the background. Is there any way we can figure out which extension the voicemail name recording belongs to so that we can properly remove it, or is it OK to just delete the file?
After restarting the system last night, the problem has disappeared, and the current memory line reads:
(uptime: 0 days 15:39:21) (47MB/2046MB 18% 8228480-0) WAV cache: 1
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The TFTP has firmware for many different models of phones in it, so it is 135Mb, and the server is also doing call recording, so the recordings directory is a few gigs. Do you think either of these point to my issue with it not playing system recordings, or am I looking in the wrong place. I just figured those memory numbers might be the issue since they look significantly different from every other PBXnSIP install, and the others are not having issues.
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I have a Windows 2003 server running 3.4.0.3202 that has an uptime of about 60 days. When checking voicemail, or calling an auto attendant, most of the time there is just silence. This seems to happen more frequently the longer that the server has been online. A reboot seems to fix the problem for a while. I have done a packet capture to verify that there is 2 way RTP. Is this a known issue with the version, or are there any workarounds available?
On a second glance I see the following line on the status page:
(93MB/2046MB 25% 1760960164-1709445284)
What do these numbers mean??? On the PNXnSIP support site it says what the 93MB/2046MB means, however the other numbers are not described (25% 1760960164-1709445284) The 1760960164-1709445284 concerns me since other pbxes that are not having an issue end in a -0
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I have posted multiple times in the past on how a park account could be created that would remain sip compliant, probably be programmed by PBXnSIP in a matter of minutes, and come very close to a squared key system on most phones, including Aastra. My suggestions have fallen on deaf ears
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I have had this happen as well. Sometimes I even have had them end up in another domain. I think it has something to do with 2 people logging in to administer the pbx at once, or having 2 coppies or tabs of internet explorer open at once to administer the PBX. It is hard to duplicate, however I am pretty sure that this is what has happened.
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I agree, blind transfer is not always acceptable, and an attended transfer does not update caller ID. What happens if the PBX were to send a re-invite to the phone after an attended transfer was completed? Would the caller ID update? Would it be a compatibility issue for some phones?
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I seconf this request, and think it should be listed at the top of most screens, since when you are reading log files, you cannot tell the exact time that the PBX thinks it is.
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I have had a similar issue, however it only happens to me when I set an extension to redirect to another extension if busy, or no answer (direct dialing does not seem to have the problem). The issue also only pops up when I prefix the redirection destination extension with an 8 to send it directly to voicemail. This is a definite bug. I am running 3.4.0.3202 Win 32
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Cisco has complete documentation for this phone at:
http://www.cisco.com/en/US/docs/voice_ip_c...e/sipins80.html
I have used it, and it works very well.
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Will the PBX need to do any transcoding for a conference if everyone is using the same codec? I am wondering if everyone is using G.729 does it need to transcode to G.711 since that is the only way it can be mixed, or can the audio natively be mixed at G.729?
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Please see my posts above, it works perfectly with DNS
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I have found that if you enter an incorrect extension number in an auto attendant, and the voice prompts you that you have done so, the redirect timeout is no longer in effect. This is problematic since if I you want to replay the greeting, you cannot do so after a certain amount of inactivity. I am on version 3.4.0.3202 Win32
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I experienced the same problem. I had a call path that went hunt group 1>auto attendant>hunt group 2>voicemail The problem is it would cut off after the 3rd leg, so it would never get to the voicemail. Adjusting the Max Loop in the PBX.xml from 10 to 20 fixed the issue, however I would like to know why the issue happened in the first place. If it was set to 10, why was it cutting me off at 3? Obviously it was the right value since adjusting it to 20 fixed the problem.
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I am thinking about adding another ITSP that does inbound calling on a per path basis. I am wondering if there is a way that I can tell the total number of simultaneous inbound calls on a trunk. I can monitor the specific trunk with SNMP, however it shows both inbound, as well as outbound call totals. Any way to get just inbound?
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there is no sure fire way. It all depends on the software that is on it. Although holding down pound while booting, and then pressing 123456789*0# when the side lights start to flash in order down the right hand side of the screen, works on most software versions. Be prepared to have all firmware available, since it erases the flash.
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I feel every extension option should be available. As soon as you do not include one, I am sure I will come up for a need for it.
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Another thing that would help streamline the process would be a template account where you could set all of the settings the way you want, and then every new account will utilize them.
Having an account that if you changed it, would apply the changed values to all existing accounts would be handy as well.
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I believe the carrier is doing it since I can see the digits in a packetcapture of the RTP stream, however I do have inband detection turned on as well.
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I would place an outbound call, and monitor it in the active calls screen. See if the call changes it's state to connected once the other party picks up. If not, it may be hanging you up based on a timer if the call never goes to the connected state.
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It is still hit and miss for me
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I am running 3.4.0.3201 (Win32) and I am having an issue where the DTMF digits are rarely recognized while the greeting is playing, however they work perfectly in the silence after the greeting is done being played. Any suggestions for improving the recognition during the greeting?
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I was looking for a way to call into the system, and be able to re-record auto attendant, and agent group greetings from off site (without a phone registered to the domain). Dialing *98+group number from a mailbox, auto attendant, or outbound call menus does not seem to work.
recording in or out?
in Call Recording
Posted
I am having the same issue. I did not with previous versions, however with 3.4.0.3202 all call recordings state that they are incoming