joso

What is the default list size on the user page? I would like to keep the call log for as long as possible. I would like to be able to go into a users page and view the call log list going back like 90 days for example. is this possible? Also, i have my cdr's going to a csv, but for some reason it isn't showing the transfered calls. Any ideas? Joso

Hi, I am using the latest version of pbxnsip 3.4.0.3201 and found something really weird. The president would call my internal extension from his internal extension (300 to 333) and he receives the auto attendant saying “to receive a callback when the ext becomes available, .....), but when he calls into the office from a cell phone, the receptionist redirects him to my ext 333, a different AA comes on with my name greeting. Shouldn’t everything be the same? Also, i have seen numerous times, people’s voicemail greeting they made disappearing and they have to recreate it. Any ideas on what is going on? Am i missing something? Joso

i have setup multiple vlan's on my dell layer 3 switches internally. Multiple subnet's reside such as 192.168.2.x, 192.168.5.x and 192.168.6.x. the pbxnsip server sits at 192.168.2.19. For testing, i set a port to use vlan 6 (192.168.6.x) and hooked up a snom 320 to it. the dhcp 66 option is set at 192.168.2.19 on vlan 6. I reset the phone and it picks up a 192.168.6.53 address just fine, but i get a prov.request when trying to provision. if the phone was on any subnet other than the pbxnsip server's subnet it would not provision. i even move the pbxnsip server to 192.168.6.19 and then it worked again. I also have xlite softphone users on several different subnets and it takes a while to register with pbxnsip and when it does, i get one way audio. Does the pbxnsip server have to be on the same vlan as all the phones? what do i have to do to make this work? Joso

In version 3.2.0.3144 i have had a couple of users say they received an email with a voicemail but when they checked the voicemail on the phone by calling their extension, there is no message. I think this was a bug in this version. Maybe pbxnsip can clarify if this has to do with the "Race condition" bug found in this version.

Did you happen to transfer the voicemail to another mailbox? I have experienced an issue were i transfer a voicemail from my mailbox to someone else's mailbox, yet the MWI is flashing on my snom phone and the mailbox says there is no message.

I was able to fix this issue by disabling "Disconnect on Broken Connection" within the audicodes. no disconnects anymore. Thanks

although when using xlite or polycom phones, i don't get disconnecte dwhen i call my cell phone and press mute.

This only happens when with a call to the outside like a cell phone. when calling another snom phone internally, it is fine. the snom phone reports this in the log: [5]30/4/2009 11:47:30: Dialog 7/8 going to trying [5]30/4/2009 11:47:30: Dialog 7/8 going to early [5]30/4/2009 11:47:33: Dialog 7/8 going to confirmed [5]30/4/2009 11:47:55: Dialog 7/8 going to terminated [5]30/4/2009 11:47:56: timeout::callback: Registering with timeout of 0 ms [5]30/4/2009 11:48:17: Settings applied! [2]30/4/2009 11:48:33: Registered at registrar as 321@localhost (Expires: 179 secs) this is what my audiocodes is reporting: 11d:14h:37m:6s ( lgr_psbrdex)(375420 ) recv <-- acEV_BROKEN_CONNECTION, Ch:2 11d:14h:37m:6s ( lgr_flow)(375421 ) #2:RTP_BROKEN_CONNECTION_EV 11d:14h:37m:6s ( lgr_flow)(375422 ) | #2:RTP_BROKEN_CONNECTION_EV 11d:14h:37m:6s ( lgr_flow)(375423 ) | #2:MANUAL_DISCONNECT_CALL_EV (send) : (f0c802a0@pbx) 11d:14h:37m:6s ( lgr_flow)(375424 ) | | #2:MANUAL_DISCONNECT_CALL_EV:(f0c802a0@pbx) 11d:14h:37m:6s ( lgr_flow)(375425 ) | | #2:Call changing states from:ConnectedState to:DisconnectingState 11d:14h:37m:6s ( lgr_flow)(375426 ) | | #2:RELEASE_EV:(f0c802a0@pbx) 11d:14h:37m:6s ( lgr_flow)(375427 ) | | #2:Call changing states from:DisconnectingState to:DisconnectingState 11d:14h:37m:6s ( lgr_flow)(375428 ) | | #2:RELEASE_ACK_EV:(f0c802a0@pbx) 11d:14h:37m:6s ( lgr_flow)(375429 ) | | | #2:RELEASE_EV(f0c802a0@pbx) 11d:14h:37m:6s ( lgr_flow)(375430 ) | |(SIPTU#1)DISCONNECT_REQ State:Connected(f0c802a0@pbx) 11d:14h:37m:6s ( lgr_flow)(375431 ) ---- Outgoing SIP Message to 192.168.2.19:5060 from SIPInterface #0 ---- I will get the wireshark trace, although i am seeing alot of "UDP Checksum errors" in wireshark at the moment is this normal?

7.3.14 Using the latest software 3.3.1.3177

i am having the same issue and it seems it has to do with snom, since my 2 polycoms conference phones work just fine. i too have an audiocodes gateway. The snom disconnects after 10 seconds. Does anyone else have an idea? Joso

I have a snom 360 for Reception and would like to add a snom 320 to the same extension for the receptionist so i can put it in another location. i am using the 3.3.1.3177 version which i heard supports multiple mac address for binging. When i add the second mac address and click save, it disappears. only the first mac address stays. is there something i am doing wrong. I put in MAC MAC, in upper case within the field. Joso

pbxnsip, if you mean network bandwidth, i am pretty sure we have plenty. I also have 2 polycom conference phones ip 6000 and ip 7000 and they do the same thing, (2 parties sounding distant) mattlandis, i will try your suggestion. Thanks! i am hoping to find what the issue is.

Maybe someone might know about what happens when 2 outside callers join the conference. Is this a getway thing to join the callers or PBXNSIP? or is it the snom phone that joins them? Thanks Joso

Hi there, we have a sidecar for the snom 360. I have set the buttons to BLF in pbxnsip so reception can transfer the call by hitting the button or she can monitor the extension if it lights up when in use. the 2 issues i have with the sidecar is: 1. occasionally when hitting the person's extension, it routes to another extension instead. (this does not occur when entering the extension manually. 2. When a user hits DND on their phone, the BLF on the reception sidecar button lights up for 5 sec and dissappears, it needs to stay lit. Any help would be appreciated. Joso

Hi there, I've been told be a couple of colleagues with snom 320's that when they conference call the 3rd party in, the sound degrades and the 2 callers on line 1 and 2 say they sound distant from eachother and also distorted. I was also told that when they use the speakerphone with 1 caller on line 1, it sounds fine. It only happens when the 3rd person comes on. I am trying to determine if this is a snom, pbxnsip or audiocodes issue. Any help is appreciated! Joso

With regards to the question on the second consecutive incoming call ringer on the receptionist's snom 360, it looks like it has not been implemented yet in the snom's firmware. i found this post in the snom forum http://forum.snom.com/index.php?showtopic=...entry2714 Any update on the DND and sidecar?

I did exactly you said and it works great! Thanks Joso

Just to clarify and explain the process i want the call to flow is : when the call comes in i want it to ring the receptionist at ext 300 during business hours. If a service flag 101 for night mode is on then i want it to call the hunt group at ext 102, which will ring all phones. in this hunt group the final destination will be the AA at ext 100, which will have the greetings based on the service flag 103. I noticed that this does work but the service flag 101 has to be set to "Set" in order for it to go to reception and when it is "Cleared" it goes to night mode. I think this is due to the hunt group having the service flag set to 101 with the number to 300. Is there something wrong with this setup?

would i put Service Flag Account="123 124" in the hunt group or the AA? the reason for the hunt group is night mode i want all the phones to ring, so put the extensions in there and the final stage to the AA. in your example, how would i include the hunt group. Also, in your examply, your mentioned Night Service Number="130 131", what would these be, hunt group, extnsions? in my scenario, how would you set this up to match your example. Thanks Joso

i used the below setting in my xml file (in the html directory) and rebooted the phone <cw_dialtone perm="">off</cw_dialtone>

Hi there, i have a reception account with ext 300. I have created 2 service flags. The first service flag is account 101 which is used for the Night Service and is manual. The second service flag account is 103 which is used for holidays and is set to Day/Night. The AA has an account 100. Within the AA IVR, I first set the *98100*1 to the 101 service flag then i put in 103 service flag for the *98100*2. First off, i have set up a hunt group (102) and all calls get routed to this account first and i set the night service like this: Service Flag Account: 101 Night Service Number: 300 The AA has this as well: Service Flag Account: 101 Night Service Number: 300 When i call in after hours, it goes to the holidays service flag greeting (103) not the 101 greeting. For a test, i remove the 103 service flag from the IVR and then it goes straight to the AA Default greeting. Any idea why? do i have this setup correctly?

It seems from another post that the DND is not working properly and when pressing it too quickly on the snom phones it affects the status of pbxnsip. i am assumming it is affecting the BLF on the receptionist's sidecar. ANy idea when this will be fixed?

I will take a look on sunday and test this and see if there is an INVITE. In the previous question about CO lines. i am wondering if i need to set it at all, regardless of the caller id. the reason why i set it was so i can dedicate the first line to incoming only and the rest for/incoming/outgoing. Since we spoke about led's, i set the snom sidecar to BLF for all extensions. When a user is using their phone it lights up and shuts off when the user is done, that's good. When the user puts their phone to DND, it lights up only for a few seconds then goes away. Any way to make the LED light up until the DND is off for the extensions? Thanks Joso

Ok. I will look at the audiocodes setup. What about the second call that comes in on the second line and ringing? the receptionist doesn't hear anything, she just see's it on screen. We want it to ring out loud. Joso

Hi there, Our recpetionist has a snom 360 phone and i enabled 4 lines on 4 buttons. All this connected to an audiocodes gateway. i enabled all 8 ports on the audiocdoes by entering the DID's, the first DID on port 1 is our main phone number and the rest are the DID's the telco gave us. When a call comes in, the 1st line button flashes, the phone rings and the correct phone number of the call comes through. Now when the second caller comes in on Line 2 at the same time the caller id is wrong (It shows the did number i put in the audiocodes), the button on line 2 is flashing and there is no ringing out loud. How do i get the caller id to show up correctly on the second, third, etc..call as well as it make it ring? This is for Canada BTW. i set it both 10 digits NANPA, and 11 digits NANPA and no difference. i also set it to p-asserted identity. this on one trunk. i also have 6 lines and set the CO Lines to line1:i line2 line3 line4 line5 line6. should i have set this for FXO? is it needed? Thanks in advance! Joso