[TCP: TOS could not be set]

What is the TOS Count not be set?

 

[5] 2010/04/04 21:15:03: SIP port accept from 99.78.45.55:2079

[1] 2010/04/04 21:15:03: TCP: TOS could not be set

[Caller Priority]

i believe it uses the caller id info to see which agent that caller id number last talked to, and based on that will send the caller to the agent

now if that i said was correct, do you have to keep call records for a year long so the feature works good?

[Call barge In *81 and Teach Mode *82 - don't work]

can you post how many active calls on your system when you had the issues? and what was the CPU performance level?

[Setting ANI on Huntgroup does not Work.]

i have also tried changing the outbound ANI but with no results its clear to me that that setting is there for that reason, but yet it never works for me

[Charge user 1026 for redirecting calls]

i looked through several trunks and i cant find any trunk that has "assume that call comes from" with any data in it

what else can i look at?

 

reference to the nextone, it is not "mine", I'm doing support for a client that has it, they need me to set up their PBXnSIP's installation

[Charge user 1026 for redirecting calls]

Hi

 

using: Version: 3.1.2.3120 (Win32)

 

Looking thru the logfile i see the following entry

 

[6] 2010/03/25 15:55:21: Sending RTP for 3c295a59f3eb-xlmjc09mlzlk#eed67afb8d to 192.168.1.102:57688

[5] 2010/03/25 15:55:21: Dialplan NexTone - Brentwood: Match 8008008000@mydomain.pbx.net to <sip:18008008000@ipaddressremoved;user=phone> on trunk NexTone

[5] 2010/03/25 15:55:21: Charge user 1026 for redirecting calls

[6] 2010/03/25 15:55:21: Sending RTP for 3c295a59f3eb-xlmjc09mlzlk#eed67afb8d to anotheripaddressremoved:57688

[6] 2010/03/25 15:55:22: Sending RTP for b8123d49@pbx#57825 to ipaddressremoved:32464

 

can anyone explain the following line "[5] 2010/03/25 15:55:21: Charge user 1026 for redirecting calls"

i see that several times int he logfile, not always is it user 1026, sometimes its user 1031 etc

[register SPA2102 on any account/domain]

I use diff SIP passwords across all account

all he did was change the proxy and extension number, and was able to place and receive calls as that extension

[Set processor affinity to 1 failed]

what does the affinity do? "lock" it down to 1 core? is it bad or good to do?

[File length]

is it okay to put a 7 minute long file for Music On Hold ? or the longer the file themore memory it will use up?

also whats the best format to have it in?

[Need polycom template for pbxnsip 3.5.0.3301]

I'm looking for thge template files used for the Polycom phones

 

also is it possible to modify it ?

if i can modify it, where would i save it so set pbxnsip to use the new settings ?

[IVR plays very fast]

i did reboot and it helped

[Snom with POE Switches]

I have never been able to make a snom 360 or 320 work on any cisco switch running spanning tree, and PoE

[Snom with POE Switches]

on a Cisco 3550, when i first plug it in it powers up but takes a while for the phone to actually connect/register its extension, it starts the countdown process repeating prov.prequest from 20 down to 0, and at about 9th attempt it will register only sometimes, other times it will keep power cycling

 

with polycom phones i dont have the problem, only with all snom phones i have tried

 

its something about when the phone drops the ethernet after receiving power

[Snom with POE Switches]

Does anyone know what setting the Snom phones are looking for when powered up using POE Switch ?

I'm currently using the value of= power inline delay shutdown 20 initial 300 on my cisco 3550 switch but it seems as if the phone doesn't work correctly

what I'm looking for is how long after it gets power is it looking to drop the Ethernet

[register SPA2102 on any account/domain]

he is not changing password, only user name and domain name

[IVR plays very fast]

looking at the PBX Status page i see the following:

Available file system space: 16%

Uptime: 2009/12/4 01:58:59 (uptime: 18 days 15:23:41) (2942623 -1274686128--1431066544) WAV cache: 14

 

and when i check the memory on the box i see this

[root@]# free

total used free shared buffers cached

Mem: 4149620 3957236 192384 0 170032 3388176

-/+ buffers/cache: 399028 3750592

Swap: 4192956 68 4192888

 

so i rebooted the machine and now the PBX Status is:

Available file system space: 17%

Uptime: 2009/12/4 02:09:04 (uptime: 0 days 00:00:31) (22107 24399552-0) WAV cache: 1

 

and the box is:

[root@]# free

total used free shared buffers cached

Mem: 4149620 393476 3756144 0 28124 226464

-/+ buffers/cache: 138888 4010732

Swap: 4192956 0 4192956

[Simple Windows Failover Solution]

Not operating on WIN but its nice to see a win failover example for other windows users.....

[Simple Windows Failover Solution]

nice work matt!!!!

[DTMF]

so you believe you had the issues with the SIP Provider? and its not related to the PBX?

[register SPA2102 on any account/domain]

3.4.0.3202 version

A client is able to log into his SPA2102 box and change the line 1 information from his extension info to another one and register without any issues, how is that possible?

all he changed was his PROXY info from companya.pbx.som to companyb.pbx.som, the display/auth/user name from 101 to 755 which is an active account on the compoanyb domains and start making and receiving calls

 

what have i done wrong that allows the user to do so?

[importing files using csv]

when i import a CSV list i have an issue that the DID for example doesn't use the java script

the java script that takes 9789789788 and changes it to (978)978-9788

 

even after i upload it and i click in the field that has the DID and then click in another field so the java script does its thing, it still doesn't work. am i doing anything wrong?

[IVR plays very fast]

Been getting complaints from our clients that when calling into our PBX (calls goes directly to a Auto Attendant) they hear the IVR Greeting speaking very fast

at first i comfort myself saying the issue happen when the call comes in from a caller using comcast, so i said "its comcast's problem"

now I'm getting add'l complaints that the problem is happening even when calling from cell phones etc

 

Ive also had issues that when someone calls my extension, it also rings my cell phone, if i don't answer and it goes to my voice mail then i get a voice message saying "you have a call from nine seven eight nice seven eight nine seven eight eight, to accept the call press one" etc

but sometimes when i hear the prompt, its being played very fast to me

 

I've also had issues where when calling the phone number, its silent, the IVR isn't heard/played by the caller

 

 

so the issue here is that at times (i cant duplicate it) the IVR Greetings are being played at FAST SPEED

and soemtimes its not being played at all

[PBX.XML]

on the PBX.XML file, what is the value of app_product? and what is it used for

 

and the 80 emails is the nightly cdr's being sent from each domain, seems as it only sends approximatly 80 emails when it should be sending about 100+

might be google apps is preventing from flooding them with emails.

[Asterisk to PBXnSIP]

has anyone on here actually ever configured it?

[Voice Mail Blasting]

i am able to "move the message" to a hunt group, but not to a agent group

in a hunt group, do all three stages get the VM file? or do i need to create a hunt group and put the people on the first stage and set up as many hunt groups aka VM blasting lists i need?

also the only way to do this now is to leave your self a VM, hang up, call back and then move it? i cant just log into my voice mail and record a message a move it right from there to a hunt group? i can to any other Extension.

is it possible to make this a little easier? maybe call a *code and enter the call blast list, or the hunt group list and record your message?