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Thread: RE: No registrations!
sup

Replies: 3
Views: 7964

No registrations! 17.06.2007 16:43 Forum: Miscellaneous

Sometime yesterday, I seem to have lost registration to my PBX, both here and in another city on 2 different types of devices. I can not register to it , and my device works with other SIP proxies. I have used the same 2 devices on the PBX for over 1 year now

Pascal, Can you please look into this? If it can not get fixed promptly, I will be abandoning the PBX completely, as the problems that seem to come and go are most certainly associated with changes being made to the server.


Mark

Thread: RE: Language Selections not working
sup

Replies: 1
Views: 7716

Language Selections not working 17.05.2007 22:55 Forum: Bugs

I have an inbound trunk and an extension both are set to spanish. Calling the DID on the trunk gives me voice mail in English. I now changed them to German, and Left them there, still all is in English!

This has been somewhat precarious since day 1 however now it seems to not work at all, all messages are in English! I even changed the PBX to spanish and no luck!

Thread: RE: Full G.729 support
sup

Replies: 33
Views: 179087

g729 16.05.2007 06:09 Forum: News

I would be willing to pay , not only the license fee, however some additional profit for PBXes.

I would however not want shared g729, I would need to KNOW that I have X number of channels available should I need them , not Erlang's law.

I also think that many users who need g729 do not use the PBX or are free users and can not post, thus g729 should be a pro feature for a fixed number of channels.

Let me ask... do any of you post on the forums at web sites you do not use? Why then if I NEEDED g729 would I ever be using the PBX?

Thread: All Circuits are Busy Now or Fast Busy
sup

Replies: 0
Views: 5092

All Circuits are Busy Now or Fast Busy 19.04.2007 07:02 Forum: Miscellaneous

When I complete a call , and hang up , then immediately attempt to make another call, I most always get Äll Circuits are Busy Now¨or a faxt busy, unless I wait about one minute between calls.

Can this be fixed? It does not seem limited to one PBXes account, rather system wide.

Thread: Just to add a comment....
sup

Replies: 4
Views: 12050

Just to add a comment.... 19.04.2007 06:55 Forum: Miscellaneous

Although in theory audio bypass works on PUBLIC IP Addresses, if thetre is any NAT between either party and a public IP address, it is likely there will be some problems with audio bypass ON.

As an example, there is an office with 10 workers. Each has a hardphone on their desk connected to PBXes.com. It is VERY unlikely that each of these users will each have a public IP with no NAT. In this scenario Audio bypass will most certainly need to be set to NO.

STUN may halp on many systems, but it is not a fix all solution.

Thread: Nice undocumented feature discovered!
sup

Replies: 0
Views: 5989

Nice undocumented feature discovered! 09.04.2007 06:05 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

I spent a fair amount of time recently trying to get PBXes to send to a SIP URI on a particular server that for this document we will call 192.168.1.1. That server does not allow open SIP calling to its extensions.

I had a trunk registered to the same server 192.168.1.1 , but I found that PBXes was sending the call unauthenticated when sent as a SIP URI. I had tried as a classic extension and found that the caller ID was not forwarded from the calling party, although the call was sent with authentication.

I took a short nap, and it came to me. PBXes is Asterisk Based and Asterisk Lets you formulate a dial command such as DNIS@trunkname, in addition to DNIS@domain.ext.

So the name of the trunk we will call ABC. Instead of sending the call as 1234567890@192.168.1.1, I changed the dial command to 1234567890@ABC. The call then arrived to the destination server as authenticated! It was now sending via the registered trunk, not to a random URI, so I suspect the difference was it is now sent with username and password.

Too bad however I am still battling getting the original caller ID passed from PBXes to the external server on the inbound call. Also, apparently also I can not ring two URIs on the same external server via a PBX ring group. It appears only one of the calls is sent..

Nonetheless, it may well be a valuable find.

Thread: RE: Caller ID
sup

Replies: 5
Views: 21146

Caller ID 08.04.2007 07:24 Forum: Miscellaneous

I have reached an point of extreme frustration with this topic, and I think I am on to the root of the problem.

Problem: I authorize PBXes directly via IP address to an external SIP Proxy, and PBXes Passes Caller ID correctly to the outbound call. However as PBXes will not perform the complex LCR I need, then I have tried both sending calls to an Asterisk (for LCR) then the appropriate SIP proxies (after Asterisk LCR) , and now I have also tried the same with a VOIPSwitch. The VOIPSwitch is currently passing caller ID on Wholesale traffic as well as devices registered to it. One SIP proxy shows the caller ID passed to it. A Sample CDR is below.
supernettel-500 15036677700 UNITED STATES Oregon Apr 8, 2007 01:12:27 AM 00:00:00 00:00:00 0.000000
In the above example it is clearly the PBX User ID that is being passed to the SIP Proxy. Normally in that first field, is the outbound caller ID.

Suspected Source of Problem. There are two fields that can be potentially used for Caller ID. VoIPSwitch makes this clear in their documentation. I believe one field is called User ID . I suspect this USer ID is what is being looked at in both the case of Asterisk (by default in the version I have used) and the VoIPswitch.

Solution:
Set USER ID field to Caller ID number, as it seems it should be anyhow. This may or may not work with caller ID name as well. This will create a greater compatibility with all SIP Proxies, and prevent users from passing invalid caller ID to Providers that may reject calls based on valid ANI (which some do). I see no valid reason to pass the PBXes "USER-EXT" to an outbound trunk, but maybe I am missing the rationale behind it.

Thread: Uniden UIP 2000 and Intellitouch
sup

Replies: 0
Views: 6341

Uniden UIP 2000 and Intellitouch 03.03.2007 05:51 Forum: Terminal Equipment

I have had the recent frustration of testing the following two IP phones on PBXes.com and wanted to post my results

Intellitouch ITC3002 also referred to as IP2006
Went off without a hitch, no problems found. Call quality was excellent. Transfer button works only when trasnsfer to number is followed by # key

Uniden UIP200
Severe frustration with this device. At times I have been able to make outbounds, and no inbounds. Sometimes the unit fails to register. Works fine when connected to some other servers. Some configurations have caused it to unregister when I attermpt to dial out.

If anyone has any advice to offer for a working config, please let me know. What follows are the sample configuration files for this device, so you may see all the options in the most current firmware


=============================================
# UIP200 Mass Configuration System Generic File
# Notes:
# 1. Lines start with '#' are comments, it is also possible to append a comment to the
# end of a line by preceding the '#' with ASCII space or tab characters.
# 2. To leave a field value unchanged (as saved on local phone), leave value to blank.
# 3. To set a field's value to empty, use '-' as value.
# 4. To NOT overwrite user local settings of: programmable key, one/two touch keys, VMA
# number, VMWILampIndicator, set "OverwriteLocalSetting = NO". Default is "YES". This
# key will ALSO affect whether or not THESE settings in uniden<MAC>.txt be used.
# 5. Any duplicate parameters exist in both unidencom.txt and uniden<MAC>.txt, MAC settings
# will be used.
# MAXIMUM FILE SIZE IS 10KB
# Current Limitation: No spaces allowed for a setting's value
# Version: BS4.77


#Overwrite user local settings of programmable keys, one/two touch keys, vma settings
#If set to no, these current settings on the phone will not be overwritten.
OverwriteLocalSettings YES # must be placed on top of config file

# Sip Settings --If only ProxyServer needed, set OutboundProxy1/Port same as ProxyServer/Port
ProxyServer 10.15.15.126 # can be an IP address or FDQN
ProxyServerPort 0 # 0 to use default port
OutboundProxy1 10.15.15.126 # can be an IP address or FQDN
OutboundProxy1Port 0 # enter a port number or 0 for default (5060)
OutboundProxy2 192.168.0.2 # can be an IP address or FQDN
OutboundProxy2Port 0 # enter a port number or 0 for default (5060)
EmergencyProxy 192.168.1.102
EmergencyProxyPort 0
Registrar1 10.15.15.126 # can be an IP address or FQDN
Registrar1Port 0 # enter a port number or 0 for default (5060)
Registrar2 10.15.15.127 # can be an IP address or FQDN
Registrar2Port 0 # enter a port number or 0 for default (5060)

RegisterExpireSec 3600
Q_Param 50
RegisterExpireLimitPercent 10
Register403RetrySec 1200
SipPort 5060
SRVRecordName - #_sip._udp.uniden.com

FailoverRetrySec 4 # For Redundant Outbound proxy server

#InterDigitTimer Value in milli seconds (minimum is 1000ms = 1Second)
InterDigitTimer 4000

# options are ON or OFF
SessionTimerSupport ON

# options are ON or OFF
SessionTimerRefresher ON
SessionTimerMin 60
TimerInterval0 300
TimerInterval1 150


# Audio Settings
G711MuTxPacketLength 20
G711MuJitterBufferLength 10
G711MuJitterBufferMax 200
G711ATxPacketLength 20
G711AJitterBufferLength 10
G711AJitterBufferMax 200
G729TxPacketLength 20
G729JitterBufferLength 10
G729JitterBufferMax 200
LongHoldAlertPeriod 360
RTPPortBase 25000

# options are ENABLE and DISABLE
DiffServMode ENABLE
DefaultDiffServParam 40
RTPDiffServParam 41

# options are TRUNKMODE, CASCADEMODE, DISABLE
VlanMode DISABLE
VlanID 1
PcVlanID 2

#SNTP Settings
# choices are YES and NO
EnableSNTP yes
# SNTP Server IP address
SntpServerIP 192.5.41.40
TimeZone -6 # -6 For Central Time Zone
# choices are YES and NO
EnableDST YES
# Time period (in seconds) before recontacting SNTP server
SntpRetrySec 1800

# preferred codec are listed from most preferred, separated by commas and NO space
PreferredCodec g711u,g711a,g729

# choices are English, Spanish, and French
Language English

#choices are Enable and Disable
CallWaiting Enable
EmergencyProxyPrefix **

# STUN: If Server IP sis 0.0.0.0 STUN is disabled
StunServerAddr 0.0.0.0 #stun.fwdnet.net
StunServerPort 0 # 3478
#Periodic update in seconds of MEDIA NAT ports (minimum is 10 and set 0 to disable)
StunServerUpdateSec 0 # 30
#Periodic update in seconds to keep SIP NAT port open (minimum is 10 and set 0 to disable)
StunServerNATKeepAliveSec 0 # 30

# ALLOW IP DIALING with # as the first digit map
DirectIpDialing Enable #Enable/Disable

# Set YES to AllowSharpAsDial to have # as your DIAL key.
AllowSharpAsDial Yes #Yes/No

#Flash Based Service Parameters:
# DTMFFlashEvent (Enable/Disable) -- Send RFC2833 Event 16 when enabled and XFER button is pressed.
DTMFFlashEvent Disable
#HotLineDisplay (Enable/Disable) -- Do not display HOTLINE on LCD when Hotline Number is set
HotLineDisplay Disable

#User Agent Name Settings. Use Character "^" for space
UseCustomizedUserAgentName No
CustomizedUserAgentName Uniden^Customized^UA


#Admin password must be numeric. Max is 6 digits. Format: oldpassword/newpassword
#AdminPassword 1234/1111

#end_of_file



=============================================

# UIP200 Mass Configuration System Mac-based File
#
# Notes: Lines start with '#' are comments, it is also possible
# to append a comment to the end of a line by preceding the '#' with
# ASCII space or tab characters.
#
# To leave a field value unchanged (as saved on local phone), leave value to blank.
# To disable a field, use '-' as value
# MAXIMUM FILE SIZE IS 10KB
# Current Limitation: No spaces allowed for a setting's value
# Version: BS4.77


# Firmware. The items listed in this Firmware section must be in this order.
# FirmwareVersion and FirmwareFileName only used if AutoFirmwareUpdate is YES
# FimrwareFileName only used if FirmwareVersion differ from firmware ver in Flash
AutoFirmwareUpdate YES #choices are YES and NO
FirmwareFileName uip200_477enc.pac
FirmwareVersion BS4.77


# Sip Settings
MyLcdDisplay 31521
MyDialNumber 31521
DisplayName 31521
UserNameForProxy 31521
PasswordForProxy uniden
UserNameForRegistrar 31521
PasswordForRegistrar uniden

# Programmable Keys. Key functionality must go before key values.
ProgrammableKey1 OneTouchDial
ProgrammableKey2 OneTouchDial
ProgrammableKey3 OneTouchDial
ProgrammableKey4 CallForward
ProgrammableKey5 TwoTouchDial
ProgrammableKey6 DoNotDisturb
ProgrammableKey7 VMA
ProgrammableKey8 Mute

# One and Two-touch keys. Must go after Programmable keys functionality definitions.
# Refer to Programmable and Fixed Function Keys for usage guide
# OneTouchKeyX value is used ONLY when ProgrammableKeyX is OneTouchDial
OneTouchKey1 18005558355
OneTouchKey2 5553456
OneTouchKey3 3456
OneTouchKey4 3457
OneTouchKey5 18175553152
OneTouchKey6 18175553152
OneTouchKey7 18175553152
OneTouchKey8 18175553152

TwoTouchDigit0 3459
TwoTouchDigit1 3450
TwoTouchDigit2 4420
TwoTouchDigit3 4421
TwoTouchDigit4 4422
TwoTouchDigit5 4423
TwoTouchDigit6 4424
TwoTouchDigit7 4425
TwoTouchDigit8 4426
TwoTouchDigit9 4427

# Hotline and vmwi numbers --Must be placed after OneTouchDial's
HotLineNumber -
VmaDirectCallNo 3685 #value associating with VMA Programmable key.
VmwiLampIndicator Enable

TimeDisplay Enable
ImportPhoneBook No

#end of file



===========================================

Thread: RE: Interoperability
sup

Replies: 16
Views: 82459

05.02.2007 05:55 Forum: Providers

Just a note about Fromuser=

I think this is determined by the trunk name.

I use one provider on the PBX that I must set the trunk name to the username or it does not work.

Mark

Thread: RE: Messagenet.it - The called number answers but immediatly the line drops
sup

Replies: 8
Views: 21687

Codec 05.02.2007 05:51 Forum: Providers

High probabbility this is CODEC related

Thread: RE: Codec Errors, Fax, and G729
sup

Replies: 1
Views: 7912

Codec Errors, Fax, and G729 28.01.2007 23:26 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

Some DIDs when directed to PBXes.com result in a fast busy, like a CODEC error. Is it possible to put an option on a trunk to select CODECs?

I had a DID that was used for Fax and it is now useless. I have others with DIDs that are now useless as fax, because they seem to want to use G729, or so it appears.

Thread: RE: Other DIDs ring to my PBX
sup

Replies: 3
Views: 8956

22.12.2006 17:26 Forum: Miscellaneous

Similar , yes, however no Large routing table , and thois is on a PBX that was previously functioning.

Thread: RE: Other DIDs ring to my PBX
sup

Replies: 3
Views: 8956

Other DIDs ring to my PBX 22.12.2006 08:29 Forum: Miscellaneous

Tonight , I started reciving calls in Japanese and Portugese that should be destined for a PBX that I set up! I was able to Glean enough out of one purtugese caller to know that is in fact who he was calling. I have had from time to time when setting up other PBXes with DIDs instead of ringing to the intendd destination they ring to MY PBX!

Perhaps an addition to the call logs to show ehere the call came from more explicitly would help.

My initial assesment of this problem was not accurate. This is NOT a DID coming intoi my PBX it is after the user makes a couple of selections on the Megasystem PBX , thy appaerently then land to "ext-group" or "aa_3" on My PBX. My logs are showing inbound calls from numbers that begin with 0 . All of the countries where I have DIDs, none start the number with 0.

I am being flooded with calls in Portugese from Japan!

on a further note, I believe the call logs are not accurately displaing the inbound trunk

this is from a call
2006-12-22 02:24:37 "Free-21488" <21488> 3 07451098 ext-group 0 sec play download

that is my freeworld dialup number that I tested calling to Supernettel account number 07447600. . The PBX says it came in on 07451098 , however that account is blocked from all inbound an outbound traffic currently. It is all leaving me quite perplexed.

Thread:
sup

Replies: 1
Views: 5959

oddities of late 30.11.2006 08:10 Forum: Miscellaneous

anyone else having any of the foillowing in the last few days?:

No ouitbound calls

cut off outbound calls

strange problems in general?

Thread:
sup

Replies: 2
Views: 8625

AstMan compatibility? 22.11.2006 19:43 Forum: Terminal Equipment

I recently found AstMan which seems to do what the grandstreams do with the BLF display. Does anyone have a clue if it can be used with PBXes.com? There seems to be no documentation and it has a sample Monitor.comf, which I assume is already on the PBX to support the grandstream BLF

files are here:
http://www.isisdev.com/astman/astman.zip

Thread:
sup

Replies: 4
Views: 11718

18.11.2006 05:48 Forum: Miscellaneous

Disabling some CODECS seems to do the trick as my SIP dialing from Xpro was unsuccessful till I diabled other CODECS. It looks like fax MUST be g711u

Thread:
sup

Replies: 4
Views: 11718

16.11.2006 21:10 Forum: Miscellaneous

I can see in the logs that PBXes is still ansewing my dedicated fax number, however the calling party hears nothing . This began after yesterday's outage.


ALso worth noting, was that I was not able to make calls , so the trasfer to the de server was not transparent. perhaps this is a problem with the way my ATAs are configured?

Thread:
sup

Replies: 4
Views: 11718

Server Was down! 16.11.2006 04:42 Forum: Miscellaneous

Just a comment, this server ,. www2.pbxes.com, as well as my server in the same data center were down for a while about 8 hours ago.

It looks like domainGurus Lost all connectivity, and I know that in the case of my server , it looked as if it had rebooted.

Since , my dedicated inbound fax number is answered with silence and I have had three calls turn into one way conversations at random points but normally after 15 minutes, I now think there may be something not quite right..

Thread: RE: Eyebeam softphone and Hold
sup

Replies: 5
Views: 15376

12.11.2006 06:27 Forum: Terminal Equipment

The latest in this on going saga is that the problem went away after making the changes above, but now has re appeared!

I am looking for suggestions.

Thread: RE: inbound routing
sup

Replies: 1
Views: 7250

ingound routing 06.11.2006 06:08 Forum: Miscellaneous

This may or may not relate to a thread that I posted earlier where the inbound route is followed for PSTN calls but not SIP calls coming from the same provider as the registered trunk.

If I have a trunk registered for example 123456@sip.supernettel.com , the server at sip.supernettel.com can successfully send a call every time, however if that server also sends calls via a second IP address, they MAY not follow the inbound route correctly. I have tested this in the following way.

inbound route trunk name is 123456 (example only) , registered to 123456@sip.supernettel.com
I can send a call from sip.supernettel.com , but dialing to the URI of 123456@www2.pbxes from xlite (registered to FWD) , yields a recording on an unrelated PBX.

This routing should work REGARDLESS of the originating IP, with the following case in point:
Many providers send calls from an IP address different from the registration IP address.

I believe what is happening is that the PBX server is allowing that inbound route to be used only from the IP address from which it is registered to for that particular trunk.

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