cwernstedt
-
Posts
103 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by cwernstedt
-
-
how did this work out? I am interested in trying this. I dont know if I will ever deploy it, to expensive for what it does.
I'm doing it with Vegastream ISDN devices. It works well, however I needed to bypass pbxnsip for incoming fax calls. I route those straight to NET SatisFAXtion through dial plans in Vegastream.
The system is very scalable, reliable, and you can send and receive faxes as email attachments.
I've got T38 transmissions to work over rather flaky Internet connections too with Net SatisFAXtion as an endpoint. (Though here, reliability is not 100% especially if the calls come from ISDN - a clocking issue, I believe.)
/CW
-
Hi,
On a client's system, I have noticed that calls that are forked and picked up at a user's mobile phone are listed in the system's call log, but do not show up in the user's web portal call log.
Call trace emails do not seem to get sent either (but I haven't tested this as thoroughly.)
Is there a way around this problem?
My customer is running 2.1.14.2498, and cannot upgrade to 3.X at this time due to very heavy use.
Regards,
Christian W
-
some of the white boxes came with a WAN port but the system didn't support it
OK. This must be the situation here.
Thanks!
/CW
-
Hi,
I can't find the web settings for the WAN port on a white CS410 w. 3.0.0.2998 . Under System/IP Setup I can only see "LAN Port".
Is there a solution to this problem?
/Chris W
-
We did that in the beginning. The problem is that (outside of America), you want to be able to call another extension directly without having the cell phone detection kicking in. In USA, most of the time the AA is sitting on the main DID number and then it is okay if the PBX detects that a user is calling.
I'd also like to chip in that CID as a way to identify a cell phone is not a 100% reliable method. When I travel outside the USA with my phone, CID from my cell phone may not come through at all, or may be represented in a way that the pbx misinterprets. Having the user enter an extension# + pin, is the only truly reliable way to authenticate. This method also makes it possible to allow a user to call into the system and be authenticated from say a client's phone or a pay phone .
/CW
-
Hi,
A client has two pbxnsip systems running, and I have set up a trunk on each pbx to route calls to the other, along with dial plan entries such that when dialing an extension which resides on the other system, the call is routed through the corresponding trunk.
A problem I have is that when an extension on one system is called from the other, the domain name from the originating system is present in the SIP URL that the receiving system (and it's registered phones) sees. When a user then wants to call back to that number from their phone, the phone will in vain try to contact the originating pbx's domain, and the call cannot be placed.
I know this could be solved by setting the "outbound proxy" parameter in the phones, but there are 50+ phones with no simple provisioning, so I'd like to solve this on the pbxes instead.
I would like to replace the domain part in the SIP urls for incoming calls on each respective "intra-pbx" trunk with the domain of the receiving PBX. (E.g user1@domainX will look like user1@domainY )
Is this possible? (Or is there some other way to solve my problem without specifying outbound proxy on all the phones.9
Regards,
Christian W
-
Try *90123 if 123 is the extension that you would like to wake up!
Works much better!
-
Hi,
I'm trying to set up two way intercom with Snom phones, but can't find much information about the intercom feature here or on the wiki. (There's a lot of info on paging, but that's not what I'm looking for.)
The first problem I'm running into is that when dialing the assigned *-code (*90 in this case), I just get a voice saying "This feature is not available at this time".
Any advice would be appreciated!
Regards,
Christian W
-
There must be something wrong with the domain email setup. What is the content of the field "email_from" in the domain (maybe do a quick check of domains/1.xml). What version?
Thanks! The cause of the problem turned out to be that the email-info had been entered in the global email settings, and not in the domain's settings.
/Christian
-
Hi,
I'm trying to make the box send email messages, but whatever i enter in the email configuration fields in the logging section, I get the error message below when the box tries to send an email.
"No valid source address for sending email to user"
I don't have this problem in the non-embedded version of the software.
What is the address format that will work on the box?
Regards,
Christian
-
The latest version (see post in this forum) uses AGC for this problem. And we keept the overall volume relatively low (that's ovciously what most analog handsets vendors do). That did the trick in our office, where we also had a nice long list of audio problems. Polarity change should be included in that version as well, so it is definitevely worth a try.
I'm using update-2084.tgz for these tests, so the polarity change may either not be included in this build, or it doesn't work in Sweden. (It's as if it works at times, but sometimes not, so it's random.)
Regarding the volume and AGC, I think things would work better here if it would be possible to lower the volume just one notch, but the problem with the line staying up when it shouldn't is what concerns me most at this point.
-
Some countries use polarity change to indicate disconnect. Maybe Sweden belongs to this group.
I'm almost sure that Sweden uses polarity reversal, so if you can put that in as an option, the CS410 will propably work in Sweden. (And most likely in the Norway, Denmark, Finland as well.)
I also noticed some overheard noise when speaking loud into the handset, which indicates that an option to lower input/output gain might be handy.
-
We worked on a better international support in the latest version (posted in the cs410 forum), it is surprising how the different countries are using the analog wires. I would say it should work in Germany and Italy, possibly in UK.
In Sweden, it doesn't seem to properly handle tearing down the line when it should. (E.g. when the remote party has put his phone on hook.)
Perhaps you need to allow for different countries' disconnect tones.
/CW
-
Of course it is a soft limit. But we don't want to end up selling every license one by one.
Yea, the US phone system works slightly different than in the UK. The first focus was on USA, but later versions also consider DTMF caller-ID and polarity change disconnect indication. Not sure what applies to UK. The good news is that the hardware should be able to do it, it is a standard FXO hardware you find in every gateway.
What would be a compatible one or two port external analog FXO box to use until the built in FXO gateway can be used in Europe?
I'm currently trying a Sipura SPA 3000, but it doesn't seem compatible with pbxnsip for placing outgoing calls.
/Christian
-
Well just SSH into the box, download it and start it. It is a deban computer...
How do I download to the box with the http-protocol?
To which directory should the file go?
How do I start it?
(No, I'm not particularly familiar with linux.)
We will include this in the next batch, for those who need this they need to manually download and start it. If you always want to start it, include a line in hte /etc/pbxnsip file - before starting everything else.Is it safe to FTP transfer the /etc/pbxnsip file to a desktop computer for editing, and then transfer it back (replacing the original)? Do I need to check and/or reset permissions?
I want to make sure that I don't break anything because this unit is in production, and I'm at an ocean's distance.
Christian
-
There is a program ntpclient at http://www.pbxnsip.com/cs410/ntpclient that you can use for this purpose. We will include it in future releases, feel free to use it now.
This tool will set the file system more or less to "now" (at least, GMT). The PBX itself is able to deal with different timezones (at the same time, because every extension may be in a different time zone). This is configured in the timezone.xml file, http://wiki.pbxnsip.com/index.php/Localization#Time_Zones.
How do I install and configure the ntpclient? Are there instructions somewhere?
Christian
-
Hi,
I have installed a CS 410 for a client. A problem is that the unit does't indicate the proper time and date in logs.
I have tried using the date command over an SSH session, but that setting doesn't take into account that summer time changes, and time is not retained over restarts.
How do I set the time and date and how do I make the CS 410 retain proper settings over restarts, and how will the appliance account for summer time and winter time? (Is this documented somewhere?)
/Christian
-
When a user receives a new voicemail, the PBX calls the users' cell phone and reads out the new message.
Of course Caller-ID is not secure. That's why you must enter a PIN e.g. if you want to go to "your" mailbox or place an outbound call. A good reason to choose a different PIN than just "1234".
Why not let the calling card function and its authentication process work properly as a means of access to the system? I know I'm repeating myself, but I am really very curious about why my suggestions in this area are not addressed. The present limitations of the calling card (e.g. not being able to call mailboxes) seems more like bugs, rather than reflecting conscious design decisions about how to support (or rather, as it stands, make life complicated for) mobile users.
/Christian
-
First of all, if you want to turn the feature completely off, change the setting "camp_enabled" in the pbx.xml file.
OK.
I'm not sure that I understand the part about mailbox readout. Can you clarify? Is what you are saying that anonymous cell phone numbers, or callers with messed up or suppressed CIDs, or callers borrowing a friends/client's phone can't be supported to access mailbox readouts when outside the office?The camp on is only offered to internal extensions and to cell phone callers (must be somewhere on the list of the cell phones). The number must be dialable anyway, otherwise also other features like mailbox readout are not working either.
I strongly recommend that you have a "default" dialplan that works without any prefixes. It is okay if you have special prefixes for special routes, but you should have always a route that works on standard numbers. For example, you need that feature for callback (*69) anyway.
My biggest customer (a multinational company) has no standard way of routing calls or identifying mobile callers such that CID without prefixes could be used to route calls according to the client's policy and needs.
Further, I think that the PBX's dependece on CID to support for mobile caller's access to the system is a big flaw. Setting aside the lack of flexibility this model creates, there is also the issue of the unreliability of CID (e.g. My cell phone's CID often becomes suppressed when I travel outside the US). Why not fix the calling card function? Basically what's needed there is for the calling card to let the mobile user do eveything permitted from a fixed IP phone (given that user's or the calling card's dial plan) after proper authentication (e.g with a user code+pin).
I know I'm whining about this, but this is a major issue that I need to solve to make my customers happy.) Look here for a list of the hurdles for mobile users.
/Christian
-
This seems to have happened with 2.0.3:
When callers get transfered to a user's mailbox they will hear an announcment saying that they have the option to either be called back when the extension becomes available, or to leave a message.
When pressing #2 to leave a message, they will hear the recorded greeting if such is available.
This new behaviour of the PBX is very annoying for the following reasons:
1) The call back option will not work with complex routing setups (e.g. if a special prefix is needed to route an external call), and it is not always desired that a caller gets this option anyway.
2) Being told that when pressing #2 one will be able to leave a message, but instead hearing the user's greeting, is illogical.
Is it possible to turn off these new "features". They create confusion for users, and are not helpful.
When the mailbox kicks in, the greeting (personalized or anonymous) should be played without any other options at that point.
/Christian
PS. The calling card system is still a mess.
-
We recently upgraded from version 1.5.2.7 to 2.0.1.1624. We needed to upgrade because we are anticipating some larger installations. In our install we use Snom 360 and 320 phones. Since we upgraded we encountered the following problem.
If the extension is on the phone talking to one party and another call comes in, if we put the first caller on hold, talk to a second caller for some time and then try to go back to the first caller, we cannot get back to the original caller. We basically get back bunch of static. It almost seems that media traffic is boken. At this point the phone is very confused and needs to be rebooted in order to work again since it believes that there is still a call on the first line. All this was working fine with 1.5.2.7 does any one have any suggestions?
Dusan
I have not seen this problem with a Snom 360 w. current firmware, but had several problems off all sorts with earlier versions of Snom 360 firmware. Do you run the latest version of Snom firmware?
-
He is probably talking about http://wiki.pbxnsip.com/index.php/Auto_Att...ct_Destinations
OK, well, I don't have any direct destinations entered.
The AA is freshly created with no special configuration except for the setting that specifies that the user must press "#" before the search can start.
/CW
-
are you sure you don;t have any overlaps in the extensions and the Direct Dial options 0-9 ... that has happened to me a lot inadvertently ...
You should document these if they are easily reproduced and send them directly to KM or CS ..
yori
Hi,
I'm not sure what you mean by direct dial options. Where do I find them, and can you give an example of how an overlap could happen? (Searching the Wiki for "direct dial" leads only to a description of the mailbox direct dial prefix.)
/CW
-
I have noticed several calling card related problems that are very annoying, and that I'd like to report:
- Calling directly to mailboxes via a calling card simply doesn't work. (It doesn't matter what the dial plan looks like.)
- When an extension has "Call Forward All" to external number, it will not work properly when called from a calling card in the following ways:
-
If the called extension has UAs registered they will ring, which is wrong -- the call should be forwarded.
-
If the called extension doesn't have UAs registered, there will just be silence with no notification to the user about what's happening. (Double wrong!)(In the latter case, the PBX log shows "Could not start call to extension 1141 because there is no registration or the extension is busy" .)
- Calling card doesn't work correctly with extensions that have a cell phone number attached. The cell phone will not ring.
- If calling an extension from a calling card, the caller will never wind up in the mailbox regardless of the called extension's mailbox time out settings.
Now, one might recommend using the new cell phone features in the auto attendant instead of calling in using a calling card. However this is not a satisfying solution for the following reasons:
- If a prefix is necessary in the cell phone number as entered in the user's settings, CID matching will not work. (E.g. the the number entered is the settings is 550019174110101 in order to route the call correctly, but the cell phone presents itself as 0019174110101 when dialing in, resulting in a non-match.)
- CID info is not always tansmitted correctly anyway. Its reliability depends a lot on which operator the phone is attached to. (For example, when I dial numbers in Europe from my U.S phones, the CID may often be suppressed.)
- A mobile user may need to dial in from a different phone than the cell phone having the number that has been entered. (e.g the battery may be dead.)
- Calling card functionality in cellphones (e.g some Ericsson phones) usually don't have provisions for navigating complex menus. Typically it is possible to send access code + the desired number + #, and that's it.
- The need to navigate menus should be kept at a minimum since mobile users may be walking or driving a car while operating the phone.
With the above described problems, I would say that support for mobile users in pbxnsip is currently lacking in several essential areas.
Regards,
Christian W
- Calling directly to mailboxes via a calling card simply doesn't work. (It doesn't matter what the dial plan looks like.)
Registration fails and the log shows "Final transport error"
in MacOS related topics
Posted
We've had a problem with 3.4.0.3201 (running under Mac OS X Leopard) such that a phone may suddenly not be able to register.
In the log the following lines are repeated as the phone tries to register.
[8] 2009/11/16 14:56:35: Timeout on UDP transport layer (1512382)
[5] 2009/11/16 14:56:35: Final transport error on 1512382: REGISTER
The pbx's operation in other aspects seem to be OK.
Re-starting the machine on which pbxnsip is running resolves the problem.
This has happened about two times over a period of three months, so it's not frequent, but causes a good deal of disruption.