Friedom-Tech
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Posts posted by Friedom-Tech
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Well, you can deal with that. But then you have to work with timeouts (in the AA, direct destinations). It works, but is not very beautiful.
well this issue is not only the auto attendant but internal users who are trying to call each other!
is there no way to make a dial plan that only calls with 1XX should use the trunk that connects the two systems and all other calls beggining with 1 IE 1XXXXXXXXXX should go out through the regular sip trunk?
please advise asap
I FIGURED OUT HOW TO DO IT...NO PROBLEMS AT ALL!
all i did was made a dial plan as follows:
Pref Trunk Pattern Replacement
100-NY 1xx
100-NY 2xx
110-Callcentric *
so if the number beginning with 1 has three digits it goes out to NY trunk and if it has more than three digits it uses callcentric.
is this good??
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1XXneeds to stay the same because the cards are already printed...do you have any options whatsoever to resolve this?
i will try the other suggestions and let you know.
This is a case of having seperate domains that want to call each other. Although they are located in different states, the case is not so much different than having the domains in the same data center, but on different CPU.First of all, you probably need two trunks in gateway mode. Both of them should trust each other, e.g. turn the accept redirect on (that should help with the question 2 below).
I would stay away from prefix 1, it has a lot of problems (1xxxxxxxxxx being one of them). Better choose something in the 3xx-6xx area, if the business cards are not printed yet... You can use pattern 3xx and no replacement to route calls that have three digits. If you want to make a routing entry for x11 then you can just give that one a lower priority.
You need to set the setting "Assume that call comes from user" for that. Then the PBX can use the dial plan of that extension, and it can also charge that extension.
Hmm. Ideas are: A. Use the direct destinations (if there is still space available). B. Set up "ghost" extensions with static registrations that point to the other system, have no mailbox and an impossible-to-guess password. Not very beautiful, but that might solve the problem if there are noo many extensions in the other location.
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Hi what would cause the system to continuously change the ports of the phones?
[3] 2008/11/04 09:22:58: Source address for sip:712@xx.xx.xx.xx has changed to udp:xx.xx.xx.xx:50702
[3] 2008/11/04 09:23:06: Source address for sip:715@xx.xx.xx.xx has changed to udp:xx.xx.xx.xx:50703
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HI;
i am in the process of setting up two offices in two different states.
i have one pbx in each location, i have successfully trunked the two pbx's but have the following issues with dial plans.
The extensions in Office #1 are 1XX and 2XX and the extensions in office #2 are 3XX.
My questions are as follows:
1) What is the best way to setup the dial plan that when office #1 wants to call office #2 they can enter their extension number only without having to add a fourth digit to the extension. (ie a dial plan with 3* and a replacement of 3*??)
in addition to this since office #1 has an issue where some of their extensions begin with 1 (which is the possible prefix to all US calls) what is the best way to maintain 3 digit extensions without running into issues?
2) I would also like to setup that if for whatever reason the SIP trunk from office #2 goes down they can backup off of Office #1's PRI line for outgoing calls...how do i set this up?
3) When someone calls into the Auto Attendant at office #1 i need for them to be able to call office #2 by dialing the three digit extension (currently it gives a message saying that "you do not have permission to call this extension")
please advise asap as i am going live this week.
Thank you;
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can anyone explain to me why when i switched the default gateway of the CS410 to the lan port instead of the wan port it stopped giving me access remotely over the WAN port?(both for setup AND remote phones)
i have a numbe rof setups like this when the default GW is the LAN port (for redundancy with a dual wan router) and i have no issues getting in remotely using the static wan ip address.
using version .3023 of cs410 firmware.
pls advise asap.
Thank you
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Well, I just tried with the personal address book of the extension and that did work as expected... Not sure what the problem could be here. Or are you using the domain address book?
i am trying to join all the phones to the domain address book...an dnow that i think of it this logically makes sense because all the phones are setup to connect to the personal address book so what string do i put in for it to access the domain address book?
pls advise.
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Option zero is to apply the latest software. I guess that has been done already.
Well, one option is to move the setup to a PC-based solution. Just save the config and restore it on the other system. This is a short-term measure to make sure the customer is happy. If you have a PSTN gateway that you can use, and the call quality is still bad, well then there is a problem with the line.
Of course the second step is to ask why the quality is bad with the CS410. What does the CPU load say on the web interface? Is there a QoS-problem in the setup (shared data/voice internet connection)? The CS410 has tcpdump locally on the system, this way you can get a PCAP trace to see what is going on.
We had a case where a web browser was constantly requesting web pages from the CS410 in a DoS style. After moving the port away from port 80/443 the traffic was much less and the box was working fine again.
Thank you for your speedy response.
they have a shred connection for data but they have 30 mb downl and 5mb upload....they had an asterisk system before pbxnsip and claim to have never had call quality issues like this...could it be possibly that i need to compress the packets? would that help? if yes then how would i do it? the system is setup with the latest verison of the software...any advise would be appreciated.
thank you
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Are you using buttons or dialog-state? I tried this here with buttons and it seemes okay to me.
Try hitting the save button on the button profile again. If you change (or create) an account that you list there after hitting the save button, references might be broken (we'll fix that problem in 3.1 as it really can drive people nuts).
so you are saying i should create the account and THEN hit save on each button configuration?
do i need to reboot the phones?
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You should really use PnP, at least for the initial config. The address book requires that you calculate a hash over the username and the password, and that is really difficult if you want to do it one your own.
i have my phones setup with PNP but the address book wont show...it only shows the list of local extensions.
what do i need to change in the string for it to work?
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Hi;
i have a client with a CS410 and 7 SNom 360 phone s all connected through PNP.
he is experiencing terrible call quality sometimes even on calls from ext to ext....he has had the system less than a month and is extremely disappointed with the call quality which is a shame because he loves the features!
please can someone advise what my options are?
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If you have a local PSTN gateway, you would use something like "<sip:12121234567@ip-of-the-gateway;user=phone>".
i use sip trunks only so how would i input my cell number?
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Checkout this wiki link http://wiki.pbxnsip.com/index.php/Manual_Registration
what string would i put in to make a static registration for my cell phone?
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Don't call the button "13", better call it "dnd".
That feature requires version 7.1.33 on snom.
I have a client with this same issue! he is setup with pnp with a snom 360...when he hits the DND button on his phone it changes the status on the pbx to do not disturb but when he turns it off it does not turn it of on the cs410.
what can i do to rectify this?
pls advise asap
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What phone model?
snom 360
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The call from sip:1732961****@sip.jivetel.com;user=phone to sip:1800624****@sip.jivetel.com;user=phone has been disconnected because of media timeout (120 seconds), 13071/19057 packets have been received/sent
What does this mean????
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The PBX displays the first 9 entries that match. Because there could be thousands, there is a limit. The point is here that you need to "drill down" by entering more digits. For example, if you search "Valerio", you would enter <Adressbook>, 8 (TUV), 2 (ABC), 5 (JKL) than then usually the list should be short enough that you can use the arrow keys to select the right entry.
Hi i am having an issue where the phones will only show the addresses of the extensions of the pbx. the client needs his address book...can someone please provide info on what string i need to put into the phones/pbx for this to work?
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I would factory-reset the phone to make sure that there is no residual configuration on the phone. Then use the BLF mode on the buttons. That should work just fine.
Hey two weeks later and still having the same problem. i am setup as follows.
cs410 with pnp to snom 360s. the inbound trunk rings to a hunt group for 15 seconds before hitting the auto attendant.
if an employee picks up the phone from the hunt group (in the first 15seconds) that call cannot be monitored by the other phones?
if i switch the trunk to ring to an agent group for the first 15 seconds i do not have this problem!!
please advise as i promised my client call monitoring and he is driving me nuts!!
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Checkout this wiki link http://wiki.pbxnsip.com/index.php/Manual_Registration
Thank you for your speedy response.
where do i put this string though? in the user account setting on the phone i am registering or somewhere in the settings of the extension? pls advise.
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No, that is not a bug. The idea that people fork potentially calls to a lot of cell phones (half of them being powered of, in the tunnel and redirected to the mailbox) just do not sound like a solution to me. The agent "mailbox" always picks up immediately, and the PBX has a very hard time figuring out if it is a natural person picking up or a machine.
Though there is a work-around. You can add a static registration to the extension, then the PBX will also include the cell phone in the group. Maybe you can try this and see if it meets your business demands.
I believe it is better to try the agents in house (potentially using hot desking), then if that fails escalate the call to a specific extension and potentially also forking it to a cell phone. If then the mailbox picks up, okay then it is last resort and even the caller cannot expect much more.
how can i setup a static registration? i have the same issue where the person on call sometimes has to step out of the office for a few minutes and needs the calls to simultaneously ring on her cell...pls advise the workaround asap.
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HEY GUYS.
i setup the pbx with the pnp functionality but the issue is still ongoing!
in addiotion to this its a real pain for the client who likes to use the buttons to monitor AND speed dial (which he can do when he sets them up from the phone itself) but when he sets up the buttons through the pbx it doesnt work as both!!!
pls advise.
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Speed dial codes should be in the range 0x-5x anyway, which does not overlap with 90. So you won't loose anything. And you can still just three digit speed dial codes, giving you around 500 speed dial entries.
Plus in the 3.1 version you can also use a intercom code which starts without a star, making 9123 possible (9 would be the star code for intercom and 123 the extension). Together with a suitable dial plan on the phone, that should get the job done quiet nicely.
i tried that and it didnt work!
is this applicable to the cs410 software too?
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Did you set up the snom phones manually using the function keys or with pnp using buttons? I would recommend to use buttons instead of the function keys. I couldn't read the attachment.
i have the buttons setup in the pbx. the setup is as follows
1) i created a button for each extension and then entered that button into the configuration profile or that user (ie button that monitors extension 600 was put in the the configuration of extension 600)
2) i logged into the snom phone and under function keys pointed "extension" to extension 600..
i am NOT using PNP. would that do the trick? why would it only not work for calls to the hunt group?
pls advise
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i am not monitoring a hunt group extension, just the phones. I called the hunt group and i saw the BLF's blinking and such
i am trying to monitor the extensions too! but hen the call comes in directly to the hunt group and someone picks up it wont monitor that call.
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works for me.
even for the hunt groups? is there a setting i have setup wrong? pls advise!
PHONES CHANGING PORTS
in Embedded
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i am using a linksys RV082 this should work no problem if setup correctly right?
can you advise which settings need to be changed?