brandywinetech.com
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Posts posted by brandywinetech.com
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nice explanation Maestro, I learned something too !!!
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'
'go into the phone and set
advanced -> audio ->
Alert Info playback: off
Audio indication for Dialog-Info Call Pickup: off
reboot ..
see if that fixes it ..
yori
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you can absolutely control OB caller ID based on the users extension , your reseller should be able to set this up for you ..
I would not bother with 10 trunks, if your Linux box goes down that much you have a bigger problem to give up excellent PBX functions for "just in case" scenarios ..
get the single trunk and do it right, make a hot swap box in case the server crashes and have the provider allow you to route calls to your cell phones if your SIP registration is lost ..
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The pbx will not route calls based on the protocol, it justs respects T38 and does with it what the SIP signaling says to ..
I generally keep fax lines out of my VoIP systems as a rule, I have found it's more headaches than it's worth .,.
trying to save the customer $40 a month for that line could cost you thousands in service calls when you are constantly going back for intermittent issues that are very hard to reproduce and become bothersome ,
Unfortunately the old Avayas and Panasonics were very good using TDM signaling to allow the customer use of the 4th line ,
what I do is using the Audiocodes or Patton gateways I cancel inbound routing for line 4 on the gateway (the fax line) but I allow outbound calls on it and take it out of the hunt group ,... thus it is available for outbound calls but only routes to the fax machine,
maybe a nice compromise ,,, as for the pbxnsip admin suggestion to use a DID, he is in Europe where ISDN is everywhere they have it good there ..
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there is a 2.1.5.2357 version out that would be worth upgrading to first ,
but you should be all set from what you are describing , you are doing it correctly ..
I would try to narrow down where the bug is, try to reduce the timeout to say 10 seconds, then try to redirect to an extension, then an Auto Attend, then maybe a cell phone and see if any of them work ..
also check the SIP logs and be sure the time and date is correct, this could possible casue an issue for you ,
and third when you test it , crank the SIP level to 9 and see what happens after the 10 seconds, if you don't see a redirect request then it is obviously the queue that is the problem (software issue with the feature) or if it fails then it could be an issue with it not liking the destination etc ..
2.1 has some major upgrades to the ACD functions and they are supposed to be much more dependable now, the old versions were rather buggy ,,
grab a quck ethereal trace off the server too and filter for SIP, it will give the best info ..
yori
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The UDP ports in your trace are RTP and not the SIP signalling ,
you are using TLS/TCP for your SIP and yet still using UDP for actual audio, which is your problem ,
I find it odd you believe Snom has TLS issues when they were one of the first adopters ...
I would utilize your VPN to get rid of the remote natting, that would solve the issue right there .. and be double encrypted ,
yori
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2 different topics,
SRV is basically how SIP uses DNS to create a failover mechanism. With MGCP, phones are given multiple media controllers that share a common database and the phones can try different registrars, with SIP, there is only one supported registrar, (at least that I have seen) ..
thus the only way to handle it is to tell the phone to go to say sbc.mypbx.com ... then DNS will give out IP addresses in priority order based on standard DNS practices , if one server is down it will go to the next in the list , better than hardcoding IP's, then you can take a server offline for repairs and DNS will resolve to the next in line .. So I can't see how this makes it more secure, a the other option is registering to 64.12.34.45 etc, then if you have an issue, good luck reprogramming 100 phones on the fly ..
TLS is the encryption of the actual SIP and RTP messages, this means I can run ethereal and intercept your call mesaaging, but can't see it or hear it in a trace as it will be encrypted and I don't have the private key ..
yori
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FYI,
when that happens the best way to find out what's up is to look at the SIP log and see, if you see 500 errors then it's a network issue and 400 errors means a password issue for the most part ,
at least narrows it down a bit, the Linksys phones have no internal logging so I use Kiwi on the server to send syslog messages to ,..
good luck and good job ..
yori
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unfortunately a crossover cable and ethereal is the only way ..
from experience,
the newer models will include the "paper clip" reset we all love and know ..
yori
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the Snom phones aren't your issue, any phone will act the same way . Routing is routing is routing, albeit windows or Linux or XP or Vista ,
if you have dual NIC's be sure only the WAN side has a gateway, the inside should be without a routable gateway and no matter what you do, the NAT wil be an issue,
opening ports has nothing to do with NAT only on the firewall , the PBX has a SBC session Border controller for resolving NAT issues,, (so stun is not an issue) but if both sides are natted that will be an issue for you
having a 192.168.x.x pricate IP on both sides of the PBX is the issue ..
yori
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I avoid faxing through the system, IMO it's more of a hassle than it's worth ..
Just an opinion ...
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Try using an xten softphone with a headset , if that does the same thing, you have an issue with the PBX, if the issue does not happen on that (or a different phone) you can at least narrow down where the issue is coming from ..
are you using a SIP provider I assume and not an analog gateway ?
yori
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We currently have over 75 pbxnsip systems deployed worldwide, 40+ users is no sweat. Use a good Patton or Audiocodes PRI gateway, good phones i.e. Polycom or Snom, and as long as the box is a Pentium 3.0 or so with a couple GB of memory, you will have no issues ..
I use the Netgear POE switches and they work quite well also ..
yori
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I want to use IVR for reading directions to customers as a choice from AA .. can I do this or should I use Auto Attendants for this ?
it just repeats the message in the IVR forever with no way to allow the caller to back out to a previous menu ..
yori
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In my AA, i want to give the user the option to back to the previous menu with # or *... I input them into "user input" and add my extension to the "desitnation" But it is over ridden by the " please enter your extension number" since most systems allow for this without having to say " please enter 9 to return to the previous menu" and eat up a digit ..
what can I do Doc ?
yori
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I ran into this today .. since the Auto Attendant follows the default dial plan, it tries to send the call out as an external call .. try putting say 8202 in the user input and then 8202 in the destination and see if it works ..
for some reason this way it recognizes it as a local extension and doesn't send 8202 as an external call to the dial plan ...
hope it works for you ..
yori
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contact me at ykasprzak@brandywinetech.com, I got the first pbxnsip SLA lines to work about a year ago ... SCA unfortunately doesn't exist yet .. we can get you a new demo key ..
yori
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in 2.0.3.1715 line monitoring of co1 co2 co3 etc works fine,
I updated to 2.1.0.2098 then it stops working ..
Windows system
Snom 6.5.10 360's ..
is this removed in 2.1 ?
yori
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The Patton gateways are quite nice ..... I have used a few for PRI/TI connections ..
Be aware that they are very CLI friendly, though it has a GUI, it is not that usable.
Patton has great support and be prepared to use it ..
I like the Audiocodes M1K, it is also quite nice and ALOT easier to figure out, but the support is not as good as the Patton ..
contact me if you need any help, I am a reseller for both products and can help you with either ..
yori
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Hi,
I need help with setting up my AudioCodes gatway with my CS 410 appliance. I can dial in and out of the CS 410 with no problems but I'm having problems getting incomming and outgoing call to work with the audiocodes gateway. I bought the equipment through atacomm would I be better off getting professional paid support from them or purchasing 1 -3 hrs of support from pbxnsip.
If you can get me remote access to the system or if you wish, contact me at 888-413-2131 .... I know the Audiocodes devices quite well ..
I don't know Atacomm's policies, but I think they advertise that they do NOT offer free support for configurations, but they may offer pay support as they sell pbxnsip and Audiocodes ..
feel free to buzz me or email ykasprzak@brandywinetech.com
yori
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bump
just want to see if anyone has any ideas on this one.
I have had luck with 2.0.3 and 6.5.10 doing the line monitoring ..
there is supposedly a "new" way of doing SLA with pbxnsip 2.0.4.x and the Snom 7.x firmware releases that is going to be more bullet proof ...
this should all be released within the week ,, hopefully some info in the release notes ..
yori
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I have always had to do this with SIP trunks using the dial plans as discussed here .. I tried for a while to get the pbx to seize a line on the Audiocodes with no luck .. it seems to be an Audiocodes limitation .. or you may be able to do it with hunt groups .. I just never had time ..
from what I hear this is doable with PAtton or Quintum gateways, but as you know I am mostly and AC guy from experience ..
call me if you need me ...
yori
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the 1.5.2.7.exe is not an installer, rather a service executable ..
rename it to pbxctrl.exe and overwrite the existing c:\documents and settings\administrator\application data\pbx\pbxctrl.exe
you have to stop the service to get permission to overwrite it ..
BTW, make a backup copy, just in case ...
make sure you have the park feature set to "strict" .. if it is set to loose, it will reject calls parked to an extension and want just vanilla *85 and *86 .,
yori
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this exists in Comdial and Nortel Systems ... it allows the user to hear the message ":real time" while the call is being screened ..
then they can grab it if it is someone they want to talk to without having to go through the whole process ..
basically an easier TDM funtion than for IP Packet based systems ..
yori
Active call status
in General Setup
Posted
I look in active calls and I see the audio quality of just one , any clue why
image.tiff