shopcomputer
-
Posts
452 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by shopcomputer
-
-
Well actually we are using DD-WRT in a Cisco router, so I think it is already prepared for that, in fact the SIP signaling ports are already working.
Still no sound during the calls.
What model router are you using? Did you forward the RTP UDP port to the pbxnsip? This is definetly I firewall/ router issue.
YYou may also want to check http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses
-
Well I checked these ports and they are from 49152 to 64512.
Also I made sure that from our router these ports are being forwarded to the PBXnSIP computer, however when from outside sombody tries to check one of those ports it appears as closed. So I wanted to investigate a little bit more; I logged into the PBXnSIP server, I made a "netstat -a" command, and I noticed that none of those ports were under "LISTENING", but the SIP ports, the SMTP ports, etc... were, so I think the origin of this issue is not the router configuration, but the PBXnSIP setup up.
Any ideas?
Thanks.
These ports are UDP, not TCP, so a telnet to these ports won't show. They must be forwarded to the pbxnsip server. Also you may want to get a SIP aware router.
-
Whow maybe Sprint is now also using VoIP and they are mixing ports up? Haha, just joking.
Another easy thing to check would be that you have enough RTP ports. Having 1000 RTP ports does not hurt, even if you have far less calls.
If the problem does not go away, you might have to start Wireshark (maybe rotation mode) on the system. This will involve some data mining, but then the root of the problem will become obvious.
The RTP range is from 49152 to 64512 which is over 15 thousand.
I don't think it is rtp crossing, it sounds more like the call is going to the wrong number, the other party always picks up hello, they must of heard ringing, which does not come from the rtp.
I first called Broadvox as I thought they are routing the calls to the wrong IP or have a backup route set to a bad phone number.
However they checked and they say they sent the calls to our number, and I do see the same in the call logs.
The problem I may have with wireshark is, the amount of time it takes to reproduce the problem.
-
I mean the following: get the call from a sip trunk and depending on the number dialed, terminate it on a local extension or route it to another sip trunk.
The thing is our company is going to be pbxnsip reseller in Russia and we have several potential clients for distributed ip telephony systems.
And we are evaluating where we can offer pbxnsip.
The typical scenario could be:
Meidum corporate (couple hundreds of subscribers). Main office and several branch offices with pbxnsip in each office. Common PSTN gateway(s) or ITSP link(s).
The incoming call must be routed to appropriate office according to telephone numer.
Incoming (from PSTN) calls should be routed on main office pbx, i mean that routing table on a gateway (and espesially on ITSP equipment) is unwanted.
I understand that one could have just one pbx in main office and create domains for branch offices, but this is not always suitable mainly because of the ip link failure paossibility.
That's why i'm asking about transit calls possibility..
Thanks in advance,
Nikolay.
You can have trunks between multiple PBXnSIP servers at various locations, and have the PBX send the extension numbers for the other offices over the trunks.
Set Office 1 with 3xx extensions office 2 4xx entensions, etc. and set the dial plan to use the corresponding trunk.
-
This is usually a sign that there is a port conflict with RTP. For some reason, two calls point to the same IP port. This might be caused by the router (if it is buggy) or maybe the carrier has a problem there (less probable I would say). Some routers have only 32 NAT entries, and if you have too many NAT bindings you may have an effect like this.
How to troubleshoot this? Difficult. If you have a cheap router, I would just try another router from another company and see if that changes anything.
They have an enterprise class router, there is no NAT involved, most complaints of this problem have been after business hours, workers calling in to check their voice mail, there were no other active calls on the server at the same time.
I had it a few times myself when I called in, it seems to forward to the same person usually, I once got someone's voicemail, however it disconnected before I was able to hear their complete phone number. It may be a coincidence that every one who complained about this problem was using Sprint cell phones.
-
We have strange issue with one client running 3.0.1.3023 (Win32), they use Broadvox as their ITSP. several times maybe once or twice a day people who try calling in from Sprint cell phones do not reach the autoattendant, a total stranger picks up the phone and would not identify himself.
The call logs shows the call came in to the auto attendant, I don't see the call being forwarded elsewhere, in the log files.
Does it sound like someone hacking in to the system and intercepting calls?
I attached a log of 1 of these calls, I can't reproduce on demand however if I try enough times it happens, this just started over the last few weeks.
-
Hmm, yea we have seen that before in another location. Is there any chance to get a SIP log from that? 2106 sounds like the timestamp is zero; maybe we should just discard such calls (ok that is just a fix).
I am not seeing anything in the logs about those calls, those I only see logs for today as we don't store logs to a DB.
-
Version 3.1, In the Active call log, I see 3 attive calls from @ to @, if I press the X it does not disconnect, it stays in the active log, what can these calls be?
These calls only show in system admin, not under the domain, and they have a future wrong date.
2106/02/06 01:28:16 @ @
After leaving the system without a reboot for a week or so, there is now 4 calls showing active. 2106/02/06 01:28:16 @ @ the x does not remove it.
-
They do have an AA but only picks up if huntgroup doesn't answer the call within 20 seconds. The reciptionists are in that HG. Also this bunch aren't very techinal and don't really like using technology unless they have too. Any other ideas? Also what happens if more than one have the same PIN?
They must get to their mailbox or autoattendant, in order to hear their voice mails by phone.
You can get a 2nd phone number that directs to an auto attendant, and they call in there to check their voice mails.
They can all have the same pin, there is no conflict, at the auto-attendant, they enter their extension #, they during the mail box greeting they would enter the pin.
You can also bypass the pin, by entering their cell phone number under the redirection tab of their mailbox. Then the auto attendant will recoginize their number and offer to take them to their mailbox.
-
Hi.We have a customer who want to know how they can have their users check their personal mailboxes when away from the office. They don't use DIDs but rather 1 main number with extensions. Is there a way to check without bothering the receptionist every time?
Do you have an auto atetndant?, they can call in to their mailbox and enter their Pin. They can also login to the mailbox from a web browser.
-
Firmware-Version: snom-m3-SIP/01.16//03-Jul-08 13:43
-
Hi,
We recently bought a Snom M3 but we cant get DTMF to work with it. Any clue on what setting we should change?
Br
Carl
Which firmware version is the M3, I think the older fimware had a DTMF bug.
-
To an extension "40"
How many trunks do you have, it looks like you have 2 trunks, is the other also set to send to 40?
-
Maybe it is a misunderstanding... The PBX generates files e.g. for Polycom on the fly. You don't have to put anything into the tftp directory. Just give it a try! If the PBX generates files, it will put them into a special directory "generated" - so that you can review the result of the automatic provisioning.
If you are using Polycom, you should check out http://wiki.pbxnsip.com/index.php/Polycom. Polycom is well supported with the PBX, so maybe you give that a try first. Grandstream is not so well supported, but maybe this is the opportunity to update the provsioning process for Grandstream phones.
It looks like he has something wrong with his install as his PNP page is blank.
-
I also just upgraded the system to 4 gigs of ram to try to buy some more time, and I noticed that the OS recognized it, however the PBX is still indicating 2 gig.
Are you by chance running 2003 web edition? There is a 2GB limit there.
Edited
Oops, I just check my own internal PBX, running 2003 Enterprise Edition with 4GB, PBXnSIP is only shown 2048MB.
-
Oops, sorry, i should have post this in "voip phones" forum (i think), not in "aastra phones"...
Fat fingers...
The configuration profiles are for the button configuration, it is not needed for TFTP to work.
Something sounds wrong with your install the PNP should not be blank.
-
Thanks for your reply..
I checked the wiki.pbxnsip but i looking for switch details and prices..
waiting your reply..
Azzami
Hi if you fill out the form on http://pbxnsip.com/find_var someone will contact you, and connect you to a PBXnSIP VAR covering your area, I am a VAR, however I only cover the US. You can also email info@pbxnsip.com. You can also contact me directly and I can put you in touch with someone at PBXnSIP.
-
I decided to try the latest version tonight after office hours in our live environment. I should NEVER have done that! All settings were deleted and the xml files were overwritten with 0 byte files.
I have never experienced this after an upgrade before. Luckily I had a copy thats a few weeks old of the PBX directory. I lost some extensions but I'm working now to restore them.
Always follow the upgrade documentation, I do, although I did is so many times without any issues, better safe than...
Step 1 in the upgrade process is backup the pbx directory.
-
Can you please login (via the ssh) to CS410 shell and check whether the DHCP daemon is running? ('ps -ef | grep dhcp')
I think there is a problem with DHCP on these boxes, I tried it 2 weeks ago and it did not work, so I set the router to do the DHCP.
-
Version 3.1, In the Active call log, I see 3 attive calls from @ to @, if I press the X it does not disconnect, it stays in the active log, what can these calls be?
These calls only show in system admin, not under the domain, and they have a future wrong date.
2106/02/06 01:28:16 @ @
-
I would not register a PSTN gateway to the PBX. If you can just use the gateway mode that is usually the easiest way to go. Unfortunately, I did not configure a Quintum gateway yet; but I believe you just need to point the outbound proxy to the PBX and on the PBX have a gateway trunk with the outbound proxy to the gateway.
We are using a Quintum gateway for to interface with an analog door phone, it works fine.
-
Where can I check email logs if such thing exists? In the past few days I haven't received my regular call log reports and I haven't received few voice mail messages. I want to see where does this get stuck?
Settings, logging, log email events, it will appear in your regular log file.
-
We have a CS410 running version 3.1, customer called several times complaining that many times approx. 1 minute in to the call the Snom 360 phone displays timeout occured and the call gets disconnected. We are using the built in gateway.
The call log shows the call however the duration of these calls are blank.
-
I miss little more extensive manual for pbxnsip on wiki.
so I would ask for explanation for next functions:
- Send message waiting indication:
- Call cell phone when new message arrives:
- Mailbox Escape Account (overrides domain setting):
- Mailbox Escape Account (when caller presses 0):
- Mailbox Direct Dial Prefix:
- External Voicemail System:
- Mailbox Explanation Prompt:
tnx in advance!
I beleive these are all very clear in the wiki.
- Send message waiting indication:
Sends a message waiting light, to the phone. http://wiki.pbxnsip.com/index.php/Extension#Mailbox
- Call cell phone when new message arrives:
It can call your cell phone to notify that there is a new message in your mailbox.
- Mailbox Escape Account (overrides domain setting):
If someone presses 0 while in your mailbox, you can set an extension where they should be transferred to. http://wiki.pbxnsip.com/index.php/Domain_S..._Escape_Account
- Mailbox Escape Account (when caller presses 0):
Mailbox escape domain setting, if not overridden in mailbox
- Mailbox Direct Dial Prefix:
if you want to transfer someone directly to voicemail, without ringing the phone, default is 8, so if your extension is 12, you dial 812, gets you directly to VM. http://wiki.pbxnsip.com/index.php/Domain_S...ect_Dial_Prefix
- :
If you don't want to usepbxnsip for voicemail, you rather use an external voicemail system, such as Exchange 2007 http://wiki.pbxnsip.com/index.php/Domain_S...oicemail_System
- Mailbox Explanation Prompt:
If it should give detailed explanation prompts in the mailbox. http://wiki.pbxnsip.com/index.php/Domain_S...lanation_Prompt
Router Ports to open in order to establish remote SIP calls
in General Setup
Posted
The ports you are forwarding TCP or UDP or both?