Parks
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Posts posted by Parks
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I dont get it... You mean that the PBX automatically provisions the IP address in the outbound proxy?
Generally speaking, the host resolving part should be done in the setting URL. There the phone will make the DNS query and locate the right PBX.
No so in the outbound proxy setting we place
sip:voip.vonvox.net:5061;transport=tls
which gets resolved to
sip:216.218.236.2:5061;transport=tls
after the snom phone reboots. So now if we wanted to move this customer to another cluster we would have to manually update this field in all their phones. Hope this is more clear.
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Is there a way to stop this. If we ever wanted to move the pbxnsip we would have to manually go into each snom endpoint and make the change. This isn't going to be possible without having users down for some time which is unacceptable.
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For several months now this hasn't been working but I not need to pnp new phones on this system. Here is what the snom logfile says:
[5]24/12/2001 00:00:09: Using GUI language English from: /mnt/snomlang/gui_lang_EN.xml
[5]24/12/2001 00:00:10: Using WEB language English from: /mnt/snomlang/web_lang_EN.xml
[5]24/12/2001 00:00:10: read_xml_settings: found dial-plan XML header
[5]24/12/2001 00:00:10: read_xml_settings: found one byte encoding: 0
[5]24/12/2001 00:00:18: DHCP: Received IP address 192.168.5.201
[0]24/12/2001 00:00:18: tcp::set_port: Default SO_RCVBUF=43689
[0]24/12/2001 00:00:18: tcp::set_port: Default SO_SNDBUF=16384
[5]24/12/2001 00:00:18: Opening TCP socket on port 843
[5]24/12/2001 00:00:23: Setting server was already set: http://voip.vonvox.net/provisioning/snom360-{mac}.htm
[5]24/12/2001 00:00:23: Fetching URL: http://voip.vonvox.net/provisioning/snom360-000413291522.htm
[5]24/12/2001 00:00:30: sip::process_challenge:No nonce found on response
[2]23/12/2001 16:00:33: start_dst(984276000) end_dst(1004839200) offset_dst(3600) offset_utc(-28800)
[2]23/12/2001 16:00:33: start DST: 03/11/2001 02:00:00 (984276000)
[2]23/12/2001 16:00:33: end DST: 11/04/2001 02:00:00 (1004839200)
[5]23/12/2001 16:00:34: read_xml_settings: found phone-book XML header
[5]23/12/2001 16:00:34: read_xml_settings: found one byte encoding: 0
[5]23/12/2001 16:00:34: Using GUI language English from: /mnt/snomlang/gui_lang_EN.xml
[5]23/12/2001 16:00:35: Using WEB language English from: /mnt/snomlang/web_lang_EN.xml
[0]23/12/2001 16:00:36: tcp::set_port: Default SO_RCVBUF=43689
[0]23/12/2001 16:00:36: tcp::set_port: Default SO_SNDBUF=16384
[5]23/12/2001 16:00:36: Opening TCP socket on port 80
[0]23/12/2001 16:00:36: tcp::set_port: Default SO_RCVBUF=43689
[0]23/12/2001 16:00:36: tcp::set_port: Default SO_SNDBUF=16384
[5]23/12/2001 16:00:36: Opening TCP socket on port 443
[2]28/3/2010 15:44:35: start_dst(1268532000) end_dst(1289095200) offset_dst(3600) offset_utc(-28800)
[2]28/3/2010 15:44:35: start DST: 03/14/2010 02:00:00 (1268532000)
[2]28/3/2010 15:44:35: end DST: 11/07/2010 02:00:00 (1289095200)
Please let me know your thoughts, thanks.
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On a side note, if you having issues with the start, please verify that the windows update has not taken HTTP port that the PBX is using (80, 443). If these ports are not available, then the PBX will not start.
I rebuilt the server again last night. So far so good. Keeping my fingers crossed. We didn't update windows through September but rather through April.
Hopefully everything is fixed.
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Are you doing automatic provisioning? Check the files in the generated directory. You should see what "TOS" is being provisioned to the phones.
We are using pnp. Where is the directory for this that we can look at the settings?
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Well, the "Windows Firewall" runs locally on the computer that is running Windows and it tends to block incoming or outgoing traffic. This should help getting trojan horses and other stuff under control ("Trust me, I am a soft phone. Please let me open the connection to the public Internet with a secret protocol and let me have access to your file system"). Microsoft Windows usually generates a pop-up that should warn the user when a program wants to open ports. I think because most users always just click okay Microsoft changed the strategy (even if the warning says "a program called 'StealAllYourData' tries to open the connection to the Internet. Do you agree with this?"). Funny story.
I believe you would look for RTP going into both directions. AFAIK the PCAP trace is already "behind" the Windows firewall, which means you see what is really going on on the cable.
I don't think you read my last post because WE DON'T use windows firewall at all. It's also not on every calls so leaving wireshark on might be hugh but guess I can always try it.
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I understand that the default is 160 per snoms wiki but it doesn't go into detail on changing this. I've even when to the gui in the phones and changed the qos diffser in the advanced tab to 25 46 but calls are still be tagged 160.
Please let me know if you have dealt with this and how to correct it.
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Could be a Windows firewall problem. If nothing else changed. I remember that some Windows update made the rules for Windows filewall more strict.
Last resort is to install Wireshark and investigate the true flow of media.
If the automatic start fails that can be a hint that there are too many CDR stored in the file system. Try to make the CDR duration shorter.
We don't have windows firewall but do use juniper and that's been fine as we have the tcp and udp ports open for the sip and rtp traffic. We have the cdrs duration set to 90 days which should be fine. I can always change sense we don't use the pbxnsip cdrs.
What would I be looking for in the pcap?
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We're experiencing this and believe it's because of a recent windows update. We reinstalled the os and it still happens. Can or does anyone have any other ideas?
Basically the pbxnsip service doesn't auto start on reboot and when starting manually can take a few tries. It goes half way and hangs and sometimes starts and sometimes gives an error message that M$ doesn't have any help with.
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We build each of our linux versions on that particular distribution so we have not had any real problems specific to one versoin or another. It is really up to what your most comfortable wiht. CentOS seems to be getting the most traction lately and we did get failover working in it so I would lean towards that on right now.
Ok. My linux and network engineer is testing with Debian because of how it handles packet management. We're going to deploy with 2 servers to start and use NFS for syncing. Thanks for the input as well.
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We want to move away from MS and into something more reliable but would like to know which version is more suggested for clusters. Any suggestions would be great.
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Also, check the countty code under the domain->settings
That is what I needed to do. Thanks all for your help.
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We are testing the newest stable release from our 3.1.x.xxxx version. We're not able to have the switch send the 1 as it always removes it before sending to our class 4 switch for routing. We have the following in the dialplans:
xxxxxxx -> 1925*
xxxxxxxxxx -> 1*
xxxxxxxxxxx -> blank
011* -> 011*
Why is it removing the leading 1 every time?
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EPID
This is very useful if you have remote employees, and multiple offices that on default all display the company’s main DID as the ANI and you would like the 911 dispatcher to call back the actual extension that called 911, not just the main business number which when being called will be an IVR or after hours greeting etc
What you need to do is
1) Create a trunk that you're going to use for 911 calls and on that trunk set the option of Generate unique extension identifier to Yes
2) Under the extension account go to registration and in the EPID window there should be a 10 digit number, if its empty then put in a * and save it and it will automatically generate the 10 digit EPID
3) Go to your dial plan and set a high preference that when dialing 911 it should use the trunk you just created
Now here is how Enable911 makes use of the EPID:
When you call 911 the call goes thru their switch and they attach a temporary DID to that EPID number, and that number is being displayed to the 911 dispatcher taking your call. they don't know your EPID number only the temp number that Enable911 attached to the EPID
When the 911 dispatcher calls you back on the Temp DID number from Enable911 it will ring to the EPID number and ring directly to the extension with that EPID number
I apologize if I didn’t come out clear with instructions, but wanted to get this out since I know some of you were waiting for it. Once I get time ill make it real nice.
I am willing to help anyone out who needs help
This scenario only applies if the 911 operator needs to call back the caller but the whole point of registering each location is to have it auto route to the nearest 911 call center. enable911 charges you if you go through the national operator an extra $100 per attempt.
If we are using 1 trunk to propagate this ANI it's always going to be the same. Am I missing understanding this? Anyway I can call to speak about it?
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i know its in 3.4 maybe it was also in 3.3
Ill post a new topic on how EPID works within the next few days
That would be great, thanks. Were would you be posting this so I can review it?
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why dont you set up a 2nd trunk for 911?
and that way you can have a DID for regular calls, and a DID or EPID for 911 trunk
What version is EPID from. We're running 3.1.2.3120.
Plus I thought that the extension ANI will override the domain or trunk ANI??? Also what happens if the company has a few remote workers this will not work.
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are you using a trunk just for 911 calls? if yes why dont you use the EPID which was designed for 911 Calls
Each domain has one trunk to our nextone switch which will route the call accordingly.
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If you do not have an ANI under the extension, then the domain ANI will be used. Doesn't that what you are asking here?
NO. We need to be able to override extension ANI when calling to 911 with a certain ANI that's registered in 911 dbase.
I cannot believe this is so difficult to understand. I'm been trying to get this now for month and no one seems to understand the laws regarding this and PBXnSIP from what I can see doesn't make it very economical for service providers making us register all DIDs with the 911 dbase.
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That is correct, the ANI in the extension field will take precedence over all, and the ANI on the domain will take precedence over the trunk
This still doesn't answer my question. All we want is to have everyone from a particular domain to use a specific ANI when calling 911. This is a feature that should be added.
This is what I am envisioning:
In the domain setting right under emergency number there would be a field for emergency ANI. And let's say that there is 1 home worker that cannot use that ANI of course in the domain right under ANI their would be emergency ANI that would override the domain one.
This seems to be the simplest why to cover all needs regarding multi tenant deployments scenarios. I'm just very surprised this really hasn't been an issue before. Registering ever number is a waste of money for both the customer and us and probably lost some customers because of it.
Would be very helpful if someone from PBXnSIP would review this and give feedback. Thanks all.
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the invite was correct however the ANI still shows the extension/account number. Many carriers will read the use it for the CID as that's what an ANI is. Some will not even complete the call if left blank, they want Not Available in the ANI to represent Private Number.
I'll be happy to email someone a cdr showing this.
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I have tried that as well on the account level and domain settings. What is pu in the ANI when it actually works?
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The Setup is Pbxnsip latest version and Polycom phones using the latest firmware.
1) TRANSFERRING CALLS INTERNALLY
When a user wants to transfer blind to his boss he can only do so by hitting transfer and then entering extension and hitting blind, in this case the boss will see the actual number of the person calling but the employee will not be able to tell him who is on the phone.
I have feedback from a number of clients that find this frustrating because although they want their secretary to screen the calls, if the emplyee does NOT hit blind (because she wants to check with boss first if he wants to talk to the caller) then once she transfers all he sees are a bunch of calls from her so when he is handling multiple calls he has no idea which is which, this may be a polycom limitation but any way to get around this now or maybe in 4.0?
2) CALL BACK WHEN USER IS ON PHONE
Right now when a caller makes and internal call he/she have no way of knowing if the person is on a call/away from their desk, even if you offer camp on then it will still ring until the camp on message picks up. MOST of my clients have requested the option to have a message popup on their screen or for the camp on message to be immediately relayed to them if another user is on a call and be given the option to continue calling the person or to wait for call back/leave a message.
3) USING THE CALL BACK FEATURE
I would also like to clarify, how the call back feature works...if the user is away from their desk and the system offers the call back option...when exactly will it call the caller back?
please advise on these issues asap.
I don't believe the call back feature actually calls the originating caller back but rather sends an email to the callee letting them know the callers number and they want a call back. We also never got it working so we turned off for our customers.
Don't know about your other issue as we don't use Polycom phones.
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Some residential customer don't want their number be published when placing calls. If left blank pbxnsip in puts the account number which is the customers 10 digit tn in our case. We want to pass Not Available to the PSTN as some carriers require this in the ANI and cannot be blank.
Any suggestions would be great, thanks.
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IMHO the biggest problem with G.729 is that the audio is compressed so much that customers have the impression that VoIP is inferior to PSTN.
Every transcoding step reduces the amount of information transmitted.
I understand that the audio is compressed and therefore the quality is less. However our customers on our SIP trunking service love it and cannot tell the difference. Most cannot tell the difference in quality especially because so many people use cell phones these days.
If every time it gets transcoded it reduces the quality then it really is only get transcoded twice once on the carrier side then on the customers side. Am I looking at that right?
Snom resolving domain for outbound proxy
in snom Phones
Posted
I'm not following understanding. Currently all snom phones have an ip address in their outbound proxy setting. If we wanted to move them to another cluster and their provisioning address is that of the old cluster we would have to do lots of manual work updating their provisioning addresses to their new server, correct?