andrewgroup
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Posts posted by andrewgroup
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We've read http://wiki.pbxnsip.com/index.php/Dial_Plan and based upon this and our own experiences trying to create seemingly easy dial plans I think it would go a long way towards full understanding with an explanation of how Dial Plans are processed. From a recent experience with a CS410 with the built-in PSTN gateway and a SIP trunk to a remote PBX it appears as if we could only get the plans pattern 8XXX blank replacement will go to the sip trunk we set this priority at 100 and the PSTN set at 100 also with Pattern * replacement * to dial out the PSTN.
It would be very very helpful for an explanation of how the preferred number assignments on the trunk are processed and what happens when the assignment values are the same priority.
I'd also like to see more examples of dial plans similar to a previous post about 7 digit dialing and replacing and inserting an area code...
I chopped this from the 7 digit dialing post and I think I've restated how it works below
The following dial plan 8([0-9]{7})@.* and the replacement was 1914\1@\r;user=phone
I understand the pattern does several things. It looks for a dialed number pattern beginning with an 8 followed by 7 digits each of which can be a 0 through 9.
I'm not sure I understand the @.* function as shown and I see the replacement is 1914 \1 (first match group would be the 7 digits dialed) and I don't understand the @\r;user=phone at the end of the replacement.
We need a Holy Grail Examples document of Dial plan logic...
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Thank you that solved the problem. I appareciate the help. Thanks.
we've been looking at forums posts about dial plans and after reading your post and the solution, I'm curious about something. If using PBXnSIP what do you have to dial 9 for an outside line? What type of gateway are you using?
Have you taken what you've learned and created improved dial plans?
Would you mind posting some in the best practices forums?
Cheers.
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Pref Trunk Pattern
25 Office1 1xx
25 Office2 5xx
I tend to avoid using complex dial plans involving stripping characters etc. unless really necassary, and in this case we don't need to do anything special.
The new CS410 come preconfigured with ext 40 through 49, prefer to use CS410 configs out of the box as much as possible for standardization
This client will likely have more than 10 remote offices (no interoffice calling required, only HQ to remote and back)
HQ uses extensions 100 through 199 and they are on a public IP, seems a sip gateway to HQ might be simple, But HQ back to remotes will require some logic..
Do you recall any dial plans that might strip and send call to a specified trunk.
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I recently posted a case in the best practices, where 1 main office and multiple remote offices...(will grow to 10 remote offices) not needing anything but hdqrts to remote and vis versa. It appears to me that several options exist. We'd be using Snom Phones with multiple identities. Since the revised CS410 comes preconfigured with ext 40 through 49 we'd just assume to use the default scheme. Might you read the post in best practices misc and comment on this? Thanks
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We are just now beginning to use the 410 for some clients. Could everyone provide the model / software in use when posting so that we can all determine if wr are affected or may have an appropriate solution or suggestion.
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using the following example, would somebody with experience addressing the need for interoffice calling expand on the following scenario?
Hqrtrs has PBXnSIP on 2.2.2.2 with FQDN of pbx.newco.com extensions 100 thru 150
remote1 office on IP 3.3.3.3 with FQDN of oh.newco.com extensions 40 thru 49
remote2 office on IP 5.5.5.5 with FQDN of ky.newco.com extensions 40 thru 49
assuming that remote1 and remote2 do not need to call each other, only the headquarters
would be best served to make trunks to and from each other?
or can each office simple make speeddials for ext@IPaddress?
If you create Speed dials to an extension VIA the IP or FQDN it appears as if you conserve accounts and accomplish the same thing.
Hqrtrs can have phones register on remote PBX to monitor or help answer calls..
Then what are the benefits of trunks in this scenario?
What might a dial plan be at Hqrtrs if you want to dial remote1 extension 40
Dial 8 to indicate a remote office call, then dial the 2 digit office ID say 10 for remote1 and then the extension 40
So you would create a dial plan that detected a 5 digit call, strip the 8, take the next to and based on the 10 send call out a specified trunk
Of course the remote1 pbx receives a call and must do some matching too...
We get this in general terms but would like to see how others have done and learned from the same requirements.
I've seen posts about reducing SIP registration times, would this be important?
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What's become of this situation?
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The ability to create a test account that can be programmed to make a scheduled call to validate an ITSP's ability to make and take calls. As a service provide, the complement to the test box account would be a test box reciever. A remote PBX could call an AA enter a test box account and the two systems could exchange some formalities to validate each was able to send a receive a call. This might create another SNMP item called Valid test Calls and Failed test calls.
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Eyebeam is using G.729 only and also on the PBXNSIP we have set G.729 as the first codec in settings and in trunk. How do we see if transcoding is taking place?
When eyebeam is directly configured with the ITSP details it is using G.729 and the voice is clear.
please check the top level settings for the PBX, I've learned from experience that any settings in the top level will supercede domain codec settings..
The best way that we confirm the codec is to capture packets and look into the SDP (session descriptor packet)
Below is a call that is negotiating Ulaw
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 4631 4631 IN IP4 208.64.32.55
Session Name (s): -
Connection Information ©: IN IP4 208.64.32.55
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 57486 RTP/AVP 0 8 18 2 3 101 Media Attribute (a): rtpmap:0 pcmu/8000
Media Attribute (a): rtpmap:8 pcma/8000
Media Attribute (a): rtpmap:18 g729/8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute (a): rtpmap:2 g726-32/8000
Media Attribute (a): rtpmap:3 gsm/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): sendrecv
Followed by the same call with A law set in the Global Codec settings
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 34655 34655 IN IP4 208.64.32.55
Session Name (s): -
Connection Information ©: IN IP4 208.64.32.55
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 64006 RTP/AVP 8 0 18 2 3 101
Media Attribute (a): rtpmap:8 pcma/8000
Media Attribute Fieldname: rtpmap
Media Format: 8
MIME Type: pcma
Media Attribute (a): rtpmap:0 pcmu/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: pcmu
Media Attribute (a): rtpmap:18 g729/8000
Media Attribute Fieldname: rtpmap
Media Format: 18
MIME Type: g729
Media Attribute (a): fmtp:18 annexb=no
Media Attribute Fieldname: fmtp
Media Format: 18 [g729]
Media format specific parameters: annexb=no
Media Attribute (a): rtpmap:2 g726-32/8000
Media Attribute Fieldname: rtpmap
Media Format: 2
MIME Type: g726-32
Media Attribute (a): rtpmap:3 gsm/8000
Media Attribute Fieldname: rtpmap
Media Format: 3
MIME Type: gsm
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
Media Attribute (a): sendrecv
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Maybe it makes sense to get an Wireshark trace and see what exactly goes back and forth. What version are you running?
Additionally since you are using an exchange server, change the default logging level form none to far more and you'll have the other end of those log files. Let's all assume that in the vrtual SMTP server settings you are allowing the IP address of the PBX to send email. Additionally download one of many free SMTP email server testers and blast a few thousand emails into the server to validate it's ability to receive.
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I'm just looking for all possible options.
We have an Asterisk server we set up a while back and with a phone on the same router / setup, it works fine back and forth. And we all know Asterisk uses port 5060 by default, just like pbxnsip. Which is why I am asking about other reasons beside the firewall. I don't want my clients when we go live to have to figure out how to port forward on their routers unless absolutely necessary.
We too have supported Asterisk in the past. However we seldom are forced to use a NATTED solutions but when forced we have used Zyxel. In all installations where we control everything we specify a Public IP dedicated to PBXnSIP for both Linux and Windows Installations. This avoids in most cases where doublenatting with home phones for the boss on some cheap cablemodem router barfing at us. I'd highly advise a Public IP in all cases.
I''ve been advised by a good friend that Asterisk (next major release) will be more sip compliant and they too will begin to experience problems common to SIP compliant solutions. Time will tell though.
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So can anyone else tell me why the phone will ring, but I can't hear on either end? Besides a firewall issue?
Thanks.
Do you want to hear that it isn't the firewall, when all evidence says it is? We specify Xyzel routers and we have yet to have a 1-way audio problem. Zywall Plus+2 makes a fine choice.
RINGING occurs as a result of a TCP message using SIP
Talking occurs on a UDP Stream once handed off between the two endpoints talking SIP to each other.
The firewall of choice is clobbering the UDP streams but passing the SIP messages as you already know since you have 5060 forwarded.
PLease post you firewall choices so others will know...(perhaps a firmware upgrade from vendor will help?)
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It would be easier to POST them using HTTP.
Post them Where?
I'm fearful the smaller clients getting 410's or small servers without internal exchange we will be installing and forced to use the ISP SMTP server will in the near future (end of year) be clobbered by the ISP's shutdown of port 25 and migration to ssl email in the anti-spam battles. My Home DSL is ATT and twice in the last 8 months they've closed port 25 even though I've signed the waiver... Not good for business use and not a lot of choices in certain areas. I suppose if we are using Win or Linux we can install an SMTP server, force the client to Static IP's, create a legitimateDNS record..We'll see how this shapes up.
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By no means am I happy with this solution, I am not seeing any negative indication from the server itself - it is only the Media CPU Usage graphed in PBXnSIP.
Would you expand on the questions regarding your platform. If you'd like to do that off-line I think my profile will drop me a line.
Diagnosing this problem should be easy since it appears as if you can predict failure based upon load. So therefore the system should be working fine with a smaller load and we can track performance and isolate the failure to the extranet or the intranet and the place to start is to packet capture all data to the failing server during a low use (working OK) and into the (Not working OK) high use period.
Using a good quality Switch that supports port mirroring capture all traffic... I also know of a low cost software package that will record and rebuild in realtime all SIP/RTP sessions and record all calls.
This will give you something to go on, an until you can starting asking questions that can be answered you will be chasing ghosts...
Happy to help.
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Well, the PBX already has TLS (which is compatible with SSL3.0). Maybe it is time to "upgrade" the email client to TLS.
If it helps to reduce the number of SPAM, we will be happy to help.
The logic in my note... A client purchases a cs410 and it's installed in a remote office and the owner of the business want's the daily CDR files sent to his email address that is beyond our control. Perhaps his personal email address like Boss@gmail.com. The way I see it from a practical standpoint the only easy to implement sending is to use the ADSL accessible SMTP server. When selling CS410's to small businesses this is going to be a great selling point, and it will not be practical or cost affective to add tls encryption to every mail server just to recieve daily CDR files. How about posting the daily CDR files into the FTP directory?
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Complaints are coming in referring to static, audio cut outs, etc... Is 2.1.5 any better, or do I need to start moving customers off of this feature server? Please advise!
Can you share whats become of this situation? You can't sweep something like this under the rug with your clients.
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AT&T, US largest DSL provider will soon require that all email sent from an DSL user account must be sent with the from address matching the master onfile email account. No big deal to match this, but they are also eliminating PORT 25 as the sender and using SSL in it's place. This will disable or require a small remote office to have their own SMTP mail sender. the chhanges are;
Change the SMTP port to 465 and check the option labeled Use an encrypted
connection (SSL).
Change the POP3 port to 995 and check the option labeled Use an encrypted
connection (SSL).
To get access to port 25, you now must accept a waiver and the fine print is legaleaze and I'm sure it's in their favor.
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You could also do this with DNS and not bother with a registered trunk.
There are many different ways to do this, it all depends on how you want configure the PBX
could you clarify what this means in a practical example, on the surface this seems to be the solution but don't understand it perfectly"
using the following example
Hqrtrs has PBXnSIP on 2.2.2.2 with FQDN of pbx.newco.com extensions 100 thru 150
remote1 office on IP 3.3.3.3 with FQDN of oh.newco.com extensions 40 thru 49
remote2 office on IP 5.5.5.5 with FQDN of ky.newco.com extensions 40 thru 49
would be best served to make trunks to each other.....or can each office simple make speeddials for ext@IPaddress?
If you create Speed dials to an extension VIA the IP or FQDN it appears as if you conserve accounts and accomplish the same thing.
Hqrtrs can have phones register on remote PBX to monitor or help answer calls..
Then what are the benefits of trunks in this scenario?
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Maybe we should come up with a use case list, and sort it by importance. I don't think that we can come up with a generic "solves all" solution, so lets take it from the user/admin perspective.
I'm not sure what a use case List is, but from a user admin perspective then an export / import accounts functions is recommended. Being able to give a client an excel spreadsheet to type in all the stuff gives them documentation and make the setup job far easier.
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So a thought could also be that the PBXnSIP software can't handle this load, as opposed to the hardware? We didn't buy cheap hardware - and have invested heavily.
Other suggestions?
Linux or Win? What Version?
disk controllers might be heaviliy used causing interupts? Consider RAM drive from Gigabit for swap on linux or or Win.
In the Win system are you familiar with perfmon or system admin, WMI etc.
I can refer you to a Voip SIP call recorder, and you can with client permissions record all calls and confirm the drop outs, cracks etc exist on the NIC ports the Servers.
Perhaps it's the QOS,DIF stuff on the circuit?
when you said, "Invested heavily" Just what is this running on?
I know 23 concurrent calls for 5 or 6 hours straight across 45 extensions for 4 weeks in a row never raised a AMD3100 over 9% utilization on a WinXP Pro box with a NIC to a PRI gateway, a NIC to the LOCAL Lan, and a NIC to the public intenret.
2.1.5.2357 (Win32)
License Status: Office 75
License Duration: Permanent
Additional license information: Extensions: 58/75 Accounts: 78/100
Working Directory: C:\Program Files\pbxnsip\PBX
IP Addresses: 127.0.0.1 192.168.1.250 192.168.100.1 216.xxx.yyy.20
MAC Addresses: 1D04ACC519C4 1D15F29BC703 1D902713C720
Calls: 2917/486 (CDR: 6322) 0/0 Calls
SIP packet statistics: Tx: 3844156 Rx: 3844269
Emails: Successful sent: 31 Unsuccessful attempts: 5591 (Warning: Last email could not be sent!)
Uptime: 8 02:06:28 (51MB/511MB 54% 13842388--6078828) WAV cache: 0
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Could someone further explain the differences between proxies and gateways and why they exist?
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That is not even a RFC-compliant packet. Not even talking about RFC3581 (http://www.ietf.org/rfc/rfc3581.txt, August 2003), which is not set to the actual port. The content-length is not set.
I know to little to challenge that statement, and I'm getting lost in all the packet captures. I do know they have Alworx, Asterisk and Hosted phones, Cisco IAD 2400's all on the switch and it's all SIP.
The ITSP has provided all of the IP information. They say while calls may come from on IP they'll hand off the media to another IP thus no traffic actually passed through the BTS10200 switch.
I have a barebones PBXnSIP on a public IP of the ITSP running WIN.
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Can you post the full invite?
No more on this topic.
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Can you post the full invite?
In the logging options for the domain, set it to log "other sip" messages.
Diagnosis is that Cisco BTS10200 has a know bug discovered in Sept 07, that a REFER statement causes. The ITS is pressing Cisco for a bug fix...
Dial Plan Theory
in Dial Plan Setup
Posted
Being of Regular People descent, I agree.
After working with some of our own dial plans and reading a variety of posts, the missing link in understanding dial plans, at least for me, is some basic concepts that I'll share to reinforce my own understanding and to help others of regular people descent.
Dial plans are a filtering system where you define exact call patterns from most restrictive to least restrictive. this means tha a Pattern 911 is exactly defined as three digits containing the exact numbers. Whereas 9XX would be a three digit call where the first number is 9 followed by three more characters. Either pattern could be used to detect an emergency 911 call. something to consider for dial plans.
In order for these restrictive filters to be processes properly, you must set the priority of each dial plan. The most restrictive such as 911 should be given the lowest numerical value, thus highest priority so that it is processed and checked against first.