Vodia PBX
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Posts posted by Vodia PBX
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I have one installation that is having a strange problem calling out, and it is intermitent. The problem only exists when calling across a trunk, internal calling works fine. When you dial the phone number, the phone just sits there, despite the dialplan telling it to dial, or pushing the dial button. In the background it authenticates with the PBX, however the PBX never sends the final sip packet containing the 200 OK. Then when I hang the phone up, and it cancels the call with the PBX, the pbx goes on to send the 200 OK and place the call through the itsp. This customer is running 3.0.0.2998 I have noticed that the pbx logs the error
Via is empty, cannot send reply
Despite the via field having content in it.
I am attaching the pcap, and logfile to this post.
The pcap was done from a remote phone, however the same problem exists within the office where there is no NAT
Is there any reason why you are using 8.3? I remember some of the 8 versions were not very good. What is the latest? 8.6? Do you have a chance to give that a shot?
From the logs I must say it seems that the phone has some problem authenticating. I does not respond to challenges and that is really strange.
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Just for clairification, I would have 1 extension (on the main server) for each different outbound caller ID number I wanted to send. The problem I see with this is then I would need need as many trunks on the 410 which has a low limit on the number of trunks that it supports. Isn't there a variable string you could use in the ANI field of the extension on the main server so you would only need 1 trunk from the 410 to the main server's 1 extension, and it would pull the caller ID off of the SIP invite message from the 410? Seems much simpler, as well as it is how most ITSPs handle it.
Well, considering that a single DID is a dollar value every month I see the registration problem relaxed. The issue is that ther eis not clear specification on how to deal with one registration and multiple identities behind it. Especially because the CS410 will probably have only a few DID per box I would say lets take it easy in the beginning and then later when the standards are more clear a trunk may have more than one DID.
Actually I know carriers who seriously provide hundreds of DID to the same client each of them with a seperate registration.
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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!!!
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.
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i have been testing out pbxnsip with Exchange UM and OCS, and things are going pretty good. I want to be able to test out dual forking with the use of office communicator and a desk phone. I currently only have a mitel 5220 SIP Phone. Can this phone work with pbxnsip? How do i go about this.
I remember testing Mitel phones some time ago (well, must be more than a year...). But anyway, it worked and I don't remember any significant problem. The only disadvantage I can think of is the PnP will not work out of the box.
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Once you have an audio session enabled you can re-INVITE with all kinds of video and image codecs. The PBX will just be the SBC between the endpoints then, and H.263 can be transmitted between the connected phones.
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Hello every one.
Here is another PBXnSIP problem.
PBXnSIP support has never actually helped fix any problem so far, so im hoping that other unfortunate system users / builders can help.
What kind of support are you referring to (http://wiki.pbxnsip.com/index.php/Trouble_Ticket_Processing)?
System Config DetailsOperating System. Redhat Enterprise Server REL 5.0
PBXnSIP version. 3.0
I assume this is 3.0.0.2998.
Scenario.Incoming calls go to an Agent Group and into the queue.
Announcements and music on hold work OK
The queue passes callers through to the agent in the correct order OK.
Note: there is only (1) agent
Problem Description
If the "agent" is not attending the phone and the call is past to that extension the phone rings continiousley IE: non stop and
WILL NOT GO TO VOICEMAIL.
A agent group does not have a voicemail. If you want that the call goes to voicemail, you have to specify a timeout value and use as destination the voicemail of an extension. Usually, you can put a "8" in front of the extension number. For example:
After hearing ringback for (s) ...: 30
... redirect the call to the destination (e.g. "73"): 8123
That means that after 30 seconds of ringing an agent the call will be redirected to the mailbox of extension 123.
Things I have tried including:1)
From the "Agent Group" Level.
A) If the caller already waited longer than (s) >>> set to 5 seconds
Redirect to the destination >>> EG: 73Result: agents phone keeps ringing. the call does not redirect.
Comment: Would not be a good fix any way as all calls in the queue would be redirected/
This setting has nothing to do with a ringing agent. This timeout is for the waiting time in the queue (where a caller would hear music on hold). 5 seconds if extremly short, a typical value would five minutes (which is 5 * 60 = 300 seconds).
2)From the Extension level in "Redirection"
A) Call foward on no answer to : EG 73
Call foward no ansertime out: EG 5 seconds.Result: agents phone keeps ringing. the call does not redirect.
Call forward on no answer is a setting that affects calls directly to the extension. If the call comes in through a agent group, that setting has no meaning (otherwise if every agent sents their own little call redirection the chaos would be complete).
NOTES:If a system extension calls that agent direct IE: not VIA the QUEUE, calls go to voicemail as normal.
After hours: All calls go directly to that same agents extension IE: not VIA the queue. Calls go to voicemail as normal.
Commment:
I have been in IT for 20 yaers and have never seen a product (Sold by a reputable wholesaler) with as may bugs and as little support for resellers as PBXnSIP.
It seems that this application has some powerfull features but it is extreamly weak when it comes to basic features, stability and functionality.
Adding features has always the danger of making things complicated, escpecially with a relatively complex thing as a waiting queue. On the one side we get the pressure to add features here and there, and on the other side we get complaints that the software is difficult to use. We try to take the advice from both sides and have more features, but at the same time still make it possible for an average person to use them. It is not always easy.
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That sounds like a good way to do it. I will look forward to seeing it in a future code set. Do you have any ideas on if you can do something similar with retrieve?
Retrieve is the same thing.
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I am not concerned about what to do when the button is pushed. That can easily be taken care of in the phone's config file. I was just wondering if the capture made it any easier to make it light up when the extension it is subscribing to the presence of is in use. (the capture was the packets that got sent to an idle phone from the call manager when the BLF was turned on, because the extension it was monitoring the presence of went into use)
Maybe you are right. Another was of generating kilobytes in order to turn a light on (after dialog).
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It would be running at all times. All of the phones in that specific office would register to it. It would then register it's trunk as an extension to my Large PBXnSIP server at my data center. If for some reason their internet connection went down, the settings on the 410's trunk to my server would be set to allow failover. The next thing in the dialplan of the 410 would be its PSTN gateway. The dialplan would also have an entry for 911 that would go directly to the 410's pstn gateway and not the trunk back to my datacenter.
Okay, but then you don't have to worry about caller-ID on the CS410. You can use ANI, or a trunk prefix or just s DID for the whole trunk. Just like any other installation.
On the hosting side, you would have one "extension" for each DID number. In NAPNA, I would say one DID one registration, that keeps things very simple.
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I have found in a few 3.0.0.x and 3.0.1.x installs that hunt groups are displaying the following issues ,
Hunt group 600
stage 1 extensions 200 201 202 203 204
final destination 601 (auto Attend)
set the ring duration to 0-29 seconds , works fine ,.
set the ring duration for Stage 1 30+ seconds and it hangs up or I get callcentric number cannot be found ,
I am attaching 2 SIP traces , sorry could not get a Pcap , working remote on a CS410
In the bad.rtf there is a call redirection turned on on the phone to 18606704381?!
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With the latest firmware on my 7961 and a newly written config file, I was able to get my 7961 to subscribe to the presence of another extension. The following is what shows up in my extension's registration field:
presence 7208 sip:211@192.168.3.130:16785;transport=TCP Cisco-CP7961G-GE/8.4.0 170
I was wondering now that I have gotten this far, if it is possible to make the BLF work. I am attaching a packet capture of the same model phone registered to a Cisco Callmanager running SIP. I hope it is of some use to see if it is possible to light the BLFs through PBXnSIP
I don't want to be too optimistic here. Presence has the problem that you might see the "presence" (whatever that is), but the PBX is not able to tell the phone what to do when the button is being pushed.
And there are a lot of x-cisco-xxx headers in the packets. Look for "extended-refer", if you want to have some fun. When the draft was written Rohan Mahy was still with Cisco, that is a long time ago. I would not build opon that...
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I recently setup the programmable buttons on my snom phones with 7.1.35 and the most recent version of 3.0 .. I compared using the buttons programmed by hand through the GUI as extensions, PnP with Buttons set as "monitor extension" and PnP with Buttons set as "speed dial"
BLF is supposed to be really stupid. It does show the status of the monitored resource. But whenever you are pushing the button, it just calls the number. Pickup is not possible with that, that is a feature because it avoids race conditions (incoming call and at the same time pushing the button will accidentially pick up an incoming call - you don't want that).
Speed dial is even more stupid. It just dials the number, no matter what.
Transfer scenarios in SIP require that the phone initiates a REFER. That is a little bit tricky with the buttons. Lets see if a firmware upgrade on the phones addresses this problem in the future.
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I have been running the numbers and thinking about offering hosted phone service to my customers utilizing the PBXnSIP product line. I was thinking that in some cases, where the customer would be interested in failover of having a CS410 at the remote office, and having one of it's trunks registered to my main high powered PBXnSIP server back at my office, and seting the 410 to fail over to it's built in PSTN gateway if the network connection to my server goes down, or 911 is dialed. It all seems quite easy until I start to think about Caller ID. Under normal operating circumstances, I would like to be able to send caller ID on a per extension basis. I can do this without a problem until the call hits the main server where it would be registered. Hou would I use the caller ID that is coming in from the 410 across the trunk, instead of the value that would be entered in the ANI field of the extension on the main server that the 410 would be registered to. Are there variables that can be entered in the ANI field to pull the caller ID off of the incoming invite, instead of hardcoding a number in the field?
I dont 100 % get it... You mean the CS410 is only for the case of failover? Or should it run at all times?
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I realize that it does not require an extension number behind the star code, however when you get to a larger install it is very problematic that it automatically assigns it to an orbit that is the same number as the extension number that is doing the parking. If there was an extended prompting radio button option next to the star code, and when it was turned on, the system would ask for the orbit number. This would allow the operator to type in any orbit number that they wanted. They could then page, and say you have a call parked on orbit XXX After that the receipient would press the retrieve button, and if the extended prompting radio button was turned on it would ask for the orbit (which they could type in), and then they would be connected.
While something similar to this could currently be accomplished if you had different buttons for different park zones, this is problematic when the average Cisco phone has 6 buttons. The first 3 are usually lost to the extension, intercom, and general message. Leaving park, retrieve, and one other. It also does not help much to assign zones in the PBX since the operator would need to be a member of all, and the person picking up the phone could be picking up from anywhere.
I think the easiest way to solve that problem is to use the setting "Explicitly specify park orbit preference". If someone put a "*" in there it will mean "ask". This way we nicely stay backward compatible with that we have now and we can even specific the behavior on per-extension basis.
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I have attached some packet captuers, as well as found some more info on the problem. It seems like when doing an attended transfer, it does not always fail. It seems lile there is a race condition, (or the pbx will only work on one of the 2 extensions) where it may work if you happen to pick up the call on one of the 2 extensions as opposed to the other. When it fails, you pick up, and you just get silence, and the other registered extension just continues to ring, and then roll to voicemail.
The latest thing I have found its it also happens in some cases where there is only a single extension registered, and you attempt to answer the call as it is forked to a cell phone. You answer on the cell, and get nothing but silence. The desk phone continues to ring, and then rolls to voicemail. I have attached a packet capture of this as well.
If you can, give 3.0.1.3016 a try: http://www.pbxnsip.com/protect/pbxctrl-3.0.1.3016.exe.
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Does anyone have any notes about setting up the m3 to auto provision with a pbxnsip server? Specifically, I'm wondering what fields need to be set in order to make it pull the config?
Check out http://wiki.pbxnsip.com/index.php/Snom_M3.
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Ah I see, I *only* tried it from my cell phone and never tried it from another line that wasn't programmed in the system i.e. a land line. Interesting feature, but just so I know, is there a way to turn it off?
There is no settings for that, but you can easily fool it by putting a prefix to the cell phone and then strip that prefix in the dial plan.
For example, if your cell phone number is 212-123-4567 you would put 999-212-123-4567 there and then have a dial plan entry with the pattern 999*.
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Just a clairification, this would be usefull for any feature that would normally require more than just the * code. I would love to see it in parking/retrieving, pickup, and so on.
Well for park/retrieve, that is not so easy. There is already a park/retrieve code that does not take an extension behind it. Plus in this case there is a useful meaning behind it - let the PBX search for you. And usually there are not so many park orbits, so that it is reasonable to put one on each key.
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Out bound calls are not effected.
What do you think i should try next.
Okay, the next step is to get packet level trace (e.g., Wireshark) to see what is going on on the cable. Then we see if it is a jitter problem, packet loss problem, SRTP problem or whatever.
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That is the problem. If the HTTP port is not able to bind to the port, you can't get into the web interface to fix anything. You are also going to have to restart the PBX to get into the web interface in the future. In my test above, the SIP ports were not able to be opened, so that is a call affecting problem.
I would agree with you that non-critical services like TFTP and SNMP are probably not a huge issue, if perhaps a big warning could be put on the Status page. As for HTTP/HTTPS, I'm not convinced that your management interface isn't critical, but the pbxctrl process definitely not start without the SIP ports being able to bind to their ports.
If there was some way to restart the management interface that would not be call impacting, then I would agree with you 100% that it is not a critical process. However, since getting the HTTP interface back would require a restart of the pbxctrl process a client, ITSP or manager could get backed into a really difficult position. (i.e. Take down 50 active calls to bring the web interface back or wait hours to make this change that a client needs now to restart when the load is much lower.)
If you aren't comfortable with having the process not start unless all the ports bind successfully, perhaps an option can be added to the Global Configuration for Strict or Loose Startup mode. Strict being, if any port bindings fail, pbxctrl will exit with an error code. Loose being, as long as the SIP ports bind, the pbxctrl process will start properly.
Next version will have a command line option "--no-check-ports" that will allow the PBX to start up even if FATAL errors are reported. TFTP and SNMP will not be fatal any more. The default is that the PBX will not start up any more if the SIP or HTTP ports cannot be used by the PBX.
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The problem, as I see it, is the pbxctrl process will allow itself to start in a potentially inconsistent manner.
Well, that is a policy question... If the PBX does not start because it does nto get the TFTP port, I would call that picky. HTTP is also not essential for making phone calls... So where is the line?
IMHO it is reasolable to start the process anyway. If you have HTTP, then you can fix the other problems through the web interface. If you have SIP, you can run the service already.
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From my pbx.log:
[0] 20080915060138: Could not bind socket to port 80 on IP 0.0.0.0
[0] 20080915060138: FATAL: Could not open TCP port 80 for HTTP/HTTPS
[0] 20080915060138: Could not bind socket to port 80 on IP [::]
[0] 20080915060138: FATAL: Could not open TCP port 80 for HTTP/HTTPS
[0] 20080915060138: Could not bind socket to port 443 on IP 0.0.0.0
[0] 20080915060138: FATAL: Could not open TCP port 443 for HTTP/HTTPS[/code]
Well, those errors are probably because the PBX does not have superturtle powers to open those protected ports.
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Is there anyone who can tell me how to setup simple pickup-groups?
I have several offices within one company with all snom 300/320 phones and pbxnsip 3.0.0.2998
The wish is to be able to pickup the ringing phone within the departement.
Now I have programmed a button with the *87 code, but then you don't know what ringing phone you are taking over....
The best way is to define hunt groups. Even if you don't really call them they form a group association.
http://wiki.pbxnsip.com/index.php/Park_and_Pickup explains the priority on how calls are being picked up.
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I am evaluating the system and have version 3.0.0.2998 with the 3 min. license. I am trying to duplicate our existing PBX system and I am getting a strange behavior with the IVR menu when called from the PSTN, but not from an extension on the pbxnsip server. I created an account with a primary name of 10 and an alias of a PSTN phone number that is being sent to the system. When I call 10 or the PSTN number from an internal extension it plays my recording and woks as expected (including my night/day service flag). However, when I call the PSTN number from my cell phone I get a message that says "To place an outbound call press 1, to go to your mailbox press 2, to go to the auto attendant press 3" instead of my recording. Besides that it should have played my recording instead of that system message, when I press 3 it plays my recording but it ignores the day night setting and never plays the other message. So it looks like I may have 2 things messed up in the config and I'm not sure what I am doing wrong. Any help would be appreciated.
Well, there is a special feature with the cell phone. When users are calling from their cell phone into the office, they get a "special treatment" (feature). It might be annoying for testing, but for the real life it is quite useful, especially when those users want to place international calls or just use the caller-ID of the PBX for outbound calls.
The feature kicks in when the cell phone is calling an auto attendant. For direct extension calls, that feature is not active.
Redirect to the destination >>> EG: 73
Mitel SIP Phone
in VoIP Phones
Posted
I don't have the Mitel phone here, but you should set up username, password, domain, outbound proxy as usual. If you plan to install a large number of handsets, we can look into PnP.