jlumby

I have an issue when if I reboot the server to do updates, all BLFs on all Aastra phones stop working until the phones are rebooted. I recreated this in the lab, and did a packet capture, and the Aastra phone responds to the PBX's notify with 500 CSeq number out of order. What can be done to fix this? I am running the latest Aastra firmware 2.6.0.66

I would find this option very usefull. If you no not need or have a license to record/wisper/bardge, why put the RTP load on the PBX. Can this be accomplished as easily as a second entry in the VIA field?

I do not have any experience with Mediacodes, however I am very happy with Patton. I have a SN4960 PRI gateway, and it works well, and I have not run across anything that it cannot do.

I have never had a problem, however I have heard rumors of old versions of firmware having issues, or if you have 2+ devices registered to the same extension, and one is subscribing to the MWI, and the other is not.

I have the same problem, it does not do or show anything

While there may not be a simple way to block a specific country, for my version 3 installs, I went to http://www.arin.net to figure out the American networks, permitted them, and wrote a rule to block everything else at the firewall. The list is not that bad since they can be summarized into 8 bit masks.

I have not heard of one, however it would be a great idea, just like the blacklists that you can query on your email server

Does anyone how how to generate a simultaneous call graph for an agent group. I am looking for something just like the one that is generated automatically at midnight, however for a specific agent group, and not the PBX as a whole

There is only 1 phone registered. I just tested placing an intercom call to it from another extension, and that worked fine. I set connect=true, and the phone still just rang. Not sure if it matters or not, but I am running 3.4.0.3202 (Win32)

Is there a variable that can be used in the click to dial URL that will cause the extension to ring as an intercom so that the phone does not need to be picked up to initiate the call?

That will be appreciated, Are you talking next minor version, or major one? I am hoping I do not need to wait for 5.0 Let me know when the version is out.

Setting the country code, and trunk did the trick. I have purpously been not setting this on version 3, because I do not want to PBX re-writing my numbers, unless I specifilly ask it to do so in the dialplan, so it is a shame that version 4 forces you to turn this feature on. As for your statement about the ANI at the domain level, you are correct that there is a domain default ANI, however I need a domain default E911 ANI Have you had a chance to look into the other issues? The park/retrieve is very important, since it is holding back upgrading a customer to Ver4, and the customer is in dire need of DoS protection. He gets taken down about once a week by the not so "friendly-scanner"

Is there a variable that can be used when setting up call recording that shows in the file name of the actual extension that took the call. I have a situation where they want to review phone performance for agents in an agent group, however the $u shows the DID that they called, and not the member extension that actually answered the call. Is there another variable that can do this?

I am trying out version 4.0.1.3499 (Win32) for the first time. I am very fluent with the latest version of 3.4, however I am noticing the following issues with 4.0.1.3499 (Win32) When receiving a call on a global trunk, it cannot find the extension unless the trunk is in the same domain as the extension, I have my send calls to extension field blank, however filling an exact extension in there does not work unless it is in the same domain. I have found that if you have * in the Explicitly specify park orbit preference: of an extension, It prompts you for the park orbit, and that works properly, however when retrieving, it prompts, and when you enter the orbit number, nothing ever happens. I know the problem is only with retrieving, since typing the full * code including orbit number, and dialing it in 1 shot still works. If failover is set on a trunk, and the call does not go into connected state before the request timeout timer, it fails over to the next trunk in the dialplan. In 3.4 it did not do this as long as it got a SIP responce from the gateway. This was usefull because if the gateway was down, you could have a short timer, and it would fail to the next gateway. In version 4 if you have a short timer it will fail to the next trunk while ringing When editing a dialplan in the text editor mode, when you use global trunks that are not in the domain it fails to add those lines, however if done traditionally, the problem does not exist. I am sure there are no typos since if I take a working dialplan copy/paste it into a new one, it leaves off all of the global trunks that are not in the domain. I think this might be related to my first issue listed above. I have also noticed that after it has failed with the text version once, if you try to add lines to the dial plan the traditional way, it fails to add them, and the dialplan must be deleted. I like the addition of the ANI for emergency calls, however I think you should be able to specify it at a domain level as well as the extension level. I like how you can type in an extension number and jump directly from 1 extension to the next, however I feel this feature should be available on all pages of the extension. It became quite combersom when I had to apply intercom rights to a bunch of extensions, and always click through the pages of the extension to get there. Previous/next buttons on each page within the extension would work well as well I have also notice when you click on the button to list extensions, it takes you back to page 1, and not the page you were on. I liked how 3.4 kept you on the same page. Please let me know of any workarounds/suggestions for these issues.

I would also like an additional parameter that would tell the system not to open the pop up window when the link is clicked.

I am very interested in testing the 64 bit version for windows

Is there a parameter that can be used so that inputting the URL does not open a new window stating: Remote Call Initiation Your phone should display the number that you have entered. Just pick up the handset to start the call now. Please see the manual of you phone to customize this behavior. Depending on your security policy, you phone may start the call immediately or wait until you acknowledge the call invocation.

I am trying to test 4.0 on an old Win 2000 server machine, and the service will not start with the error of Could not start the pbxnsip PBX service on local computer Error 193 %1 is not a valid win32 application. Is it supposed to run under 2000, or is something newer required.

I can understand how all of the different standards can be problematic, however you said that what is clear is what is in the from header is what should be on the display. In my packet captures I noticed that the from header changes based upon the type of Remote Party/Privacy Indication selected, is this a glitch, and if so, can it be fixed? Just wondering since I can come up with a workaround with my second ITSP if the from field always indicated the original caller id (like it does for PPI and PAI) They stated that if they did not receive a PPI or PAI, then they will fall back to reading the from header. So therefore I could set it to no indication (or remote party id), and that would cause them to read the from header. The problem now is the from header changes when you select no indication or remote party ID vs PPI or PAI

I have just completed a bunch of packet captures, and uncovered what seem to be some inconsistencies in the fields when you select different types of caller ID on the trunk With the trunk set to P Preferred when forking a call, the fields display the following: From: Original Caller ID Contact: Original Caller ID P Preferred: Extension ANI field With the trunk set to P Asserted when forking a call, the fields display the following: From: Original Caller ID Contact: Original Caller ID P asserted: Extension ANI field With the trunk set to none, when forking a call, the fields display the following: From: Extension ANI field Contact: Extension ANI field With the trunk set to remote party id when forking a call, the fields display the following: From: Extension ANI Field Contact: Extension ANI Field remote party ID: Original Caller ID With the trunk set to 3325 don't hide when forking a call, the fields display the following: From: Original Caller ID Contact: Original Caller ID P Preferred: Extension ANI field All of these packet captures were done on 3.4.0.3202 win 32. While I do not know the RFC behind them, I am going out on a limb, and assuming that the From: and contact: fields should remain the same, and the Remote Party/Privacy Indication: should be the only thing that is different in the packets. Ontop of that, I would expect that it would show something consistently like always show the original caller ID, or always show the ANI field, and not change based upon the type of Remote Party/Privacy Indication: selected on the trunk. Personally I wish I could always show the original caller ID, however I believe that to make a flexible product, there should be the option to choose on a per extension basis if you want to send the original caller ID, or if you want to send the extension ANI on forked calls. The inconsistencies I am having come from if I have 2 trunks, and someone turns on call forking, if their call goes out the trunk that supports remote party ID, they get the original caller id, however if that trunk if full, or the second trunk (P preferred identity trunk)is prefered in the dialplan, then they get their Extension ANI on forked calls. This inconsistency is causing a real customer service headache.

I just wanted to see how caller ID should be handled when forking to a call phone. In the past all of my ITSPs have sent the original caller ID along when forking to a cell phone. However I now have a situation with a different ITSP where it always sends whatever is in the ANI field. I have tried all of the different caller ID settings on the trunk, and it always sends the ANI. I have verified that they will allow me to send any ANI I want by filling a random 10 digit number in the field, and it will send it on through to the cell phone. I have looked at packet captures, and I see both the ANI, and the original caller ID going out in the invite packet from the trunk. What value should be getting used, as well as what is the intended way for PBXnSIP to work?

In this specific case, I was reluctant to clone the server because of problems with the server. I wanted a clean install. One of the reasons I only trusted the domain portion of PBXnSIP was at the system level it was holding on to 55302 CDR records even though the retention was set to 2d (the customer averages 150 calls per day) I believe it was causing the service not to start in the 30 seconds that windows gives it. Now after the clean install it starts in about 2 seconds.

I had a case where I needed to backup a domain on a machine running 3.4.0.3202 so I used the system to create the backup tar file for the domain. I then needed to format/rebuild their machine. Upon re-installing PBXnSIP on the machine (same version) and then importing the domain, all DID numbers were lost. I remember a problem where something similar to this would happen in 3.1 however it would take restarting the service before they were lost. Now with 3.4.0.3202 they do not show up at all. Luckily this customer did not have very many DIDs, however I see this as an urgent issue since it means that backup procedures I have given to other customers are not fully effective.

I would be happy to, however I cannot figure out a setting to make all of the phones ring instantly, and at the same time.

I have clients that would like to be able to log their extension out of a hunt group similarly to how you can log in, and out of an agent group. Is this possible?