Carlos Montemayor

Got it! As I had suspected, the software did have what was needed to present things as the carrier wanted. It was just a matter of selecting and writing the right stuff at "Custom Headers" together with selecting the right choice in the Rewrite Global Numbers what did the trick This is great! Thanks!

I tried to set the title in such a way that may help others. Not an easy task, but well, I tried. This is weird. For the first time, a call that I placed got the clock on the display of the phone started, as when it actually starts a connection and there will be conversation. The estrange thing, is that the phone kept ringing. No one had actually answered the phone. So, was the call connected or not? Since I was able to talk to the other party later on, I found that there had not been a connection. In the call log, I saw that the pbx had actually recorded the call as a successful one, although, it was not. Moreover, in the sip messages, I was able to see the following messages regarding that particular call: Invite Authentication Required Ack Invite Trying Session Progress Invite OK Ack Bye Ok The pbx is a 5.1.3 running on Debian 64 and the phone is a Yealink that has its configuration file thanks to a wan based plug and play. Is there something wrongly set in the trunk? or what can it be?

Yes indeed, and I really do not want to take that route. Perhaps I did not fully understand your previous post. Is it about sending you a PM with the things as need to be so this supplier can be taken care of? I will do it right away.

That sounds very promising, however, in this particular case, I was able to adjust all the headers just as the supplier wanted them thanks to the ability of the pbx to allow for custom headers. Since they are not asking for any format on the dialed numbers, I imagine that we already have that option as well and I feel that it is I who do not know how to produce them. I suspected that either the option "Do not Rewrite Global Numbers or for ROW 0 and 00 would do the trick. I agree that once you start "cooking" something, the options can be endless, however, in this case, they do not want any cooking. Just raw as it comes from the store. I bet we already have that option. Don´t we? Regards

Hi, I recently added a new SIP trunk supplier and I have not been able to present the digits as they need them in the INVITE. Having set the country code and area code in the Domain Settings, the PBX presents the digits to the carrier starting with the "+" sign and country code. I need the PBX not to do that. The way this carrier needs the digits is just plainly as the end users dial on the phones. No more, no less. I understand that on the Trunk level, there is a way to control how the digits are presented to the carrier and that is the parameter "Rewrite Global Numbers". In my case, the value that would seem logical or best bet for me would be: "For ROW (0 or 00 format)". I tried it but found that it had no effect on the digits on the INVITE. Just to be sure, I tried every other value there but none worked for me. Perhaps I need to set some other value at some other parameter in order to make this Rewrite Global Numbers to work for me. Is that so? At the moment, the only way for me to present the digits as I need them is to erase the country code and area code from the domain fields, however, doing that will make me loose all the other nice things like inter-domain calls and CDR standardization that I have come to love. So, is there something that I am missing somewhere? Regards,

By way of trial and error, during the wee hours of Sunday, I was able to find out that if I change the value of the parameter "Rewrite Global Numbers" to "ROW (0 and 00 format)", the outbound trunk calls work fine with my carriers. That was the only value that would work in my case. Fortunately, now I can have inter domain calls as well as good outbound trunk calls. Now I have a new question. Is there a way to present the DID as the displayed number for the inter domain calls? It seems that we are showing the extension. Regards

Hi, I am trying to replicate this scenario, because I also need to do some inter domain calling but I am getting the "Temporarily Unavailable" error message, In order not to affect current active users, I set two "test" domains, each with only one extension with a 10 digit DID, Following what I understood from this thread, the following is my checklist: 1) Loop back detection is set to off in sip settings 2) Both domains have a Sip Gateway trunk with the proxy address of 127.0.0.1. In both cases, the trunk is Global and set to send calls to the Request Address in the URI 3) Both domains have a dial plan where Try Loopback is the highest priority trunk with "*" in both fields 4) Both domains have the country code set In the log file, I can see that the system does find the right extension (thanks to the DID) and it seems to be sent through 127.0.0.1, However, I only get the Temporarily Unavailable message. What am I missing?

Hi, I put the country code in all domains but had to remove it immediately. The issue is that my carriers started to get digits that they do not need and that messed up calls. In other words, the chain of digits that are produced from the dial plan, were fine without the country code, but started to get the country code at the beginning (after adding it in the domain) and distorted what the carriers were expecting. I would like to allow for inter-domain calling, but, obviously, normal outbound calls need to continue as always. How should I configure things? Thanks in advance.

Following this tread... what about trunk used? I can see the column with the header "Trunk", but the data is not there. How can I get the trunk used on each call to be shown in the Call History again? Regards

Hi Bria works fine on iPad. I have used it with no problems. As for the blue tooth headset, well, I have not tried it yet, but if the iPad supports it, it should work too. Cheers

Got it. Thanks!

Hi So the main trick was to issue the command once positioned in the right directory. However, I had to write the command as previously recommended, as follows: tar -zcvf backup20140123.tar.gz /desired-directory That was the way that I was able to get my file. Thanks! Regarding excluding things, If I would like to keep everything exception made of the recorded conversations the command would be: tar -zcvf name-of-backup-file.tar.gz /desired-directory --exclude /recordings/pbx.company1.com --exclude /recordings/pbx.company2.com and so forth? Regards

Agreed. Could you please outline the procedure to do the file system backup? I mean, I accessed the server using SSH and gave the tar command but it did not work. I have my backup file now thanks from the help of Vodia support, however, since this is something that should be done every so often, I should be able to do it by myself. How should I use the tar command? Regards

Done. Regards

I got the following response: Cowardly refusing to create an empty archive Try "tar --help" or "tar --usage" for more information

Hi, I am trying to make a backup file, but I am getting a 0 KB TAR File as a result. I figured that the actual backup must be larger that the allowed limit, so I deleted all the recordings of conversations and increased the limit of the backup file to the maximum of 10 MB, however, that did not help. I continue to have a 0 KB tar file. What can it be? I am using version 5.1.2 64 bit debian hosted. Regards

Got it. Thanks!

I was under the idea that they did. After Reading your answer, I did some research on the web and I am gathering that they have discontinued support. It may or may not be available depending on the kind of account and when that account was activated. So, I really do not know now. So, to have ActiveSync of contacts with the pbx directory one should use Outlook or 360?

Hi, I am trying to use Activesync to populate the personal phone directory but I have not been able to do it yet. As a matter of fact, I am unsure what exchange address to specify. My contacts are in a Google account and I am using version 5.1.2 hosted.

Had not tried before, but I just did after reading your post. It hurts not to have thought of that, but mighty glad that you did. It works!! It only misses the message (the one that tells you that you have placed the call on park orbit), which would be nice but that is something one can live without. Thanks!!

Hi I just noticed that I am unable to do a group pick up. When I try, I get a "not found" message on the phone display and in the log file of the pbx I see a "undirected call pick up failed" line, then I see that the *87 call is evaluated against the dial plan. Curiosly, if I do a directed call pick up (*87 plus the extension) it works just fine. I am using version 5.1.2 (Debian64) hosted version and Yealinks with the latest firmware. I imagine I have some configuration issue, but I have not been able to get it right. What can it be? Regards

Well, I do believe that there is room for a system in that category. I mean, I have been building around atom based systems and they are over dimensioned for a small office and around here people do appreciate the lowering of the investment. It very well be that the BeagleBoard does run Debian out of the box. It is just a matter of trying and it may be worth doing it. If it does not work for our purpose, we may find something else to do with it. Cheers!

If one were to try the BeagleBoard, which installation package should be downloaded? I no longer find the ARM based version in the release corner. Should one try the regular Debian version?

Is there something we can change in the configuration of the phone? or a workaround? Is it something that could be fixed in a future firmware release? Thanks

It works with the snom 710 but not with the Yealink. I have tried with and without dialing the exact park orbit. In my case, I used account 66, so I tried *8566 ok which did nor worked. As I said earlier, *8666 ok, did work (pulling the call out of the park orbit. If it were not that pulling the call out of park orbit (as well as listening to voice messages) was working using the Yealink, I would say that the probable cause of the problem would be a wrong DTMF configuration. I cannot say that because retrieving the call and other DTMF managed functions do work. So, I would suspect of something that has to do with DTMF but that only happens during a call being on hold (the procedure indicates to first put the call on hold and then dialing *85). The other thing would be the permissions. I have not written anything there, although it does not prevent the snom 710 of doing the park. What should I do?