Carlos Montemayor

Hi, I am also interested in this feature. I am currently using a hosted system, version 5.2.3 and I can see the limit for the Global System (all domains) and also per extension, but not for a domain. Am I not looking in the right place?

Fixed! After reading your post, for some reason I thought on changing one parameter in the "behavior" section of the Auto Attendant. Under "Behavior", Extension input, the parameter was set to "When extension matches" and I changed it to "User must hit pound". I do not quite understand why the direct destinations would work for extensions, but not for IVR nodes with the first parameter and with the second, extensions as well as IVR nodes can be used for direct destinations in the Auto Attendant. The good thing is that it all works now. Knowing the "why" would be nice too. Regards

And help it did indeed! Thanks! I had not tried typing anything. The window saying "empty" had lead me to believe that it was not working. It only needs the first letter to display everything that starts with that letter. Thanks again!

Exactly right. The AA does not take me to the IVR node. However, I can reach the IVR node from an extension. On the other hand, the AA does take me to all other extensions that I have there as direct destinations, only the AA to IVR is not working for me. What can it be?

Hi, By reading the online documentation, I figured that an IVR Node should be reachable from within a direct destination of an Auto Attendant. I tried that and was not able to make it work. When I dialed the IVR number (its account) from an extension, I did get to the IVR node, no problem there. What could be happening that is not allowing to reach the IVR node from the Auto Attendant? Regards

Hi, I see that after plug and play, all the section related to LDAP on my Yealink T22P got populated. I mean, it has the server address, user, password, port and the rest of the seemingly correct settings. It also has the correct flags to enable LDAP. So, I figure that LDAP, or at least some functionality of it will work with a Yealink series T phone. That encouraged me to enable one soft key of the phone (under DSSKey, Programable Key) as LDAP. Now, pushing that button shows a display saying that found the address book empty. I had put just three addresses in the global directory (domain level directory) that I do can see in the Console (if I log in as a user). The data that is populated in the LDAP section of the phone are as follows: Enable LDAP Enabled LDAP Name Filter (](sn=%)(gn=%)) LDAP Number Filter (](telephoneNumber=%)(mobile=%)) Server Address xxx.xx.xxx.xx # which is the right IP address of my server Port 389 Base ou=people Username pbx.xxxxxxxx.com\xx # domain name\extension password ******** Max Hits 50 LDAP Name Attributes cn sn givenName LDAP Number Attributes telephone Number mobileTelephoneNumber LDAP Display Name %cn Protocol Version 2 LDAP Lookup for Incoming Call Disabled LDAP Sorting Results Enabled Did something go wrong in the population of this data? Is there something that needs to be adjusted in order to be able to use LDAP on the phone? I am using version 5.2.3 on a debian 64 bit with a hosted license Thanks in advance

Oh my! I promise I had looked over the menu options and I did not see it, but yes, it is there. Sorry and thanks!

Hi How can we prevent users from changing their web passwords? Regards

Hi, I just encountered some weird behavior. It is as follows: I can have good interdomain calling by using Loopback, and I can restrict outbound calls based on PIN (through CMC authentication and checking the P box at the trunk on the dial plan). However, I cannot have them both. I mean, if "Try Loopback" is enabled as an option in the highest priority of the dial plan, the PIN will not be required to use the outbound trunk. I do not see the connection, but it is happening. Is there something else to configure here? Regards

It did help. Thanks indeed! The dial plan only had a bunch of "x". I did not count them. So I changed it to the following: *xxxxx]xxxxxxxxxxxxxxxxxxxxx (Well, I did not count the number of x in the second part of the "or"). I suppose that we should also change something in the PnP template for the Cisco phones. This need appeared because we were unable to record an auto attendant after doing PnP (we had to dial *98 plus the account number of the Auto Attendant) Regards and thanks again.

Hi, I need to set up an Auto Attendant. First step is to record the greeting but the phone, which is an SPA303 that did successfuly a WAN based PnP does not allow the use of the "*" (star). Right after inputing the "*" key, it says "invalid number". What should I change in order to be able to dial * codes on a SPA303? Regards

It certainly helps. That is the setup and in such a way, I will have the CDRs that I also need. Thanks!

After thinking this over, I do realize that it cannot be done as proposed. However, the pbx does provide for a very nice work around. The CPE pbx can send all the outbound external calls to an extension on the hosted pbx. A SIP Trunking solution. In this way, the hosted pbx would be writing the CDRs on its CSV because it is actually terminating those calls. In this new scenario, what would be the way to go? I mean, should the CPE sip register a trunk as an extension on the hosted pbx? or should it try as a SIP Gateway or interoffice trunk? The hosted pbx is on a public fixed IP, facing the Internet directly and the CPE is behind NAT, although there is a fixed public IP assigned in the router (The CPE pbx is an item on a LAN, I meant to say). Regards,

Hi, I have a hosted system that is handling most of my deployments and it made sense to collect CDRs for billing purposes in its disk. Currently the CDR URL indication in the hosted system is simply: fileto:$C.csv and it is writing a csv file correctly under /usr/local/snomONE. The hosted system is on the public Internet facing it directly (it is not behind anything). Now. I am about to start some CPE based systems and I would like those systems to write their CDRs in the CSV file that is maintained in the hosted system. I imagine that such a thing should be able to be done. Is that right? And if so, what would be the correct syntax for the CDR URL on the CPE systems? Regards,

Hi, The value of being able to evaluate, or at least to have a yardstick to assess the effectiveness of a SIP Trunking supplier (for call termination) is rather obvious (or so I think), so I will not elaborate on that. There are a few parameters to do so and some of them we can rather easily get from the PBX. The ones that can give an idea and we can get rather easy are the following: 1) Average call duration 2) Estimated MOS of calls 3) ASR (Average Seizure Rate, which is just the percentage of completed calls of all those attempted) The one that would be very useful and the ingredients to have it may already be there but I do not know how to get is the following: 4) NER (Network Efficiency Ratio). This one, is like ASR, but taking away the calls that did not complete because of reasons not accountable to the carrier (or SIP Trunking Call Termination Supplier). Those reasons to not complete a call that are not the responsibility of the supplier are: 1) The call was ringing, but no one answered 2) The recipient was busy 3) The receiving carrier answered with an error code (or recording) of their own This NER would be the truer measure of the effectiveness of a call termination supplier. The percentage of successful calls terminated taking away the effect of those calls not terminated because of reasons out of the suppliers reach. I bet that the PBX has the ingredients to estimate NER, or at least allow us to gather them so we can estimate it. I believe that all those things can be gathered from the JSON CDRs, but I do not know how. Currently, we have been successful getting sufficient information from JSON CDRs for billing purposes, but we are rather lost about getting other kind of information from it and I suppose we could. Can we get guidance on these topic? Regards

Got it. So, one less thing to worry about. That is great news. However, in this particular model, there is one different behavior, the language of the GUI. After provisioning, models 20 and 22 get set to whatever language one is using in the pbx, which is great, but on model 21, it gets set to Chinese, which at first, is scary.

Hi, I noticed that the model T21 is not mentioned in the list of Yealink models that one can make PnP with. T20 and T22 are listed. I have already tried it and it works, although it does come with a very small problem that can be easily fixed. The problem is that the GUI language gets set to Chinese. Changing the language is not really a problem, one could do it after PnP. The reason I bring this topic up is only because the model T20 is no longer being offered in my region so model T21 would be the next logical choice as an entry level phone and, if so, I would worry that beyond the evident GUI language problem could be other ill configurations that I may not be seeing just because the features have not yet been required by the end users. Is there a reason not to do PnP with a Yealink T21? Could we have it in the list of PnP pones? Regards,

Well, this is a bit embarrassing, but what the hell, shame would be to get caught stealing in flagrancy. Sorry! I was not aware that the section regarding customization exists at admin level, for all domains, as well as in the domain. I was tweaking in the admin level and the file actually being used was the one at domain level. Now I wonder when would the files at admin level would be used since when MAC based PnP is used, it always gets its configuration file from the section at domain level. I was also a victim of a terrible translation of the release notes from Chinese to English. Now, I do not know what they were talking about regarding configuration files on version 72, Sorry again and thanks

Hi, I was trying to make some changes to the PnP file for Yealink pones. Something that I use to be able to do. However, although I am generating a new edited yealink_common.txt, It is not being taken by the phone during plug and play. What I believe that may be happening here, is that the newer version of the firmware (version 72.0.30) is not allowing for erasing of the configuration file by doing "reset to Factory defaults". Has somebody else from the forum come across this issue? If it is not that, I do not know what else I may be doing wrong. Regards

Hi, Thanks for the reply. It was hitting all across the board, I mean, in different sites as well as with different phones (Yealinks and snom's). I increased the setting of "SIP connections per second" and "total number of sip connections" and that did help a lot. I was unaware that I had to increase those parameters as my deployment was growing. Those settings are there to protect us against of external attacks, which is great, but had to be adjusted as we were growing. Now that we are looking into those settings, the total number of sip connections should be rather simple to set, it should be just above the total actual number of sip registrations that one has because of the current users, I guess, however, the number of sip connections per second is another thing. Is there a common sense rule regarding what to put there? Also, regarding how to be aware of a problem with registration, yes, email helped me. However, I understand that the pbx has the feature of snmp and I imagine that as things grow, it would be a better solution to have something like a control panel that could be getting its information through snmp. Is there a recommendation regarding how to implant snmp for the pbx? Regards

Hi, I have been favoring TLS to register extensions on our hosted offering, however, something weird has started to appear lately. It has to do with failure to register with the pbx. That one method of transport work where other do not, is not new or estrange to me. What puzzles me, is that it appears not to follow a pattern. For example, in the same site, where some phones can work with tls, others need udp, and to compound the perplexity, the brand of the phone does not matter either, some Yealinks can and some do not, some snom phones (710s) and some do not. Also, deployments that had been working fine for several months with tls, are starting to require the change to udp to keep registration. And it is not that all the phones all a sudden need it. It is happening every now and then, but it seem to start to happen more often. The most extreme example, is my own phone, which is a Yealink T22P, which has 3 accounts. The first two currently have extensions that are using tls, but I could not register the third one, it had to go with udp. What can be happening? Regards

I got it! It had to do with digit presentation. Once I chose the right combination of "how to rewrite global numbers" and source of caller ID (from, or p-asserted) it started to work fine. I was getting confused because 4.5.1 and 5.2.2 have different presentations of those options. Cheers!

Thanks for the prompt response. The scenario is perhaps an easier one. I would like to allow for outbound calls from the mini into the 5.2.2 using a final trunk that is at the 5.2.2. I already created the extension at the 5.2.2 an in the 4.5.1 I created a trunk that registered as that extension in the 5.2.2, however, when calls are made in the 4.5.1 the phones get a "Forbidden" message.

Hi, I just came across a scenario where it would come handy to have a CPE PBX using a sip trunk from a hosted pbx. It seems to be pretty simple and I successfully configured that using a 5.2.2 version for both pbx (the hosted one and the CPE one), however,when I tried to do it with a CPE that is a snom mini version 4.5.1, I could not make it to work. Registration is done with no problems but the calls cannot go through. I am getting a Forbidden message. Everything else is being held the same, I mean, in both scenarios I am using the same final trunk, same phone and same dial plan. Is there something that I should adjust? Actually, where I really need to do the sip trunking is in the snom mini 4.5.1 (it is with an actual customer), and the CPE with the 5.2.2 version is in my lab (where it worked nicely). Regards,

Yes indeed. I have updated the ticket with that info. By the way, I saw that using the text version of the trunk settings produces the value "row" for the rewrite global numbers parameter. The one that I saw that worked, was "row (0 or 00 format)", the other possible choice, which was row (00 format), did not work. Cheers!