Carlos Montemayor

It is not in the list yet, but willing to enter. Shall I PM you with a SIP account of theirs with its corresponding user, password, domain and so forth and such like?

Hi, I have the same problem. I need to redirect always a call to a cell phone and the user needs to see the original number, not only to know who is really calling but also to be able to return a missed call. In my case, there is absolutely no restriction from the carrier side, which is in turn an ITSP. Where can I instruct the pbx to present the call with the original number? Regards

Well, Just to unveil the mystery. The problem was not the syntax of the interfaces instructions. The issue was the age of the operating system at the customer premise. It is so old, that neither Linux nor Mac equipment can get Internet Access from the LAN. So, I will have to base my system on windows if I want to serve this particular customer (there is no way to know when are they going to update their operating system). Regards

Pablo, I just followed your exact advice in an identical system that I have at home and it worked just fine. Hopefully, I will be able to do the same at the customer premise, however, over there, it is always possible to have a problem (the IT guy can assign the system a fix IP that is in conflict with another address, or what have you). Thanks and regards

Where should I look for those generated files? In the pbx or on the phones?

Well, I just saw that the firmware is not the latest. The m9 is running 9.5.6 a and I tried to upgrade to 9.6.2 a with no luck. In the log I can see a 408, request time out. I imagine that a firewall did not allow me to upgrade. Nevertheless, I tried a simple "reboot" and that did the trick. They can dial 110xxxxx numbers now.

I just did. The box for emergency numbers is empty. Nothing there. Users tell me that they used to be able to call numbers starting with 110, but all of a sudden they cant What else can it be?

Hi, I am running a pilot with a customer and I have set several snom 3XX, 7XX and a couple of M9r. Everything is working smoothly exception made of the inability of the M9r handsets to place calls to certain local phone numbers. When somebody dials 110, the call is dropped immediately. This does not happen with any of the desk phones. Everything has been configured the same for the desk pones and the handsets. I mean, they are just extensions with the same dial plan. Does 110 mean something special to the handsets? Can I change something so the M9r can call local numbers that start with the digits 110? Regards

Will do. Regards

Hi, I feel a bit ashamed as I am writing this, but sorry, I need the help. I am setting my systems based on Debian Linux just because it is so stable. Once set up correctly, you can almost forget about it. I have learned to install Debian and snom ONE on top of it as a cooking recipe, and so far so good. Problem being when something in the environment is different, then I do not know how to adjust. The problem at hand has to do with a network fixed address as opposed to dhcp. In my last installation, there was no dhcp server and I had to set a fixed address for my box. I tried to do it but I got lost in the woods. How should I do it? Can I get an example on how to pass those instructions to the system? Thanks in advance

Hi, To compound the problem, I believe that one should take into account not only the distance from the base system and the cloud hosted service, but also where the sip trunk provider is located. The majority of my customers are in the northeastern part of Mexico and the SIP trunk provider is in Central Mexico. It could work that the cloud be in Dallas, I believe it does, but one should run a pilot test because with IP traffic, the actual flow of packets not necessarily follows a straight line. Many moons ago, I negotiated a "peering" arrangement between the local incumbent carrier and a new competitive one. If one customer of the competitive carrier wanted to visit a web page hosted with the incumbent, the packets were to go first to the USA (where the incumbent carrier linked with a tier 1 Internet provider) and then back to the country, to the incumbent. After the peering arrangement, the packets followed a more logical path. So, things like "latency" (the time it takes to the packets to travel from server to server and back), variability of such latency (jitter) and packet loss, are the main parameters to watch. And to make things even more complicated, some measurements, like packet loss, can be tolerable with one codec, but not with another. This is because, proportionally, more data is lost when a high compression (and hence efficient) codec looses a packet. In short, one has to do tests. If it works, great! And here there is a characteristic that snom phones have and I feel that is has not been touted enough yet. At user agent level, the IP phone, has failover between "identities", so, one identity can be registered to the local pbx, and a failover identity registered to a PBX in the cloud. If no backups had ever been done, the users may not have access to their stored voice mail, directory and things like that, but they would be able to continue making and receiving calls. Coming back to "how far is to far", latency is the keyword. The number of milliseconds that a packet takes to make a full round trip. In my short experience, below 50 milliseconds, you are going to be praised for the quality of the voice, from 50 to 150, most likely everybody is going to be just happy, above 150 you will probably start to have some complaints, first by the more discerning users and as latency approaches 250 ms by probably everybody. Above 250, probably registration will become a problem in itself.

Thank you Pablo. I will certainly have a look into it.

Following up on your third paragraph (physical machines instead of virtualization). If the "production" system goes down and it has a fixed IP address of the local network (not DHCP) wouldn't it be easier to start the "failover" system with the backup configuration file that one should have generated recently from the production system? In that way, I suppose, everything, including its fixed IP address would be the same, exception made of the MAC address which would only cause the need to reset the license key which in turn could be done promptly too? Regards,

Well, I believe that being the pbx a solution for business and that for a business, regardless of its size, communication is always a mission critical service, the handling of fast recovery from a disaster is something that has to be addressed. As integrators or value added resellers, we will be encountering some end users that will have knowledgeable IT people who can do some of the errands and some that will not. Budgets will differ as well, so, I think I will have to learn about the several ways to handle this. If downtime can be only minutes and such a thing will happen every other year or so, I will try to privilege and promote the solution that is easier and more economic to implant and that seems to be the one outlined on your second paragraph. I do not dislike virtual machines, I am just not acquainted with them, but that can be fixed. Browsing the web, I just found a product called: "VMware vSphere Hypervisor", it claims to be an entry level and free solution. It may be a good place to start to learn about this things unless there is another tool you would suggest. Cheers!

Hi, I would like to follow on your last recommendation. Currently, I am building my own little systems based on Debian, and they have been performing just fine. However, as larger customers (with mission critical uses) start to show up, the need for physical failover becomes a reality. For what I have seen in my area, building two simple (but good) systems would cost about one third of what could cost a server with redundant power supplies, redundant hard drives and a CPU that will surely have more processing power that we will ever have use for, and for what I gather, will render just the same kind of reliability. So, how could we set up such a couple of boxes? Thanks in advance

Hi, After being quoting and installing around 20+ users systems, the first larger beast is rearing its head. I have just been asked to quote a system for 120 users. It will have to coexist with a digital E1 to interact with a carrier and with some 10 or so FXO lines, as well as be prepared to one day use SIP trunks. Reliability is the number one characteristic required and, of course, it should have enough processing power to take care of the 120 users generated traffic. Now I wonder, it makes sense to have redundant power supplies, redundant fans, but, how about redundant hard drives? Would that actually work? I mean, license wise I suppose that if the hard drive where the system is residing "dies", it would not help to have another one instantly attempting to do the job, it would lack the proper license, or not? What would be the recommended way to address the issue of achieving the highest levels of reliability and resiliency?

Hi, I was attempting to do the remote Plug and Play and although the phones said: "Request sent", I did not see that anything changed. Neither I was able to see anything related to that in the log of the pbx. However, while I was struggling with that, something happened in the site where the phones were located because I lost connection of the VPN and when it came back, the phones had switched their local and private IP addresses (the ones assigned by the router). It seems that having renewed their local IP addresses was what was needed since the phones started to receive calls as before. Regarding upgrading the firmware, I read in the release notes of the newer version that something that they improved had to do with PnP and provisioning. So, I am tempted to try again, however, the procedure to update its firmware is not as straightforward as it is with snom phones. I understood that I would need to use a TFTP server and some correct templates. It is not just adding a new file and presto. I am not 100% sure that the port forwarding had been done right, because I do not have access to the router, the companies IT guys did it for me. So, it may have been that, or it may have been that the phones need the latest firmware. I do not know at the moment. Regards

I think that finally I understood the procedure of the wiki. Now I only wonder if I need first to change the firmware of the phone or should I try first to do it as is. Regards,

The plug and play of this Yealinks happened in a LAN even without my intention. Out of the box and even before I had turned on the pbx where I intended to do my testing. This happened because I had forgotten that I had a free versión in my laptop that I had forgot about and the Yealinks instantly and automatically found the pbx in my laptop and did plug and play. I had nothing to do with it. Now I learned that I should stop the pbx in my laptop to avoid the plug and play or deactivate listening to multicast. So, PnP in a LAN with Yeallink happens out of the box. I already have versión 5.0.10 running in the production pbx and know that PnP woks in a LAN out of the box. I will have the port forwarding done in the router to reach the pbx via http and set such same port in the pbx. What would be the next steps in order to achieve manual WAN based PnP with those Yealinks if at all possible? Thanks!

I was able to upgrade to versión 5.0.10. In the realse corner 5.0.10a is not referenced there yet. Can I get its download address? The PBX is under a UTM (Unified Threat Management) Router/Firewall, Cyberoam Brand on the HQ of the customer. Placing the PBX on a public IP would be possible, but it would loose points at the eyes of the IT people. Port 80 is being used by the Cyberoam itself and port forwarding is possible. They already did once to allow me to manage remotely the pbx (through https) On the snomONE wiki page I found a step by step procedure to do the manual PnP for remote phones when they are snom models, however, I could not find such procedure for Yealinks, only for a local LAN. Is there a procedure for Yealinks?

Yes indeed. I will update the firmware and attempt WAN based plug and play (since the phones are away from the pbx). If port 80 is already in use in the local network of the pbx, do I need to change it and also do port forwarding in the router? This is going to be my first WAN based plug and play. Regards

Hi, I am trying to make a remote extension work. The phone is a Yealink T22P Firmware version 7.61.0.80, The PBX is running version 5.0.8 (Debian64). The phone registers as an extension and is able to make calls with no problem. However, receiving calls is another story. The PBX receives the call but the result is the extension voice mail. This happens in both types of calls, inbound trunk or from another extension. In the lofgile I could see the follwing: [8] 2013/05/30 15:30:35: Incoming call: Request URI sip:8780047537@192.168.1.1:5060;transport=udp;line=8f14e45f, To is <sip:1401258111071219@200.76.112.13:5070;user=phone> [8] 2013/05/30 15:30:35: Set the To domain based on To user 210@alfa777.mx [9] 2013/05/30 15:30:35: Using outbound proxy sip:192.168.1.254:38470;transport=udp because of flow-label [7] 2013/05/30 15:30:35: Call 2d85356f@pbx: Clear last INVITE [5] 2013/05/30 15:30:35: INVITE Response 480 Temporarily not available: Terminate 2d85356f@pbx [5] 2013/05/30 15:30:35: set codec: codec PCMU/8000 is set to call-leg 414 Please advice. I ran out of ideas and need to make them work as soon as possible. Regards,

Hi, I am afraid that the Dialog 4222 and 4223 phones are proprietary and work only when connected to a same brand PBX, like the MD 110 they are hooked to now. If so, I will have to quote the snom IP PBX, plus SIP compatible phones, plus a Gateway for the carrier side (the E1) and in that regard, I may acquaint myself with a Patton or Vega. Hopefully the price tag does not go as high as to kill the deal. Regards and thanks for the prompt reply

Hi, I have had success with analog Gateways (both, FXO and FXS), however, I am about to have my very first encounter with the digital world and I need guidance. The potential customer has an old MD 110 originally from Ericsson (today Aastra) and I have already talked them into moving to snom. However, they will like to keep as many as possible of their phones, which are of the model dialog 4222 and 4223. On the carrier side, they will also want to keep their current E1. What Gateway or Gateways would work? Which one would mean the least pain for configuration and installation? Thanks in advance!

Steve, You are right. I am not even sure if one is faster that the other. I am relying on a benchmark test where we do not even know exactly what kind of processes they put the chips to do so they can grade them. The key process that we are interested on in our application, would be the ones that allow a higher number of concurrent calls. That parameter would really be the one to use to judge them. With the information that I have at the moment, I would intuitively suspect that the Celeron would have more "throughput" thanks to its memory management capabilities despite of its slower clock frequency. Just as a big fan can move more air than a little one, despite the fact that the little one may have more RPMs, but I am guessing and again, relying on a benchmark test which may be scoring processes that may be irrelevant to our application (what it takes to be able to handle more concurrent calls). I hope that someone who has had hands on experience with these two processors or who knows of their inner workings will give us more info. Regards!