Nikolay Kondratyev

Hi All, I noticed that pbxnsip does not keep UA registrations across restart. And phones are not available for incoming calls untill reregistation occurs. And one must reboot all the phones manually. This affects, for example, upgrade procedure. Meanwhile phones simply does not know that their registrations do not exist any more... Is there a workaround for this? Is it possible (may be for the future versions) to keep registrations information (say in a file), so that pbxnsip will read that file at the startup, and if registration is not expired, use it? This feature would simplify restart (upgrade). Thanks, Nikolay.

Hi, We use pbxnsip version 3.0.1.3023 (Centos). I have set up a trunk with outbound proxy "m600.lab.nstel.ru". Pbxnsip does NAPTR lookup for "m600.lab.nstel.ru" and gets two records: [8] 20081212172627: DNS: Add dns_naptr m600.lab.nstel.ru 20 10 s SIP+D2U - _sip._udp.m600.lab.nstel. ru 25 15 s SIP+D2T - _sip._tcp.m600.lab.nstel.ru My understanding of rfc 3263 and 2915 is that a client must select a transport acoording to the lowest 'order' field in the NAPTR record. In my case it should be UDP, nevertheless pbxnsip uses tcp when sending calls to m600.lab.nstel.ru. My opinion is that this behavoiur does not correspond to the standards. Am i correct? Please clarify... Thanks in advance, Nikolay.

Another crazy idea: possibly it'll be more economical to develop a separate product - "pbxnsip-proxy"... It can be statefull proxy (not b2bua). And will serve just for tandem calls, consolidating several pbxnsip servers into single network...

Please investigate this task/problem. First, i was quite happy when i found that regular expressions can be used in the dialplan. The second time i was happy was when i found that i can assign separate dialplan to each trunk for incoming call analisys (using "Assume that call comes from user" field). And i was disappointed, when i found that it does not work (i described the problem earlier in this topic). Anyway i'm sure, that after a some time you (i mean pbxnsip team) will arrive at the same idea: tandem/transit routing is a must (if you are going to play at medium corporate market...). By the way, one of the side effect could be as following: P-Asserted Identity, P-Prefered Identity etc. header is unwanted for such tandem calls... (I mean that pbx should not insert it's own PAI header in such calls). If i'm allowed to tell my point of view, i would say, that there should be a possibility to select dial plan for incoming calls directly from the trunk configuration page, without reference to an extension. Though, i don't know if it is possible... Thanks and regards, Nikolay.

Do you mean, that i should redirect all calls to the same number? Or do you mean that i should use separate extension for each destination number, that is not a local extension? (It means that customer will have to pay for each destination number, which is being routed to different sip server.) What i'm talking about is quite standard task: routing of incoming calls between several offices... Routing is making a decision (depending on dialed number) if incoming call should be sent to local extension (phone, vm, aa, acd queue... any type of extension) or it should be sent to another sip trunk (trunk selection should also be possible based on dialed number). Many other working features of pbxnsip are quite greate, but...Without transit calls routing pbxnsip just does not suite for customers with several pbxes. So i have to ask again: can pbxnsip route transit calls? If not, may be it's worth planning this feature for the nearest development? Thanks and regards, Nikolay.

Thanks for teh reply. But dial plan is used for outgoing calls (i.e. originated from a registered extension). Incoming call does not even go through any dialplan. But i need to terminate incoming call on a local extension, or route it to another trunk, depending on the dialed number. The only exception, as far as i understand, when incoming call goes through dial plan, is when "Assume that call comes from user" field in the trunk configuration is used. I tried to use this field. And i was able to do what i want, but i found a problem. Let me describe: I created a special extension 9999, with a special dialplan. Put 9999 into the "Assume that call comes from user" field of a trunk named "m600". The dialplan for ext. 9999 is named "incoming_dialplan" and looks as following: pref trunk pattern replacement 100 "call extension" ^633([0-9]{4})@.* \1 120 m600 ^632([0-9]{4})@.* sip:\1@\r When the call arrives from trunk "m600" it is sent into "incoming_dialplan" (i can see it in the log). When the dialed number is 632xxxx the call is successfully routed back into the "m600" trunk. When the dialed number is 6333801 the call is successfully routed to existing and registered extension 3801. So ... it looks like i got what i wanted, but... When the dialed number is 6333808 (3808 is configured, but not registered extension) the calling party hears "comfort noise" instead of voicemail prompt. When 3801 calls 3808 the voicemail prompt is ok. When "Assume that call comes from user" field is empty and "Send call to extension" field is used, and external caller dials 6333808, he can hear voicemail prompt. Could it be configuration problem? Or is it a software problem? Should i send any additional info? Please advise what should i send, log file or trace, or something else... I use 3.1.0.3043 (Centos) version. Thanks in advance, Nikolay.

I mean the following: get the call from a sip trunk and depending on the number dialed, terminate it on a local extension or route it to another sip trunk. The thing is our company is going to be pbxnsip reseller in Russia and we have several potential clients for distributed ip telephony systems. And we are evaluating where we can offer pbxnsip. The typical scenario could be: Meidum corporate (couple hundreds of subscribers). Main office and several branch offices with pbxnsip in each office. Common PSTN gateway(s) or ITSP link(s). The incoming call must be routed to appropriate office according to telephone numer. Incoming (from PSTN) calls should be routed on main office pbx, i mean that routing table on a gateway (and espesially on ITSP equipment) is unwanted. I understand that one could have just one pbx in main office and create domains for branch offices, but this is not always suitable mainly because of the ip link failure paossibility. That's why i'm asking about transit calls possibility.. Thanks in advance, Nikolay.

How about importing users via LDAP from exchange or other ldap server? It would be convinient feature for new deployments... Thanks and regards, Nikolay.

Does PBXnSIP support transit calls? If yes, how do i setup it? If not, is it planned for future development? Thanks in advance, Nikolay.

I would say in another way: i mean using dial plan for incoming call routing. Not just single regular expression in the "send call to extention" field. The wiki page ( http://wiki.pbxnsip.com/index.php/Trunk_Settings ) says: The setting "Assume that call comes from user" is used for trunks that accept redirects (see above). The settings must be an extension in the domain of the trunk. This setting is necessary in order to determine what dial plan to use; and it is also necessary to charge a user on the system for the call. For regular trunks, you should leave this field empty. So, i make a conclusion that one can route incoming calls through a dialplan, where further flexible routing to different sip trunks (or flexible DID mapping) can be achieved. Of course From/To header must be preserved in this case. If all calls are routed through one extention, then - ok, since From/To headers are the same. And as far as i can see in log, the field "Send call to extension" is just ignored in this case. I tried to do this and i was able to route a call to pbxnsip extension and to another trunk..., though i found some problems... So the question is: is it a proper way to route incoming calls? Thanks in advance, Nikolay.

My idea was: one can create a special fake user, and a special dialplan, which is to be used for incoming call analysis. So one will be able to use several regular expressions to route incoming calls... Each range may be routed via it's own dialplan entry... In particular, this alows transit routing.. Is it bad idea? (Why?) Rgds, Nikolay.

i'm new to pbxnsip but... could it be done using "Assume that call comes from user" setting on the inbound trunk and thats user personal dialplan?

I also installed latest 2.1 version on my windows pc just for comparison. I found that PnP page indeed contains a list of parameters on this 2.1 windows installation. Looks like these parameters are descibed in pnp_parm directory. I have this directory with more than 20 files in it on my 3.1 centos installation, but PnP page is blank. And, again, there is no pnp.xml file on my windows 2.1 installation too. "html" directory is present, but it is empty. Should pnp.xml be there? Thanks and regards, Nikolay.

There was no "html" directory after installation. My version is 3.1.0.3043. I just downladed the binary, audio_en, audio_ru and started the service as wiki recomends. It is working, i can hear english and russian audio prompts, but no html directory. I created it myself and put pnp.xml (example from the wiki) there. rgds, Nikolay.

Aha... looks like i'm starting to understand how it works in pbxnsip... But unfortunately i don't have policom or snom phone right now... I found your message http://forum.pbxnsip.com/index.php?showtopic=277 with files for aastra attached (aastra.txt and aastra_mac.txt). Do i understand right, that there should be a corresponding entry in pnp.xml for those two files? Can you please provide me with this pnp.xml entry for aastra? I think i'll be able to finally understand how it works and will try to write my own files for, say, grandstream gxp2000. By the way, what about blank PnP page? is it ok? Thanks and regards, Nikolay.

Thnaks for the reply. I created tftp directory with the same permissions (1755) as other directories. And restarted pbxnsip service. But there is no files there... And admin settings->pnp page is still blank... Is it possible to make pbxnsip generate config files for phones? As far as i understand wiki automatic_provisioning page - yes. But how do i tell pbxnsip, that for extension xxx it must generate polycom configuration file, and for extension yyy it must generate snom config file? And moreover, is it possible to make pbxnsip generate, say, grandstream config file? If yes, what do i do for that? Thanks in advance, Nikolay.

Thanks for the reply. I just put the binary into /srv/pbxnsip dirctory, made pbxctrl symlink, and started the service using startup script from the wiki... Did i miss something? Has somebody saw the same problem (blank PNP page) in 3.1.0.3043 on centos? Thanks and regards, Nikolay.

Oops, sorry, i should have post this in "voip phones" forum (i think), not in "aastra phones"... Fat fingers...

Hi all, we are evaluating pbxnsip, i have 3.1 version on centos with demo licence (calls length is limited by 3 min). I'm new to pbxnsip but have some experience with sipfoundry ... My understanding of how dhcp/tftp provisioning works is as following: A phone gets tftp-server-name via dhcp, and then gets config file from the tftp-server-name tftp server. But pbx must generate the config file... First i'd like to try to configure dhcp/tftp autoprovisioning for snom phones, then i'm going to try to autoconfigure grandstream phones... I tried to follow instructions at http://wiki.pbxnsip.com/index.php/Automatic_Provisioning (created pnp.xml as in the wiki example) but alas... I still have nothing to in select accouts->registration->configuration profile... (i expected snom320 and snom360 and polycom there) And of course no config files are generated... I also noticed that i have nothing in admin settings->pnp page... just header, comment and save button. What am i doing wrong? Is there a more detailed description of dhcp/tftp autoprovisioning? Could it be a problem in 3.1 version or restriction of my demo licence? Thanks in advance, Nikolay.

Hi, all. We are evaluating pbxnsip. I'm not able to find out if it is possible to setup direct rtp between the phones on the lan. Is it possible? How to configure it? Will such setup affect some functionality? Thanks in advance, Nikolay.