Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <[email protected]> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <65070> Message-ID: <[email protected]>
Firstly I would like to say many thanks for this software. My first experiences with a few other servers quickly taught me that everything seemed to want to speak it's own dialect of sip rather than properly conforming to standards, hence looking up this project and trying to teach myself how to use it. I have been messing around for a while now and have numerous observations which I suspect either are issues with the software I have found, or if I am doing it all wrong, please feel free to slap me I'm not too familiar with the protocols with submitting bug reports for open source software so I thought it may be an idea to mention these on the forum first, just in case I am wasting people's time seriously investigating lots of user errors. I have only played around with various versions of 4.4.0, and I have reproduced all of these issues consistently except the voicemail issue. Anyway, here goes: • Should there be a file in /var/sipxdata/configserver/phone/profile/docroot containing a directory to be served to the phones? (could not find any reference to this in the docs but saw it mentioned when googling) - or is there some other mechanism to serve directories to phones? ***it uses tftproot for profiles, not docroot • Discover devices - does this work with linksys/cisco/snom as I have had no joy and can't see any mention of compatible phones. *** This is no longer functional and is being removed. More phones are autoconfigured by sipx if the device supports it, no discovery is necessary. • Create a linksys spa 9xx device. Server is then unable to create/edit any ciscoplus devices until rebooted. Same works the other way around. *** consider opening a discussion with the developer of the plugin, or assume responsibility for the plugin and fix it would be appreciated. • Profiles generated by ciscoplus templates do not register without tweaking, then calls cannot be heard without more tweaking, they contain references only needed for cisco callmanager environment, and contain items that cause non fatal parsing errors on phone (using 7970 with 8-4-2 - latest f/w that works with sipx. F/w 9-2-1 will register if we use ip rather than dns srv lookup so not used, previous 9.x.x firmwares will not register. Also previously tested only 8-4-2 with 7945 with same results). *** consider opening a discussion with the developer of the plugin, or assume responsibility for the plugin and fix it would be appreciated. Also profiles could be optimised to allow g722 support and enable wideband headset/handset and to turn on display when incoming call if display turned off. Also worth changing default reregistration time as phones can fail to reregister. Also have a list of more minor simple optimisations. *** Most phones have a sdp offering that would include g722 if it is supported, how it is listed in priority by default is another thing. You can always group phones and set this by group to optimise it the way you want. Consider opening a discussion with the developer of the plugin, or assume responsibility for the plugin and fix it would be appreciated. (Before anyone mentions that these devices are a pita, I seem to have mine working quite nicely, I prefer the sound of these compared to any other voip phones I've tried, and they look impressive to end users and more importantly the people who will be paying for them). • Also incoming calls come in with the itsp's domain appended to the phone number. Trying to call these numbers back on the cisco from the call register, the server will reply back with a 100 Trying, then less than a split second later, it will then send a 403 Requires NoAccess to the phone without any attempt at contacting the itsp. Should it really be doing this? *** You can either provide information about the ITSP (this is not normal and shouldn't be considered a shortcoming of sipx. At the same time, you can easily craft a dialplan rule in sipx to strip that on outgoing calls, but you are not doing so. • Voicemails sent via email become very quiet and inaudible. Also once forward voicemail to email enabled, voicemails retrieved via phone for extensions with voicemail forwarded to email also quiet and inaudible. (only tested in one setup around 4-5 months ago and currently unable to test). *** Post back when you can test. Make sure you are on latest 4.4 release/patch. Never had the issue so won't comment. • If server ip is changed via gui this breaks the server (sipxbridge xml rpc exception) even if rebooted. *** Correct. Changing the IP address is a PITA. I typically export and build a new system. Alternately you can grep for the IP in all files and replace the IP and restart the server and hope you found everything. • Creating new/custom permissions just results in permission errors if we use these permissions, even if all permissions granted to all users. *** Provide an example... • Active calls page under diags too slow to update to have usable purpose (not talking about 30 second delay, even if manually refreshed). My current test server is a c2d 2.93 / 3gb only handling one extension which still lags a few seconds. *** This is normal behavior. CDR is always a few seconds behind, by design. • All documentation I can find says media is direct between endpoints, and if server is unplugged, active calls will continue. If I wireshark the traffic, I can see one rtp stream between the itsp and server and another between the phone and server, which makes me suspect this only applies for setups using a SBC which contradicts the information regarding calls being setup directly between endpoints and only signalling being rewritten by server. If I am correct, it may be an idea to mention this where relevant in documentation. *** It only applies for calls where the media is not anchored by sipx... Sipx anchors calls to mediaserver/aa and when sipxbridge/media relay is used. It's always been that way. SBC's and other gateways anchor media too, but if the call is between the gateway and mediaserver/aa unplugging the ETH on sipx results in breaking that too. • Also noticed I have my phone set for g722 and calls to ivr, etc use g722. If I call externally I can see the phone tries to use g722 then drops to g711 - I'm guessing to match the itsp. If the server actually is transcoding media, should it be converting format? (not sure if there is anything to gain in this particular example but if itsp is trying to use a codec incompatible with phone......) *** The phone prefers g722 in its sdp offer. None of the media server files are encoded for g722 so it doesn't apply here. Hardly any ITSP offers G722 or will let is pass through anyway. • Also I would suggest not putting a value as a default for the stun server - I remember when first experimenting with sipx I got caught out as the default stun server was dead. I've noticed this has recently been changed to one that appears to be run by ezuce that also recently died - most itsp's operate a stun server and I'm guessing for a lot of users it may make more sense to use the itsp's stun server. *** The default stun server points to a public stun server. I always use static IP so I don't get bitten by this. But everyone's results vary. Feature requests: ***You would find it easier to subscribe to the mailing list if you want to have your questions reach a wider audience. Even in your feature requests you are asking for something that might already be in the system. (i.e. sip logging, see sip trace in wiki). • If a directory is created/populated on the server, is it possible to rewrite sip requests showing the name so this is visible on handsets? (would be nice for users looking at call registers on the phones) • Detailed sip logging - I remember seeing this feature on snom 300's which made life easy, although I would suggest either this log be rotated once an hour/6/12 hours or even a day or maybe logging only if manually enabled. • Linksys SPA 9xx devices - the wiki mentions manually changing the provisioning profile rule to /spa$MA.cfg to allow the phones to be provisioned by server. If the phone is on latest firmware and we have a file on the server with the relevant name for the model of device being used with the following contents, that step can be automated. (I haven't had a linksys to play with for a good few months so I am going off memory and this may need a little tweaking) <flat-profile> <Profile_Rule>/spa$MA.cfg</Profile_Rule> </flat-profile>" • Localisation files - I eventually managed to track it down after lots of links that don't work just to find it is incompatible. Somehow I doubt you want me to record my voice, but without any documentation on how to create the files, I can't see anyone else doing so either which is a shame. (If I was still employed at my last firm for example, it is quite likely I would have requested one of the receptionists to have helped me with this). • Reboot button in gui. I've taught myself to ssh in and /sbin/reboot -n, but this could be very helpful for the numerous windows converts. -- ===================== Tony Graziano, Manager Telephone: 434.984.8430 sip: mailto:[email protected][/email][/email] Fax: 434.984.8431 Email: mailto:[email protected][/email][/email] _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
