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Henry Castillo

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Everything posted by Henry Castillo

  1. Call-id is the identifier for the call. Caller ID is a different thing. In the latest version call-id was added, Thanks! But with the record location variable is also sent which make things much easier. Henry
  2. Thanks for testing YMSL, seems to be better but not quite perfect yet. Will keep waiting for the fix build for Linux. Henry
  3. HI, Even though we set all recording options to ON, Call recording does not work for inbound calls ( not going thru Agent or Hunt group). Previous or current version 3.3.1.3177 have the same problem. Outbound calls are recorded fine. I think it has to do with the new more granular options for recording implemented in version 3. It all worked before. Please speed up the fix for that. Henry
  4. Paging is a nice add on to any PBX. Nowadays Paging is also IP based, mostly IP multicast in which you can have dozens of endpoints (local or remote) simultaneously playing the audio but a single audio stream in the network. Pbxnsip offers a great solution for that when combining it with the right IP multicast paging speakers or IP phones that support it. Snom phones do but if the customer needs overhead (louder) paging we tested good interoperability with Cyberadata: check it here We are close to have a solution that truly integrates Paging into the PBX but we are missing a feature for background music. Specifically We need an option in the paging account to continuously (or schedule based) stream the audio from soundcard’s line in/mic (like the option used in MOH) to the configured multicast IP/Port. Henry
  5. We are using SIPfire from OPC marketing, it is an affordable sophisticated solution for call center kind of setup in which the agent has a client that pops up a form to update the info of the called person. Henry
  6. Does anybody know how to enable SRTP?. It should work with Snom phones. TLS works but would like to have the media encrypted too. Henry
  7. Please add call-id variable into call recording options. Now there is no effective way to relate CDRs captured using the SOAP CDR and recordings. Henry
  8. The active calls screen shows correctly but the SNMP value is completely off. The offset is about 20 calls now since last restart about a month ago. If this is in memory then I should be prepared for bad performance and a crash sooner or later. Any fix for this?
  9. When pressing the "Record" button using a Snom phone the PBX does not email the recording anymore. The pbx does email voicemails fine so the email part is not the problem. This was working before. The traces do show the Snom phone sending the Info message with the record=on option and PBXnsip responds with 200 OK. Seems like a bug to me, please check Henry
  10. Good to see that part of this is already implemented in new versions, thanks
  11. Henry Castillo

    snmp

    The counter for the OID 1.3.6.1.4.1.25060.1.1 (calls) seems to be broken. After reboot it is fine but over time it presents a growwing off set in number of calls. Henry
  12. There seems to be a glitch in the click to dial function. When triggering a call using an URL it works perfect if the phone that will generate the call is answered, if not or the call is denied the call will show in the calls screen till system is rebooted. Might be just cosmetic problem but I don't know what happens behind scenes and if that happens multiple times would affect performance. Henry
  13. The agent group has a very nice way to merge moh and custom messages. I believe we should take advantage of that feature (already in the system). I've seen a post regarding not playing ringback but keep the queue moh+messages. I believe that is something that would be appeciated. An option to configure the Agent group to keep playing moh even when the agent is ringing is needed. But lets extend that concept to all extensions and hunt groups where user can choose to play regular ringback or moh (a dynamic moh like in the queue, not just a .wav file), it would be just great. Henry
  14. The wiki says that while a caller is in the waiting queue the pbx will play messages 1 to 9 in round-robin fashion. In many cases it is better having it playing a random message or at least start the round-robin from a random message. Many companies have same customers calling repeatedly, in that case the customers will most of the time get the same messages... boring. Henry
  15. The email we receive with a voicemail has a nice feature that is a link that will initiate a call back to the caller. It is just a click to call url (nice option available) however, this link does not work for us for the followng reason: The links contains the public Ip mapped in the "IP Routing List" field. Which is good in some cases but It should contain the domain name instead. Many routers/firewall/nat won't allow connections to its own public from the LAN.
  16. I've seen the same. In one of the updates Pbxnsip changed the directory where it looks for the files. i.e. Before it was /html/img , I believe now it is in /img I'll request pbxnsip updating that information in the wiki or put the way it was before if it was a programming glitch. Same question for the custom emails, what would be the directory to put them?
  17. The problem is that Broadvox ( and some other ITSPs) pass the number the call is going to in the TO field of the incoming INVITE. The Requested URI contains always the company's main number. This becomes a problem in a company with multiple DIDs and when some DIDs need to be routed to some extensions directly (very common scenario). Is there a way to make pbxnsip route the inbound sip trunk call based on TO field?
  18. I believe Pbxnsip started with its B2BUA model when it turned impossible to only Proxy SIP messages and hope that the rest of the world would follow all standards (also Many new nice features were then feasible with B2BUA). Following the same idea and to make life easier to all of us, Pbxnsip should make available the option to replace the “From” header for outbound calls thru the trunk. It would work with almost every ITSP and Voip gateway without having to dig into standards to find out who to blame. The new "ANI" setting seems promising..... In the meanwhile take a look at this link where I show screenshots on how I implemented it: external Caller ID in pbxnsip
  19. Yes, Linux in a quadcore server. What worried me is the "failed" part. Anyway, is there any way to avoid that shifting around procesors and introduced jitter? (of course taking advantage of multicore for a higher number of calls capacity)
  20. after service reboot the log show this : [2] 2008/06/06 18:20:50: Set processor affinity to 1 failed Does Anybody know the meaning of that?
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