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Everything posted by cwernstedt

  1. How could that happen? Bug? Server wasn’t touched except for a reboot.
  2. After reboot, the call history shows all calls as having been made on 12/31/1969 7:00:00 PM . Version is 63.0.1 (Debian64) . Any idea of what happened here and how it can be prevented in the future? (Fortunately this time, the log isn't needed.)
  3. DNSmadeEasy provides a user name, and two keys: API Key and Secret Key . From these three, what should go into the two fields provided on the pbx (user name and password) ? [Solved on my own: Should be API Key and Secret Key for user name and password ]
  4. To clarify, this is the situation, and maybe there is a misunderstanding: Twilio utilizes, for example, the below IP-addresses and with network for Media being put into use on Monday next week. Is it perhaps the case that only the signalling IP-addresses need to be configured in the Vodia trunk settings, and that media will flow regardless of if the originating Media IP-addresses match trunk settings? If so, Twilio users should be fine...If not, there's a lot of trouble coming. Signalling IPs: which translates to: Ports: 5060 (UDP/TCP), 5061 (TLS) Media IPs: Port Range: 10,000 to 20,000 (UDP)
  5. I have this problem now with Twilio as they just added a class C network , and things will stop working next week unless I add it. (Trunk doesn't get properly identified on incoming calls without the IP-addresses specified.)
  6. Ok. Where do I configure these things in this version? Ideally I would like to prefix +423 to all numbers lacking a 00 prefix (local numbers) and replace 00 prefixes with + for international numbers. So 2371336 would turn into +423 2371336 and 0013343456666 would turn into +1 3343456666 . (Incoming calls only)
  7. Hi, Well, it's the inbound (not dialed) that I'm looking to change. In other words, the number that is presented on calees' phones and in logs etc. The trunk provider presents local numbers without the correct international prefix, which makes it difficult to process these calls correctly.
  8. One of our SIP-trunk providers isn't particularly good about how they present the CIDs on inbound calls. Is there a way to rewrite these according to some rules, for example replacing 001 with +1 , or adding a country code to local numbers. Note for clarity: I'm talking about how numbers are presented in the From field in CDRs, logs, etc.
  9. Can you take a look at the packet captures zipped and attached to the earlier message? ( )
  10. Initiate call back after hangup is "No" for the called extension in the tests and the calls are not made from a cellphone or POTS-phone, but from a registered extension.
  11. I assume it's the mailbox, since there is no agent group configured, but mailbox doesn't explicitly offer callback as an option, and "offer camp on" is disabled. The sequence of events from the point of view of the caller seems to be like this: 1) caller places a call to the extension 2) Ring tones are generated . 3) "Something" (probably voicemail) at the PBX picks up and it immediately (before even playing a greeting) for unknown reasons adds the caller to the callback list. 4) The caller hangs up. From the callee's point of view, the phone rings, but when picking up, it's the PBX offering various options. Not sure what it is. (See the file rec_115_13.wav in the attached ZIP. I'm also attaching PCAP files captured on the PBX.) When this happens there is always two calls registered in the log, starting at the same time. A big question is if the callback list option can somehow be triggered before the mailbox has kicked in . It almost appears to be the case, because the mailbox really shouldn't have come into play here. (The callee account had a live registration generated by Counterpath, and the callee picked up well before the call should have landed in voicemail.) weird call with stretto.zip
  12. Looks like I'm closer to pinning down what's going on... Seems like my test call generator, somehow, sometimes, manages to make some noise that is misinterpreted as the starcode that gets the call added to the callback list. I wonder, what are the conditions for this feature to be activated by a caller? Can it, for example, happen only when the caller reaches the mailbox, or is there some other situation when it can happen, such as when ringtones are being generated before the callee or voice mail has picked up? Can the feature be turned off?
  13. The setup is now the standard one where the client and the registration server alternate registrations, but potentially with some brief overlaps (option 0 below). Not sure this is what is causing problems since Vodia supports multiple registrations for the same account, and since most of the time, the handover appears to work fine. In any case, seems like you are suggesting that 2 would be more stable? /CW 0: Standard: Your VoIP service provider supports multiple registrations. In this mode, there may be short overlaps of registration where both the Push server and the Bria client are registered to the SIP server. 1: Single Device Emulation: Your VoIP service provider does not support multiple registrations. Bria and the Bria Push server must unregister before the other one can register. 2: Continuous: Your VoIP service provider support multiple registrations. The Bria Push server is always connected to the SIP server. Bria is only connected to the SIP server when Bria is in the foreground. When Bria is in the foreground, Bria ignores push notifications so you do not receive duplicates. 3: Single Device Takeover: Your VoIP service provider does not support multiple registrations. Bria and the Bria Push server take over registrations from each other without unregistering first.
  14. Thanks! Upgraded with no issues, and could restore PRACK after that. A remaining problem with Counterpath and its push/handover mechanism is that some calls behave very strange. Every now and then this happens: The caller lands in voice mail, but the Bria Stretto user will hear the phone ring. If picking up, the callee will hear nothing OR will hear what seems to be the Virtual Private Assistant. The PBX registers these calls as two calls in the call history. I wonder what can be done to troubleshoot this. No trunks are involved. This is just between registerred SIP-clients and the PBX.
  15. Ok. Excellent. I have hesitated to upgrade because I don't want to break anything that is working, but maybe 63 is a safe one? Rolling back seems difficult due to the password encryption.
  16. FYI everyone: PRACK needed to be turned off for CounterPath's push solution to work with v62.0.1. (Not sure if this will have some unforeseen consequences as I have only tested successfully twice.)
  17. Regarding CounterPath as an alternative for iOS clients, I wonder if the Voida PBX supports CounterPath's handover model for receiving calls when the client is not in the foreground. The way it is supposed to work is described here: https://www.counterpath.com/assets/docs/guides/mob/mob_Retail_Ios_5.5.3/UG/Default.htm#clients/UserGuides/Mobile/Reference/mobBriaPushService.htm . However, with Vodia PBX v62.0.1 I can't get this to work "out of the box" at lest. The PBX sends caller to voice mail if no other extensions are registered, maybe due to some unexpected behavior of the Bria Push Server. Would be nice if this could work.
  18. Why isn't the below working? from-trunk:from trunk-ani (If coming from trunk, use from; otherwise use trunk-ani if configured.)
  19. Hi, Your suggested string (the below) causes the ANI from externally originating calls to now show (instead some other ANI is shown, extension or trunk ) from-cell:ext-ani from-cell:ext-ani from-cell:dom-ani from-trunk:from acd-ani ext-ani trunk-prefix dom-ani trunk-ani So from our point of view the same as having nothing in the Regular ANI rule field. This string, on the other hand, causes the same bad behavior as before (internal extension numbers are presented as ANI to the PSTN) : from-trunk:from acd-ani ext-ani trunk-prefix dom-ani trunk-ani Note that this is the same as your rule, but with the from-cell clauses removed. Why from-cell clauses in the rule field changes behavior in this way at all is a mystery as none of the test calls originate from a user’s cell phone. They originate from either an internal extension or from some random external number. It looks like something is seriously wrong with the code here with regard to the parsing of the ANI rule field. Note that what I’m trying to do worked perfectly before and this isn’t a weird scenario. (One would certainly want to pass on the original ANI if the trunk provider is OK with it.) Cheers,
  20. When upgrading from 60 to 62.0, ANI behavior stopped being good, I'm not sure how to make it work. Desired behavior (worked with version 60): 1) If the call originates from trunk, then the original caller ANI should be used when calling out to an extension's cell phone . 2) if the call originates from internal extension then the extension's (if available) or the trunk's ANI should be used instead. I'm trying to make it work with v62, but whereas the ANIs of calls originating from a trunk are presented correctly, calls from internal extensions are presented incorrectly (the internal account number is presented instead of the extension's outbound number ANI or the trunk ANI . (So all calls from say, internal extension 999 would be presented as 999 instead of say +46314343455.) On the trunk I'm trying this setting which seems logical to me: ani_regular: from-trunk:from trunk-ani I needed to put in "from-trunk:from" to make criteria #1 above work, but this also results in the bad behavior for calls from internal extensions. How can I replicate the behavior of v60 so that things work again?
  21. Is there a way to get automatic blacklisting to block all attempts from an offender IP address instead of blocking IP+originating port as separate instances? As it is now it appears that, say, and are blocked as separate entities, so if tries with thousands of other additional originating ports, the black list will be filled with thousands of entries. (Plus my email address being hammered with thousands of notifications...)
  22. Is there a way to periodically and automatically change the participant access code (PIN) of a conference using a script that accesses the API or manipulates settings files? It's an "ad-hoc" conference that users regularly access. I have studied the API documentation, but I can't find the name of the attribute to change, and it's unclear if a new access code can be submitted via the API at all. /CW
  23. OK. I think to move things forward a bit faster than trial and error, I should pay for some consulting from someone skilled with setting up auto-provisioning with Snom and Vodia. (Should be fairly standard scenario.) To make this happen, do you recommend that I purchase some premium support tickets?
  24. I don't PnP the phone yet. It's not in the same LAN as the PBX so not sure how to do it properly at this point, but I'd like to solve this when the installation grows. Is PnP necessary for as-feature-event subscription?
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