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Department of Informatics - Communication Systems Group

Known issues

This page contains a list of current known issues with the overall system:

  • Due to problems with the university firewall, we are forced to have a rather unconventional setup with two gateway IPs, depending on the location of the user (inside university network/connected via VPN or outside university network).
    To handle this, we have two SRV entries for each protocol (SIP, IAX2), where only one will actually work, depending on your location. While this should work in principle, not all clients/gateways handle this correctly - some for instance will only look up one of the two, consider a failure to connect final and stop trying. In this case, the only workaround is either a change of software or the usage of the appropriate DNS entry for the correct interface, instead of (as also returned to ENUM queries) (analogous to the information under the QuickStart Guide).
    We are working on resolving this issue in cooperation with Informatikdienste, but it is not clear yet as to when it will be solved.
  • Entries for external PSTN redirections set up in the web configuration menu currently require an additional 0 in order to work.
  • Usage of the MeetMe (Conference) Phone Menu (accessed by '*') crashes Asterisk when the caller dialed in from PSTN.
    This apparently is due to a bug in either MeetMe or the ISDN channel driver. It does not happen when the menu is used over SIP or IAX2.
    This issue is scheduled to be addressed by update of both Asterisk and the channel driver in the upcoming maintenance on Mar. 14, 2008.
    (Currently, this issue should have been worked around and should not happen... please report, if it does!)
  • Calling a UZH participant who activated his Siemens Hisax (regular UZH telephone system) VoiceMail or configured a diversion crashes Astersik.
    The Siemens system signals the diversion to the ISDN card, which reports it to Asterisk. Asterisk coped with it in V.1.2, but does not (bug) in V1.4.
    Due to this circumstance, this feature has been disabled in the most recent driver, which is to be installed on Mar. 14, 2008, hopefully rendering the system stable again.
    Unfortunately, there's no intermediate workaround, so this issue will remain until the maintenance day.
    Update: unfortunately, this bug seems to be an additional issue, which will have to be resolved in cooperation with the developers, so the bug is still present. :( We're trying to solve it as fast as possible. Sorry for the inconvenience.
    Update: After the most recent driver update, asterisk does not crash anymore upon attempts to connect to the Siemens VoiceMail, but the connection itself still fails.
  • New Voice messages in the IFI VoIP VoiceMail system are now signaled and sent via Email to the participants. However, some mail provider reject these Emails, so they are not delivered yet.
    This is due to a misconfiguration of the VoIP server's MTA and will be corrected during maintenance on Mar. 14, 2008.
  • When a VoIP user calls an external number and cancels the call before the other party answers, the external line is not hung up and the receiving side keeps on ringing until it is either picked up, or until a timeout occurs.
    We're currently communicating with the developers to solve this issue
    We've solved it on the backup machine and the fix will become effective soon. (effective Mar 23, 2008)
  • The ENUM script is currently broken due to an API change.
  • Asterisk lacks an IAX2 SRV lookup implementation. Therefore, external Asterisks will be unable to resolve the IFI IAX2 ENUM entries.
    In order to facilitate IAX2 calls from the outside, we decided to change the ENUM entries to incorporate the full external address, until IAX2 SRV lookups are implemented in Asterisk.
  • Asterisk's SIP SRV lookup implementation does not consider the use of multiple SIP SRV entries for failover scenarios. Instead, it simply accepts the first received entry and does not bother testing the second ones. This breaks the concept of having SIP SRV entries for both the internal and external interface to allow ENUM lookups for connections both from inside UZH network, as well as outside UZH network.
    As ENUM lookups from inside UZH network are rather unlikely/rare, while lookups from the outside are frequent, we decided to eliminate the internal interface entry from the SIP SRV entries, until Asterisk's SIP SRV implementation is completed.

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