http://blog.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41778 <pre>Hello, I would like to find out if there are any recommended products for backing up a SIPX instance installed on a VM Server. Is Granular Backup an option? Thank you Saad </pre> S.K.- G 2012-05-26T12:20:37 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41776 <pre> I am getting a extension not valid message if I enter a number during the AA Greeting. If iI enter a number and get the message 3 consecutive time the next greeting is a backup AA greeting that is not active. At this point entering any extension fails. I am running 4.4 on Centos 5.7. Thanks </pre> wgreen 2012-05-25T23:07:31 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41775 <pre>Just a note for those installing latest snapshot. I'm looking into it, but resulting system seems ok, I've been playing with error reporting and introduced this Benign error looks something like this: ... Enter SIP domain name [ press enter for 'hubler.us' ] : Enter SIP realm [ press enter for 'hubler.us' ] : Configuring system, this may take a few minutes... Error. Could not examine file "/var/sipxdata/cfdata/servers" in readfile. System error for stat: "No such file or directory". Could not examine file "/var/sipxdata/cfdata/servers" in readfile. System error for stat: "No such file or directory". Could not examine file "/var/sipxdata/cfdata/servers" in readfile. System error for stat: "No such file or directory". Could not examine file "/var/sipxdata/cfdata/servers" in readfile .... </pre> Douglas Hubler 2012-05-25T18:19:44 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41762 <pre>When trying to create a conference call using the flash key on an analog clearone maxattach speaker phone connected to an audiocodes mp118, it rings to the operator. Scenario: Dial first number - call established hit flash - phone immediately rings the operator The maxattach also has a conference key that does nothing when pressed. Is there a recommended setting in the AudioCodes for hookflash timing? Currently set as "Flash Hook Period 700" server 4.04 AC 6.20A Thanks, Paul Needham Telecom Engineer Pr Phone: 856-840-2620 Fax: 856-840-2621 SIP: pan&lt; at &gt;nj.sip.qad.com Email:pan&lt; at &gt;qad.com www.QAD.com * * </pre> Paul Needham 2012-05-24T18:20:39 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41758 <pre>DNS, IP tables, NTP and DHCP are among the few services that some folks configure separately on sipxecs 4.4 or older systems. Starting with the 4.6 release these services are integrated in a much tighter way. In order not to conflict with any custom configuration methods, these select services now have a "Unmanaged" setting you can set which allows you to configure the services yourselves. George and I realized that for each service, an unmanaged state can have different consequences depending on what the service does or how it's configured. So in short, there is no common specification for how unmanaged services are dealt with, so George and I urge you to test out 4.6 and see if you can still configure the systems as you once did. Don't worry, there will *always* be a way to hack want you want together in 4.6 because all the rules are now in editable text files, but the goal is to lower the level of hacking you would have to do to make system easier to setup out of the box and easier to maintain thru a system's major and minor upgrades. </pre> Douglas Hubler 2012-05-24T13:10:26 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41742 <pre>We need to implement faxing. At the moment, I am most concerned about inbound faxes. My understanding is that sipXecs can do a T.38 to e-mail conversion and mail inbound faxes to users' mailboxes. Is that correct? Does anyone have a SIP trunk provider with good T.38 support that he or she would recommend? If so, are there any peculiar config settings required to work with that provider? Thanks. </pre> Mike Pinkerton 2012-05-23T11:59:27 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41733 <pre>I need to host a sipx server and was wondering if there might be some suggestions on the list about doing that. It would be a fairly low volume server. </pre> mike< at >grounded.net 2012-05-22T21:22:20 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41730 <pre>Customer has sipx 4.4 and is experiencing infrequent transfer failures. Scenario: External call A in to SNOM phone B. SNOM phone B consults external phone C C answers and B completes transfer. About half the time the call disconnects after approx 30 seconds. From the attached trace we can clearly see that an ACK is missing back to the SIP trunk in the failing case. But looking in sipxbridge.log I cannot see any obvious reason for this. I have a felling it is a timing issue of the re-invite, but I cannot really see the concrete problem. Regards, Sven *Sven Evensen, Operations Consultant* *OnRelay* Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │ mailto:sven.evensen&lt; at &gt;onrelay.com &lt;sven.evensen&lt; at &gt;onrelay.com&gt; │ www.onrelay.com This electronic message transmission contains information from OnRelay, Ltd., that may be confidential or privileged. The information is intended solely for the recipient and use by any other party is not authorised. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information or any attachment, is prohibited. If you have received this electronic transmission in error, please notify us immediately by electronic mail (info&lt; at &gt;onrelay.com) and delete this message, along with any attachments, from your computer. Registered in England No 04006093 ¦ Registered Office 1st Floor, 236 Gray's Inn Road, London WC1X 8HB </pre> Sven Evensen 2012-05-22T12:39:53 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41725 <pre>10 AM EDT - 11 AM EDT Topics/Questions - Developers and users, please add to list as you see fit. http://www.google.com/moderator/#15/e=1fbcb9&amp;t=1fbcb9.41 You need me to add you to my circle me to hangout, but not to view live stream. Circle me and I'll circle you back. +Douglas Hubler https://plus.google.com/u/0/109516887366518833202 Setup - You'll have to install gtalk plugin in your web browser, highly recommend chrome browser, but i think firefox works too. Note: we're thinking of running this hangout the last non-friday of each month. Douglas </pre> Douglas Hubler 2012-05-21T15:14:37 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41718 <pre>Hi, I've got the following situation, mabye someone can shine a light on this: - SipXecs pabx - VOP (Voice Operator Panel) - Yealink phones - Bria for Windows Presence on all devices is enabled based on RLS. VOP shows status of Yealink and Bria, no issues here. Yealink shows status of VOP and Bria, no issues here. On Bria though, no presense status is shown of any devices. Bria has been provisioned by SipXecs and this results in working XMPP status. SIP presence however is not working. Does anyone have experience with Bria and RLS? How should this be configured for Bria to show presence of other devices? Kind Regards, Met vriendelijke groet, Elwin Formsma Telecats BV ----- Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 44 | Fax: 053 488 99 10 | E-mail: e.formsma&lt; at &gt;telecats.nl&lt;mailto:e.formsma&lt; at &gt;telecats.nl&gt; | </pre> Elwin Formsma 2012-05-21T11:28:10 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41684 <pre>I am having problems getting the XCAPI product from TE-Systems to register on the sipXecs server. This is a FoIP software product for the GFI FAXmaker server. Does anyone use this product and if so has anyone been able to get the SIP setup to work with sipXecs? Thanks everyone, Robert Schroeder NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain confidential and/or privileged information. Any unauthorized review, use, print, save, copy, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. </pre> Robert Schroeder 2012-05-17T18:09:04 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41679 <pre>We have an Ezuce customer that uses another vendor for network support. We are trying to set them up with some demo trunks off our account at voip.ms. The customer has a SonicWall firewall installed, model TZ100. The IT vendor seems confused as to the 1 to 1 port request for NATing, and at this point, is the only thing standing in the way of the test as the trunk registers(albeit on port 16990 currently). I have googled the 'sipx sonicwall config' and get some bits and pieces of other peoples work like 'disable SIP ALG' etc. Does anyone have a definitive set of configuration settings for this firewall I can pass on to the IT vendor? Thanks, Mark W. Wood office: (760)202-0224 X2010 Direct: (760)459-1981 [cid:image001.png&lt; at &gt;01CD3408.4D38C040] www.redphonetech.com </pre> Mark Wood 2012-05-17T15:46:39 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41669 <pre>Hi All, If I set speed dial and with subscribe in following order in 4.4 build(Admin user page) Speed dial(without subscribe to presence)--user 200 Speed dial with subscribe to presence--Park 777 Speed dial with subscribe to presence--user 202 or Speed dial with subscribe to presence--Park 777 Speed dial(without subscribe to presence)--user 200 Speed dial with subscribe to presence--user 202 But in polycom phone it always shows the speed dial with subscribe to presence as first line after user-id line and speed dial at last..So its not reflecting as per admin or user selected.... Speed dial with subscribe to presence--Park 777 Speed dial with subscribe to presence--user 202 Speed dial(without subscribe to presence)--user 200 Any suggestions? Regards, Kumaran T </pre> Kumaran 2012-05-17T09:17:30 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41665 <pre>Is there a way to restart just this particular service on a scheduled basis, or do I need to restart the entire SipX stack? </pre> Robert B 2012-05-16T16:25:10 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41662 <pre>Hello, as far as I understand &lt;http://wiki.sipfoundry.org/pages/viewpage.action?pageId=6520862&gt; my IM state should follow to the one I set with the softkey "My State" on the phone. Regardless what I set on the phone (PolyCom Soundpoint IP 650 with firmware 3.2.6.0314) it is always "available" and not even the extended information changes. It seems that this even won't work the other way. If I set "do not disturb" on my IM client (pidgin 2.10.4) a call isn't forward to the mailbox (the IM settings are enabled for this to work). Is anyone using sipXecs + IM with success? See my previous msg about the IM state with "On the phone". I think something is seriously wrong with my setup or something is broken inside sipxecs. </pre> Claas Hilbrecht 2012-05-16T12:35:56 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41658 <pre>Hello there. It seems to me that currently any user can intercept any call with simple INVITE, replacing to and from fields. You can test it with VOP or any other capable phone. Since I'm not currently a guru in sipXproxy internals (I'm not even sure if it's fully stateful) I have a question - is it possible (theoretically, without re-writing whole proxy) to add a Permission there? It would be nice to have groups which users were able to intercept users from other groups (like with Intercom). Current behavior already looks bad for some of our extra-paranoid clients. </pre> Михаил Родионов 2012-05-16T09:09:05 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41650 <pre>Hello Experts, I wonder if there's a way to set the IM status to busy or away if someone is on the phone. Today I configured the IM feature in sipXecs and start playing with that. I can see that a phone is ringging or that someone is one the phone with pidgin in the extended status information. But I think it would be much easier to see that someone is on the phone if the status "busy" or something else is used. I tried to changed the message "On the phone" to something else and restarted the phone and IM server but the message remains "On the phone". What system needs to be restarted to get the setting change active? </pre> Claas Hilbrecht 2012-05-15T15:16:27 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41645 <pre>All: I'm new here -- so apologies if I breach protocol. I am trying to run sipxecs on a standard install of Fedora 16. I tried to install the latest stable, which apparently is a build for Fedora 14, which is past its end of life. That set of packages has dependency issues on Fedora 16. Not finding a stable set of Fedora 16 packages, I installed a Fedora 16 snapshot. I was having a bit of difficulty provisioning phones, but was trying to work through those issues when a routine Fedora yum update included the sipxecs repo, which I had inadvertantly not disabled. That set of packages, which included some nightly builds, immediately killed the web configuration interface -- any attempt to load it resulted in a 503 - server down for maintenance issues. Log files said something to the effect of something killing the worker process -- I'm at a different location today and don't have the exact log message. My questions: 1. What is the best way of alerting the developers to this regression in the latest snapshot? 2. What is the best way to get a stable set of packages onto a stock Fedora 16 box? Thanks. </pre> Mike Pinkerton 2012-05-15T12:59:59 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41642 <pre>In sipx 4.4, we have a customer (with their own SIP trunk) where a user has set forwarding same time to a parallel hunt group with 4 members. I would expect to see one 180 from the called user and one 180 from the hunt group. Instead we are seeing a burst of a number of 180s, up to ten or more. This is causing the SIP trunk to cancel the call. And the odd thing is that all the 180s are identical. Any ideas? Is the hunt group supposed to give one single 180, or will 180 come from every single member of HG? In the capture 10.100.10.106 is sipx and 193.246.242.135 is SBC with CS2K behind. Sven </pre> Sven Evensen 2012-05-14T16:33:43 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41641 <pre>I'm running into some troubles with dialplan and schedules. I was trying to define some rewrite rules , so that I can forwards an extesion (like 1000) to 2030 in some hours or to 100 in other. So i define a rule with a specific schedule and then a rule scheduled always with lower priority. Is this supposed to work? thank's Domenico Chierico </pre> Domenico Chierico 2012-05-14T16:32:38 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/41633 <pre>Hello, after searching for something to solve our BLF problem I found that polycom offers the "Polycom® Productivity Suite" for anything but the qualitiy monitoring for free. I tried to add the license as a custom configuration option but the license is not properly installed to the phone. How can I add the free polycom site license to sipXecs? http://www.polycom.com/products/voice/applications/index.html </pre> Claas Hilbrecht 2012-05-14T14:05:44 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.sipfoundry.general