gmane.comp.telephony.pbx.asterisk.devel

IMAP_STORAGE issue with app_voicemail

Brendan Martens — 2008-10-06T21:33:38

Hello, I have scoured the internet and asked a few times in IRC but I can't figure this out. I'm hoping someone here can shed some light on this seemingly odd behavior. I have asterisk 1.6.0 compiled with the IMAP_STORAGE set for voicemail, that all seemed to compile up just fine. The error I am getting in * cli is this: [Oct 6 17:28:43] ERROR[20802]: app_voicemail.c:1999 mm_log: IMAP Error: Login aborted [Oct 6 17:28:44] ERROR[20802]: app_voicemail.c:1757 init_mailstream: Can't connect to imap server {mail.crosscomm.net:143/imap/notls/user=vmail< at >crosscomm.net }INBOX [Oct 6 17:28:44] ERROR[20802]: app_voicemail.c:1517 messagecount: Houston we have a problem - IMAP mailstream is NULL The output I am seeing on the imap server it's trying to connect to is this: Oct 6 17:29:49 gabriel imapd: LOGIN: DEBUG: ip=[::ffff: 64.105.202.244], command=LOGOUT Oct 6 17:29:49 gabriel imapd: LOGOUT, ip=[::ffff:64.105.202.244] Here is my voicemail.conf, I removed the commented items for readability here: [general] imapserver=mail.crosscomm.net imapflags=notls [imap-voicemail] brendan => brendan,System's Mailbox,,,imapuser=vmail< at >crosscomm.net| imapsecret=password I'm quite baffled at this point and have no idea how to troubleshoot this. The very odd thing to me is that the imap server seems to be receiving a LOGOUT command first? Thanks for any help you guys may be able to offer figuring this out. Brendan Martens _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

How to keep the connection with AMI interface active?

caif~ — 2008-10-06T08:48:42

_______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Asterisk 1.6.1 IAX oddities

David A. Bandel — 2008-10-06T03:50:33

Folks, Installed 1.6.1 for testing. Have noticed 2 issues: 1. Have a IAX/IAX connection between 2 * servers (1.4.22 and 1.6.1). Turn on dnsmgr. Start *, after 300 seconds, iax2 show peers goes from reachable (using port 4569) to unreachable and the port has changed to something else (like 5017). Turn off dnsmgr, restart * and all is well forever. dns srv record looks OK: _iax._udp IN SRV 20 0 4569 pbx.pananix.com. The 1.4.22 system has dnsmgr on and does not show this problem. 2. Peer connection from 1.6.1 to 1.4.22 does not show (T) trunk. From 1.4.22 to 1.6.1 it does. Each has an identical (but opposite) peer/user stanzas with trunk=yes. (1.6.1): office/office 190.33.255.69 (S) 255.255.255.255 4569 OK (40 ms) (1.4.22): bagala/bagala 172.16.12.101 (S) 255.255.255.255 4569 (T) OK (21 ms) Ciao, David A. Bandel

FastAGI server

Gnu Devel — 2008-10-05T21:56:43

Are there any open source FastAGI server writing in C or C++ ? Thanks in advanced. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Asterisk on Windows (Subsystem for UNIX)

Alex Dubinsky — 2008-10-05T20:44:11

where are aliases processed?

Tzafrir Cohen — 2008-10-05T09:17:08

I was testing something with both asterisk 1.4 and trunk. I had 1.4 running, and then installed trunk over it. I tried to tell it to stop: $ asterisk -rx 'stop now' No such command 'cli quit after stop now' (type 'help cli quit' for other possible commands) So aparantly the client-side asterisk was smart enough to know that "stop now" is deprecated (or not supported anymore?) and translated it to something that is more civilized. Sadly the version on the server side is not the same. Which is a nice way to ruin an upgrade.

Asterisk 1.6.1-beta1 Now Available

Asterisk Development Team — 2008-10-03T22:07:23

The Asterisk.org development team has released Asterisk 1.6.1-beta1. While Asterisk 1.6.0 was in the beta and release candidate stage, the development team was also working on merging new things for Asterisk 1.6.1. Now that 1.6.0 has been released, the testing cycle for Asterisk 1.6.1 has begun. To see the list of new features in Asterisk 1.6.1, see the CHANGES file: http://svn.digium.com/view/asterisk/branches/1.6.1/CHANGES?view=markup For a full list of changes that have gone into the development of Asterisk 1.6.1 so far, see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.6.1-beta1/ChangeLog?view=markup To download Asterisk 1.6.1-beta1, visit the Digium downloads site: http://downloads.digium.com/pub/telephony/asterisk/ Please report all issues to the issue tracker, http://bugs.digium.com/. Thank you for your continued support of Asterisk! _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Asterisk 1.4.22 and 1.6.0 Released

Asterisk Development Team — 2008-10-02T18:56:17

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. ================================================================= === Asterisk 1.4.22 ============================================= ================================================================= Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22 also includes support for DAHDI. For more information about the transition from Zaptel to DAHDI, please see the following help file: http://svn.digium.com/view/asterisk/tags/1.4.22/Zaptel-to-DAHDI.txt?view=markup For a full listing of changes in this release, see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.4.22/ChangeLog?view=markup Asterisk 1.4.22 is available for immediate download from the Digium downloads site: http://downloads.digium.com/pub/telephony/asterisk/asterisk-1.4.22.tar.gz ================================================================= ================================================================= ================================================================= === Asterisk 1.6.0 ============================================== ================================================================= Asterisk 1.6.0 is the first official release of Asterisk 1.6. ----------------------------------------------------------------- --- Upgrade Information ----------------------------------------- ----------------------------------------------------------------- Asterisk 1.6 no longer supports Zaptel. It only contains support for DAHDI. For more information on this transition, please see the following help file: http://svn.digium.com/view/asterisk/tags/1.6.0/Zaptel-to-DAHDI.txt?view=markup There are a number of other important changes to be aware of when upgrading to Asterisk 1.6.0 from previous versions of Asterisk. For a listing of those things, please see UPGRADE.txt: http://svn.digium.com/view/asterisk/tags/1.6.0/UPGRADE.txt?view=markup ----------------------------------------------------------------- --- New Features ------------------------------------------------ ----------------------------------------------------------------- Asterisk 1.6.0 contains new features that were not previously available in an Asterisk release. For a full listing of the features that are included in Asterisk 1.6.0, please see the CHANGES file: http://svn.digium.com/view/asterisk/tags/1.6.0/CHANGES?view=markup A verbose listing of each individual change that was made in the development of Asterisk 1.6.0 is also available: http://svn.digium.com/view/asterisk/tags/1.6.0/ChangeLog?view=markup ----------------------------------------------------------------- --- Release Management ------------------------------------------ ----------------------------------------------------------------- The Asterisk.org development team has decided on a new release management style for Asterisk 1.6. Previously, a release series was strictly feature frozen for its entire lifetime. The release management guidelines for Asterisk 1.6 were inspired by the Linux Kernel, among a number of other projects. Asterisk 1.6 is not feature frozen. Features will be added in point releases. However, effort will be made to ensure that the number of large changes is minimized in a single point release. Even as 1.6.0 is being released, the Asterisk development team is already working on what new things will be in 1.6.1 and beyond. To take a look ahead to see what features have been added for releases that have not yet been made, take a look at the trunk version of the CHANGES file: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup Even though new features will be added in point releases of Asterisk 1.6, that does not mean that any deprecated functionality will be removed as has been done between major releases in the past. In fact, we have decided that maintaining backwards compatibility is of the utmost importance for configuration and external interfaces. C API and ABI compatibility is not guaranteed between point releases. However, things like dialplan applications, functions, and AGI commands will not disappear just because there is a new and better way to accomplish the same thing. With Asterisk 1.4, once Asterisk 1.4.N is released, Asterisk 1.4.X is no longer supported, where X < N. With Asterisk 1.6, the development team plans to maintain a total of 3 releases at a time. For example, the development team will support Asterisk 1.6.0, 1.6.1, and 1.6.2 until 1.6.3 is released. This means that for the time that 1.6.0 is supported, there may be 1.6.0.1, 1.6.0.2, etc. releases that include fixes for regressions found in 1.6.0. With Asterisk 1.4, the goal has been to make releases every 4 to 6 weeks. With Asterisk 1.6, we aim to release updates in a similar time frame, but it is likely that it will be closer to 6 to 8 weeks for Asterisk 1.6 due to a more strict beta and release candidate testing cycle for each point release. The number of releases supported and release time frames are not yet set in stone. The development team is interested in adjusting these policies to help meet the desires of the Asterisk community. ----------------------------------------------------------------- --- Download ---------------------------------------------------- ----------------------------------------------------------------- Asterisk 1.6.0 is available for immediate download from the Digium downloads site: http://downloads.digium.com/pub/telephony/asterisk/asterisk-1.6.0.tar.gz ----------------------------------------------------------------- --- Reporting Issues -------------------------------------------- ----------------------------------------------------------------- The bug tracking process is one of the best places to get started in the Asterisk development community. Please report all issues to http://bugs.digium.com/. Live discussion about current issues is done on the Freenode IRC network, in the #asterisk-bugs or #asterisk-dev channels. ================================================================= ================================================================= Thank you very much for your continued support of the Asterisk project! _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Pickup Bug in Ver. 1.6, 1.4 and also 1.2

Stefan Schmidt — 2008-10-02T18:29:02

hello, i have found the following problem with the pickup command. i´ve tested this with ver. 1.2.29 (with and without multiple pickup patch), 1.4.21.2 and also 1.6.0rc6. the problem is, after making an unsuccessfull pickup the call hangup and the further lines in the exten are ignored. for example: exten => *8,1,Noop(trying pickup) exten => *8,n,PickUp(123< at >somecontext) exten => *8,n,Noop(pickup failed) in version 1.2 and 1.4 comes the message that pickup has failed and the noop isn´t executed. in version 1.6 with this example the result is the same but when i add the following: exten => *8,1,Noop(trying pickup) exten => *8,n,PickUp(123< at >somecontext) exten => *8,n,PickUp(123< at >somecontext) exten => *8,n,Noop(pickup failed) both pickups are executed and the noop will be executed too when the pickup fails. i´ve found this problem by building a global pickup function in a hosted pbx system with 2000 sip users and around 3000 different extensions grouped under different DIDs in a realtime enviroment. I wanted to build, a global pickup function to pickup a ringing line without knowing which extension exactly rings. I tried this by making a Pickup in a while loop trying for every possible extension for this DID. Call- and pickupgroups wont solve my problem cause of the limitation of 63 groups. Maybe there is a wrong return code of the pickup so the call is terminated. best regards steve smith _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

dahdi-linux 2.0.0 and dahdi-tools 2.0.0 released

Asterisk Development Team — 2008-10-02T18:07:52

The Asterisk development team is pleased to announce the first offical release of the Digium Asterisk Hardware Device Interface (DAHDI). The list of packages released today includes: dahdi-linux 2.0.0 dahdi-tools 2.0.0 dahdi-linux-complete 2.0.0+2.0.0 Both dahdi-linux and dahdi-tools are required to enable DAHDI support in your system. You will need to install dahdi-linux first, then dahdi-tools, and finally you can configure and make Asterisk. dahdi-linux-complete is both dahdi-linux and dahdi-tools combined into one download as a convenience. You still need libpri for PRI support with Asterisk if you are using DAHDI. DAHDI is supported by Asterisk 1.4.22 and Asterisk 1.6.0. More detailed information about each of the packages is below. ================== dadhi-linux-2.0.0 ============================== This is the first release of the DAHDI Linux kernel modules package, which replaces the kernel modules from Zaptel. The primary purpose of this release is to rename the package from Zaptel but in addition to several bug fixes some of the new features are: * Echo cancelers can now be applied per channel and selected at configuration time. * Channel memory allocation changed from one large block into smaller blocks in order to reduce out of memory errors on a system that has been running for some time. * Layout changes to support binary packaging. * Neon MWI support added to the wctdm24xxp driver. * Dropped support for Linux Kernel 2.4 as well as the torisa and wcusb drivers. For information on upgrading from Zaptel to this release, please see: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt Known Issues * Reference counting is not currently done on echo canceler modules, and therefore it is possible for an administrator to unload an echo canceler module that is in use which could result in a crash. It is recommended to use /etc/init.d/dahdi start|stop to load and unload your drivers to eliminate exposure to this issue. http://bugs.digium.com/view.php?id=13504 * Cannot compile with CONFIG_DAHDI_NET or use DAHDI for data connections. http://bugs.digium.com/view.php?id=13542 =================== dahdi-tools-2.0.0 ============================== This is the first release of the DAHDI userspace tools package, which replaces the userspace components of Zaptel. The primary purpose is to rename components from Zaptel to DAHDI and support binary packaging. The names and layouts of the configuration files have also changed. Please see UPGRADE.txt for more information. http://svn.digium.com/svn/dahdi/tools/tags/2.0.0/UPGRADE.txt =================== dahdi-linux-complete-2.0.0+2.0.0 =============== This release combines dahdi-linux and dahdi-tools into a single download, one-package installation process. Users who are installing DAHDI for the first time don't have to download and install the dahdi-linux and dahdi-tools packages separately. _______________________________________________ swdev mailing list swdev< at >lists.digium.internal http://lists.digium.internal/mailman/listinfo/swdev _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

AOC Information

Jay R. Worthington — 2008-10-01T18:17:25

Virtual Modem Pool

Brad Silen — 2008-10-01T15:20:24

Thank you to (almost) everyone for your many thoughtful and helpful responses. I agree the VOIP pathway does not make sense. We are expecting high volumes and I wanted to make sure to examine even those alternatives which were a bit outside the box. We are examining leased services like Level3, Quest,... And, we will compare the economics against setting up our own dial-in pool based on legacy hardware now flooding the market. While it does not make sense to use the VOIP pathway for high volume of data if someone finds this thread needing to move data this article gives very specific technical detail related to faxing: http://www.voip-info.org/wiki-Asterisk+fax _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

DeadAGI warning on Asterisk 1.4

Giorgio Incantalupo — 2008-10-01T08:41:46

Hi, we are making some tests trying to port our DeadAGI-based sw from Asterisk 1.2 to Asterisk 1.4 but we get a lot of warnings about DeadAgi: res_agi.c: Running DeadAGI on a live channel will cause problems, please use AGI What does it mean? Which kind of problems? Thank you. Giorgio Incantalupo. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Virtual Modem Pool

Brad Silen — 2008-09-30T15:30:39

We are looking to deploy thousands of hardware devices connected to the PSTN which will upload data and download firmware updates using v.90 modems. It will be deployed to a demographic which does not have Internet access. We are hoping to avoid setting up an old fashion modem pool, POTs or T1-PRI, and hope to access the PSTN through a SIP Trunk or IAX2. This solution would be both cost effective and scale to handle peak loads; For example, when a firmware download is required. Ideally we would like our application servers to send/receive using TCP/IP sockets with the virtual modems which are being driven by the VOIP infrastructure. The network might look like: Device <--> POTs <--> VOIP Gateway <--> IAX2 <--> ??? <--> Clear Text on TCP/IP Socket Solve for ??? Has anyone used Asterisk in this way? Is there any reason why the VOIP Gateway (SIP Trunk or IAX2) data path would prevent modem communication? Is there any similar solution terminating a v.34 connection (aka Fax)? A Fax solution would verify the ability to send data via the VOIP pathway and offer sample code as a starting point. Would we extend the Asterisk concept of an "extension"? For example, instead of forwarding the traffic to a SIP Phone the virtual modem would be an "extension" which converts the data stream to ASCII clear text. Or, what would be the suggested architecture choice in Asterisk? Note, I am very open to better, easier, or more clever solutions. If there are service providers offering this type of virtual modem pool please have your shameless commerce division email me directly. I suggest they not respond to the list since I am concerned it would violate the rules of this list. I have not been able to find a solution and expect to contribute. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Queue and presence Asterisk integration withOpenFire XMPP server

2008-09-26T15:41:40

Hi all, here is my question regarding making asterisk a generic contact center having both voip and IM contacts. Let's suppose i want the operator to receive a call or a chat but not together in the same time. I would like to integrate for example the OpenFire XMPP server capabilities with the asterisk ACD feature. In particular i would like to know if is possible to have for Asterisk a similar plug-in like the Asterisk-IM already available on OpenFire. In this case i need to communicate from asterisk->XMPP server the presence information (that i can do with asterisk-IM), and from XMPP -> asterisk the chat presence information (so that operators result busy when in a chat). Is it possible to have APIs to set into asterisk the chat user presence coming from an external XMPP server? Thank you Donato F. ---------------------------------------------------------------------------- Vuoi essere presente online? Vuoi dare voce alla tua attivita`? Acquista un dominio su domini.interfree.it. A partire da 18,59 euro ---------------------------------------------------------------------------- _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

T38 fax gateway announcement

Daniel Ferenci — 2008-09-26T09:20:49

h323 channel and tcp/udp port ranges

Alex Samorukov — 2008-09-25T23:43:02

Hi. While customizing firewall rules for my asterisk box I found that it always uses ports range from 1024 to 65535 for the H.245 channels. While looking into openh323 sources I found that there is a way to specify tcp port range using SetTCPPorts functions. I tried to put it inside MyProcess::Main(ast_h323.cxx) and this works correctly. Now i want to create patch to allow user change this values inside h323.conf I think that smth. like h245tcpstart and h245tcpend should be a good options for this. But also I found rtp.conf with rtpstart/rtpend settings for rtp, and now i`m not sure, may be its better to place this settings here? This is the reason why I decided to ask in maillist before creating the patch. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

upstart script for asterisk

Tzafrir Cohen — 2008-09-25T04:17:26

Aperantly Upstart has been included in Ubuntu for quite some time and now in latest Fedora. http://www.netsplit.com/2008/04/12/upstart-05-job-lifecycle/ reads "Since Upstart forks and supervises its own processes, it generally prefers that daemons do not fork() and remain as the pid they were given when started." Anybody from those two distributions working on an Upstrart script for Asterisk?

asterisk 1.6-rc6 zaptel/dahdi mismatch?

Klaus Darilion — 2008-09-24T13:49:04

Hi! I just wanted to install 1.6-rc6 with zaptel 1.4.12.1 as recommended on http://www.asterisk.org/downloads: > Version 1.6 > > Asterisk 1.6.0-rc6 > Zaptel 1.4.12.1 > Libpri 1.4.7 > Addons 1.6.0-rc1 but that does not work - in rc6 there is chan_dahdi which I guess requires dahdi! I guess somebody should fix the website replace zaptel with dahdi for 1.6 release. Further I could not find any link for downloading dahdi - there is no dahdi directory in http://downloads.digium.com/pub/ regards klaus _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

GUI Development Framework & Asterisk...

Darren Schreiber — 2008-09-23T17:11:23

dahdi dummy

Mark Spowage — 2008-09-23T16:07:14

Has a way been opened to use the 2.4.20 kernel without USB via some RTC code ? if so then can the dahdi dummy code be used to fix the ztdummy code ? why reinvent a fix ? surely the ixp425 under 2.4.20 has many timer resources to satisfy the needs of zaptel ? without usb simply cherry pick some code from dahdi to zaptel ? to smarten up ztdummy thx !