Re: [asterisk-users] Intro to DECT vs IP
On Fri, Jun 29, 2012 at 10:42 PM, Michelle Dupuis mdup...@ocg.ca wrote: Do the C610H and C300IP use an international standard for frequencies? I can't even find gigaset sold in USA/Canada... Gigaset C610a (base + handset combo) are widely available, even on Best Buy and Amazon. You can add additional handsets each with its own registration. Seems to work well, haven't stopped using them like I have all the wifi phones. Range seems decent, although handsets feel hollow and lightweight they seem reasonably solid. Sometimes when dropped the batteries fall out and although the box indicates Made in Germany, the power adapters are made in China. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues with Digum Repos / AsteriskNOW Bad Packages
Anyone have an update as to when Digium will ship a working package? -- Forwarded message -- From: Andrew Joakimsen joakim...@gmail.com Date: Wed, Mar 23, 2011 at 23:53 Subject: Issues with Digum Repos / AsteriskNOW Bad Packages To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with Digum Repos / AsteriskNOW Bad Packages
I have used those packages: [Apr 7 01:09:51] WARNING[27966]: loader.c:434 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/asterisk/modules/app_voicemail_imapstorage.so: undefined symbol: copy [Apr 7 01:09:51] WARNING[27966]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. You're now apparently running into issue #18718 (https://issues.asterisk.org/view.php?id=18718), which was a regression introduced in Asterisk 1.4.39 or so. This specific issue won't be fixed in a normal Asterisk release until 1.4.41. However, packages are in a slightly better position, since we can sometimes apply a fix and just rebuild the packages. I'll do that today, and you will see 1.4.40-3 shortly (I'll also send you a note when they're available). I can only imagine how frustrating this is for you... Unfortunately, app_voicemail.c is written in an overly complicated way, and it's difficult to catch issues like these. I've talked to the person that manages our test platform, and we'll be taking steps to watch for these types of issues in the future. On Mon, Jun 6, 2011 at 14:31, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 06/06/2011 08:07 PM, Andrew Joakimsen wrote: Anyone have an update as to when Digium will ship a working package? According to https://issues.asterisk.org/view.php?id=18748 new packages should already have been pushed. If not perhaps you could join #asterisk or #asterisk-dev on irc.freenode.net and ask Qwell (aka Jason Parker) about this. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
I am still using Asterisk 1.4 because of the Asterisk GUI. I don't understand why it was ever dropped, it's easy to setup (no SQL databases), quick, works well and in my experiance it gets along with manual config file changes. The only real issue I've encountered with 1.4 is Digium can't seem to properly build RPMs... Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receive faxes
It isn't any better than the so called t.38 support in Asterisk that only drops calls. Gee I wonder why, maybe so they can sell their fax product? On Wednesday, May 4, 2011, Steve Edwards asterisk@sedwards.com wrote: On Wed, 4 May 2011, vip killa wrote: screw that i just got hylafax to work with IAXMODEM...i refuse to pay digium a dime... supposed to be open-source right? Great attitude. Should be worth about a bazillion bad karma points. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with Digum Repos / AsteriskNOW Bad Packages
On Thu, Mar 24, 2011 at 15:34, Kevin P. Fleming kpflem...@digium.com wrote: On 03/23/2011 10:53 PM, Andrew Joakimsen wrote: I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? Yes... but this indicates that the module that was built appears to be broken. I'll let the package maintainer know. Bug 0018748 is closed a few days ago, but I don't see any new RPM... -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with Digum Repos / AsteriskNOW Bad Packages
On Thu, Mar 24, 2011 at 15:34, Kevin P. Fleming kpflem...@digium.com wrote: On 03/23/2011 10:53 PM, Andrew Joakimsen wrote: I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? Yes... but this indicates that the module that was built appears to be broken. I'll let the package maintainer know. There's a bug report 0018748 against Asterisk 1.8 using the yum repository and I've added my messages from 1.4 using the AsteriskNOW disc. -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues with Digum Repos / AsteriskNOW Bad Packages
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testing from where number is...
2011/3/3 Piotr Górski pi...@prnet.pl: Something free? If your provider provides a proper rate table you will pretty much know which is mobile and which is fixed line and assuming their rates are accurate I assume your company wouldn't care if you allowed the mobiles billed at fixed line rates. -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] alarm POTS lines
On Thu, Dec 2, 2010 at 11:58, Jeff LaCoursiere j...@sunfone.com wrote: we have a low-cost Atom based PBX and a fax relay setup locally with hylafax/iaxmodem to solve that issue, and it is working very well. We don't however, have a solution for their alarm lines. You would desire the entire path to be UL listed if you are doing anything other than facilitating the phone call to the central station. There is app_alarmreciever in Asterisk, and furthermore the ContactID protocol is pure DTMF so that should work without issues. But why use phone lines at all? Recently I installed a DSC T-Link TL260GS which uses internet and GSM, there is no phone line plugged into the alarm panel at all. The problem is of course that modem calls over VoIP are flaky at best. Even though these alarm calls are low baud rate, when we test with the alarm company we only pass about 30% of the time (ulaw from customer site to our central switch, then out a T1). To be fair there is no QoS on their Internet links yet, and that certainly plays a role. SIA format is 110 or 300 baud, ContactID is (rapid) DTMF. -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
On Sat, Feb 12, 2011 at 07:31, ast guy ast...@gmail.com wrote: Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Polycom phones are still working well and durable as a brick. Gransdstream phones still feel cheap. Cisco still have the same shenanigans going on with their firmware downloads. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Park by EFK
Has anyone gotten one-touch call parking to work on Polycom phones via the Enhanced Feature Keys feature working? I've looked at various examples, they appear correct, but the phones (501, 3.1.x firmware) show the Park button while in a call but this does not actually cause the call to be parked. Doing a SIP debug, I don't see that anything is transmitted as a result of pressing the call park key. My understanding of the below configuration is it should cause the DTMF sequence #70 to be sent across the SIP channel -- but it isn't. efk version efk.version=2 / efklist efk.efklist.1.mname=callpark efk.efklist.1.status=1 efk.efklist.1.label=Call Park efk.efklist.1.use.active=1 efk.efklist.1.action.string=#70 efk.efklist.2.mname=blindxfer efk.efklist.2.status=1 efk.efklist.2.label=Blind XFer efk.efklist.2.use.active=1 efk.efklist.2.action.string=$P1N10$$Trefer$ efk.efklist.3.mname=daynight efk.efklist.3.status=1 efk.efklist.3.label=Day Night 1 efk.efklist.3.use.active=1 efk.efklist.3.action.string=*281 efk.efklist.4.mname=pageall efk.efklist.4.status=1 efk.efklist.4.label=PageAll efk.efklist.4.use.active=1 efk.efklist.4.action.string=800 / efkprompt efk.efkprompt.1.status=1 efk.efkprompt.1.label=Extension: efk.efkprompt.1.userfeedback=visible efk.efkprompt.1.type=numeric efk.efkprompt.2.status=1 efk.efkprompt.2.label=PIN Code: efk.efkprompt.2.userfeedback=masked efk.efkprompt.2.type=numeric efk.efkprompt.3.status=1 efk.efkprompt.3.label=Password: efk.efkprompt.3.userfeedback=masked efk.efkprompt.3.type=numeric efk.efkprompt.4.status=1 efk.efkprompt.4.label=Conf ID: efk.efkprompt.4.userfeedback=visible efk.efkprompt.4.type=numeric efk.efkprompt.5.status=1 efk.efkprompt.5.label=Extension: efk.efkprompt.5.userfeedback=visible efk.efkprompt.5.type=numeric / /efk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why does Digium not respect their own development guidelines?
As recent as 2008 Asterisk 1.4 is feature frozen if that is the case how come now CallingToken support is added? I don't really know what this is but all I know is: 1) Callingtoken adds new options to the config files 2) Callingtoken is some new protocol in IAX? 3) Upgrading asterisk 1.4 breaks previous IAX connections. So why was this added? I have 1 machine that is set not to log these messages on the console and am pulling my hair out after upgrading Asterisk 1.4 to a new release. It is not anything I would look into normally since Asterisk 1.4 is feature frozen so why would I look to troubleshoot a feature that isn't supposed to be there? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Youmail RDNIS
I don't see why it does not work. Setting RDNIS and calling most GSM mobile phones produces a forwarded call annoucement, so why would the do it any different? We get RDNIS in a SIP field and use it to keep the same voicemail for a desk phone and cell phone, also can forward ILEC and most CLEC remote call forwarding and get the correct info, or forward a cell phone to RCF to DID and I see the entire route. If you have a T1 with many DIDs and a provider that supports it you can have all DID forward to a single DID elsewhere and still be able to route by dialed number. All if this done with RDNIS. Are you sure your provider is consistantly sending it? On Wed, Aug 11, 2010 at 17:54, Karl Fife karlf...@gmail.com wrote: into the voicemail account belonging to the RDNIS value. In practice I find that YouMail, when presented with a redirected call as described above ignores the RDNIS value and prompts me for an subscriber account number. By contrast, when various MNO's do the redirection, YouMail is able to determine the redirecting subscriber account number--presumably some other way. Does anyone know the mechanism(s) by which this is normally done? Is there even a 'normal' way to do this (such as a QSIG call transfer message), or is truly home-spun and carrier-specific, such as a Q.931 facility message. Any advice on the subject would be much appreciated! Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues running Asterisk + Iaxmodem + Hylafax on same machine
I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on the same machine. After rebooting the iaxmodems don't register to asterisk. Stoping and starting the relevant services gets it working, but what is the point of using init scripts if it does not work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it should be starting in the correct order. I've previously done this so Asterisk is on one server and IAXmodem and Hylafax on another, I am stumped. [r...@pbxserver ~]# rasterisk Asterisk 1.6.1.20, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.6.1.20 currently running on pbxserver (pid = 2218) Verbosity is at least 3 pbxserver*CLI iax2 show pee peer peers pbxserver*CLI iax2 show peers Name/UsernameHost Mask Port Status ttyIAX0 (Unspecified) (D) 255.255.255.255 0 (E) UNKNOWN ttyIAX1 (Unspecified) (D) 255.255.255.255 0 (E) UNKNOWN ttyIAX2 (Unspecified) (D) 255.255.255.255 0 (E) Unmonitored ttyIAX3 (Unspecified) (D) 255.255.255.255 0 (E) Unmonitored 4 iax2 peers [0 online, 2 offline, 2 unmonitored] pbxserver*CLI exit [r...@pbxserver ~]# /etc/init.d/hylafax stop Shutting down HylaFAX queue manager (faxq):[ OK ] Shutting down HylaFAX server (hfaxd): [ OK ] [r...@pbxserver ~]# /etc/init.d/asterisk stop Stopping safe_asterisk:[ OK ] Shutting down asterisk:[ OK ] [r...@pbxserver ~]# /etc/init.d/iaxmodem stop [r...@pbxserver ~]# /etc/init.d/asterisk start Starting asterisk: [ OK ] [r...@pbxserver ~]# /etc/init.d/iaxmodem start Starting IAXmodem: [ OK ] [r...@pbxserver ~]# /etc/init.d/hylafax start Starting HylaFAX queue manager (faxq): [ OK ] Starting HylaFAX server (hfaxd): [ OK ] Restarting HylaFAX modem manager (faxgetty): [ OK ] [r...@pbxserver ~]# rasterisk Asterisk 1.6.1.20, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.6.1.20 currently running on pbxserver (pid = 4175) Verbosity is at least 3 pbxserver*CLI iax2 show peers Name/UsernameHost Mask Port Status ttyIAX0 192.168.3.28(D) 255.255.255.255 4570 (E) OK (1 ms) ttyIAX1 192.168.3.28(D) 255.255.255.255 56869 (E) OK (1 ms) ttyIAX2 192.168.3.28(D) 255.255.255.255 38332 (E) Unmonitored ttyIAX3 192.168.3.28(D) 255.255.255.255 56565 (E) Unmonitored 4 iax2 peers [2 online, 0 offline, 2 unmonitored] pbxserver*CLI -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Switchvox vs Asterisk codebase
Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
This works for me using DNSMasq: dhcp-host=00:04:f2:*:*:*,net:polycom # creates a 'polycom' group for all equipment with MAC prefix of 0004f2 dhcp-range=net:polycom,192.168.1.151,192.168.1.180 # dhcp range for 'polycom' group dhcp-option=net:polycom,66,http://pbxserver/gui/phoneprov; # polycom bootserver HTTP URL is asterisk provisioning URL. I assume an FTP URL can work fine. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using the PBX Directory from a Blackberry
It is a problem with Windows mobile phones as well, there is *NO* way to dial a number e.g. 800-CALL-ATT. On my Nokia S60 phone (E71) I can dial the number but it is not possible to dial letters when the call is connected. This affects everyone. When I call American Express it asks me to enter my mother's maiden name, which is not possible for me. On Thu, Jul 2, 2009 at 10:53, JR Richardsonjmr.richard...@gmail.com wrote: Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure is there is actually tones generated when the alpha characters are pressed, it just doesn't work). Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. That is how folks can use a Blackberry effectively with the PBX Directory application. Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk w/ Nokia e Series Handsets
By any chance does that also login to the hotspot for the ones that have a walled garden? My ISP provides free WiFi with my connection, but it is a hassle to manually login unless I actually have to use the connection. But that is a clever program, it sounds like it would let me connect my Mail for Exchange between WiFi and 3G automatically. On Thu, May 14, 2009 at 02:02, Remco Barendse aster...@barendse.to wrote: On Tue, 12 May 2009, Andrew Joakimsen wrote: Overall, given the limitations of WiFi, it works rather well. I've never had to reboot my E71 or play with the settings after it was setup. Something I can't say about other WiFi (only) phones I have used. And VoIP on Windows mobile phones is crap. I installed a program called WeFi on my phone. To the phone it appears as one single access point, while WeFi handles connections to all access points automatically. It solves the problem of creating one SIP profile for each access point. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk w/ Nokia e Series Handsets
On Mon, May 11, 2009 at 11:24, Cory Andrews c...@voipsupply.com wrote: Anyone using Nokia “E” Series handsets with Asterisk? I’m trying to deploy some e71’s and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won’t allow me to save the profile, saying that devices in the same “realm” must have identical username and password. Anyone have a workaround for this to add a second Nokia phone under the Asterisk “realm” with a different userid and PW? If you do mean a 2nd phone, it should not have anything to do with the 1st phone. Make sure your settings in the proxy server menu and registrar server menu are the same. Another thing, it is practically required to use the Nokia SIP VoIP settings program (download from europe.nokia.com, it is not listed at nokiausa.com). Unless you use that program to set count of voip digits (I set to 11) and ignoring domain section to digits only the call logs are a mess -- by default if you dial a VoIP call in the call log it shows it with @server at the end. If you then go out of WLAN coverage you can not redial that call. Overall, given the limitations of WiFi, it works rather well. I've never had to reboot my E71 or play with the settings after it was setup. Something I can't say about other WiFi (only) phones I have used. And VoIP on Windows mobile phones is crap. On Mon, May 11, 2009 at 11:27, Steve J. Douglas stev...@moij.biz wrote: In my case, I was trying to add more than one SIP profiles for the same user account, but with different access point. In that case, just setup the 1st profile and then select WLAN found (or WLAN scanning off, twice) from the home screen, select search for WLAN and once you are connected you will see the option Connect to PROFILE in that same menu and it will automatically create it for you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
I use these cards and they work pretty well. FWIW when Digium sold them they were also just winmodems with a resistor removed to change the PCI device ID. Later on the Zaptel driver included the device ID of the winmodem. I used to be able to get the winmodem itself for under $10, but I think they are discontinued now. Ambient = Intel, FWIW. If you want I'll dig out out and give you the details. If you need a large quantity I would try to find the winmodems that are compatible. On Wed, May 6, 2009 at 08:43, Vincent vincent.delpo...@bigfoot.com wrote: Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the computer's CPU? 2. Are they all bad, no matter what chipset is used (Intel, Motoral, Ambient)? If not, which offer good enough quality to handle a single POTS line? 3. Why are they often bad quality? Because the driver itself is badly written? Because PC's don't have enough speed to handle the tasks using their own CPU (hard to believe, but I don't know)? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64bit: any problems with asterisk?
On Sat, Apr 25, 2009 at 06:03, sean darcy seandar...@gmail.com wrote: We're getting a new server. I'm considering installing 64bit fedora rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any issues we should expect? I have been using Asterisk on 64-bit and 32-bit openSUSE for quite a few years with no problems. They both work flawlessly. Most recently I have just been using the RPM packages from the openSUSE build service (network:/telephony now in network:/telephony:/asterisk), again with no problems on either platform. I have been using exclusivly AMD for 64-bit, FWIW. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US Caller ID
The *BEST* solution would be to have Verizon switch you over to a PRI. On Wed, Apr 29, 2009 at 17:29, Daniel Hazelbaker dan...@highdesertchurch.com wrote: Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0 signaling a ring. 2) A short time later, via ZapScan() I can hear the FSK tone. 3) About the same time I hear the FSK tone I see the Starting simple switch line in the Asterisk console. 4) Next I see the second ring trigger in zttool and then Asterisk say ss_thread: Got event 18 (Ring Begin). Caller ID never shows up. I have tried cranking the rxgain up thinking maybe it was too quiet for Asterisk to detect but that did not help. My caller id settings in zapata.conf are: usecallerid=yes callerid=asreceived cidsignalling=bell cidstart=ring signalling=fxs_ks Is there any existing debug options I can turn on, or do I need to add some to try and figure out what is going on; or does somebody have an instant answer for me? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New system for recording - SCSI, SAS or SATA?
There are RAID controllers (hardware, of course) that have battery backup, so the risk in very minimal in using write cache. Just one (random) example: http://h18000.www1.hp.com/products/servers/proliantstorage/arraycontrollers/smartarrayp400/index.html SAS controllers support SAS and SATA drives, FWIW. On Fri, May 1, 2009 at 08:35, Benny Amorsen benny+use...@amorsen.dk wrote: t...@softins.clara.co.uk (Tony Mountifield) writes: I'm in the process of specifying the hardware for some new Asterisk systems which will be running a substantial number of conferences with recording. I was wondering what there is to choose between SCSI, SAS and SATA disks, in terms of performance for this kind of application. Modern SCSI, SAS, or SATA drives don't perform differently because of the interface type. You can't get 15kRPM SATA drives because the market for those is too small though. If you record 1 channel in Alaw, you need 2 x 64kbps disk bandwidth, or 16kB/s. If you record 1000 channels, you need 16MB/s from your disks, which should be easily achievable with even the cheapest disks. However, that depends on doing sequential writes. You can only do (best case) 120 random writes pr second on a 7200RPM disk without write cache, and you can reach that limit with just 2 channels, if you have to do a seek pr packet. The solution there is write cache; 1 second gives you 120 channels and 5 seconds bring you up to 600 channels. If you are unlucky and the files are placed widely spaced on the drives, the performance will be lower than those numbers. So, to get decent performance from many streams, you need a lot of disk write cache, either on the disk itself (with the risk that a power failure destroys data), on the controller, or in memory. You can gain a factor of 2 by going to 15kRPM disks, and another factor of two by doubling the number of spindles (if you get the layout right). The Linux write cache can be tweaked for this purpose, but again you risk that a power failure destroys data. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Verifone-Agi
Could you explain better what you want to do? The VeriFone terminal can talk to the merchant processor via a phone line or via Ethernet (TCP/IP). Why do you need to interpret the incoming information from the VeriFone? What do you intended to do with that information? On Tue, Apr 28, 2009 at 13:00, Juan Miguel Quiros Arrieta jqui...@sistemasanaliticos.net wrote: This week in another project I have to develop an application using the VeriFone vx510 device and I read this device needed or could use a PPPoE connection in order to validate and send all information collected from the end user. My question is if I can use the asterisk and the IVR I built to interact with the VeriFone. I mean, VeriFone-E1 or FXs-o-Asterisk-MyAgiIVR. Obviously I have to adapt my IVR to interpret the incoming information from the VeriFone and can I return information to the VeriFone device in real-time?. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cheap CHEAP ata
Google shows one result for low cost ATA: http://www.trixbox.org/forums/vendor-forums-non-certified/linksys-cisco/linksys-pap2-and-rt31p2-low-price Buyer beware! Those are probably counterfeit! On Fri, Apr 24, 2009 at 19:15, Wilton Helm wh...@compuserve.com wrote: Have you checked ebay? Just beware that there are a lot of ATAs on Ebay that are locked to Vonage or similar providers. While they are not impossible to unlock, it requires considerable time and good Linux networking experience, as the process generally involves creating an isolated world (with its own DNS, etc) that mimics the provider and then updates configuration files. If you want a lot of cheap ATAs it might be worth your while to set up such a system, as most of the work would be for the first one and the rest would be relatively easy. On the other hand, if you weren't anticipating this problem, you might get stuck with a bunch of useless paperweights, which wouldn't make the total cost of the solution very cheap. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
On Mon, Mar 16, 2009 at 20:26, Marc Charbonneau timebandit...@gmail.com wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply Or the Polycom 320 -- same phone as the 330, both have PoE support, 320 has 1 Ethernet port, 330 has two ports (built in switch) Now that I have to reply to your message, may I suggest telephonydepot.com. They have the 330 for $106 and the 320 for $83. FWIW VoIP supply are horrible (and overpriced.) They took 6 months to RMA a phone, and even then they didn't do what I requested. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Communicator with Asterisk/Trixbox
Is it the Windows software, or other? I noticed the Nokia E71 mobile has an option for Cisco IP Communicator (besides the built-in SIP client) On Wed, Mar 4, 2009 at 22:32, Dorien K. Takeshi dorien.take...@webhad.co.nz wrote: Hi guys, Has anyone had any luck with getting the Cisco IP Communicator working with your Asterisk or primarily, Trixbox installation? I've tried searching the net for information, and found someone said to set it up like the 7970 hard phone, which I have, and I'm just running into the problems with it saying Error Verifying Config Info. Any and all help is appreciated. Dorien ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Vs AMD
On Mon, Feb 23, 2009 at 03:10, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Feb 23, 2009 at 08:00:38AM +, Gordon Henderson wrote: On Sun, 22 Feb 2009, Doug wrote: Interesting shopping list - I've just built a new server for my co-lo and it's an Intel Atom mobo. Normally I do use AMD though, but right now, power consumption is an issue and hen AMD's low power chips are mainstream, I'll look at usin them.. My dual-core Atom mobo with 1GB RAM, 2 x WDC drives consumes just 42 watts. Which means it will run at room temperatures without any issues with no case fans. (although the co-lo has 2+1 AC) If power consumption is a concern, also concider some Via CPUs. AMD also has dual-core 45w CPU's. Not as efficient as Intel Atom or VIA, but quite a bit more powerful. I use on on my desktop PC -- with other systems I notice the room gets warm, but not when using a 45w AMD. For my deployed Asterisk servers, they are pretty old -- either AMD Opteron 939 (I think those are at least 89, if not 110 watts!) pins dual core or 478 pin Pentium 4 (2-3ghz) -- but they run just fine. I think I would concentrate more on getting high-quality mainboard, power supply and RAM than what the CPU speed is, unless you plan to have heavy traffic. Who cares how fast the CPU is if you have a PCCHIPS board with generic RAM and an off-brand 500 watt PSU (that really gives off 200 watts) that all make the system unstable? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
On Wed, Feb 18, 2009 at 12:50, bilal ghayyad bilmar...@yahoo.com wrote: And is there a bank accept to give such kind of communication? The user was able to dial his card number and the amount from his phone (or IP Phone registered with Asterisk), and Asterisk communicate with the bank or company credit card provider? How the user will enter $50.25? What about expiration date of the credit card? Where there are two solutions: 1) The bank provides the service... you do nothing but call the number they provide. 2) The bank provides some sort of API (authorize.net is common in the United States of America) and you write code (an AGI script) that a) accepts the input via the phone b) communicates with the bank using the API, probably via the internet using some sort of encryption (HTTPS is pretty common) Answers to your questions: 1) Probably just by entering 5025 2) Probably just by entering MMYY (month, month, year, year. e.g.: 1210 = December of 2010) This is rather simple since the format is known. Currency usually has two decimal places and years are again a standard format. If using option 2) above it would be wise to provide a confirmation (user dials 5025 and then a prompt would say You entered fifty dollars twenty-five cents. Is that correct?, etc.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
On Tue, Feb 17, 2009 at 15:09, Jeff LaCoursiere j...@jeff.net wrote: On Tue, 17 Feb 2009, Jerry Jones wrote: Most alarm systems around here use bursts of dtmf - not an actual modem to communicate with alarm central. Yes I have seen these have many issues with voip in the path. You mean they communicate with an IVR? Seems like that could be made solid with the right DTMF options enabled on the ATA. FWIW that makes a lot more sense than a modem connection. No, it's not an IVR. It's a protocol called ContactID. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net wrote: Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look into reducing the baud rate on the machines, but sadly the bank has them password protected and wants to charge a reprogramming fee :( They make credit card terminals with Ethernet -- use that instead. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (Fwd) New problem: They disconnect your service for no reason
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote: Your service is still up and working, Because Suzanne Bowen has better judgment than you. You did charge back on the payment to us, That is correct. There is $86 balance in my account I did not expect to get back by just asking for it. We are being nice to you and you do not understand the meaning of nice? Your actions are not nice. You threaten to cut off service if a customer discusses issues... issues which you were actually given plenty of time to solve... in public. Like I said in my other post, can you imagine Level3, Global Crossing, ATT or Verizon doing that? Why don't you spend time correcting the flaws in your service, instead of policing the internet for people talking bad about your service? I am not even making this up, I posting FACTS, not lies like you and your employees posted in my tickets. What is wrong with you ? If you didn't know, I have too much free time. You don't mess with me because I have plenty of time to mess back with you. I will not be done until every person that uses VoIP knows how terrible your service is. Do you want me to really close it up ? Yes, as we discussed with Suzanne, the service will be closed up on 28th February, 2009 at 5:00 CT. Or are you going to change your mind again? FYI I do not have a problem in you complaining to me, you can complain a million time, and you will get result, it is your public posting about the problem and discussing with people who do not undertand the issue is the reason we can not do business with you. I did not get results waiting almost a month for the feature to get fixed. The provider I use now we had an issue. I reported it yesterday, within 5 minutes they confirmed there is an issue. Within 1 hour they had resolved the issue with their upstream provider. THAT is how you provide good service. There is an issue, I am not going to go around talking bad about you. But there is an issue, you deny there is an issue, and you take a month.. then I will talk bad about you because you do not care about the customer. Sorry, it's true. If you told me that you found a programming issue (which is what I think the problem is) and that you programmers will fix it in the next release, I can understand that answer and have patience. But to be told multiple times that everything is working when that is not true, I can not deal with lies. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Beware of DIDX Super Technologies
I assume most people here know what a joke DIDX is -- but in case you didn't already know, please avoid these people. Basic features of their service don't work, their tech support refuses/drags their feet to fix them for a month and if you post publicly about them, they terminate your service. Instead of investing their effort in reading mailinglists to terminate customers maybe they should invest their efforts in fixing the issues with their service first. This is all despite the fact that they don't control the numbers they sell -- that I understood and can deal with (it never was an issue since most of the numbers we had with them were from Global Crossing -- Vendor # 701534 in their system) Hell, I was planning to get off their service anyways, if they would have allowed me time to properly port out the numbers, they would not have created an enemy for life. -- Forwarded message -- From: Rehan Allah Wala re...@supertec.com Date: Sat, Jan 10, 2009 at 13:56 Subject: Your DIDX account To: Andrew Joakimsen joakim...@gmail.com Cc: muneeb @ supertec. com mun...@supertec.com, suza...@supertec.com Thank You for this email Andrew, Please move your numbers in next 3 days somewhere, we will close your account as per your request on Tuesday. Rehan On Sat, Jan 3, 2009 at 13:09, Trixter aka Bret McDanel trix...@0xdecafbad.com wrote: On Sat, 2009-01-03 at 12:14 -0500, Andrew Joakimsen wrote: Can you look at ticket # 702556000194? This is very simple: apparently it isnt. Asterisk is down, I am simulating that with the command stop now, Calls should then go to the failover SIP address, but they do not. I have been back and forth for weeks with your support and they do not figure it out. I am not even sure they understand what I am saying. is this related to the below request for a non-profit doing a telethon? If it is I am confused by it. If it isnt, I am unsure what ticket system you refer to. Additionally I am unsure what your setup is since you havent even provided more information. Odds are the equipment that is supposed to do the failover isnt even asterisk. Further I do not think that its a business list question (unless you are asking for a consultant to fix your problem with failover). Or was this supposed to be a private email to someone at supertec that you posted to a public mailing list? On Tue, Dec 9, 2008 at 15:34, Suzanne Bowen suza...@supertec.com wrote: There is a Northwest Florida organization http://www.linkingarms.org who wants to have a telethon using open source telephony technologies. If anyone reading this is interested in talking with the executive director Kenny about this, please email me OFF the list so we won't bother the list with further details. You are right, the message was sent to the wrong person. But now that this is out in the open, let me explain: DIDx has an alternate ring-to feature. So if our Asterisk server is down, the calls can roll-over elsewhere. This feature is not working. The calls do roll-over, but there is no audio (even using DIDx's own carrierx.us proxy) and drop a few seconds later. Testing from a landline, the caller hears a few seconds of silence and then a reorder tone. If we set the alternate ring-to proxy as the primary ring-to, it does work. Clearly the issue is on DIDx's end. I conducted my testing by issuing the stop now command at the CLI and called the DID number from an ATT landline phone. I was promised last week this would be looked into, fixed, etc. But no response thusfar. I am going to stop using DIDx as it has been one big headache. On top of this not working, the CDRs on their website are incorrect -- either they have inept programmers or they do absolutely no QA testing on their code -- but they'll gladly sell you a copy for upwards of $20,000. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz Rehan Ahmed AllahWala Msn/Yahoo/GoogleTalk/Email: re...@rehan.com http://www.supertec.com/ - Internet Telephony Solutions Http://www.DIDX.net - DID Number Market Place. Don't Remember Me ? Visit http://www.Rehan.com ~~~ First they ignore you, then they laugh at you, then they fight you, then you win. By Gandhi. Live as if you were to die tomorrow. Learn as if you were to live forever. - Gandhi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] u-law file header ?
On Mon, Jan 12, 2009 at 16:15, Karl Fife karlf...@gmail.com wrote: QUESTION: Who's in the wrong: I recently saw an example of a u-law file with a metadata header on the file. The asterisk playback function 'PLAYED' the ascii header values as if they were audio data, creating an audible 'click'. After realizing the click was coming from metadata (and fixing it), I became curious: Which is 'correct? In other words: 1. Is it considered incorrect to ever include metadata on a u-law formatted file? or 2. Is it considered incorrect for a playback engine to fail to check for metadata headers before playback? or 3. Is it unspecified, and therefore considered incorrect for someone (ME) to FEED a u-law stream into any playback engine without FIRST making sure that somebody's two-bit application didn't tack on an unsolicited header? :-) In other words, should case 1 or 2 be considered a software defect? No, it is not a defect. You need to distinguish between the file format and audio codec. They are independent of each other. You can store ULAW in a WAV file. The default asterisk sounds (.g729, .ulaw, .alaw) are raw audio files, thus there is no header. If Asterisk is expecting a raw format file, it will playback the entire contents. If I save a WAV format file encoded in ulaw, name the file .ulaw, and then play it back in Asterisk, the exact scenario you describe happens. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLA and Polycom
On Wed, Jan 7, 2009 at 23:39, Noah Miller noahisaacmil...@gmail.com wrote: Hi Mark - You really want to do SLA with all 23 lines of the PRI? That's a lotta lines to be shared. You'd need two sidecars for each phone (Cisco or Polycom). Actually there will be multiple PRI's :) This customer is a multi-tenant situation so each tenant will have a few trunk SLA's and maybe some extension SLA's. Aha. That makes more sense. This is, they will if a) it's do-able b) it works on Polycom as I don't see anything coming back from the phone when I designate a line key as shared. I don't believe that Polycom's version of SLA does anything with Asterisk. You have to use asterisk's SLA implementation (http://www.asterisk.org/node/48342). I believe that SLA on the Polycom phones is based on the Broadsoft SIP implementation. You can read more about it here: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=11688 Mark, you will be very dissapointed with the Asterisk SLA feature. Caller ID does not work, neither will the redial or call logs on your phone. In the sense that the line is shared, yes it is. But that's where the function ends too. I would say the feature is in alpha testing right now -- it can not be used in a production environment. You will not find much information about it, I assume because those that have actually gotten it to work realized its more of a joke than anything and gave up on trying to use it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith, Music-on-Hold Parody--outstanding.
On Wed, Dec 31, 2008 at 22:09, Paul Hales pdha...@optusnet.com.au wrote: Karl Fife wrote: Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes. I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!! -Karl Any chance of us non-us citizens hearing it? (podcast, download...) I put up a recording here: http://app5.netjdn.com/~joako/karl.wav I hope Karl doesn't mind. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-11X configuration to work with Asterisk
On Tue, Dec 30, 2008 at 00:25, Jeff LaCoursiere j...@jeff.net wrote: On Mon, 29 Dec 2008, Andrew Joakimsen wrote: On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote: What does Audiocodes release under GPL? j The MP-202 is running Linux. At first they said no it's not and later they admitted it did, but refused to supply the source code. Oddly enough, the Linux distribution is OpenRG, which itself had GPL problems a while back. I don't know about any other products, but I have never used them either. Of course, if they use GPL software they probably have the same attitude towards it. They shipped me the devices from their offices in Israel, so I could not just go to small claims court to get the code from them, I just gave up and never used their products again. Too bad, because the product was very nice. Oops - I take it back: http://www.audiocodes.com/gpl-lgpl Looks like they are at least attempting to comply... did you follow these steps? j No -- that information did not use to be there. I could have sworn even a week ago a Google search for audiocodes GPL did not find that page. This is interesting: http://www.audiocodes.com/bsd-bsds No mention of OpenRG... OpenRG is a sort of embedded Linux distribution/SDK. I don't see why its mention would be relevant. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-11X configuration to work with Asterisk
AudioCodes blatantly violates the terms of the GPL by not distributing the source code even after requesting it. Please don't use their hardware. On Thu, Jul 24, 2008 at 07:34, Frank Tarczynski ft...@mindspring.com wrote: I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-11X configuration to work with Asterisk
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote: What does Audiocodes release under GPL? j The MP-202 is running Linux. At first they said no it's not and later they admitted it did, but refused to supply the source code. Oddly enough, the Linux distribution is OpenRG, which itself had GPL problems a while back. I don't know about any other products, but I have never used them either. Of course, if they use GPL software they probably have the same attitude towards it. They shipped me the devices from their offices in Israel, so I could not just go to small claims court to get the code from them, I just gave up and never used their products again. Too bad, because the product was very nice. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP URi dialing
On Mon, Dec 22, 2008 at 16:30, amit salunkhe amitsalunkh...@gmail.com wrote: i need to implement Inward SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me First, you must understand the security implications if this is not correctly configured. But I'll assume you have a proper system setup and have already addressed the security matters. Basically, you want all the local extensions (but none of your providers trunks) setup into one context in your extensions.conf. Then, assuming you have kept most of the default settings, it should be a matter of insuring these two lines in sip.conf are set: allowguest=yes ; Allow or reject guest calls (default is yes) context=default ; Default context for incoming calls This will send all calls that do not match a valid defined peer to the default context, which in my case looks like this: [default] include = localusers So, in effect, this configuration allows all the defined users to access their accounts as they normally do, but then allows unknown/unauthenticated peers to only dial into those extensions defined in the localusrers context. Misconfiguring your diaplan or SIP could very well allow unknown users to dial calls that will cost you money or cause data breaches within your organiszation, or those that you host services for, i.e. you DO NOT want any MixMonitor() or your paid providers to be accessible this way! As for the DNS, there is nothing unusual there, simply set an A record for whatever host you want to use to point to your Asterisk server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cut Through DTMF caller ID on SIP phon
On Fri, Dec 19, 2008 at 12:08, David fire ddf...@gmail.com wrote: set(CALLERID(number)=000) David Keep in mind that with doing that, you would loose the caller ID number for the CDR -- thus there will be no record of the caller ID anywhere (asterisk-related, at least). I believe if you use a Local channel (Dial(Local) it will not cause this concern, but you would have to test, because I sure didn't. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP Origination with RDNIS
I am looking for a VOIP provider that can offer origination and provide the RDNIS with each call. I am not looking for any large volume commitment. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring back tone
On Fri, Dec 12, 2008 at 18:57, Eric ManxPower Wieling e...@fnords.org wrote: Philipp Kempgen wrote: michel freiha schrieb: I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way to play the asterisk ring back tone all the time? Dial(,,r) ? Much like violence and herding of llamas, the r option to Dial (and the Ringing app) almost never solve the problem they are intended to solve and frequently cause more, usually unforeseen, problems. Just say No! to r. If these are inbound calls you are answering, r can be acceptable. But ONLY on the final leg where the call might actually be ringing. But for outbound I fully agree. Let the carrier generate the ringing. A second or 2 of dead air is acceptable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about connecting with Mobile Base Station
On Tue, Nov 18, 2008 at 22:30, mark morreny [EMAIL PROTECTED] wrote: Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. There are various devices. PCI GSM card, GSM to Ethernet, or the most basic is GSM to analog, then you connect it to asterisk with e.g. X100 card or SPA3000. Either the PCI or Ethernet devices should work very well -- since the call from the GSM network continues to be digital. An analog adapter will have a slower call setup time, can not support SMS or data and might have echo issues and by definition of a digital-to-analog and subsequent analog-to-digital conversion the quality of the call will be worse (but probably not noticeable). Here is one example: http://www.junghanns.net/en/GSM-PCI_produkt.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ALL of DIDx Down?
Anyone else notice all of DIDx is down? Calls on their 3rd-party DIDs do not go through, but the website is up. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Card Noice issue
On Mon, Nov 17, 2008 at 11:55, Bipin [EMAIL PROTECTED] wrote: Hello all, I am facing as serious problem when running asterisk in HP server.We are developing application to make the outbound calls in PRI lines .We normally uses IBM machine as our servers ,and it was working fine for all installation.For the cost reduction we this time tried with HP server. Model(HP proliant ml110). When we make the calls the there is a lots of disturbance in the sound even if we make a single calls the issue persist .I found in google that these issue normally comes by the load or by the line or by the IRQ . As in my case i am making a single call the 1 st case wont occur here.Also i tested it with one smoothly working E1 to the same card and still the problem came.so I guess the problem is with IRQ. But when i tried it with a normal PC with pendium 4 processor it was working fine. My question is whether the Digium card had any hardware compatibility issue with HP proliant ml110 server.Why the sound has issue in HP server when it working fine in a normal pC with pendium processor...?? When i switched to Asterisk now it is very much ok. can any body explain why it have when using with ubuntu??? You might want to try Sangoma cards, instead of Digium. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey
I am confused now. I called Polycom early October and was told to submit a ticket for the latest firmware. I submitted a ticket and was told to fuck off. Dave posted a link and I can download firmware 3.1.1 from there. The Firmware Table shows 3.1.0RevB as the newest firmware. 3.1.1. says released 10 November 2008 so I am going to assume this is a mistake and post a copy of the firmware here incase I misplace it: http://app5.netjdn.com/~joako/spip_ssip_3_1_1_release_sig.zip Am I thinking aloud? Am I the only one that noticed some problems with 3.1.0RevB? Here's my problem, the vendor *CAN NOT* Provide a reliable download site, last time I requested firmware it was 28th April and they took until 01 May to get it to me. So every time I need to contact VOIP Supply, wait a bit, follow up, wait a bit. Just not acceptable. Hopefully the current bug on the Polycom site is never fixed. Or maybe they finally noticed their support cost of people calling in, being told to submit a ticket, responding to the ticket, angry customers, people like me badmouthing them, etc cost more than just giving out the firmware. And I strongly believe that especially the opensource community will be more prone to purchase Polycom if updates are freely available. On Mon, Oct 27, 2008 at 07:39, Dave Fullerton [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Sunday 26 October 2008 21:28:34 Andrew Joakimsen wrote: Other vendors, including Cisco, will provide the firmware directly. I no longer deploy Polycom (unless someone really wants them) due to this. Yes I can get it from the supplier but it takes a few days. I would rather just go to Polycom.com and get the firmware when I want to. There is no excuse for Polycom's behaviour. I don't see what is the benefit, nor what anyone has to gain from it. I believe your anger is misplaced. I was able to get to a direct download of Polycom firmware, from their homepage, within 4 clicks, with no login whatsoever required. http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip501.html While Polycom at one time may have had a policy of only providing firmware to distributors and resellers, that is no longer the case. Their firmware is freely available now to all comers. Not quite, you'll notice that the most recent version they allow you to download is 3.0.4. If you look at the SIP Downloads matrix, the latest release is actually 3.1.0RevB which is only available through your supplier. From their site: NOTE: At this time, end-user customers can only download previous software. Please work directly with the Polycom Certified VoIP Reseller you purchased the products from to obtain the most current and appropriate software. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MS Exchange IMAP Voicemail
On Sun, Oct 5, 2008 at 8:04 PM, David Backeberg [EMAIL PROTECTED] wrote: Isn't IMAP IMAP? Does MS not actually follow the protocol? Why would it be different? When I setup my voicemail.conf for IMAP Asterisk does not work right. sip show peers only shows 1 peer. The CLI is freezing up, etc. When I turn off the IMAP voicemail these problems go away. I configured everything how it should be so I am wondering if someone can post a configuration they know works 100% with Exchange IMAP server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cheapest 4 port FXO
On Sun, Oct 26, 2008 at 4:51 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Sat, 25 Oct 2008, Joseph L. Casale wrote: X100P. Yeah I saw these but they are single port and I need at least 2 ports. I only have 1 free pci slot as well. OpenVox. Those look great, and on top of the price they are 100% TDM400P compatible it seems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey
On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote: The 3.1.0 firmware allows you to create up to 10 custom softkeys. This is all documented in Polycom's SIP 3.1 Admin Guide. Should I post some examples? Which would be great, if Polycom weren't the Firmware-Nazis that they are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Emerging dilema? DID forwarding meets SMS
On Fri, Oct 24, 2008 at 10:09 AM, Drew Gibson [EMAIL PROTECTED] wrote: Can anyone clarify how SMS to non-mobile numbers are generally handled in North America? Is it possible to have SMS delivered direct to your landline DIDs? Then have Asterisk relay it to the actual mobile DID. When I send an SMS from a SprintPCS phone to a landline it gets delivered via voice, pretty much how Gordon describes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey
Other vendors, including Cisco, will provide the firmware directly. I no longer deploy Polycom (unless someone really wants them) due to this. Yes I can get it from the supplier but it takes a few days. I would rather just go to Polycom.com and get the firmware when I want to. There is no excuse for Polycom's behaviour. I don't see what is the benefit, nor what anyone has to gain from it. On Sun, Oct 26, 2008 at 7:53 PM, Darrick Hartman [EMAIL PROTECTED] wrote: If you buy your phone from a reputable place they will be able to provide the firmware. --Original Message-- From: Andrew Joakimsen Sender: To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey Sent: Oct 26, 2008 5:45 PM On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote: The 3.1.0 firmware allows you to create up to 10 custom softkeys. This is all documented in Polycom's SIP 3.1 Admin Guide. Should I post some examples? Which would be great, if Polycom weren't the Firmware-Nazis that they are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Returning to Voicemail after returning call
No, it is not possible. I submitted a bug report[1], because it has been bothering me too. [1]http://bugs.digium.com/view.php?id=13781 On Thu, Oct 23, 2008 at 4:36 PM, Mark Wiater [EMAIL PROTECTED] wrote: Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
If you are VoIP-only then you need a SIP provider that offers T.38. On Wed, Oct 22, 2008 at 11:17 PM, Brendan Martens [EMAIL PROTECTED] wrote: I am using 1.6.0.1 and we are going to be pure voip. I know it has pass through and termination, but that is useless if I don't have a way to transform the analog t.30 to t.38 before it gets to me. That is where my confusion lays, is there some way of doing this that I am not aware of? Brendan Martens On Oct 22, 2008, at 3:02 PM, Jonn R Taylor wrote: What version of *? Are you going all VOIP for your voice or are you using a T1/E1? *? 1.4 has t38 pass-through and 1.6 has pass-through and termination, but 1.6 was just release and I would not suggest using it in a production environment unless you can tolerate problem or even outages. If you are planning on using a T1/E1 then send incoming calls to iaxmodem/hylafax or to an ATA/FXS card. Either works very well. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Brendan Martens Sent: Wednesday, October 22, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fax / t38 gateway I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with normal fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Polycom IP330 user problem
Could it be DND? I noticed the other day on my 501 that if I set do not disturb the phone still rings -- it is just silent. This could be caused by the configuration, I am unsure. On Sat, Oct 18, 2008 at 3:16 PM, Bill Michaelson [EMAIL PROTECTED] wrote: I recently sent this email to a user in response to a problem report of phone calls going to voicemail without the phone ringing. I'm wondering if I've covered all bases, or whether there is some logical explanation I haven't considered, and generally what others' opinions/experiences are that relate. This is an Asterisk system, of course. --- I looked at the server logs for the phone call missed by . They indicate that the call came in at 15:32:25, and was routed to her telephone at 15:32:32. This timed out after about 25 seconds as it should if unanswered, and was sent to voicemail at 15:32:58. I called BB and asked her to check the phone display. She told me that the phone logged an unanswered call at 15:32:32, precisely in accordance with the server log. This leaves two possible conjectures: * The telephone, for whatever reason, did not ring in response to the incoming call signal which it obviously received. * The telephone ringer was not audible or noticeable to for some other reason. For the first possibility, I can think of three circumstances that would cause this: * If the handset is slightly ajar, i.e., off-hook, the phone will make no sound, but log the call. Upon receipt of the message waiting notification, it will start blinking. Eventually, the phone reverts to on-hook status by itself even if the handset is still ajar. * If the alert code for silent ring is set, the line annunciator will flash silently to indicate the call coming in. * If the phone is malfunctioning anything can happen. There is no indication that silent ring alert was set, nor is there any current configuration setting that should cause this. That leaves three bullet points for us to consider. I can follow up with one: I will research this as thoroughly as I can to see if there are any reports of malfunctions by Polycom IP330 phones that conform to this behavior, or if there are any other possible explanations for the events that I've overlooked. If you would like to follow up in any other way, let me know what I can do to help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latency woes, qos the fix?
On Sun, Oct 19, 2008 at 12:31 AM, Stephen Reese [EMAIL PROTECTED] wrote: My latency is kind of high and the voice delay is noticeable. Then pretty much all you can do is lower the latency to lower the voice delay, or use a connection to th e PSTN that has a marginally lower delay if you have no other options. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Telrad Analog CID
Does anyone know if I have an older Telrad PBX if I can get CallerID to Asterisk when the connection is via analog FXO-FXS? I only need 1 or 2 lines so T1 is an overkill. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cli commands missing
I've seen something like that (in your next post you show 20-some modules, a stock install will be 100 modules) when using the openSUSE distribution Asterisk package along with asterisk-addons package. What happens is it gets stuck on some module that is not configured I think one of the ones relating to database or CDR. You will notice if you do an asterisk -vc that you did not get Asterisk is ready at the point where your modules are missing. I think after a few minutes everything loads (module times out starting with its default config) If this is the case you should just disable any modules you don't use from /usr/lib/asterisk/modules or /usr/lib64/asterisk/modules you can move them out of there at first and eventually add them as noload to modules.conf (so when you upgrade to a newer version you don't run into the same issue). On Sat, Oct 11, 2008 at 11:17 PM, Eric Fort [EMAIL PROTECTED] wrote: I just loaded a new asterisk install (1.4.19) and found that the sip, iax, and extentions commands are missing from the cli and are not listed in help either? Any idea what could have happened or where these commands may have gone? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setup for fax machine
On Sun, Oct 12, 2008 at 5:17 PM, sean darcy [EMAIL PROTECTED] wrote: Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten = fax,1,Dial(DAHDI/1) ; the fax machine exten = fax,2,Hangup() exten = s,1,Answer() exten = s,2,Dial(DAHDI/2) ; internal extension . Would this work? I'll need another TDM410 card to do this, so I'd like some reassurance before I go purchase it. Another thing you can do is get a comswitch -- they are pretty cheap and have been around for years. http://www.commandcommunications.com/products.php IMO they work pretty well. I use these as a backup. I use VoIP only at some sites and the fax line is used for the DSL. The comswitch is connected to the line and then connect the asterisk machine (using a cheap winmodem), fax machine and a red WECo 2500 clone. So if the DSL goes down the calls get routed through that phone line and the comswitch routes them to the Asterisk machine and the IP phones ring like normal. If the PBX goes down or the power for a long time, they can use the Red phone The main motivation for going to this setup was a low cost way to keep 911 working while trying to stay 100% voip . The comswitch was installed so in case 911 needs to call back it won't go straight to the fax. The use of the line as a backup ended up being an unintended consequence, but it seems to work well. This setup works very well, but if you are using this on a main phone line then be advised there will be an extra 1-2 ring delay -- the comswitch actually answers the call when it comes it and does fax detection. I also see these installed in many retail stores -- normally they will have 3 lines in a hunt group -- the 3rd line is shared between the fax and the hunt group using the comswitch. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Budge Tones pick up wrong calls
Are you using NAT? On Fri, Oct 10, 2008 at 4:24 PM, Paul Douglas Franklin [EMAIL PROTECTED] wrote: We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For example, 2501 might respond to the calls for 2518. After a reboot, it might decide to respond to 2501 as it should. Or it might respond to 2536. The phone it responds for will not respond. I don't know whether to look in the settings on the phone or in an Asterisk setting, and what setting to check in either place. Has anyone seen this behavior before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] registration limit
Maybe you can write your own patch that will allow this based on the useragent somehow mapping it to 2nd peer based on the useragent? But this feature is not there now. What will happen when host=dynamic is the last registration will be the one used, so if you have two SIP devices trying to register at the same time they will fight for the registration. Both will probably be able to dial outbound, but only one will get the inbound calls. The easiest way to accomplish what you want is to setup 2 SIP friends for each user. In your dialplan, setup as follows: Dial(SIP/USER1-1SIP/USER1-2,90,r) Where you have USER1-1 and USER1-2 in your sip.conf. You simply can append more SIP (or IAX or ZAP or even LOCAL/) endpoints using the ampersand (). If one device is offline the other will still ring as normal. On Wed, Oct 8, 2008 at 12:11 AM, Nhadie [EMAIL PROTECTED] wrote: Hi, Is there a way to limit only one registration for each user at a time? meaning if a user tries to register, but that user is already registered. i will deny? or is it possible to for a single user at the same time, and when someone calls that user, it will ring both phones? Just want something whereby a user can assign his extension on an IP phone in the office, and assign the same thing maybe to a softphone on his laptop or maybe a sip client on a mobile phone. so that whenever he leaves the office he can still be reach on his extension via the sotphone. thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automatic call pickup
See the documentation for DISA, it is restricted by context. So assuming you already have your dialplan configured securely, there are no security implications. Be aware that the behavior of the phones change when you dial through DISA, you can no longer use features such as redial. That is because the SIP call the phone places is to that extension, and DISA takes the DTMF signals after the call is already answered. Another option could be to auto-dial call pickup and make the users dial on-hook (if the phone can do that). I have something like that on my Polycom 501 for shared line appearance (i need to take it off -- SLA currently is very poor, caller id is broken, for one). When I pickup my Polycom it dials the shared line. When I dial on-hook it calls without using the shared line. On Wed, Oct 8, 2008 at 3:58 AM, Vieri [EMAIL PROTECTED] wrote: --- On Tue, 10/7/08, Andrew Joakimsen [EMAIL PROTECTED] wrote: I am not sure if it is possible to somehow invoke a function to pick up the call via dialplan, if it is a combination of that function and DISA should do what you need. Thanks! I could configure the ATAs to auto-dial a custom destination which would then call the command Pickup() passing it the appropriate parameters by checking the CallerID. After the Pickup() cmd has been executed I would call DISA without a password. There could be security issues of course. The only check I would do is based on the caller's ID (basically the extension number). Can a caller ID be spoofed? What other filtering logic can be applied for dialplan security? I've read this doc: http://www.voip-info.org/wiki/view/Asterisk+security+dialplan but is there more information elsewhere? Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No reply to our critical packet
]' Method: ACK On Mon, Oct 6, 2008 at 8:26 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: The odd thing is on this particular phone it only happens when you call voicemail. It is certainly a bug in Asterisk, not the UA. Asterisk is trying to send to 192.168.1.x which obviously is not possible. Something in the NAT support is not working right. Hi, You should get SIP traces to see why Asterisk is trying to reply to 192.168.1.x. To do this, enter sip set debug on in asterisk CLI, and post us a log of call reaching voicemail and disconnecting. Regards, Atis On Mon, Oct 6, 2008 at 3:06 PM, SIP [EMAIL PROTECTED] wrote: This message is usually caused by Asterisk not receiving an ACK after about 30 seconds of attempts. There are countless misconfigured UAs and proxies out there that don't handle ACK well, so it would be nice to be able to turn this 'feature' off. What's annoying is that the explanation has always been If we can't get an ACK, we can't send any RTP data. This is patently false, as the RTP will often work fine even if ACK handling is misconfigured (we see it all the time). But alas. As far as I can tell, there's no way to disable this check. I suppose I could code around it, but not being the world's most proficient C coder, I'm always afraid I'll break something else. ;) N. Andrew Joakimsen wrote: I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this critical packet -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops app5*CLI sip show peer 17865221569 app5*CLI * Name : 17865221569 Secret : Set MD5Secret: Not set Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags: Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : CENSORED MaxCallBR: 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 74.CENSORED.213 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:[EMAIL PROTECTED] app5*CLI core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
Re: [asterisk-users] Help with remote users
Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asteriskt38.com
Since when is there a T.38 Gateway in Asterisk 1.4? On Tue, Oct 7, 2008 at 3:01 AM, Daniel Ferenci [EMAIL PROTECTED] wrote: Hi, fax gateway isn't just a packet bridging. It does the mediation between T30 (voice) - T38 (fax over ip) protocols. It does work for asterisk 1.4, asterisk 1.6, asterisk svn head. If it doesn't please send me a bug report and I'm going to fix it. Best regards Daniel. On Mon, Oct 6, 2008 at 7:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: That isn't real T.38 support, it's just Packet2Packet bridging that works correctly. Still need to use a Cisco gateway to support sending the faxes somewhere on the PSTN. But it does work and it is reliable, I use it every day. On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote: Actually it exists. 1.4 had passtrough mode ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Efax from Agi script
I recently did something similar using fax1.com. If you can send an email you can send a fax that way. On Tue, Oct 7, 2008 at 9:19 AM, Riccardo Cupardo [EMAIL PROTECTED] wrote: Hi all, i wrote a script agi, sking for a code, after that it sends an email now i need to send a fax... any hints or tips for that? Ty in advance. -- Riccardo Cupardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing passwords
The value is not Authenticate ID; From the config file: # Authenticate ID P36 = 8000 # Authenticate password P34 = If you look at the HTML source of the webconfig the form field you need to edit will be marked P34. On Tue, Oct 7, 2008 at 5:30 PM, Ken Zarifes [EMAIL PROTECTED] wrote: I have a question about changing passwords. When I change the secret field in sip.conf for a Grandstream phone, and then use the browser to change the Authenticate ID field of the phone to match what's in the sip.conf file, I can no longer make calls on the phone. Any ideas? Thanks for any help, Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
Load the firmware of www.dd-wrt.com on that WRT54G and then put all the VoIP devices directly behind it. It MIGHT work to set the first NAT router to have the 2nd NAT router in the 1st's DMZ... but I prefer to do things The Right Way. On Tue, Oct 7, 2008 at 7:24 AM, Steve Anness [EMAIL PROTECTED] wrote: I have just confirmed that they may be having a problem with double NAT. They have two ATAs, and they have two different DSL connections. One set-up goes from the first DSL Modem (NAT Wirless are disabled on the DSL Modems) to a Linksys WRT110 and then there is a WRT54G hooked in to the 110 that has the ATA plugged into it. The other ATA is configured from a DSL Modem (again, I was told NAT Wireless were disabled on the modem) to a WRT600N and the ATA is plugged in there. I have the same issues on both ATAs. I have no idea why their network is as poorly designed as it is, the bad part is I have to make sure the phones work there and try to troubleshoot from 3000 miles away. Any work arounds for a problem because of double NAT? A quick and dirty solution for them to get their phones working right? Steve Anness On 10/7/08 2:12 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
On Tue, Oct 7, 2008 at 6:00 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Philipp Kempgen schrieb: Klaverstyn, David C schrieb: Mysql for CentOS 5.2 is the mysql client tools. mysql.i386 : MySQL client programs and shared libraries. Does anyone have any other suggestions? http://www.centos.org/modules/newbb/viewtopic.php?topic_id=13097 Or just download Debian at http://www.debian.org/ :-) SCNR Or SuSE at http://software.opensuse.org/ ... IMO the best package management of any distro. ...You would think PNAELV or Cent would have developed a better tool by now... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help no ring on caller side
Try making sure you use the r option in your dialstring. You should *NOT* be answering a ringing channel, as Steve suggested, FWIW (if it doesn't work any other way that is another story) On Tue, Oct 7, 2008 at 5:04 PM, Nhadie [EMAIL PROTECTED] wrote: Hi, Got this weird problem that the caller does not hear a ring. The issue is it's specific to the local telco: Using telco 1 (mobile), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. Using telco 1 (landline), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. But using Telco 2, my phone is ringing, caller does not hear a ring on his side, and i dont answer call hangs up instead of going to voicemail where should i start tracing the problem? TIA regards nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automatic call pickup
I am not sure if it is possible to somehow invoke a function to pick up the call via dialplan, if it is a combination of that function and DISA should do what you need. On Tue, Oct 7, 2008 at 8:37 AM, Vieri [EMAIL PROTECTED] wrote: --- On Tue, 10/7/08, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: regarding your combination of analog phones and ATAs I would look for the auto-dial functionality in the ATA. I am pretty sure I saw it in one web-interface or the other Thanks! I actually found the option. I'm using Grandstream's GXW4008. The option is Offhook Auto-Dial and I set that to *8. It seems to work fine. There's just one drawback: if I don't need to pick up a call but just place one then I need to press the R(Flash) key to get dial tone. Otherwise, *8 leaves me with a hung up tone and I can't dial out. This behavior may be even worse... so I may have to look for another solution. Thanks anyway. Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Destinations
What do you do to get that message? On Tue, Oct 7, 2008 at 8:45 AM, Mr surfit [EMAIL PROTECTED] wrote: Very new to Asterisk, on my console it says there are 47 bad destinations...What is the best way to track these down and resolve them ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asteriskt38.com
That isn't real T.38 support, it's just Packet2Packet bridging that works correctly. Still need to use a Cisco gateway to support sending the faxes somewhere on the PSTN. But it does work and it is reliable, I use it every day. On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote: Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive. Hopefully it works. The one in CallWeaver doesn't. On Mon, Oct 6, 2008 at 8:12 AM, Daniel Ferenci [EMAIL PROTECTED] wrote: and there is a new application called fax gateway (http://bugs.digium.com/view.php?id=13405) that can do gatewaying between T30 and T38 and vice versa. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asteriskt38.com
Maybe it works in more recent versions? I don't know. Anyways this is getting rather off-topic. On Mon, Oct 6, 2008 at 2:23 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Mon, Oct 6, 2008 at 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Hopefully it works. The one in CallWeaver doesn't. How do you mean - it doesn't? We currently use CallWeaver - Asterisk 1.4 - SIP Provider for sending and receiving faxes. Whenever we'll switch to 1.6, we plan to get rid of CallWeaver, as it has T.38 support in SendFax and ReceoveFax. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No reply to our critical packet
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this critical packet -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops app5*CLI sip show peer 17865221569 app5*CLI * Name : 17865221569 Secret : Set MD5Secret: Not set Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags: Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : CENSORED MaxCallBR: 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 74.CENSORED.213 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:[EMAIL PROTECTED] app5*CLI core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No reply to our critical packet
The odd thing is on this particular phone it only happens when you call voicemail. It is certainly a bug in Asterisk, not the UA. Asterisk is trying to send to 192.168.1.x which obviously is not possible. Something in the NAT support is not working right. On Mon, Oct 6, 2008 at 3:06 PM, SIP [EMAIL PROTECTED] wrote: This message is usually caused by Asterisk not receiving an ACK after about 30 seconds of attempts. There are countless misconfigured UAs and proxies out there that don't handle ACK well, so it would be nice to be able to turn this 'feature' off. What's annoying is that the explanation has always been If we can't get an ACK, we can't send any RTP data. This is patently false, as the RTP will often work fine even if ACK handling is misconfigured (we see it all the time). But alas. As far as I can tell, there's no way to disable this check. I suppose I could code around it, but not being the world's most proficient C coder, I'm always afraid I'll break something else. ;) N. Andrew Joakimsen wrote: I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this critical packet -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops app5*CLI sip show peer 17865221569 app5*CLI * Name : 17865221569 Secret : Set MD5Secret: Not set Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags: Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : CENSORED MaxCallBR: 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 74.CENSORED.213 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:[EMAIL PROTECTED] app5*CLI core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
I've used the smaller ones, I think 8pt with 4pt PoE stuck in drop ceilings and such to power ORiNOCO APs and never had an issue. As for the larger switches I've used Linksys SRW224P. I have a few running for a few years without issues. They have GB uplink but the individual ports are 100M. On Mon, Oct 6, 2008 at 12:12 PM, Karl Fife [EMAIL PROTECTED] wrote: If you happen to be looking for a SMALL poe switch for a home or lab: Think twice before you buy a netgear FS1xxP. While they're great because fanless, I've had 2 Netgear FS116p POE switches, and so far BOTH have developed one or more 'dead' POE ports. The manufacturer has a LIFETIME warranty, but they have an advance-replacement charge, plus you have to pay for your own shipping. $60 so far this year on warranty replacements. According to support there is no 'Second Gen' hardware design to fix the problem so I expect it will happen again. Has anyone else seen this? -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold for sub tenants
Yes, you can set moh in sip.conf or zapata.conf. The options are mohinterpret= mohsuggest=. I think last time I used them (1.2.x) they were just moh= but it seems mohsuggest=class will do what you want it to. On Sat, Oct 4, 2008 at 2:57 PM, carl Lougher [EMAIL PROTECTED] wrote: This seems to be related to inbound calls. So would this work for music on transfers within that context as well as hitting the hold key on calls? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t1 cards
How much further than 300m? It might be very well possible to just lower the speed to 10M and just use that If you already have some quality Cat5 cable between both points it's worth a shot. I support some sites with this arrangement and I've had to find 10M hubs for replacement hardware (the previous guy insisted that only a particular model HP print server would work, coincidently that model only has a 10M Ethernet port)... it's not something I would advise someone to setup but if cost is a concern I wouldn't rule it out -- it certainly can work and be reliable in the real world. On Fri, Oct 3, 2008 at 3:14 AM, Eric Fort [EMAIL PROTECTED] wrote: yes, more than 300 meters (longer than copper based ethernet allows). Yes to E1, as I understand it, it's just a config change on many cards anyway. I'm specificly looking at pci based t1/e1 cards because I'm finding single port cards on ebay going for 100-200 usd. in some cases I may want to drive a channel bank at the far end, thus t1/e1. anyone have experience on how far these pci based cards will drive when wired back to back? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco VAD and Asterisk recordings
Yes. Disable VAD in your Cisco as Asterisk does not (fully) support it. On Wed, Oct 1, 2008 at 9:21 PM, Gabriel Ortiz Lour [EMAIL PROTECTED] wrote: Hi all, I'm experiencing problems with VAD activated on a cisco router doing the bridge between an PBX and de asterisk server. The calls are all rights, but on the recordings the silence from the cisco end point doesn't get recorded, so the audio is completely wrong (the words and phrases from this side are all 'glued' togheter and the other (native SIP) are OK) Anyone experienced problem like this? Gabriel Ortiz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MS Exchange IMAP Voicemail
Has anyone successfully used the IMAP voicemail storage with Microsoft Exchange 2003? Can someone provide a working example configuration? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MS Exchange IMAP Voicemail
Yes, IMAP is IMAP... at least it is supposed to. But not all IMAP servers use the same configuration. Not all IMAP servers will use the same Master User IMAP setup, what works in Dovecot might not work in UW or Exchange due to a prefix or some other fairly trivial setting. Remember there are two pieces of software that need to be configured for this to work properly. So I am asking if someone has a configuration that they *know works* with Exchange 2003 and if they could please share that. On Sun, Oct 5, 2008 at 9:04 PM, David Backeberg [EMAIL PROTECTED] wrote: Isn't IMAP IMAP? Does MS not actually follow the protocol? Why would it be different? On Sun, Oct 5, 2008 at 8:38 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Has anyone successfully used the IMAP voicemail storage with Microsoft Exchange 2003? Can someone provide a working example configuration? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asteriskt38.com
I was going to write a blog once about the non-existent T.38 support in asterisk hence my purchase of the above domain. It expires in 10 days. T.38 support in asterisk still does not exist but I don't have any time. If someone wants this domain I will offer it for free and can send push it to your enom account since I was going to allow it to expire anyways. The only condition would be that you do not use it for a commercial use, i.e. you don't try to sell a t.38 module for 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: real 2 line phone vs. 1 line and call waiting
On Tue, Sep 30, 2008 at 9:23 AM, Lyle Giese [EMAIL PROTECTED] wrote: 1) a two line phone can register with two different * servers or sip carriers. Many phones/ATA with multiple lines only allow 1 server and multiple registrations! On Tue, Sep 30, 2008 at 6:29 PM, Lyle Giese [EMAIL PROTECTED] wrote: I have never been convinced that VM via email is a convenence. You have to use the loudspeakers on the PC or headphones, which is not as convenient as a handset. Not to mention the privacy issues/problems using loudspeakers for VM. Do you want your kids/wife overhearing your customer that is upset with you? I find it very convenient because I use a Windows Mobile phone with an Exchange server. So if someone leaves a message while I am out of the office 1) I am (pretty much) instantly notified 2) I can listen to the message (after download which takes 2-3 seconds normally) without having to place a phone call, which avoids using airtime and is just faster than placing a call, going through the menu, listening to all other messages, etc. And I know who the caller is beforehand so I know if the message needs attention right then and there or if it can (or should) wait until later. each to his own I suppose. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 3.1.0RevB
Could someone please tell me where to download Polycom 3.1.0RevB? Polycom.com is not possible. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No reply to our critical packet
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this critical packet -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops app5*CLI sip show peer 17865221569 app5*CLI * Name : 17865221569 Secret : Set MD5Secret: Not set Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags: Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : CENSORED MaxCallBR: 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 74.CENSORED.213 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:[EMAIL PROTECTED] app5*CLI core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G723 on asterisk 1.4.1
On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cheap FXO Card?
I have many of the Intel PCI modems in the field working for some time, but I am trying to find a source for more of them. IMO places like x100p.com are a rip off -- $40 for a PCI modem? I recall getting the AMI modems a few years ago for $10. So does anyone know where I can find the PCI WinModem that is detected as X100P or X101P for a better price? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] credit card processing
On Sat, Sep 27, 2008 at 6:52 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi Guys On the website, we already accept credit card by sending users to paypal website where we have an account. PayPal does have a service that is more like a traditional merchant service. I don't know if they have a real API that you can integrate into your system, however. Now, we want to do the same with an IVR where people can call a number, enter their credit card number and expiration date. This should be rather easy. Any traditional online merchant account. When you obtain a merchant account there are (simplified version follows) two parties involved, the bank that process the transactions and the gateway that accepts the transactions from the merchant (you) and sends them to the bank to be processed, in real time. Authorize.net is a very popular gateway supported by most e-commerce software. The point is that the Authorize.net API is a very popular system -- just about any pre-built e-commerce software supports it. It should be rather simple to create an AGI script which takes the credit card information and interfaces with the Authorize.net. They publish many examples and detailed API documentation so this should be a breeze for any skilled programmer. I strongly recommend that you use the CVV2 and AVS as a minimal means to reduce fraud. But I don't see any service or credit card procession company that offers this. What I want basicly is a service where I can send the credit card number I collected and expiration that and their charge the number and give me a status back. Do you know any company that do this ?? That's exactly the purpose of the Authnet API! Further information can be found here: http://developer.authorize.net/ Authorize.net also sells their gateway service under another name (I cant recall it right now), but everything else is the same. Also, some other gateways support Authorize.net emulation. Chris Bagnall wrote: Most credit card processing gateways require you to have the user's name and address for AVS verification when you perform customer not present transactions. Easy enough to do over a website, but a bit more tricky on the phone. AVS simply verifies the street number and zip code, nothing else. If I live at 123 Maple Street in zip code 77099 and I steal the credit card from someone at 123 Test Ct. in the same zip code I can have things mailed to me and it will pass AVS. Either way, when you are not shipping a physical product the rate of fraud rises dramatically -- you should carefully investigate fraud prevention for your system. Authorize.net provide a service which claims to flag/reduce fraudulent transactions. One of the merchant services I deal with, CDG Commerce (I highly recommend them, their customer service is top notch, but I dont think they will process for a VoIP/calling card service), has another similar system for no cost. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
On Thu, Sep 25, 2008 at 7:34 AM, Rizwan Hisham [EMAIL PROTECTED] wrote: The fax is originated from a fax machine connected to an ata which supports t38. That would be great if Asterisk had true T.38 support. It can pass the T.38 packets it receives to another SIP endpoint (it will do this even if the other device doesn't suppor tT.38 -- which cause the call to drop) but it cannot originate nor terminate T.38 traffic. If you have a VoIP provider or Cisco gateway that support T.38 then that's all you need but if you want to terminate the calls yourself on a T1/E1 T.38 does not help when using 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro [EMAIL PROTECTED] wrote: ATAs work OK I guess, just make sure to use a loss less codec such as ULAW. Since the OP stated he is using E1 lines then he should probably be using alaw instead. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail from an unknown caller
When I get a voice message from an unknown caller it will say Message from telephone number and just not say any number. I was wondering if I can manually set the caller ID in this case to be something that the Voicemail app will recognize so it will read out Message from an unknown caller ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fallback on a fallback
I have two sites running Asterisk PBX. Normally the inbound calls go
through a 3rd (colocated) server and are routed via IAX to the site
(the site registers with the main server)
I created a macro that tries to ring one location and then another.
Each site explicitly Answer() the call even though it will only ring
all the sip phones at the relevant location. When fall back is in
effect it goes to the other location and then the other PBX server
rings the same phones (they register to both servers, the server at
the other location via VPN). It was implemented this way because at
the time there was a hardware stability issue.
Now I want to add a 3rd failback via a PSTN line. This will be done
from the main colocated server so even if the internet at the
location is down calls go to the PBX via the PSTN and if the PBX
server catches fire we setup some Weco 2500 clones (in red) as further
protection. But the issue here is that if we tell it to ring (in this
order) site 1, site 2 and then PSTN and the internet to site 1 is down
it will go to site 2 and be answered but since the internet is down
so is the VPN and the call drops there. I can change that, but if only
the PBX server is down (and not the internet or VPN) then I don't want
to use the PSTN line because capacity is only 1 call inbound or
outbound and any subsequent callers would get a busy tone. I also
don't want to send the call out of site 2 directly due to bandwidth
concerns.
Does anyone have a suggestion on how to implement this?
Current setup is exactly as follows
MAIN:
;exten = 13057221371,1,Macro(welcome-message)
;exten = 13057221371,n,Macro(site-fallback,site1/4997|site2/4997|7|7)
[macro-site-fallback]
; ${ARG1) Dialstring 1
; ${ARG2} Dialstring 2
; ${ARG3} Ringtime Peer 1
; ${ARG4} Ringtime PEER 2
exten = s,1,Playtones(ring)
exten = s,2,Dial(${ARG1},${ARG3},m)
exten = s,n,Goto(s-${DIALSTATUS},1)
;exten = s-NOANSWER,1,
;exten = s-BUSY,1,Macro(all-circuits-busy)
;exten = s-BUSY,n,Hangup
exten = _s-.,1,GoTo(s-BACKUP,1)
exten = s-BACKUP,1,Dial(${ARG2},${ARG4},m)
exten = s-BACKUP,n,Goto(s-BACKUP-${DIALSTATUS},1)
exten = s-BACKUP-NOANSWER,1,Macro(no-answer)
exten = s-BACKUP-NOANSWER,n,Hangup
exten = s-BACKUP-BUSY,1,Macro(all-circuits-busy)
exten = s-BACKUP-BUSY,n,Hangup
exten = _s-BACKUP.,1,Macro(network-error)
exten = _s-BACKUP.,n,Hangup
Site 1 or 2 (they are basically identical) but FWIW this is the config
of site 2 for failover of site 1:
exten = 4997,1,Answer
exten = 4997,n,Set(CALLERID(name)=CM Fallback Service})
exten =
4997,n,Dial(SIP/401SIP/402SIP/403SIP/404SIP/405SIP/406SIP/407SIP/408SIP/409SIP/410,90,r)
exten = 4997,n,Playtones(ring)
exten = 4997,n,Wait(1)
exten = 4997,n,VoiceMail(499|u)
pbxserver-sitetwo*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
410/410192.168.12.111 D 5060 Unmonitored
409/409192.168.12.100 D 5060 Unmonitored
408/408192.168.12.116 D 5060 Unmonitored
406/406(Unspecified)D 0Unmonitored
405/405192.168.12.223 D 5060 Unmonitored
404/404(Unspecified)D 0Unmonitored
403/403192.168.12.248 D 5060 Unmonitored
402/402(Unspecified)D 0Unmonitored
401/401192.168.12.119 D 5060 Unmonitored
210/210192.168.0.253D 5060 OK (39 ms)
209/209192.168.0.106D 5060 OK (40 ms)
208/208192.168.0.190D 5060 OK (40 ms)
207(Unspecified)D 0UNKNOWN
206/206192.168.0.194D 5060 OK (38 ms)
205/205192.168.0.105D 5060 OK (43 ms)
204/204192.168.0.173D 5060 OK (39 ms)
203/203192.168.0.126D 5060 OK (37 ms)
202/202192.168.0.187D 5060 OK (39 ms)
201/201192.168.0.176D 5060 OK (40 ms)
501/501(Unspecified)D 0UNKNOWN
20 sip peers [18 online , 2 offline]
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[asterisk-users] openSUSE Asterisk Packages
Does anyone know who maintains the asterisk packages in the openSUSE buildservice? They are not updating Zaptel with their kernel updates and I want to get that matter corrected. I submitted to them a bug report but they seem to not care... https://bugzilla.novell.com/show_bug.cgi?id=407408 ... usually within 24 hours a bugreport is assigned or some sort of comment is made. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spam Filter
Does anyone know of a spam filter that will work with 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spam Filter
Doesn't have to be Voip-originated.. One guy out of his apartment in Texas constantly orders phone lines, disconnects them after a few days of continuous dialing. http://customcampaigns.net/political.html He is not calling with a political message but instead simply harvesting phone numbers. What purpose Mr. Maxi does this for is irrelevant, it's still illegal. Point is SpamAssassin analyzes certain values in an email before its delivered to determine if its spam or not. I am looking for something that works in the same fashion but for phonecalls be they transmitted via the PSTN, PRI, ISDN, SIP or IAX (etc) If the telemarketers followed the laws this would not even be a concern. On Mon, Jun 30, 2008 at 12:16 PM, Brian J. Murrell [EMAIL PROTECTED] wrote: On Mon, 2008-06-30 at 12:03 -0400, Andrew Joakimsen wrote: Does anyone know of a spam filter that will work with Asterisk? What does spam have to do with Asterisk? Or do you mean spit perhaps? http://en.wikipedia.org/wiki/VoIP_spam ? Probably the same techniques such as whilelisting, blacklisting and greylisting are going to have to be applied. It will be much more difficult however as there is no digital form of SPIT that can be analysed before delivery and reported to clearinghouses. Then again, isn't SPIT just telemarketing and regulated by the same (albeit jurisdictionally local) rules such as Do Not Call lists and so on? b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding funcionatlity to asterisk?! is it possible?!
Right now the issue I see is you are using overlapping extensions
so maybe that's not working as expected?
you have in context sipura line exten 201, exten 201 included from
context spa and also exten 2xx included from context spa.
What you want to do with sending calls elsewhere if they are not
completed look at DIALSTATUS, e,g,:
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - SIP DEVICE
; ${ARG2} - ringing seconds
; ${ARG3} - vm-box-Nr.
;
exten = s,1,Macro(docid)
exten = s,2,Dial(SIP/${ARG1},${ARG2},r); Ring
the ${ARG1} interface, ${ARG3} seconds maximum
exten = s,3,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Playback(silence/1)
exten = s-NOANSWER,2,Voicemail(${ARG3},u) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,3,Goto(s-${VMSTATUS},1)
exten = s-USEREXIT,1,Playback(cancelled)
exten = s-USEREXIT,2,Playback(goodbye)
exten = s-USEREXIT,3,Hangup
exten = s-SUCCESS,1,Playback(goodbye)
exten = s-SUCCESS,2,Hangup
exten = s-FAILED,1,Playback(sorry-youre-having-problems)
exten = s-FAILED,2,Playback(please-try-again-later)
exten = s-FAILED,3,Playback(goodbye)
exten = s-FAILED,4,Hangup
exten = o-CHANUNAVAIL,1,Goto(o-BUSY,1)
exten = s-BUSY,1,Playback(silence/1)
exten = s-BUSY,2,Voicemail(${ARG3},b) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,3,Playback(goodbye) ; If they press #,
return to start
exten = s-BUSY,4,Hangup
exten = o,1,Goto(o-${DIALSTATUS},1)
exten = _o-.,1,Goto(o-NOANSWER,1)
exten = o-BUSY,1,Goto(s,2)
exten = o-NOANSWER,1,Playback(please-try-again)
exten = o-NOANSWER,2,GoTo(s-NOANSWER,2)
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else
as no answer
exten = a,1,Playback(this-is-the-voice-mail-system)
exten = a,2,VoicemailMain(${ARG3}) ; If they press *,
send the user into VoicemailMain
For the directory, there's a directory application built into the
voicemail system. You might want to check that out, if it fits your
needs then it's probably the simplest solution.
On Sat, Jun 14, 2008 at 5:56 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.
i want to add a step for asterisk to follow which is, when a sip extension
doesn't answer or its offline, instead of immediately transferring to voice
mail, i want it to dial that sip holder's number so it transfers the call to
his cellphone for example. and if he didn't answer his cellphone its then
that i want it to direct it to voice mail.
i want to add another item to the operator menu, instead of just receiving
the call and telling the caller to either dial extension or 100 for
operator, i want asterisk to offer the caller an additional option like for
example pressing 2, would direct you to a list of key personnels with their
respective extensions.
please find below my extensions.conf:
[sipura-line]
exten = 201,1,Answer() ; Answer inbound calls
exten = 201,2,Playback(silence/1)
exten = 201,3,Background(simzy1) ; input an extension
exten = 201,4,Wait(8)
include = spa
exten = 201,n,Hangup()
[spa]
exten =_201,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED])
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED])
exten = _2XX,3,HangUp()
exten =_01,1,Dial(SIP/200)
exten = 203,1,VoicemailMain
exten = _2XX,1,Dial(SIP/${EXTEN},15)
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Re: [asterisk-users] Logitech DiNovo Mini keyboard with myth
On Fri, Jun 6, 2008 at 9:03 PM, OCG Technical Support [EMAIL PROTECTED] wrote: Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) Is that in extensions.conf or chan_dinovo.conf? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DTMF dialing
On Feb 12, 2008 10:40 AM, Ian [EMAIL PROTECTED] wrote: Hi all, its been quite a busy few day with pc's packing up etc, I recompile my whole asterisk today using zaptel 1.4.7.1 and now the problem is miraculously fixed, I will be sending this report to Digium bugs as well. Just a quick heads up for the order in which I had to recompile in order for this to work Recompile Zaptel Restart Asterisk, asterisk doesn't pick up the zap channels Recompile Libpri Retart Asterisk, still no zap channels Doing the thing I was hoping to skip, Recompile Asterisk Everything in working order Did I miss something for me to have to only recompile zaptel, or is that the way of doing things? Thank you all for your support Please scroll down to see the answers to my own stupid questions :-) Asterisk depends on Zaptel (well chan_zap and the respective codecs do) so always make sure to install first LibPRI, then Zaptel then Asterisk FWIW in the wav recording you sent there is alot of static. I am playing back with amaroK 1.4.7 of openSuSE. On Feb 12, 2008 11:50 AM, Andres Jimenez [EMAIL PROTECTED] wrote: I am having similar problems running the same versions of Asterisk, libpri zaptel. The Asterisk bug (http://bugs.digium.com/view.php?id=11855) was supossed to be related to FXO only, but I am having issues with a PRI line and Digium's TE120P. Do you guys think it can be the same issue? Maybe it is related but with PRI Asterisk does not generate any tone it sends a signal regarding your keypress. If you are using SIP phones make sure the dtmfmode in use is RFC2833. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
