Re: AW: [asterisk-users] 7970 sip success
I also have nat=no qualify=no I haven't checked to see if they're necessary. I think I've read some suggestions that the phone needs to be on the same subnet as the asterisk server but I haven't been able to check that either. On Fri, 2007-04-27 at 09:19 +0200, René Enskat wrote: Mmm i have set it in my MySQL Database in the row: Variables buggymwi = yes But can't see MWI Regards rene -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Zachary Whitley Gesendet: Freitag, 27. April 2007 00:09 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] 7970 sip success MWI also works with Asterisk 1.4.2 with buggymwi=yes in sip.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7970 sip success
I managed to upgrade the phone to 8.2.2SR1 after renaming jar70sip.8-2-2ES1.sbn to Jar70sip.8-2-2ES1.sbn but the phone would continually say Registering and the red X next to the phone icon. The phone would eventually time out and couldn't make incoming or outgoing calls. Then I disabled registering with the proxy with the following line in the config: registerWithProxyfalse/registerWithProxy The Registering line didn't appear but the red X was still there. I could make outgoing calls but couldn't receive them. Next I deleted the following lines. backupProxy192.168.20.2/backupProxy backupProxyPort5060/backupProxyPort emergencyProxy192.168.20.2/emergencyProxy emergencyProxyPort5060/emergencyProxyPort outboundProxy192.168.20.2/outboundProxy outboundProxyPort5060/outboundProxyPort and changed the registerWithProxy back to true as follows: registerWithProxytrue/registerWithProxy The phone no longer got stuck with Registering, the red X is gone, and I can make and receive calls. I'm not sure if there are other settings that are critical to this working but this was the last thing I tried before it started working. I'll post this to the wiki shortly. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7970 sip success
MWI also works with Asterisk 1.4.2 with buggymwi=yes in sip.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7921G running linux
I was just watching the informational video on cisco's web site about the 7921G and they guy mentions that the phone is running Linux. Anyone know if they've released the source code? This page confirms that the phone is running Linux http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0900aecd80601788.shtml The phone doesn't support sipyet ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 / SIP tftp configs
On Wed, 2005-08-24 at 12:44 -0400, Asterisk User Group wrote: I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? Change it to UNPROVISIONED 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? messages_uri: 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** line1_shortname: Home The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small form factor system w/PCI slot
I've been using a Compaq Deskpro EN SFF. They're small, have 3 pci slots, and you can get them up to 1Ghz PIII w/ 512MB of ram on ebay for under $100. Great for testing. When you're done with it throw in a PCI-PCMCIA adapter and turn it into a wireless AP or throw in a network card. They make a great router. You can't beat the price. Corporations bought them by the hundreds and they're all coming off of corporate lease. A TDM400 fits just right. If you want an even smaller package the Compaq Deskpro EN ultra SFF has 2 full size pci slots. --Zach On Fri, 2006-06-09 at 09:34 +0200, Jens Vagelpohl wrote: On 9 Jun 2006, at 02:04, Leo Ann Boon wrote: Jens Vagelpohl wrote: Hi everyone, I'm trying to buy a small form-factor PC system for use with Asterisk and Hylafax in conjunction with a Eicon DIVA Server single-port ISDN card (needs full-size 5V PCI 2.2 slot, but PCI-X compatible). Use is very light - at most a single call at any one time. If the Mac Mini had a PCI slot I'd try to use that one, but oh well ;) You mean PCI-E? If you really need PCI-X, then you're out of luck. PCI-X is only available on server boards. For a single port ISDN, one of those Mini-ITX boxes should work. I built something similar using a Mini-ITX (1GHz CPU) with an AVM Fritz! PCI ISDN card using chan_capi. IIRC, Xorcom has a TS-1 which is a SFF Asterisk server for $500. BTW, I don't think the Mini-ITX mobos can support PCI-E. It's a normal 5.5 V PCI slot, the card can also deal with PCI-X slots as the documentation claims. I'll take a look at Xorcom's offerings, thanks. jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speex QoS
On Mon, 2005-08-08 at 22:20 +1000, Mark Edwards wrote: speex is a codec. it's not a network protocol or a service. you need to be looking to be providing QOS for RTP data, over which the speex encoded data is sent. cheers, Mark On 8/8/05, Adam Robins [EMAIL PROTECTED] wrote: Can anyone out there please tell me what ports Speex uses? I want to set up QoS on switches but I can't seem to find this information anywhere. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. See http://www.speex.org for more information. (They developed it) Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications by providing a free alternative to expensive proprietary speech codecs. Moreover, Speex is well-adapted to Internet applications and provides useful features that are not present in most other codecs. Finally, Speex is part of the GNUProject and is available under the Xiph.org variant of the BSD license. I know you probably can't control it but I hate that psudo corporate lawyer shit at the end of emails. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk rpms (was: Does anyone run Asterisk on FC4? with Digium's TDM40B cards)
There are already some bug reports at bugzilla.atrpms.net on enhancements and bugs in the packages, see http://bugzilla.atrpms.net/buglist.cgi?query_format=advancedshort_desc_type=allwordssubstrshort_desc=long_desc_type=substringlong_desc=asteriskbug_file_loc_type=allwordssubstrbug_file_loc=bug_status=NEWbug_status=ASSIGNEDbug_status=REOPENEDemailassigned_to1=1emailtype1=substringemail1=emailassigned_to2=1emailreporter2=1emailcc2=1emailtype2=substringemail2=bugidtype=includebug_id=votes=chfieldfrom=chfieldto=Nowchfieldvalue=cmdtype=doitorder=Reuse+same+sort+as+last+timefield0-0-0=nooptype0-0-0=noopvalue0-0-0= Thanks! ___ I didn't know that there was a bugzilla site setup for atrpms. Thanks for the great repo. I'll make sure to post anything that I find. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P - All extensions have same CallerID
As you can see, the channels are set properly. One thing I did notice is all of the ;; in front of the [ext] sections. Does that seem correct? I removed them and it didn't change anything. Other files that you would like to look at? Thanks, Mike Looks a bit more complicated than it needs to be but I don't think there's anything wrong with the zapata.conf (and friends). Anyone know if modifying configuration files by hand breaks AMP? I'm assuming that that AMP is going to expect a particular setup. The ';' is just a comment character. Everything after a ; is a comment including the other ;'s. the [xxx] I'm guessing is there just to let you know what extension AMP is using for that channel. I think that it is very confusing to use the configuration file syntax in comments. It makes it hard to see what is a comment and what isn't but that's just my opinion. I think the next thing to look at is the extensions.conf file. head wrapped around all of this. The good thing is once I know how to do it, I don't need to ask again. Give a man a configuration and he'll make calls for a day. Teach a man how to configure and he'll make calls for a lifetime ;) I tried usinig AAH first too but found that it got you going with something that sort of works quickly but then trying to work backwards from a complex system was too difficult. There were too many confounding variables. Is it A, is it B is it A and B I've found it easier to start with rpms. You don't have to worry about compiling and installing the system and you get to start from a simple state and work your way up. There are many great sources of info but here are a few I've found helpful: This list google voip-info.org asteriskdocs.org VoIP Telephone with Asterisk by Paul Mahler The sample config files included with asterisk. (Search for theConfigFileYouWantToCheckOut.sample) Please feel free to add to this list if you know any good sources of documentation. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
I've posted my config files in Adobe pdf format at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I think you're either going to get complaints about the pdf files or people are simply going to ignore your question. Is there any reason you chose to post pdf's instead of just posting the ASCII files? And you're really going to hear it when people follow your link and find the file isn't there. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice prompt repository
I was wondering if there would be any interest or support out there for an IVR voice prompt repository, a la atrpms but for voice prompts instead of rpms. I was thinking of something that collected the meta data such as spoken text, gender, file size, speaker ID, language, duration, encoding, MD5, etc. prompts could also be organized into collections almost like IVR themes where a complete set of standard base prompts are collected so you could make one change in your configuration file and all prompts are changed to the new speaker. There could also be a rating for quality of recordings and links to professional services if you needed better quality or specific recordings, etc. It could be like pod casting for IVR. Suggestions, comments, questions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it - unableto cteate channel iax
If you find a wiki page that is incorrect, incomplete or needs any other editing, do it! The rest of the community will be thankful for your help. I don't want to get in the middle of this but what wiki are we referring to? voip-info.org/wiki-asterisk ?? I would be willing to contribute if I knew were to go. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] z-machine + asterisk = fun!
On Sun, 2005-08-07 at 14:59 -0500, Tim Connolly wrote: Wow! Not sure what else to say. This ranks right up there with my ability to open my garage door from asterisk... Sarcasm or serious? Sounds cool to me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote: Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Kumara, Yes, Without problems. Can u install RH9 on ur box? Cheers, ~Madhawa I'm assuming that Madhawa is suggesting that you install RH9. I've installed Asterisk on FC4 with very few problems. Start with a standard FC4 installation then install the following rpms from atrpms.net: asterisk-addons asterisk-sounds zaptel zaptel-devices If you already have it set up in yum you can just use: yum install asterisk asterisk-addons asterisk-sounds zaptel zaptel-devices If the atrpm repo isn't set up in yum just copy the following to /etc/yum.repos.d/atrpms.repo ---CUT--- # # [atrpms] name=Fedora Core 4 - i386 - ATrpms baseurl=http://dl.atrpms.net/fc4-i386/atrpms/stable failovermethod=priority # # requires stable # [atrpms-testing] name=Fedora Core 4 - i386 - ATrpms testing baseurl=http://dl.atrpms.net/fc4-i386/atrpms/testing failovermethod=priority enabled=0 # # requires stable and testing # [atrpms-bleeding] name=Fedora Core 4 - i386 - ATrpms bleeding baseurl=http://dl.atrpms.net/fc4-i386/atrpms/bleeding failovermethod=priority enabled=0 ---CUT--- One little problem. Maybe it's been fixed but last time I checked it wasn't. In the /etc/init.d/zaptel the path to ztcfg is incorrect. Find all references to ztcfg and change them to = /usr/sbin/ztcfg You can copy the sample configs from /usr/share/doc/asterisk-1.0.9/configs/ to get you going. Running asterisk -c -vvv will let you know which ones you need. The rest is going to be specific to your hardware and setup. Good luck. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
On Sat, 2005-08-06 at 14:21 +0100, Julian J. M. wrote: Run memtest86 from the boot menu. You may have faulty RAM. I had the same problem installing CentOs 4... Julian J. M. On 8/6/05, Kumara Jayaweera [EMAIL PROTECTED] wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara Where in the install sequence is it hanging? Installation of Asterisk or installation of FC4? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM400P - All extensions have same CallerID
On Sat, 2005-08-06 at 11:36 -0500, Larry Shields wrote: Check your zapata.conf file. Your terminal profile options section under [channels] should be ended by adding the associated channel, i.e. channel = 1 Sample two port config: [channels] ; ; Default language ; language=en ; Default terminal profile FXS PORT1 ; context=internal signalling=fxo_ks usecallerid=yes callerid=Line 1 2001 callwaiting=yes callwaitingcallerid=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 mailbox=2001 group=1 channel = 1 ; Default terminal profile FXS PORT2 ; context=internal signalling=fxo_ks usecallerid=yes callerid=Line 2 2002 callwaiting=yes callwaitingcallerid=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 mailbox=2002 group=1 channel = 2 Mike Putnam [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... I've been searching the forums and on the list to see if this has been addressed. If it has, could someone point me to the thread to fix or at least acknowledge it is an issue and what is causing it. Posting to the list was last resort as I couldn't find a solution anywhere else. Setup: [EMAIL PROTECTED] 1.3 (this is my first system, so path of least resistance) Digium TDM400P (2 FXS on ports 1 2, 2 FXO on 3 4) My FXS extentions are 201 and 210. Both of my FXS extentions report being the same extension when doing a *65. It is always the last extension configured as well, in this case 210. In fact, when recording prompts, even if I'm on extension 201, I have to tell the system that I'm on extension 210. I deleted the second extention and 201 started acting properly. I added a new extension (x400) and both are being reported as 400 now. To make things even more confusing, calls can be placed to the proper extension, so it seems to be something with CallerID, but I can figure out what it is. As stated, this was my first system so I used AAH to get up and running fast and then work backwards to learn and use the system. I started with AAH 1.1 and everything worked fine for about 2 weeks, then I noticed a problem when I started having messages on x201 but the light wasn't blinking. To the best of my memory, when I started out I didn't have this problem. Both extensions were acting normally. I didn't put on any AAH or Asterisk updates. I'm running AAH 1.3 now just to see if it would fix the problem. Any and all suggestions are greatly appreciated. Mike After messing around with the zapata.conf file I have a few questions. What determines the scope of commands in the zapata.conf file? In zapata.conf we're defining channels, correct? Since the config file is kind of flat. (everything is under [channel]) What is the official start and end of a channel block? Does it start with context= and end with channel= ? Why isn't there separate blocks? ie. [channel1] ... ... ... [channel2] ... ... ... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On Sat, 2005-08-06 at 13:02 -0400, Doug Lytle wrote: Andrew Kohlsmith wrote: On Friday 05 August 2005 21:31, Doug Lytle wrote: exten = s,1,Dial(SIP/PHONE1,15,rt) exten = s,2,Dial(SIP/PHONE4,15,rt) Using 'r' flags makes baby Jesus cry. Stop doing that. Excuse me? r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM400P - All extensions have same CallerID
On Sat, 2005-08-06 at 12:34 -0500, Andrew Latham wrote: No real start, Channel ends and the following is assumed to be the next channel. Ok, so the scope of the configuration is from channel= to channel= statement with the configuration for the channel coming before the channel statement. As in... these=are configs=for the=first channel=1 these=are configs=for the=second channel=2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
On Sun, 2005-08-07 at 02:54 +0600, Madhawa Jayanath wrote: Zachary Whitley wrote: On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote: Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? --cut-- Hi Zachary! I'm not suggesting installing RH9 :) I mean, whether he can install RH9 on same box without any problems. He said he couldn't install @least FC4 on the box, do you have any idea? problems of IRQ sharing with the card? I guess the first question is does it still freeze if you remove the TDM40B cards? Where in the installation process does it freeze? Have you tried a text only install instead of the graphical? What method are you using to install? nfs? ftp? local cdrom? What file system are you using? What hardware? Need info ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: TDM400P - All extensions have same CallerID
On Sat, 2005-08-06 at 20:55 -0400, [EMAIL PROTECTED] wrote: As I recall, should channels start as channel=2 and not channel=2? I have all mine config'ed channel = 2 and it works fine... Greg That's correct. My mistake. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
