[asterisk-users] Re: System from AMI
Richard Lyman wrote: if you are unable to get '!' to work... there are other ways of doing this. manager originate can do this, use a local channel and point it at either a context/exten with an echo/system call, You mean like this? [testdelete] exten = myexten,1,System(rm /tmp/test.txt) exten = myexten,2,Hangup Action: Originate Channel: SIP/987987 Context: testdelete Exten: myexten Priority: 1 Callerid: 987987 Timeout: 3 ActionID: uniqueID Can I use this in production use? I need to execute this up to 1000 times a day (in period of few hours). or with the application/data method. Can you please show me some example? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Linksys SPA 3102 causing me problems
Friday, March 30, 2007, 5:02:08 AM, Matt wrote: On 3/29/07, Alan Chandler [EMAIL PROTECTED] wrote: I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next to fix it. Any ideas? I dont know if you have done this but run a sip show peers and make sure that its registered with asterisk. Sounds like it is not registering with asterisk which would allow you to call out but when it tries to call you it dosent have an ip to contact you at. Wehh... He activated the DND function of Linksys. It can be activate with *78 and deactivate with *79. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] Re: System from AMI
Richard Lyman wrote: Action: Originate Application: System Data: /path/to/script Channel: Local/[EMAIL PROTECTED] Context: dummy Exten: 2 Priority: 1 In extensions.conf [dummy] Exten = _X,1,Wait(2) Exten = _X,2,NoOp fyi: manager originate is channel + context + exten + priority OR channel + application + data not both. So, you are saying that this should look like this? Action: Originate Channel: Local/[EMAIL PROTECTED] Application: System Data: /path/to/script And this is all that's needed? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Re: Multi-registration ?
Hi Benny, 29 Mar 2007 12:36:21 +0200, Benny Amorsen [EMAIL PROTECTED]: Indeed, this is the limitation of Asterisk which we hit most often. In theory, it isn't a big problem, because you just put Dial(SIP/fooSIP/bar) in the dial plan. Next: someone with that solution needs to be able to join or leave queues. Ok, a bit of playing around solves that. Then you discover that it's annoying to have one phone ring while you are talking on the other, so you make call groups to handle that problem. Whoops, you need to recognize outgoing calls too, to get them in the call group as well -- but transferred calls should not be counted. Hackety-hack, fixed that one as well. After a while you end up with 10 lines of dial plan per phone. Not nice. It would be much nicer to deal with if the dial command could be told to return BUSY when just one of the devices it calls is busy. Even better would be multi-registration, of course. That's exactly what I meant by simpler : maybe, you can work around single-registration but it seems to me that things would be much more natural with multi-registration : it's up to your hardphone to offer convenient and flexible behaviour in multi-registration : return BUSY when it has to, allow per-registration call forwarding or logout, MWI, ... Snom phones support multline registration. If Asterisk could also support it ... Cheers ___ --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] Re: Is it possible to install CCM on a Linux platform ?
Olivier wrote: Hi, I know this question doesn't exactly relate to the core of this list but I thought it does relate to its hacker spirit. Is it possible to install a Cisco Call Manager 5.X on a non-Cisco appliance ? A friend of mine working for a Cisco VAR told me his colleagues couldn't make it, even for testing purpose. Do you agree ? Regards This really isn't Asterisk users question. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Security on long distance calls
Hello, which kind of method could you use to inhibit long distance calls to _some_ extensions? Is there a way to do it with freepbx or you have to do it manually in the config files? I wouldn't like to set a route password, because that is not confortable for the pbx operator. I just would like the operator being able to call whatever number, while the extensions should only be able to make local calls. Thanks Stefano ___ --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] Security on long distance calls
For operator: [longdistance] include= local include= International for extensions: [localcalls] include= local now assign longdistance context to operator and localcalls context to every other user for whom you want to restrict intl calls [local] should include all local extension codes [International] should include all international extension codes you get my point? On 3/30/07, Stefano Corsi [EMAIL PROTECTED] wrote: Hello, which kind of method could you use to inhibit long distance calls to _some_ extensions? Is there a way to do it with freepbx or you have to do it manually in the config files? I wouldn't like to set a route password, because that is not confortable for the pbx operator. I just would like the operator being able to call whatever number, while the extensions should only be able to make local calls. Thanks Stefano ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] Call dies when I press *
What kind of phone are you using? some devices which connect you pstn phone to voip network have their own special dtmf keys defined inside their configuration to perform special actions, maybe thats your problem. also check for your dtmf setting. dtmf settings should be same on both sides. On 3/29/07, Mike Diehl [EMAIL PROTECTED] wrote: Actually, it turns out that sometimes I can't get ANY DTMF to work. I can call a local phone number and log into my voicemail system at work. But my wife is unable to dial a toll free number and use their IVR. Hope this helps. On Wednesday 28 March 2007 16:58, Mike Diehl wrote: Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] bugetone 200's
From: [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 18:29:07 +0100 how do these phones perform? ok for office use? work well with asterisk? any info would be appreciated. I have a bugetone 100 at home. It appears to work with Asterisk and basic quality seems to be OK (G.711) including speaker phone. But I use none of its built-in features such as transfer or even hold so the elaborative right-hand side buttons are pretty much useless. One annoying problem about buttons: it doesn't have a Redial key, but has a Send key to please some annoying VoIP system. And if you need to redial, press that Send key. The designer must be out of his or her mind. A lesser button issue: the mute key is marked as Mute/Del. As such, could easily be overlooked by a casual user looking for Mute key. I would much appreciate a stand-alone Mute key, and won't mind having a combined Flash/Del key. You guessed it: Flash stands alone. (I do appreciate the corner location of the Mute/Del key. But Del is really not that useful to qualify for this premier location.) Yuan Liu ___ --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] Call Waiting problems
If you are using sip then you should look for the call-limit option in sip.conf file. On 3/30/07, Lachek Butalek [EMAIL PROTECTED] wrote: Situation, simple home setup: * Trixbox 2.0 * Feature Codes installed * GNet PA-168V based ATA * Cheesy cordless analogue phone From what I gather, dialing *70 from the handset should activate Call Waiting. All it seems to do is change the message The person at extension is on the phone to ring ring The person at extension is unavailable. The person speaking on the phone at the time of the second incoming call hears no indication that another call is incoming. Part of the problem is that I have no idea how the feature should work when it's functional. Could someone help me troubleshoot this, or point me in the right direction? It seems as though, as a very basic feature, not a lot of documentation is written about it. Thanks! ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] Setting rxgain per channel
From: Delca [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 18:39:37 -0300 How do I set rxgain per channel on zapata.conf? I've a TDM400 with 2 FXS. Does FXS even use rxgain? To set rxgain for an FXO channel, simply put the entry before saying channel =. Hope this helps. Yuan Liu Thank you! Santiago del Castillo ___ --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] Asterisk-Addon-1.4.0 MySQL
I can't find anything about Asteirsk-Addon-1.4 MYSQL problem from googling. I thought it would be my error but surely not just tried asterisk 1.2.17 with addon 1.2.5 and it work. Does anyone else having problem make res_config_mysql, cdr_addon_mysql and app_addon_sql_mysql in addon-1.4? Thanks for sharing There are no res_config_mysql and cdr_addon_mysql module after. /configure make all make install in asterisk module directory. It would be great if someone can give me some hint. I never experienced this before with 1.2 releases. Is there something changed on 1.4 releases? Or am I missing something. I am about to pull my hair out after many hours looking at the monitor. uname -a Linux 2.6.20-1.2933.fc6 #1 SMP Mon Mar 19 10:42:48 EDT 2007 i686 i686 i386 GNU/Linux rpm -qa | grep -i mysql mysql-5.0.27-1.fc6 php-mysql-5.1.6-3.4.fc6 mysql-devel-5.0.27-1.fc6 perl-DBD-MySQL-3.0007-1.fc6 mysql-server-5.0.27-1.fc6 *CLI core show version Asterisk 1.4.2 on a i686 running Linux on 2007-03-28 05:45:27 UTC *CLI show modules like mysql Module Description Use Count 0 modules loaded Thank You K -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.22/739 - Release Date: 3/29/2007 1:36 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.22/739 - Release Date: 3/29/2007 1:36 PM ___ --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] Off Topic: Open Source USB Softphone
On Fri, 30 Mar 2007, laurent schweizer wrote: openwengo.org Looks good - however question/answer 1 in their FAQ: http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsNg#a1 Can WengoPhone 2.0 be used with any SIP provider? Not right now. However, it is the item with the highest priority on our todo list, apart from having a 2.0 release. So expect to see this feature implement right after the first NG release. But it is open source, so should be easy to hack something in :) Gordon Laurent 2007/3/29, Gordon Henderson [EMAIL PROTECTED]: On Thu, 29 Mar 2007, Luis Claudio Santos wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? Maybe not quite what you want, but I used a Yealink USB phone with Linux and there was a driver for it that would let you read the keyboard, and program the display - it wasn't a bit-mapped display though. Gordon ___ --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] maximum simultaneous calls
From: Mark Quitoriano [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 23:05:57 +0800 Hi, what could be the maximum simultaneous calls can asterisk do? i read about the asterisk business edition review[1] and it can only handle 120 simultaneous calls? i'm using 1.2.x branch of asterisk and i use more or less 90 simultaneous calls. [1] http://www.voiptalk.org/products/Asterisk+Business+Edition What about http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning? People reports all kinds of numbers above 120. The answer partially depends on your hardware. Simultaneous calls can also mean very different things under different circumstances, as the page will tell you. If there is no transcoding/NAT'ing/in-band signaling, simultaneous calls can mean SIP set-ups only. You can see extremely high numbers even on ancient equipment. If everything is in-band and you are using CPU-intensive CODECs, the number will drop sharply. It also varies with types of channels, i.e., whether you use PSTN, IAX, SIP, H.323. But still, I don't think 120 is any limit. Yuan Liu ___ --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] web based sip phone
hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and .. best MAni Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097 ___ --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] Asterisk 1.4 with Digium B410P - Timing problem
Hi list, we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686 kernel. The server has 2 B410P cards plugged in. No other card. We installed Asterisk 1.4 trunk with zaptel trunk, ran make b410p, the install mISDN1.1.0 (for bug 9064) configure and compile Asterisk with chan_misdn, all is fine. In misdn-init.conf we have added option=1,master_clock. Asterisk is up and running, voicemail, echo test, demo, MOH, everything works well. Now we want to add conferences with meetme. So we load zaptel module who created /dev/zap/channel|ctl|pseudo|timer|transcode. Asterisk is running under asterisk:asterisk,we added this user to dialout group to have the good rights on those files. Problem: we can't open conferences, we always have [Mar 30 10:55:37] WARNING[28302]: chan_zap.c:896 zt_open: Unable to open '/dev/zap/pseudo': No such device or address [Mar 30 10:55:37] ERROR[28302]: chan_zap.c:7631 chandup: Unable to dup channel: No such device or address [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:758 build_conf: Unable to open pseudo channel - trying device [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:761 build_conf: Unable to open pseudo device What's wrong? The B410P being a Digium card should gave us a timing source? Do we forgot to compile some module (no one wxxx from zaptel directory is activated)? More generally, how you get the timing working with B410P cards and 1.4? We tried another way, ztdummy: by loading the module, the meetme application start to work but ... no more audio in all applications (voicemail, meetme, MOH, echo-test, demo,...)! In log we found Mar 29 22:40:36 SrvPhone kernel: Zapata Telephony Interface Registered on major 196 Mar 29 22:40:36 SrvPhone kernel: Zaptel Version: SVN-branch-1.4-r2351 Mar 29 22:40:36 SrvPhone kernel: Zaptel Echo Canceller: MG2 Mar 29 22:40:39 SrvPhone kernel: ztdummy: RTC rate is 1024 Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:40:39 SrvPhone kernel: Registered tone zone 0 (United States / North America) Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:41:10 SrvPhone last message repeated 1514 times Mar 29 22:42:11 SrvPhone last message repeated 3050 times Mar 29 22:42:57 SrvPhone last message repeated 2348 times Removing ztdummy give audio back but no more meetme :-( Before opening a bug, we would like to know if some of you have a similar working setup with B410P and meetme. Thanks for your feedback. -- Daniel ___ --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] Unsetting Global Vars
From: Johann Hoehn [EMAIL PROTECTED] Date: Wed, 28 Mar 2007 16:45:28 -0500 How do I clear a global variable for good? I have a situation of needing to use global variables to aide in channel communication, but will be changing the name within a defined scope. Not sure if I understand what clear a variable mean. I don't think there is a concept like unset in Asterisk. If you want to make sure a used variable does not cause side effects, simply set it to null string. Additional Background... I want to get a variable from a channel (child) that is created by another channel (parent), however the execution of the parent channel does not continue until the child channel is gone. So I want to use a global variable as 'scratch' space and later the parent to grab it. Basically I need to be able to do the opposite of variable inheritance. I need to propagate a variable status up the channel chain instead of down. I feel the need to propagate a variable up the chain from time to time. But I still don't understand why this is necessary in your case, much less how this relates to the need to unset. Maybe you can give more specifics, even pseudo code. Yuan Liu -- Johann Hoehn Project Coordinator, Administration Direct: 270-707-2040 x 4011 Ecommerce Corporation (www.ecommerce.com) ___ --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] Asterisk 1.4 with Digium B410P - Timing problem
did you modprobe ztdummy? On 3/30/07, Administrator TOOTAI [EMAIL PROTECTED] wrote: Hi list, we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686 kernel. The server has 2 B410P cards plugged in. No other card. We installed Asterisk 1.4 trunk with zaptel trunk, ran make b410p, the install mISDN1.1.0 (for bug 9064) configure and compile Asterisk with chan_misdn, all is fine. In misdn-init.conf we have added option=1,master_clock. Asterisk is up and running, voicemail, echo test, demo, MOH, everything works well. Now we want to add conferences with meetme. So we load zaptel module who created /dev/zap/channel|ctl|pseudo|timer|transcode. Asterisk is running under asterisk:asterisk,we added this user to dialout group to have the good rights on those files. Problem: we can't open conferences, we always have [Mar 30 10:55:37] WARNING[28302]: chan_zap.c:896 zt_open: Unable to open '/dev/zap/pseudo': No such device or address [Mar 30 10:55:37] ERROR[28302]: chan_zap.c:7631 chandup: Unable to dup channel: No such device or address [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:758 build_conf: Unable to open pseudo channel - trying device [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:761 build_conf: Unable to open pseudo device What's wrong? The B410P being a Digium card should gave us a timing source? Do we forgot to compile some module (no one wxxx from zaptel directory is activated)? More generally, how you get the timing working with B410P cards and 1.4? We tried another way, ztdummy: by loading the module, the meetme application start to work but ... no more audio in all applications (voicemail, meetme, MOH, echo-test, demo,...)! In log we found Mar 29 22:40:36 SrvPhone kernel: Zapata Telephony Interface Registered on major 196 Mar 29 22:40:36 SrvPhone kernel: Zaptel Version: SVN-branch-1.4-r2351 Mar 29 22:40:36 SrvPhone kernel: Zaptel Echo Canceller: MG2 Mar 29 22:40:39 SrvPhone kernel: ztdummy: RTC rate is 1024 Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:40:39 SrvPhone kernel: Registered tone zone 0 (United States / North America) Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:41:10 SrvPhone last message repeated 1514 times Mar 29 22:42:11 SrvPhone last message repeated 3050 times Mar 29 22:42:57 SrvPhone last message repeated 2348 times Removing ztdummy give audio back but no more meetme :-( Before opening a bug, we would like to know if some of you have a similar working setup with B410P and meetme. Thanks for your feedback. -- Daniel ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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] web based sip phone
From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT) hello is any web based sip phone? The easy answer is yes. Search for Java SIP phone. Some of them can be deployed on the Web. Yuan Liu for example: a user after logining in, view a configured sip phone, and .. best MAni ___ --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] web based sip phone
On 30 Mar 2007, at 10:05, Pezhman Lali wrote: hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and .. I know you asked for sip, but we have one for IAX - which makes it much more firewall/NAT friendly. (It is a java applet - so cross platform and it is lightweight and skinnable in HTML) If you have a Tesco Internet Phone account, you can try it on: http://www.phonefromhere.com If you really want SIP, there are a couple of activeX Sip components like vaxvoip - but they don't work on linux/macs/firefox. If you dig around on http://www.voip-info.org/ You will find a long list with a (very) few entries that might meet your requirements. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] just on my LAN
From: Josu Lazkano Lete [EMAIL PROTECTED] Date: Wed, 28 Mar 2007 09:59:16 +0200 hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 Depends on what you use it for. You certainly don't need Libpri. You may need Zaptel (specifically the ztdummy driver) if you want to run meetme, i.e., conference (some document also mention music on hold). Not sure what sounds and addons are for. (Basic sound files are included with distribution at least up to 1.2.16.) Yuan Liu thanks ___ --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] UK PRI and outgoing CLI FYI
On 29 Mar 2007, at 23:33, Steve Kennedy wrote: On Thu, Mar 29, 2007 at 10:40:50PM +0100, Julian Lyndon-Smith wrote: We only present the 6 digits ... and they give us 6 digits. For our outbound calls, for the the numbers 01702 1234[00-99] we have to present 1234[00-99]. BT isdn pri line. Weird, seems they're inconsistant or there's some oddity at the driver level? For NTL it is negotiable. Basically we send as few digits as needed to unambiguously identify the number from the set of possible numbers. So when we ported some numbers in from a different exchange we had to increase the number of digits we sent! Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Asterisk 1.4 with Digium B410P - Timing problem
Marco Mouta a écrit : did you modprobe ztdummy? Marco, thanks for your answer. The second way we tried -see end of message- was with ztdummy, so yes, it was modprobed ;-) On 3/30/07, *Administrator TOOTAI* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi list, we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686 kernel. The server has 2 B410P cards plugged in. No other card. We installed Asterisk 1.4 trunk with zaptel trunk, ran make b410p, the install mISDN1.1.0 (for bug 9064) configure and compile Asterisk with chan_misdn, all is fine. In misdn-init.conf we have added option=1,master_clock. Asterisk is up and running, voicemail, echo test, demo, MOH, everything works well. Now we want to add conferences with meetme. So we load zaptel module who created /dev/zap/channel|ctl|pseudo|timer|transcode. Asterisk is running under asterisk:asterisk,we added this user to dialout group to have the good rights on those files. Problem: we can't open conferences, we always have [Mar 30 10:55:37] WARNING[28302]: chan_zap.c:896 zt_open: Unable to open '/dev/zap/pseudo': No such device or address [Mar 30 10:55:37] ERROR[28302]: chan_zap.c:7631 chandup: Unable to dup channel: No such device or address [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:758 build_conf: Unable to open pseudo channel - trying device [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:761 build_conf: Unable to open pseudo device What's wrong? The B410P being a Digium card should gave us a timing source? Do we forgot to compile some module (no one wxxx from zaptel directory is activated)? More generally, how you get the timing working with B410P cards and 1.4? We tried another way, ztdummy: by loading the module, the meetme application start to work but ... no more audio in all applications (voicemail, meetme, MOH, echo-test, demo,...)! In log we found Mar 29 22:40:36 SrvPhone kernel: Zapata Telephony Interface Registered on major 196 Mar 29 22:40:36 SrvPhone kernel: Zaptel Version: SVN-branch-1.4-r2351 Mar 29 22:40:36 SrvPhone kernel: Zaptel Echo Canceller: MG2 Mar 29 22:40:39 SrvPhone kernel: ztdummy: RTC rate is 1024 Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:40:39 SrvPhone kernel: Registered tone zone 0 (United States / North America) Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:41:10 SrvPhone last message repeated 1514 times Mar 29 22:42:11 SrvPhone last message repeated 3050 times Mar 29 22:42:57 SrvPhone last message repeated 2348 times Removing ztdummy give audio back but no more meetme :-( Before opening a bug, we would like to know if some of you have a similar working setup with B410P and meetme. Thanks for your feedback. -- Daniel ___ --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] web based sip phone
thanks Yuan I was search the best result is sipfoundary.org but it's client is not spesific for my purpose, but it will be. is any better answer for this searching? best Mani --- Yuan LIU [EMAIL PROTECTED] wrote: From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT) hello is any web based sip phone? The easy answer is yes. Search for Java SIP phone. Some of them can be deployed on the Web. Yuan Liu for example: a user after logining in, view a configured sip phone, and .. best MAni ___ --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 Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 ___ --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] Re: Friday asterisk users live conference/podcast at 12:30PM EDT
Just one last reminder to please join us at 12:30 PM EDT today, Friday
to talk about asterisk, ask questions, talk about your solutions,
share your experience as users, developpers or providers at the first
weekly Asterisk Users Conference/Podcast:
Check http://x2z.eu for technical info. The basic idea is:
${EXTENSION},1,Dial(SIP/[EMAIL PROTECTED],60,D(22622#0123456789#))
Where 0123456789 is your 10-digit PIN from Talkshoe.com. (You can
choose your own PIN as long as it's not already taken. Most people use
their phone number.)
The conference will last as long as there are people talking.
Theoretically there's no limit to the number of participants. That
will be interesting to test :)
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RE: [asterisk-users] FAX mISDN
Ohm. So funny man. I know, sorry. Ok, let me do it again. Hello everybody. I'm working on Asterisk 1.2.15 with a mISDN (no the embedded module, the external one) and also works with NVFaxdetect. I've just sent the DDI from incoming context (exten == 93XXX,1,Dial(SIP/XXX,,tT))to a SIP. But the problem is that I've just set a PAP2 up to receive FAXes, and nothing happens. It's so impossible (now) to receive a FAX. It rings and so, but nothing at all. Anyone knows how to do it? THANKS (better??) Saludos, Lukassky. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Gergo Csibra Enviado el: jueves, 29 de marzo de 2007 20:20 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] FAX mISDN Thursday, March 29, 2007, 7:18:43 PM, LKS wrote: Hi folks! Does anybody know how to receive send faxes throw mISDN? It's almost impossible! Describe your problem, but read this before: http://www.catb.org/~esr/faqs/smart-questions.html It works for me, in 3 places, the analogue fax machines connected to a Linksys PAP2. Everything is the default settings, comes from make samples, only edited the sip.conf and extensions.conf -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] Can I generate random SIP traffic?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: Wed, 28 Mar 2007 09:53:14 +0100 Hello, I would like to generate a peer-to-peer or a server/client SIP traffic between two or more Openwrt access point, to make some statistics about QoS. I tried some SIP traffic generators for OpenWrt, but I didn't find nothing of satisfactory. Now I wonder if asterisk can help me generating random SIP traffic. I'm googling since yesterday without results. Can you help me plz? Thanks and sorry for the disturb. Since no one seems to have specific information, let me try generic. You can certainly program Asterisk to generate random SIP traffic. Or maybe you really mean SIP+RTP traffic. Either way, Asterisk can do it, just like you can program C or Perl to do so. The real question is: what is unsatisfactory about SIP traffic generators you have tried that you hope Asterisk to help? Yuan Liu ___ --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] FAX mISDN
THANKS Lee. Saludos, Lukassky. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Lee Howard Enviado el: jueves, 29 de marzo de 2007 19:43 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] FAX mISDN LKS GMAIL wrote: Does anybody know how to receive send faxes throw mISDN? It's almost impossible! I know that IAXmodem users are doing it. They have to get the right version of the mISDN stuff, though, I think. Lee. ___ --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] Linksys SPA 3102 causing me problems
On Friday 30 March 2007 08:11, Gergo Csibra wrote: Friday, March 30, 2007, 5:02:08 AM, Matt wrote: On 3/29/07, Alan Chandler [EMAIL PROTECTED] wrote: I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next to fix it. Any ideas? I dont know if you have done this but run a sip show peers and make sure that its registered with asterisk. Sounds like it is not registering with asterisk which would allow you to call out but when it tries to call you it dosent have an ip to contact you at. Wehh... He activated the DND function of Linksys. It can be activate with *78 and deactivate with *79. No - thats not it (I don't think). First off DND Setting: is set to no in the User1 page. Secondly I did the *79 to deactivate it if it was set, and then tried again. Still no joy -- Alan Chandler http://www.chandlerfamily.org.uk ___ --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: Fwd: [asterisk-users] Multi-registration ?
From: Drew Gibson [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 15:08:51 -0400 Olivier wrote: I tried using multiple accounts from one phone to separate call centre traffic but the phones (Aastra 480i) would default all calls from the phone to the account with the highest line number. This made it impractical for my purposes. Drew Do think this limitation comes the phone or from Asterisk ? Cheers The phone, it selects the outgoing account to use. Logistically, there is no way for the phone to know which outgoing account YOU want it to use, unless you press extra buttons like on old style PBX phones or multi-line phones. Short of having custom made phones, you can play with dial plan and use, for example, a special prefix or postfix to indicate which personality you want to present when outgoing. Is this practical? Yuan Liu regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --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] how to define a pilot number
From: Lito Lampitoc [EMAIL PROTECTED] Date: Tue, 27 Mar 2007 14:28:25 +0800 Hello all, is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Thanks. Lito Telco calls this line rollover. No it cannot be done with Asterisk or any PBX. It can only be configured on the telco side. Yuan Liu ___ --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] web based sip phone
From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:34:24 -0700 (PDT) thanks Yuan I was search the best result is sipfoundary.org but it's client is not spesific for my purpose, but it will be. is any better answer for this searching? Have you tried JAIN SIP applet? It requires an application server to deploy (JBOSS does fine). But if you are desperate :-) (Well it didn't fit my need then but my requirements were rather bizarre.) Part of the answer also depends on your requirements. For some, a CGI/AGI Web interface constitutes a Web based phone. (Think Jahjah.) Such does not require any remote deployment and can be made very sophisticated. (You can even write a streaming Applet without running anything SIP on client machine, and let server do the SIP work.) On the other hand, with appropriate Active-X permissions, you can also deploy nearly any thick application. Yuan Liu best Mani --- Yuan LIU [EMAIL PROTECTED] wrote: From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT) hello is any web based sip phone? The easy answer is yes. Search for Java SIP phone. Some of them can be deployed on the Web. Yuan Liu for example: a user after logining in, view a configured sip phone, and .. best MAni ___ --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] snom led not working with asterisk 1.4.1
Jamie Heckford schrieb: Just to let you know that this doesn't work with the latest SNOM firmware. With the latest FW 6.5.8 the LED work in most cases (for Unavailable , see below) on my SNOMS. BUT: They don't show the correct incoming account/line: only the LED for the first accounts lights up. ciao, Carsten BTW: There is Port of app_devstate (bristuff) for Asterisk 1.4 available: http://www.voip-info.org/wiki/view/Getting+app_devstate+to+work+on+Asterisk+1.4+without+rest+of+bristuff+patches -Original Message- Steve Murphy wrote: On Wed, 2007-03-21 at 15:00 +0100, Giorgio Incantalupo wrote: Hi Steve, as you know if you type show hints inside asterisk console you can see phone status. When a phone is not connected, Asterisk says it is Unavailable. With Asterisk 1.2.9.1 my SNOM leds worked well so I knew when a phone was not available but with Asterisk 1.4.1 is not possible anymore. This is one of the functions which I'm trying to keep from Asterisk 1.2.9.1 to 1.4.1 . Pardon my ignorance! I am new in this area. I have not used my SNOM 360 with anything but 1.4. When the monitored extension is busy, the LED is on; when the extension is ringing, the LED flashes. What does it do for you in 1.2, when the line is unavailable? The LED is also on. I noticed a change from 1.2 to 1.4: channels/chan_sip.c, 1.2.13 : case AST_EXTENSION_UNAVAILABLE: statestring = confirmed; local_state = NOTIFY_CLOSED; pidfstate = away; pidfnote = Unavailable; Asterisk 1.4.2 channels/chan_sip.c Line 6892 function static int transmit_state_notify(...) case AST_EXTENSION_UNAVAILABLE: statestring = terminated; local_state = NOTIFY_CLOSED; pidfstate = away; pidfnote = Unavailable; The var statestring has changed. I changed it back to confirmed and the phone shows the unavailable state. ciao, Carsten ___ --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] bad case of buzzing
Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using SIP to our central PRI-equipped asterisk. Phones are Polycom 430, 601, Cisco 7960, 7912 all to the latest firmware. We tried everything: changing the switch, network cards, auditing every network drop with fluke, re-certifying our wan, swapping some phones to no effect. Has anyone gone through that ordeal? ___ --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 30VIP Phone
On 3/29/07, Jason Parker [EMAIL PROTECTED] wrote: - Chris Nighswonger [EMAIL PROTECTED] wrote: That is the conclusion I came to and was confirmed today in a very brief chat with one of the individuals listed as a developer on the chan_skinny module. He said that they could be implemented. What I would like to know, and do not understand, is the relationship between the code in chan_skinny.c which sets up the softkeys which are implimented and the actual key positions on the phone. With this info, I can hack the code to impliment other of the keys (ie. speed dial, etc.). Search the code for 30VIP, there are only like 2-3 places where it's referenced. Right. I have done this. It should be immediately obvious how it works. Maybe to some who have been in on the skinny/cisco conversation for awhile. I am not new to c or c++, but am to * and cisco ip phones (this is probably more of my problem) and it is not at all obvious (to me). I would like to possibly contribute, however. I have also been working with chan_sccp which I understand supports these phones more fully than chan_skinny. I am surprised with a $1000+ bounty on the porting of chan_sccp features to chan_skinny that no one has taken time to do it yet. Please forgive me if this sounds ungrateful. I am thankful for the help and for the great product that * is. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Refresher course needed!
From: Brad Sumrall [EMAIL PROTECTED]
Date: Tue, 27 Mar 2007 00:06:13 -0500
Hello everyone
My name is Brad, I am an old Asterisk Vet of the very early days just
coming
back to join the group.
Ok, for starters, I feel like the monkey with the light bulb looking at
extensions.conf and sip.conf.
It has been some time.
A friend ask me to set up a asterisk server that records phone calls.
FC4
Asterisk 1.4
And all the latest and greatest
Problem number 1
Some good get back into the grove literature.
I work CLI only, never much for graphics and gui's
Asterisk 1.4 still has CLI. I don't think many people here use GUI.
voip-info.org is a good starter. Another really good restarter? CLI help!
Problem number 2
We have asterisk logged into teliax but cannot see the inbound call come up
on the CLI
Tethereal says this;
1660 3.829799 207.174.202.4 - 66.109.17.92 SIP Status: 100 Trying(1
bindings)
1661 3.831357 207.174.202.4 - 66.109.17.92 SIP Status: 200 OK(1
bindings)
Asterisk says this;
*CLI
Nothing, notta!
How did you start Asterisk or remote console? Have you tried core set
verbose 10? (Just kidding. Most often I go 3.) Have you tried
sip set debug?
My extensions.conf
(yes, I loaded the samples)
[general]
static=yes
writeprotect=no
clearglobalvars=no
;#include filename.conf
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;From here is brads stuff
exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten = YOURNUMBER,1,Answer()
exten = YOURNUMBER,1,DIAL(SIP/user,20)
Getting more confused about what inbound call you did not see after reading
the sample conf. Did you put a context title before brads stuff? What is
your sip.conf/user.conf if you expect incoming call from SIP?
Ah. Feels good to teach grandma cook milk:-)
Yuan Liu
Thanks to all!
Brad
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RE: [asterisk-users] outbound call
From: Karthik Arumugam [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 21:35:45 +0530 HI All, I am new to asterisk. i want to make outbound calls from asterisk. I tried with many times with the given settings but in vain In vain says vain. Exactly what does not work? Any messages? Errors? This is my scenario: I have a *user A* who has registered with sip server(ONDO), I Is *user A*'s user name with the server 'test' as your dial plan suggested? made asterisk to register as a sip client with ONDO, I want to make a call to user A from an extension. What is an extension's context? Is this extension dexter as your config suggested? You can get much better response if you can help others understand what your problem is. Yuan Liu My configurations sip.config [general] context=default register = raja:[EMAIL PROTECTED]/1234 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to ( 0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls [authentication] auth=raja:[EMAIL PROTECTED] [*192.xxx.xxx.xxx*-out] type=peer ; we only want to call out, not be called secret=adsi6677 username=raja ; Authentication user for outbound proxies fromuser=raja ; Many SIP providers require this! fromdomain=*192.xxx.xxx.xxx* host=*192.xxx.xxx.xxx* - Ignored: context=outgoing [dexter] type=friend username=dexter secret=password host=dynamic context=outgoing extensions.conf [outgoing] exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]) Here *192.xxx.xxx.xxx* is my sip server host ip (ONDO). Please correct me where i am going wrong in this scenario. I was able to receive incoming calls to dexter from user A, Thanks in advance! Regards karthik ___ --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] cutting hash in dial app
From: René Enskat [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 20:29:16 +0200 hello, isit possible to cut off the hash behind a dial string? coz we have a provider who gives us an error 600 Declined if ther is a hash in dial command. for example: Dial(SIP/x.x.x.x-b7d2d870, SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED] x) and i have to cut out: -b7d2d870 regards rene Cutting out part of a string is very easy to do - CLI show function CUT. But the dial command you cited looks really strange. Don't look like correct syntax at all. So maybe you need to fix that first. CLI show application dial Yuan Liu ___ --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] queue priority
On 3/29/07, Jordan Novak [EMAIL PROTECTED] wrote: What is the most stable version supporting queue priority. I have had many crashes, I am using 1.2.11 and have set the weight in queues.conf. is there a better way or a patch. I can't seem to find much. Any suggestions? Do you have a bug open in mantis with a backtrace of the crash? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Linksys SPA 3102 causing me problems
Is it behind a router? -t- On Mar 29, 2007, at 6:26 PM, Alan Chandler wrote: I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next to fix it. Any ideas? ___ --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] Re: UK PRI and outgoing CLI FYI
In article [EMAIL PROTECTED], Steve Kennedy [EMAIL PROTECTED] wrote: On Thu, Mar 29, 2007 at 10:40:50PM +0100, Julian Lyndon-Smith wrote: We only present the 6 digits ... and they give us 6 digits. For our outbound calls, for the the numbers 01702 1234[00-99] we have to present 1234[00-99]. BT isdn pri line. Weird, seems they're inconsistant or there's some oddity at the driver level? It probably depends on the setting of pridialplan and/or prilocaldialplan in zapata.conf. And then perhaps also internationalprefix, nationalprefix, localprefix, privateprefix and unknownprefix. The presentation of numbers in Q.931 has a type-of-number (TON) field which is affected / interpreted by the dialplan settings, and tells the other end what portion of the number you are presenting. e.g. maybe you can present 6 digits for CLI if you have pridialplan=local or something like that. I don't know of any Asterisk documentation on this, and it's probably best to experiment with PRI debugging turned on, and check out the source code! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Speed Dial Application in *
Hi all, Is there a speed dial type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar feature? Thanks, Chris ___ --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] Speed Dial Application in *
You can build it. I put one in my systems that uses an mysql table that the users can edit via a web interface and then the dialplan does a lookup and dails the number. On 3/30/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Hi all, Is there a speed dial type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar feature? Thanks, Chris ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Speed Dial Application in *
You can build it. I put one in my systems that uses an mysql table that the users can edit via a web interface and then the dialplan does a lookup and dials the number. On 3/30/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Hi all, Is there a speed dial type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar feature? Thanks, Chris ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Sipura SPA2000 Transfer Call
Hi List, I have a Sipura SPA 2000, and I am trying to get call transfer to work. I am using an old version of Asterisk, and as far as I am aware I have feature.conf disabled in the dialplan (I am happy with this do far). So I am trying to get the SPA to do the transfer. It looks like *98 is the transfer code, but it just seems to ignore this. I read somewhere about having to do a hook flash first, but this is a UK phone, which button would that be? Have I got something in the SPA disabled or just going about it the wrong way? Any pointers appreciated. Chris -- Chris Blunt ___ --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] RE: Polycom 501 + Asterisk +Edit buttons
I have this same feature enabled for one of my clients. You can't edit the soft buttons on the phone, but you can edit the hard keys. I remapped a speed dial to an unused hard key. When this key/speed-dial is pressed, it plays back a sound file that says whether or not night service is on. Noah, Please share a snip of what the Polycom config file looks like for re-mapping a hard key. I was under the impression that was not possible, just moving the keys around. So you can actually change the function of what the key does, like send DTMF tones back to Asterisk? Thanks. JR ___ --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: Fwd: [asterisk-users] Multi-registration ?
Yuan LIU wrote: From: Drew Gibson [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 15:08:51 -0400 Olivier wrote: I tried using multiple accounts from one phone to separate call centre traffic but the phones (Aastra 480i) would default all calls from the phone to the account with the highest line number. This made it impractical for my purposes. Drew Do think this limitation comes the phone or from Asterisk ? Cheers The phone, it selects the outgoing account to use. Logistically, there is no way for the phone to know which outgoing account YOU want it to use, unless you press extra buttons like on old style PBX phones or multi-line phones. Short of having custom made phones, you can play with dial plan and use, for example, a special prefix or postfix to indicate which personality you want to present when outgoing. Is this practical? Yuan Liu My phones know which line I want to use, they have a button for each one. I have since found that there is an issue with the way that the Aastra 480i registers the separate lines/accounts with Asterisk leaving Asterisk unable to differentiate between the lines on outbound calls. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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] Re: Problem converting a Cisco 7960 to SIP
Salvatore Giudice wrote: You can go directly to 5.2 and then move on to 7.x and 8.6. 5.2 allows you to upgrade to the newer firmware releases that have an app loader, which Cisco added in later releases. Beware that some cisco non-sip loads can not generate the proper firmware filename to download from tftp when they read the version numbers from the version text. I always go directly to 7.2 and it works fine for me. If anyone needs the 7.2 firmware, let me know off list. ___ --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] Which IP Phones have buttons can be assigned to functions with Asterisk
Hi List; Can someone advise me which IP Phone model that has buttons that can be assigned to do specific functionalities (call pickup, call formward, call appearance) and a transfer button and hold button? Which is the best of the following (that has buttons can be assigned to specific functions): Cisco 7970 or 7960 Polycom 501 Grandsream IP Phone Budge Tone 1001 or 1002 Linksys SPA 942 or 922 Aastra 9133 i or 480i Anyone can advise? I heared that polycom needs adaptor for the power as it does not provide standard PoE, also I do not know this. Regards ITS Bilal Ghayad Functional Consultant Mobile: 00965 9849460 Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA. http://answers.yahoo.com/dir/?link=listsid=396545367 ___ --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] Setting rxgain per channel
I'm sorry, I wanted to say FXO :P Thank you! On 3/30/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Delca [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 18:39:37 -0300 How do I set rxgain per channel on zapata.conf? I've a TDM400 with 2 FXS. Does FXS even use rxgain? To set rxgain for an FXO channel, simply put the entry before saying channel =. Hope this helps. Yuan Liu Thank you! Santiago del Castillo ___ --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] SIP NAT
If I have several local networks, can I specify that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, March 29, 2007 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP NAT Mike Hammett wrote: I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server - MT w\NAT - PPPoE - MT - Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk -- PSTN works great. Audio Asterisk -- PSTN does not. That would be expected since you did not forward the ports used for RTP. See /etc/asterisk/rtp.conf A sample is in the Asterisk source. Did you also set localnet= and externip= options in sip.conf [general]. SIP works just fine with NAT if you have it correctly configured and your server is on a static IP address. ___ --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] error in FreePBX
disable voicemail for that extension .. apply settings .. re-enable voicemail .. re-apply settings . this helped me once before. ___ --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] CallerID + Name
That did it. I didn't realize it needed that extra minute to wait for the info to come in. I figured if it came in later, it would just send to the phone the new/updated info when it got it. Guess not. :) Now, if we could only get transfer call caller id working in asterisk and the polycom 501, that'd be great! Trevor Peirce wrote: Rob Schall wrote: We have the caller id with name option enabled with our provider, however, our polycom 501 phones will only display the number of the incoming call. Is there a way to see the callerid name from the cli when the call is coming in (like a print in the dial plan)? I'm not sure if the problem is with asterisk or our phones. I did turn on the calleridpres option in zapata, but I'm unsure what else needs to be set. I bet you have something like this: s,1,Dial(SIP/polycom) When you should in fact have something more like s,1,Wait(0.5) s,n,Dial(SIP/polycom) Caller ID Name information takes a little bit longer to come in, on a PRI as I'm assuming you're using. HTH, Trevor ___ --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] pickupgroup / SIP / Cisc phones
List, I have successfully setup a couple of pickupgroups, and all works as it should (with *8#). Is there a way to configure my Cisco 7940/7960 phones, so one of the soft buttons send the *8# signal, thus picking up the call? Many thanks, Mike ___ --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] Polycom 501 + Asterisk +Edit buttons
I also would like to see how you did that. :) Rob Noah Miller wrote: Hi Rob - I have setup a special extension in asterisk which toggles whether our Night Service is turned on. However, I would like to have 2 things happen on our Polycom 501 (or 601)s. Where there are buttons which say New Call and Forward, I would like to add a button called NS. Is there a way to edit its software? The second feature would be to change the display to read Night Service On if it is. I have this same feature enabled for one of my clients. You can't edit the soft buttons on the phone, but you can edit the hard keys. I remapped a speed dial to an unused hard key. When this key/speed-dial is pressed, it plays back a sound file that says whether or not night service is on. - Noah ___ --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] SIP NAT
According to sip.conf.sample the answer is...well, I guess you can look in /path/to/src/asterisk/configs/sip.conf.sample and see for yourself. Mike Hammett wrote: If I have several local networks, can I specify that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, March 29, 2007 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP NAT Mike Hammett wrote: I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server - MT w\NAT - PPPoE - MT - Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk -- PSTN works great. Audio Asterisk -- PSTN does not. That would be expected since you did not forward the ports used for RTP. See /etc/asterisk/rtp.conf A sample is in the Asterisk source. Did you also set localnet= and externip= options in sip.conf [general]. SIP works just fine with NAT if you have it correctly configured and your server is on a static IP address. ___ --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] xten web phone
hi xten.de produced an activex for web phone. but I can not find any link for download. can u help me ? best Mani Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 ___ --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] Sipura SPA2000 Transfer Call
Chris Blunt wrote: I read somewhere about having to do a hook flash first, but this is a UK phone, which button would that be? It would be called RECALL. ___ --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] SIP NAT
I checked into it and it seems to recognize multiple entries as debug displays it. --Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Friday, March 30, 2007 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP NAT According to sip.conf.sample the answer is...well, I guess you can look in /path/to/src/asterisk/configs/sip.conf.sample and see for yourself. Mike Hammett wrote: If I have several local networks, can I specify that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, March 29, 2007 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP NAT Mike Hammett wrote: I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server - MT w\NAT - PPPoE - MT - Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk -- PSTN works great. Audio Asterisk -- PSTN does not. That would be expected since you did not forward the ports used for RTP. See /etc/asterisk/rtp.conf A sample is in the Asterisk source. Did you also set localnet= and externip= options in sip.conf [general]. SIP works just fine with NAT if you have it correctly configured and your server is on a static IP address. ___ --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] DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp
Justin Tunney wrote: rfc2833compensate=yes Why do you have this turned on? This setting is _ONLY_ for receiving RFC2833 DTMF from pre-1.4 Asterisk servers, it should never be used for any other SIP endpoint. ___ --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] Set(CALLERID(all) not working with 'unknown' call?
Christoph Fürstaller wrote: exten = _3072,n,SetCallerPres(allowed) And this also fixed my problem of many months, between an older Definity G3r Thanks for the info! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 30VIP Phone
- Chris Nighswonger [EMAIL PROTECTED] wrote: On 3/29/07, Jason Parker [EMAIL PROTECTED] wrote: It should be immediately obvious how it works. Maybe to some who have been in on the skinny/cisco conversation for awhile. I am not new to c or c++, but am to * and cisco ip phones (this is probably more of my problem) and it is not at all obvious (to me). I really don't know what to say then.. It's a simple switch statement on the phone model, with some assignments to set what the buttons do. What, specifically, do you not understand? -- Jason Parker Digium ___ --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] Re: System from AMI
Tomislav Parcina wrote: Richard Lyman wrote: *snipped fyi: manager originate is channel + context + exten + priority OR channel + application + data not both. So, you are saying that this should look like this? Action: Originate Channel: Local/[EMAIL PROTECTED] Application: System Data: /path/to/script OR Action: Originate Channel: Local/[EMAIL PROTECTED] Context: dummy Exten: 2 Priority: 1 In extensions.conf [dummy] Exten = _X,1,System(*some command*) remember your permissions ___ --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] SPA3102 PSTN fallback
Hi - I got a SPA3102. I've set it up without to many problems. If the unit looses power, calls to the PSTN are bridged which is nice. However, if the Asterisk server is unavailable (I turned it off to test), calls out are not bridged to the PSTN. I've rebooted the SPA3102 with the asterisk server off, but still it gives me no dial- tone. Under the configuration, Auto PSTN Fallback is set to Yes. Is there anything else I need to do? thanks Todd ___ --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] call file vs. originate
I'm having trouble getting the manager interface to behave properly;
specifically the Originate event.
If I create an originate event as below, the calling phone will
auto-answer (as it's supposed to) but the receiving phone never rings.
It will timeout at 20 seconds.
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201
This is rather odd, as if I create a nearly identical call file in
/var/spool/asterisk/outgoing (below) the receiving phone rings correctly.
Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201
They both use the same contexts, but the result is drastically
different. Any thoughts on how to remedy the problem?
Here is are the two contexts from extensions.conf:
; from sip lines
[from-sip]
include = internal
[from-sip2]
exten = _X.,1,SIPAddHeader(Alert-Info: AA)
exten = _X.,n,Dial(SIP/${EXTEN},200,o)
exten = _X.,n,Hangup()
; generic interal route
[internal]
exten = s,1,Answer()
exten = 500,1,Macro(voicemail)
include = parkedcalls
include = cac-ext
include = sip-ext
include = intertel-ext
include = to-ptsn
(cac-ext, sip-ext, intertel-ext and to-ptsn route the calls to our
channel bank, sip phones, intertel pbx, and the outside world respectively.)
Below lies the results given over the manager interface:
Response: Success
Message: Originate successfully queued
Event: Newchannel
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
State: Ring
CallerID: unknown
CallerIDName: unknown
Uniqueid: 1175271459.2289
Event: Newchannel
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
State: Down
CallerID: unknown
CallerIDName: unknown
Uniqueid: 1175271459.2288
Event: Newcallerid
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2288
CID-CallingPres: 0 (Presentation Allowed, Not Screened)
Event: Newcallerid
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2288
CID-CallingPres: 0 (Presentation Allowed, Not Screened)
Event: Newcallerid
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2289
CID-CallingPres: 0 (Presentation Allowed, Not Screened)
Event: Newexten
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
Context: from-sip2
Extension: 201
Priority: 1
Application: SIPAddHeader
AppData: Alert-Info: AA
Uniqueid: 1175271459.2289
Event: Newexten
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
Context: from-sip2
Extension: 201
Priority: 2
Application: Dial
AppData: SIP/201|200|o
Uniqueid: 1175271459.2289
Event: Newchannel
Privilege: call,all
Channel: SIP/201-08217eb0
State: Down
CallerID: unknown
CallerIDName: unknown
Uniqueid: 1175271459.2290
Event: Dial
Privilege: call,all
Source: Local/[EMAIL PROTECTED],2
Destination: SIP/201-08217eb0
CallerID: 201
CallerIDName: Fake Name
SrcUniqueID: 1175271459.2289
DestUniqueID: 1175271459.2290
Event: Newstate
Privilege: call,all
Channel: SIP/201-08217eb0
State: Ringing
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2290
Event: Newstate
Privilege: call,all
Channel: SIP/201-08217eb0
State: Up
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2290
Event: Newstate
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
State: Up
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2289
Event: Link
Privilege: call,all
Channel1: Local/[EMAIL PROTECTED],2
Channel2: SIP/201-08217eb0
Uniqueid1: 1175271459.2289
Uniqueid2: 1175271459.2290
CallerID1: 201
CallerID2: 201
Event: Newstate
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
State: Up
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2288
Event: Newexten
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
Context: from-sip
Extension: s
Priority: 1
Application: Answer
AppData:
Uniqueid: 1175271459.2288
Event: Rename
Privilege: call,all
Oldname: SIP/201-08217eb0
Newname: SIP/201-08217eb0MASQ
Uniqueid: 1175271459.2290
Event: Rename
Privilege: call,all
Oldname: Local/[EMAIL PROTECTED],1
Newname: SIP/201-08217eb0
Uniqueid: 1175271459.2288
Event: Rename
Privilege: call,all
Oldname: SIP/201-08217eb0MASQ
Newname: Local/[EMAIL PROTECTED],1ZOMBIE
Uniqueid: 1175271459.2290
Event: Unlink
Privilege: call,all
Channel1: Local/[EMAIL PROTECTED],2
Channel2: Local/[EMAIL PROTECTED],1ZOMBIE
Uniqueid1: 1175271459.2289
Uniqueid2: 1175271459.2290
CallerID1: 201
CallerID2: 201
Event: Hangup
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1ZOMBIE
Uniqueid: 1175271459.2290
Cause: 16
Cause-txt: Normal Clearing
Event: Hangup
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
Uniqueid: 1175271459.2289
Cause: 16
Cause-txt: Normal Clearing
___
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[asterisk-users] One way intermittent static to PSTN
We are having intermittent problems where the person we call reports static when we place an outgoing PSTN call. Only the person called hears static, to us the conversation sounds fine. Never happens on inbound calls. It doesn't matter what channel you call from (IAX, SIP, or Zap). We have a Sangoma A108D with hardware echo cancellation with 2 PRIs to Level3 and 2 PRIs to a Nortel Option 61c and then several IAX trunks running into this box as well. Box is HP DL 385 G2. I've ruled out bad cables, bad port on sangoma and bad port at Level3 rack. When under load and while the problem is occurring zttest is never less than 99.987793 and is usually 100. Nothing showing up in any logs anywhere. Not sure it is related, but I'm noticing a very loud click when an incoming or outgoing call is initiated that I don't remember in the past. I'm stumped. Anyone ever experience this? Suggestions for further trouble shooting? Thanks, Jeremy ___ --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] call file vs. originate
Nathan Bell wrote: I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/[EMAIL PROTECTED] Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe 201 This is rather odd, as if I create a nearly identical call file in /var/spool/asterisk/outgoing (below) the receiving phone rings correctly. Channel: Local/[EMAIL PROTECTED] Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe 201 *snipped depending on your manager.c you will find that manager originate need 'Exten: ..' not 'Extension: ..' meaning, if you attempt to use 'Extension: ..' it will autofallthru (if set) to 's' extension in dialplan. good luck ___ --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] SNOM 360
Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Couldn't load variables.txt?aldope=xxxxx
Hi Carlos, I suggest you to use /init/op_panel_debian.sh script inside oppanel tar file. Put it inside /etc/init.d and then as root type: *update-rc.d op_panel defaults * to setup the script for boot. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgi thanks for all, it works. Your opinion is correct, my op_server was not run, and i run him. I'm using Ubuntu Dapper and i want run the op_server when the machine starts, and i add a line in the file rc.local like this: cd /var/www/html/panel/; su -c /var/www/html/panel/op_server.pl but this not works when i start the machine. Please say me the changes i will have put in this line. thanks 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, if you have not op_panel.pid in /var/run/asterisk this means the panel server is not working. I do not know where freepbx puts oppanel files (usually they are in /usr/local but not always). Just find them and exec the file *op_server.pl* in stand alone mode (just type ./op_server.pl inside its directory) so you can see it is working (you'll see a lot of messages). If you cannot find the oppanel dir this means it is not installed. You could download it from www.asternic.org and install it following the instructions on the site. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio. when i type: ps -A -F | grep panel: # [EMAIL PROTECTED]:/home/hernandezz# ps -A -F | grep panel 1000 5378 1 0 8596 15480 0 11:26 ?00:00:02 gnome-panel --sm-client-id default1 1000 5433 1 0 5903 10900 0 11:26 ?00:00:00 /usr/lib/gnome-panel/clock-applet --oaf-activate-iid=OAFIID:GNOME_ClockApplet_Factory --oaf-ior-fd=37 root 7363 6544 0 723 812 0 12:19 pts/000:00:00 grep panel # when i type: [EMAIL PROTECTED]:/home/hernandezz# /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p # Can't open perl script /usr/local/op_panel/op_server.pl: No such file or directory [EMAIL PROTECTED]:/home/hernandezz# # what's the problem..? sorry i try search but i'm not a linux expert. thanks. 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, type: *ps -A -F | grep panel* You should see something like: root 14851 1 0 2700 8164 0 11:01 ?00:00:01 /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p /var/run/asterisk/op_panel.pid This means that tha panel process is running. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio, sorry but how do this? how i verify the server it's running, and if not runnig how i put this running. Thanks 2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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 ___ --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] SNOM 360
Hi UxBoD, just create a voicemail for your extension and Asterisk will do the rest!!! Giorgio Incantalupo --[ UxBoD ]-- wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA ___ --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] SNOM 360
exten = 123,1,Dial(SIP/123|20) exten = 123,n,Voicemail(u123) would be a start, you can have all kinds of fun... On 3/30/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting
Dear all, In my Asterisk 1.2.17 architecture different levels of permissions are established using different contexts that hierarchically include more permissive contexts until default context is reached. In default context there are only local extensions, only in more restricted contexts there are the PSTN access. So, if some user dials some number, Asterisk looks which context that user belongs to in sip.conf and sends that call to that context in extensions.conf. Call flow goes successively including other contexts along the hierarchy until some established filter matches, and than that call is routed to the destination. If no match is found after call flow has descend until the default context, Asterisk hungs up the call. Problem arises when The problem is that the phones I've deployed in my site have the optional feature of unconditionally redirecting incoming calls to other phone number by sending a 302 Moved Temporarily SIP message back to Asterisk, carrying the new contact that should be dialled by the server. When this happens, Asterisk seems to send this 302 message to the default context. If the new contact is some internal extension, it matches some rule in the default context, and Asterisk dials that extension with no problem. If the new contact is some PSTN number, Asterisk can't find a successful matching rule in default context because only upper hierarchy contexts match PSTN numbers, and call is hung up. To solve this, I can include PSTN numbers matching rules in default context (or include upper hierarchy permission contexts in default), but than, every one without PSTN dial permissions would be able to dial PSTN numbers!! Is there any way that I can make that 302 message be dropped in the context to which the user that redirected the call belongs to, and not the default context, because, this is the one that should be charged for the forwarded accounting? And like this, the redirected call would only take place if the user that redirected the call has PSTN permissions to do that! Thanks in advance, Ricardo. ___ --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] Remote host can't match request NOTIFY to call
Looks like, the SIP NOTIFY message is getting a 481 Call leg does not exist response. You can ignore this message. But it will be interesting to see the full sip debug output to see what is going on. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Richard Klingler Sent: Saturday, March 24, 2007 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Remote host can't match request NOTIFY to call Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick ___ --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] forwarding loop not detected
Asterisk 1.2.16 I have an extension 102 with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 Moved Temporarily back from 206.83.240.18 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/exten-mycontext-102-094c2c08) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/exten-mycontext-102) in new stack -- Called exten-mycontext-102 -- Got SIP response 302 Moved Temporarily back from 206.83.240.18 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/exten-mycontext-102-095bfef8) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/exten-mycontext-102) in new stack -- Called exten-mycontext-102 -- Got SIP response 302 Moved Temporarily back from 206.83.240.18 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/exten-mycontext-102-09495990) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/exten-mycontext-102) in new stack -- Called exten-mycontext-102 Looping for a long time then the next entry in the dial plan kicks in (Voicemail) after a ton of those Dialplan: exten = 102,1,Dial(SIP/exten-mycontext-102) exten = 102,n,Voicemail([EMAIL PROTECTED]) Forwarding to other extensions and outside numbers works fine, just not to itself. How can I protect against such loops? Bill ___ --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] SNOM 360
Hmmm, okay. But surely it will just try and ring the extension? Or do you mean setup a seperate extension ie. exten = 1001,1,Dial(sip/1001,20) exten = 1001,2,VoiceMail([EMAIL PROTECTED],u) exten = 1001,3,Hangup() exten = 1001,101,VoiceMail([EMAIL PROTECTED],u) exten = 1001,102,Hangup() exten = 2000,1,VoiceMail([EMAIL PROTECTED],u) exten = 2000,2,HangUp() So on pressing the DND it will send all calls to extention 2000 ? TIA On Fri, 30 Mar 2007 12:57:16 -0400 Andrew Latham [EMAIL PROTECTED] wrote: exten = 123,1,Dial(SIP/123|20) exten = 123,n,Voicemail(u123) would be a start, you can have all kinds of fun... On 3/30/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] SNOM 360
The snom360 DND button forces the phone to give a 480 do not disturb response. Bails --[ UxBoD ]-- wrote: Hmmm, okay. But surely it will just try and ring the extension? Or do you mean setup a seperate extension ie. exten = 1001,1,Dial(sip/1001,20) exten = 1001,2,VoiceMail([EMAIL PROTECTED],u) exten = 1001,3,Hangup() exten = 1001,101,VoiceMail([EMAIL PROTECTED],u) exten = 1001,102,Hangup() exten = 2000,1,VoiceMail([EMAIL PROTECTED],u) exten = 2000,2,HangUp() So on pressing the DND it will send all calls to extention 2000 ? TIA On Fri, 30 Mar 2007 12:57:16 -0400 Andrew Latham [EMAIL PROTECTED] wrote: exten = 123,1,Dial(SIP/123|20) exten = 123,n,Voicemail(u123) would be a start, you can have all kinds of fun... On 3/30/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] call file vs. originate
After fixing some issues with our pbx dial plan that worked great. Thanks, Nathan Bell IT Engineer Action Target, Inc. Richard Lyman wrote: Nathan Bell wrote: I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/[EMAIL PROTECTED] Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe 201 This is rather odd, as if I create a nearly identical call file in /var/spool/asterisk/outgoing (below) the receiving phone rings correctly. Channel: Local/[EMAIL PROTECTED] Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe 201 *snipped depending on your manager.c you will find that manager originate need 'Exten: ..' not 'Extension: ..' meaning, if you attempt to use 'Extension: ..' it will autofallthru (if set) to 's' extension in dialplan. good luck ___ --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] SNOM 360
Ahhh, awesome. Thank you :) On Fri, 30 Mar 2007 18:47:30 +0100 bails [EMAIL PROTECTED] wrote: The snom360 DND button forces the phone to give a 480 do not disturb response. Bails --[ UxBoD ]-- wrote: Hmmm, okay. But surely it will just try and ring the extension? Or do you mean setup a seperate extension ie. exten = 1001,1,Dial(sip/1001,20) exten = 1001,2,VoiceMail([EMAIL PROTECTED],u) exten = 1001,3,Hangup() exten = 1001,101,VoiceMail([EMAIL PROTECTED],u) exten = 1001,102,Hangup() exten = 2000,1,VoiceMail([EMAIL PROTECTED],u) exten = 2000,2,HangUp() So on pressing the DND it will send all calls to extention 2000 ? TIA On Fri, 30 Mar 2007 12:57:16 -0400 Andrew Latham [EMAIL PROTECTED] wrote: exten = 123,1,Dial(SIP/123|20) exten = 123,n,Voicemail(u123) would be a start, you can have all kinds of fun... On 3/30/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Polycom Power
I was reading somewhere that Polycom cables only work when the power is over pins 1236 (signal pairs). I won't swear to it, but that's what I read. Anyway, most PoE injectors (not all, but most) inject power on pins 4578 (non-signal pairs), meaning it won't work with things that need the power on the signal pairs. For that, you'll need a switch that supports PoE. For simple testing, Netgear has some cheap unmanaged switches that provide PoE that's 802.11af-compliant. We're using a couple of them in really small satellite offices, and they've been holding up pretty well. But I wouldn't use them in a large-scale deployment, since they don't support VLAN trunking, QoS, and what-not. On 3/29/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Noah Miller wrote: Hi Mike - I have a 501 with traditional power and a 301 with PoE. I rightfully assumed that the traditional power from the 501 would work on the 301. How do I get the PoE to work? Do I use the Polycom PoE cable in addition to whatever PoE injection method I use? I have a Cisco PoE injector that works on my Cisco AP350 and my 7960. No combination of this injector, the Polycom cable, and the phone result in success. I have 18v PoE injectors that I use for other things, but I hear that 802.3af is 48v, therefore probably wouldn't work. How do I use Polycom PoE? You'll probably have to get different injectors, or a new PoE switch. The newer Cisco PoE switches do speak 802.3af, but many of the older Cisco PoE products do not. The original Cisco PoE implementation was proprietary and does not conform to 802.3af. Polycom has cables available to support Cisco PoE and 802.3af PoE. They are, however, different cables. ___ --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] Couldn't load variables.txt?aldope=xxxxx
It's works. a lot of regards. thanks for all. 2007/3/30, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, I suggest you to use /init/op_panel_debian.sh script inside oppanel tar file. Put it inside /etc/init.d and then as root type: *update-rc.d op_panel defaults * to setup the script for boot. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgi thanks for all, it works. Your opinion is correct, my op_server was not run, and i run him. I'm using Ubuntu Dapper and i want run the op_server when the machine starts, and i add a line in the file rc.local like this: cd /var/www/html/panel/; su -c /var/www/html/panel/op_server.pl but this not works when i start the machine. Please say me the changes i will have put in this line. thanks 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, if you have not op_panel.pid in /var/run/asterisk this means the panel server is not working. I do not know where freepbx puts oppanel files (usually they are in /usr/local but not always). Just find them and exec the file *op_server.pl* in stand alone mode (just type ./op_server.pl inside its directory) so you can see it is working (you'll see a lot of messages). If you cannot find the oppanel dir this means it is not installed. You could download it from www.asternic.org and install it following the instructions on the site. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio. when i type: ps -A -F | grep panel: # [EMAIL PROTECTED]:/home/hernandezz# ps -A -F | grep panel 1000 5378 1 0 8596 15480 0 11:26 ?00:00:02 gnome-panel --sm-client-id default1 1000 5433 1 0 5903 10900 0 11:26 ?00:00:00 /usr/lib/gnome-panel/clock-applet --oaf-activate-iid=OAFIID:GNOME_ClockApplet_Factory --oaf-ior-fd=37 root 7363 6544 0 723 812 0 12:19 pts/000:00:00 grep panel # when i type: [EMAIL PROTECTED]:/home/hernandezz# /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p # Can't open perl script /usr/local/op_panel/op_server.pl: No such file or directory [EMAIL PROTECTED]:/home/hernandezz# # what's the problem..? sorry i try search but i'm not a linux expert. thanks. 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, type: *ps -A -F | grep panel* You should see something like: root 14851 1 0 2700 8164 0 11:01 ?00:00:01 /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p /var/run/asterisk/op_panel.pid This means that tha panel process is running. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio, sorry but how do this? how i verify the server it's running, and if not runnig how i put this running. Thanks 2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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 ___ --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 -- Carlos Jerónimo ___ --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] wireless desktop phones
Am Mittwoch, den 28.03.2007, 12:32 -0400 schrieb Brian Capouch:
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The computer
network is all wireless so the phones would have all the bandwidth.
Some of the Wifi phones--at least under the relatively stable conditions
I have here--work very reliably.
I have 3 Starcom F1000s, and a) if they don't have to roam and b) they
don't have to connect dynamically to different servers, work just fine.
FYI. YMMV.
I still have problems with mine, in a non-roaming, fixed-server setup. I
cannot recommend using them in an office environment.
Depending on the cabling, 10MBit should do for VoIP. As an alternative,
you could still use analogue phones with a FXO/FXS card (sorry I use to
mixup those, I don't have too much analogue phone hardware anymore). Of
course this would give you the full market bandwidth of available
analogue phones.
BR
Anselm
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[asterisk-users] Multi-Level Queue
I am trying to setup a queue in a very specific way and I can't quite figure it out. I'm sure someone else has done this. I want calls to come into a queue and do a ringall on a number of phones (let's say 3). So ring them for 20 seconds or so. If there is no answer, I want it to ring a second set of phones for 20 seconds. If no answer, then go back to the first set of phones. I've seen where you could do two queues and do this, but I don't want to have to setup a second queue. I would like it all in one queue. The second part is that I want queue members to have to hit a key to accept a call. The third part is that I don't want agents to have to login. The reasoning behind all of this is that I want to ring desk phones and then if they don't answer, I want to ring cell phones. If I ring the cell phones too long, someone's voicemail will pick up, which I don't want. So if I set it up where they have to ack it, I can ring the cell phones and if someone's vm picks up, it is no big deal. Any ideas? ___ --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] Realtime call-limit
Does anybody know the sql type for the call-limit field under sip peers? Everything on voip-info is missing that entry. ___ --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] Lucent TNT - ring timer
Hi, I've got a Lucent TNT that I'm using for a gateway. Its working fine, but I have one problem. I cannot find any place to set a ring timer, or number of rings. The calls seem to timeout (Goes to all circuits busy) after about 15 seconds - which isn't enough time for some voicemail boxes to pickup. I found a setting called ringing-timer under sip-options, but it doesn't seem to have any affect. Any ideas? Thanks, Brent ___ --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] Polycom 501 + Asterisk +Edit buttons
Hi Folks - Please share a snip of what the Polycom config file looks like for re-mapping a hard key. I was under the impression that was not possible, just moving the keys around. So you can actually change the function of what the key does, like send DTMF tones back to Asterisk? Well, until fairly recently, you would have been limited to just moving keys around. With the release of the 2.x firmware series, Polycom implemented remapping a speed dial to a key, and this gives us more options. But, there are still limitations. The biggest one: if you do remap a speed dial, when you press the key, it will always create a new channel (if one is available). I'd love it to be able to send strings of DTMF digits inside an existing call, but this is still not possible. Anyway, here's an example of remapping. It uses the keys element in sip.cfg: keys key.scrolling.timeout=1 key.IP_500.37.function.prim=DialpadPound key.IP_500.32.function.prim=DialpadStar key.IP_500.31.function.prim=SpeedDial key.IP_500.31.subPoint.prim=1 key.IP_600.37.function.prim=DialpadPound key.IP_600.30.function.prim=DialpadStar/ For IP 500's, this example remaps the Transfer Hard Key to '#' (Configured as Asterisk Blind Transfer), the Directories Hard Key to '*' (Configured as Autopark), and the Services Key to '701' (Configured as Park Pickup). For the speed dial, you can set it up on the phone, but I'd recommend creating a directory file on your Polycom's FTP/HTTP server. Here's the simple one that I use: ?xml version=1.0 encoding=UTF-8 standalone=yes? !-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05 $ -- directory item_list item fn701/fn ct701/ct sd1/sd /item /item_list /directory Have Fun! - Noah ___ --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] Call dies when I press *
Hi Mike - What kind of phone are you using? some devices which connect you pstn phone to voip network have their own special dtmf keys defined inside their configuration to perform special actions, maybe thats your problem. also check for your dtmf setting. dtmf settings should be same on both sides. Also, if you're using SIP phones, check and make sure you've got the right signalling (rfc2833, inband). The wrong setting can make DTMF entry really crazy, depending on your phone. - Noah ___ --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] Dell poweredge 860 acceptable forofficeenvironment ?
joe a. wrote: The tomshardware-guys (no gals would do this...) have removed the fans, and immersed the innards of the computer in a sealed cabinet filled with cooking oil. So they have a completely silent machine in 40C warm oil. Amazing... It certainly is. And, I suppose, this will work, for a while, as long as: The sealed cabinet, has enough expansion capacity for the oil to expand from heating; Has enough surface area to effectively radiate the heat that is continuously generated, (else the temperature of the oil will continue to rise . . .(sound familiar?)) The components can withstand continuous (24x7x365) immersion in a heated liquid. A solvent, basically. The oil will eventually compromise the integrity of most, if not all of the components. You might be misunderstanding the meaning of the word solvent; A solvent is any (generally liquid) substance which can dissolve another substance; the mere fact of the cooling medium being mineral oil doesn't automatically make it a solvent. It depends on what you're applying it to. The oil is non-conductive, so even if it penetrates a chip enclosure, it's not going to cause a short. It's also non-corrosive; and the suitability of the oil depends also on the heat capacity, which is a function of the volume. Whether the insulators and plastics would be affected by a mineral oil is an open question; I don't see vegetable oil causing a problem, though it will start to smell after a while. Oil is a better conductor of heat to plastic than air is, so you need to consider not only the air-exposed surface but all the others as well. Not the best of ideas, IMHO It's a fine idea, but whether it works well or not comes down to the quality of the execution. Personally, I wouldn't do it for three reasons: - it makes adding/removing expansion cards messy or impractical - a leak in the enclosure would be a catastrophe (a big mess and almost guaranteed failure) - it would be bloody heavy -Stephen- ___ --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] Multi-Level Queue
Peder @ NetworkOblivion wrote: I want calls to come into a queue and do a ringall on a number of phones (let's say 3). So ring them for 20 seconds or so. If there is no answer, I want it to ring a second set of phones for 20 seconds. If no answer, then go back to the first set of phones. I've seen where you could do two queues and do this, but I don't want to have to setup a second queue. I would like it all in one queue. This doesn't sound like a queue at all, but rather just Dial()-ing the desired extensions for that period of time. Are you really to have multiple callers (like a queue would be) or just have incoming calls ring all these phones in this pattern? This can be done with a single queue, but it will take some fancy configuration to make it work. ___ --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] Multi-Level Queue
Peder @ NetworkOblivion wrote: I want calls to come into a queue and do a ringall on a number of phones (let's say 3). So ring them for 20 seconds or so. If there is no answer, I want it to ring a second set of phones for 20 seconds. If no answer, then go back to the first set of phones. I've seen where you could do two queues and do this, but I don't want to have to setup a second queue. I would like it all in one queue. Why not use 2 queues? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Off Topic: Open Source USB Softphone
Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --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] Realtime call-limit
Peder @ NetworkOblivion wrote: Does anybody know the sql type for the call-limit field under sip peers? Everything on voip-info is missing that entry. Realtime reads/writes everything as a string. But you may as well use a tinyint default null. It's important to name the field call-limit (not call_limit). Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Realtime call-limit
Philipp Kempgen wrote: Peder @ NetworkOblivion wrote: Does anybody know the sql type for the call-limit field under sip peers? Everything on voip-info is missing that entry. Realtime reads/writes everything as a string. But you may as well use a tinyint default null. unsigned tinyint default null :) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] SNOM 360
bails wrote: The snom360 DND button forces the phone to give a 480 do not disturb response. Bails --[ UxBoD ]-- wrote: Hmmm, okay. But surely it will just try and ring the extension? Or do you mean setup a seperate extension ie. exten = 1001,1,Dial(sip/1001,20) exten = 1001,2,VoiceMail([EMAIL PROTECTED],u) exten = 1001,3,Hangup() exten = 1001,101,VoiceMail([EMAIL PROTECTED],u) exten = 1001,102,Hangup() exten = 2000,1,VoiceMail([EMAIL PROTECTED],u) exten = 2000,2,HangUp() Here is a better fix... If extension 1000 is unavailable whether in DND or just not there... Call rolls over to extension 2000 with the caller ID 1000 Unavailable so the person at 2000 will know so and so didn't answer their phone because 1000 was wasting their life away on youtube. exten = 1000,1,Dial(SIP/1000|30|tr) exten = 1000,2,Set(CALLERID(name)=1000 Unavailable) exten = 1000,3,SayDigits(1000,f) exten = 1000,4,Playback(vm-isunavail) exten = 1000,5,Goto(SIP/2000,20|tr) So say user @ 1000 is named John, you could change the caller ID to John UA (UA short for the obvious (unavailable) as well as the fact there isn't enough space for the entire string). exten = 1000,1,Dial(SIP/1000|30|tr) exten = 1000,2,Set(CALLERID(name)=Transferred Call) exten = 1000,3,Wait,4 exten = 1000,4,SayDigits(1000,f) exten = 1000,5,Playback(vm-isunavail) exten = 1000,6,SIPAddHeader(Alert-Info: http://somesite/ringer.wav) exten = 1000,7,Set(CALLERID(name)=John UA) exten = 1000,8,Dial(SIP/2000|30|tr) ... Works like this... If user John transfers the call... Whoever he transfers it to will see its a transferred call. If John (extension 1000) doesn't answer, the obvious occurs. (unavailable) I currently use this scheme for one client using Snom 320's and 360's. The caller ID works for most phones I've tested. Polycoms, Aastra's however, don't expect Aastra's to play the wav file. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Multi-Level Queue
The reasoning behind all of this is that I want to ring desk phones and then if they don't answer, I want to ring cell phones. If I ring the cell phones too long, someone's voicemail will pick up, which I don't want. So if I set it up where they have to ack it, I can ring the cell phones and if someone's vm picks up, it is no big deal. Also the cell will answer with VM if it is turned off, out of range, etc ___ --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] Paging
First off, A lot of thanks to this list. I have learned ton from reading through the posts this past year. I need some advise. I have two group of phones connected to a single server. Group1= SIP/2503SIP/2504 Group2=SIP/3501SIP/3502 I'd like to be able to dial an extension and page a certain group of phones only if ChanIsAvail returns 1. I am not sure how to go about programming this. I though to write a AGI script that reads a list of phones (one list per group), checks ChanIsAvail then Pages the phone. I will have about 60 extensions per group to Page. Will there be lag until all the phones get paged and the script finishes? Then I thought maybe a Macro in the dialplan to dial a global var of the group of phones, but that won't work. If phone isn't available, none will get paged. Has anyone done this before? I just don't know where to start. Thanks -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --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] Re: Paging
Forgot to mention. We are using Polycom phones on asterisk 1.4.2 I tried the allpage agi, but it checks for all SIP peers connected to the server. On 3/30/07, Forrest Beck [EMAIL PROTECTED] wrote: First off, A lot of thanks to this list. I have learned ton from reading through the posts this past year. I need some advise. I have two group of phones connected to a single server. Group1= SIP/2503SIP/2504 Group2=SIP/3501SIP/3502 I'd like to be able to dial an extension and page a certain group of phones only if ChanIsAvail returns 1. I am not sure how to go about programming this. I though to write a AGI script that reads a list of phones (one list per group), checks ChanIsAvail then Pages the phone. I will have about 60 extensions per group to Page. Will there be lag until all the phones get paged and the script finishes? Then I thought maybe a Macro in the dialplan to dial a global var of the group of phones, but that won't work. If phone isn't available, none will get paged. Has anyone done this before? I just don't know where to start. Thanks -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --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] Redirect failed, channel not up.
When I use the Asterisk manager interface to redirect a call (Action: Redirect) I get an error with the message Redirect failed, channel not up. This is especially troubling as it looks like this message was added to the code for the rather recent 1.2.x release. A quick google search implies that I'm not the only one experiencing this problem with 1.2.17, but me and kenw on the digium forums are the only two. Has anyone else run into the same problem, or know of any solutions? My asterisk set up isn't live yet, so if the only solution is to wait for 1.2.18, then I can just be patient. Nathan Bell IT Engineer Action Target, Inc. ___ --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] Multi-Level Queue
Kevin P. Fleming wrote: Peder @ NetworkOblivion wrote: I want calls to come into a queue and do a ringall on a number of phones (let's say 3). So ring them for 20 seconds or so. If there is no answer, I want it to ring a second set of phones for 20 seconds. If no answer, then go back to the first set of phones. I've seen where you could do two queues and do this, but I don't want to have to setup a second queue. I would like it all in one queue. This doesn't sound like a queue at all, but rather just Dial()-ing the desired extensions for that period of time. Are you really to have multiple callers (like a queue would be) or just have incoming calls ring all these phones in this pattern? This can be done with a single queue, but it will take some fancy configuration to make it work. There are a couple of reasons for what I want. 1. I want callee's to have to ack to receive the call, in case someone's cell vm picks up. 2. Yes, there could potentially be 2-4 people calling at any given point in time, so I want a sort of overflow to mobile's. 3. I don't want 5 minutes of ringing, I prefer where they get queue updates like you are the 2nd person in the queue and they hear music, rather than ringing. I guess I could have two queue's and just have it bounce back and forth between office phones and cell phones, but won't they get updates like you are th first person and then they switch to the other queue and you are the second person I also had a question about acking a call. It appears that acking a call is under agents.conf. I want to specify members as SIP/1234, etc, rather than having users login all the time. I don't want to have to login from my cell, I would prefer it to just know that my cell number is always a member. Is it possible to force an ack of a call if I define members as SIP/? ___ --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] Meetme cant handle more than 5 users in a call?? hmmmm
We've had MeetMe conferences here in testing with a couple of hundred users on. I haven't read the article yet, but only 5 users in a MeetMe bridge sounds WRONG. Matthew Fredrickson On Mar 28, 2007, at 8:56 AM, Matt wrote: Yikes! While I will agree I think Digium needs to do a little better QA (let's not start that war again), this kind of FUD doesn't do anything for the community. I've had Asterisk running with meetme no problem with many more then 5 users. On 3/28/07, Dean Collins [EMAIL PROTECTED] wrote:Meetme cant handle more than 5 users in a call?? H http://www.voip-news.com/feature/asterisk-voip-pbx-right-choice -032707/ hmmm I'm all for commercializing a product, but this FUD from Fonality seems to be taking it just a little too far Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph ___ --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] switchtype and signalling query
Hi Guys I'm configuring a TE212P card and have the following two entries in my /etc/asterisk/zapata.conf switchtype=dms100 signalling=pri_cpe When I reload asterisk I get the following messages: -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: Ignoring switchtype [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: Ignoring signalling However pri show span 1 shows the right values set for both: ast1*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, In Alarm, Down, Active Switchtype: Nortel DMS100 Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Should I be concerned as to the Warnings ? I'm not quite at the stage where I can test my setup yet and wanted to check before I get there. Many thanks for your time. Simon ___ --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
