Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
Thanks for the feedback, Ira. It makes me very sad to hear what you say and I hope that we can get more resources from the community to assist in the process to make it more friendly. We want to get those bug reports. The one thing I hate to hear when I'm travelling at conferences is that oh, I known that bug for a long time but did not bother to report it. Apologies for your experience with the bug process. Indeed, it seems as though there might be a problem of discoverability of how to report issues. Is it too burdensome to suggest attaching this link (along with a short description) to the footer of list e-mails? http://www.asterisk.org/developers/bug-guidelines That does a fair job (though not perfect, and I think suggestions for improvement are welcome) of detailing the process. It's probably also incumbent upon us all, as a community, to do a better job than just report it on Mantis. I'm quite certain every one of us would like the most stable, bug-free code in Asterisk as is possible, and if it takes an extra minute or two of our time to help get the issues reported in the first place it will be time well-spent. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mark G Thomas Sent: Friday, May 06, 2011 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer Hi, On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote: On 11-05-05 05:14 PM, Mark G Thomas wrote: Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the U options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non- existent destination for gosub: (Context:screen, Extension:s, Priority:1) You might want to have a look at: https://issues.asterisk.org/view.php?id=18910 Thanks. This is it. If I'm reading this right, it describes the change which broke things for me, but no solution applicable to my Dial() command U flag, which is invoking my AEL GoSub (Macro). Switching the Dials back to the M flag doesn't fix it either. It sure seems to me this change to AEL has had unexpected consequences in terms of breaking things in dialplans. I was under the impression that this had been fixed, although perhaps it's not yet in a release. Is there a chance you try with the latest 1.6.2 branch from SVN? - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on ways to activate voicemail light on polycom
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, May 06, 2011 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on ways to activate voicemail light on polycom Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry Yes, use the MinivmMWI application. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, March 09, 2011 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why? On Fri, 4 Mar 2011, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? So I got 1 'vote' for each. Surely more than 2 users use OpenSIPS or Kamailio. I guess Friday afternoon is not the best time to post an open question :) Probably not, no. :) I'll throw my vote in for Kamailio. I've been using it (and OpenSER before the fork/rename) for about 5 years now, and have never had an issue that wasn't my own fault (misconfiguration, etc.). - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, March 09, 2011 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints 2011/3/9 Russell Bryant russ...@digium.com - Original Message - I tried to work around this by centralizing DND requests in Asterisk and sending back a short (You're in DND mode) text to Polycom's screen (using sipsak for this). This was rather disappointing as Poycoms redirect text messages to an Instant Messaging mailbox and do not keep them visible on screen. Maybe, some king of XML magic would be a better mean to return current DND status to users. Any suggestion ? One solution that I had come up with for this situation was to use a softkey and use custom device state to have the LED on or off based on whether DND was on or off. I documented it here: http://ofps.oreilly.com/titles/9780596517342/ch14.html#usingCustomDeviceStates -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA jabber: rbry...@digium.com -=- skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org - This is interesting but though a LED is perfect binary status such as DND, in fact, I'm more after a status line also showing Screening or Forwarding destination (for instance, Fwd = 12345, Fwd = Cellphone ...). Polycom phones have a custom Status window with which you can pick Forwarding settings but, to my knowledge, it can't used to let Asterisk manage those settings (I would be very happy to be proven wrong). Another option would be to use Custom: device state like Russell suggest, but instead of a softkey remap the Do Not Disturb button to a speed dial that is configured to be an Enhanced Feature Key macro that includes toggling of DND as well as dialing the extension that changes the Custom: device state. Off the cuff, assuming the number to dial for the device state mojo is 1234, it would probably be something like: $FDoNotDisturb$1234$Tinvite$ Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Implying that the Asterisk developers (which is itself a fairly nebulous statement since those who contribute to Asterisk are many and come from different companies/countries/etc.) are not in it to make a good product but to make a profit is not only highly insulting but a complete mischaracterization of what you were told on IRC. What you were told was that there are essentially three choices (and this goes for pretty much any open source software, as already stated). You may either fix it yourself (if you have the skills), pay someone to fix it for you (if you can or must trade money for expediency), or wait for someone else with the skills and/or money necessary to fix it. Regards, - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: My bad - natively means using the Queue command from the dialplan. Since the powers that be are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: Do you use the Queue command natively or from the AGI? In the example you gave, if you did a core show channels, I assume that Agent007 would be idle, but ineligible for Queue activity. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: I use Polycom 501's and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
You are still focusing on ONE of the choices given when that isn't your only option. It is simply untrue to say that the answer to it's broken was pay us. You were (now on multiple occasions) told how it would come to pass that a resolution will come about. You choose to ignore precisely two-thirds of the options available to you in order to continue to grind your axe. I am convinced you are either trolling or simply myopic. You have choices, they are yours to make. Stop trying to say that the entire Asterisk development community is simply in it for money, because that is patently false. - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) It's simple, if a product is broken shouldn't it be fixed? In this case the answer is for a price which is absurd because it is an open source product. If there was a decent community of developers surrounding this open source project, it would be fixed simply because it's broken, no questions asked. On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley bradley.watk...@compuware.commailto:bradley.watk...@compuware.com wrote: Implying that the Asterisk developers (which is itself a fairly nebulous statement since those who contribute to Asterisk are many and come from different companies/countries/etc.) are not in it to make a good product but to make a profit is not only highly insulting but a complete mischaracterization of what you were told on IRC. What you were told was that there are essentially three choices (and this goes for pretty much any open source software, as already stated). You may either fix it yourself (if you have the skills), pay someone to fix it for you (if you can or must trade money for expediency), or wait for someone else with the skills and/or money necessary to fix it. Regards, - Brad From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: My bad - natively means using the Queue command from the dialplan. Since the powers that be are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: Do you use the Queue command natively or from the AGI? In the example you gave, if you did a core show channels, I assume that Agent007 would be idle, but ineligible for Queue activity. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: I use Polycom 501's
Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London
Wait, is 70k US for an experienced engineer supposed to be adequate? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, December 22, 2010 2:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London Wouldn't that be 70K USD? Or should we REALLY be worried about the British economy? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Wednesday, December 22, 2010 12:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Wednesday, December 22, 2010 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 95550845 225 5189 0207 788 66000161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com --
Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Friday, November 12, 2010 7:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum br...@woollum.com wrote: I'm having an issue where Asterisk continuously sends out a GAZILLION SIP NOTIFY messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. What version of 1.6? I _think_ this may have been a bug, that was fixed. Don't hold me to that. I agree with Paul, this sounds like a bugs that's been fixed. What does the 'Mailbox :' line look like when you do a 'sip show peers'? My guess is that there will be multiple entries of the same mailbox, and that's why you're receiving a bunch of NOTIFY messages. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail customizing
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benoit Panizzon Sent: Thursday, November 11, 2010 11:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoiceMail customizing Hello We would like to customize the voicemail menues. So the intro should not be played if some user has recorded an own greeting message and we would also like to remove some options from the menue. Is this all hardcoded or is it somehow possible to redefine the voice menues and the order how messages are played via voicemail.conf? Unfortunately, regular Voicemail/VoiceMailMain does not have customizable menus. However, if that is something you need/want, look into MiniVM. It's definitely a build-it-yourself approach, but will accomplish what you are looking to do. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open vm-INBOXs
This is indicative that you have set the channel's language to something that expects there to be a singular and plural version of the 'new' (as in 'one new message' versus 'five new messages') sound. According to the code, that includes Dutch, Spanish, Portuguese and Greek. If you have one of these set as your language (I'm guessing Dutch), then the sound file set you have is incomplete. Regards, - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, September 22, 2010 6:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Unable to open vm-INBOXs Hello list, it seems that a sound file is not present on my system, although I have made a standard install... [Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File vm-INBOXs does not exist in any format [Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to open vm-INBOXs (format 0x8 (alaw)): No such file or directory I do not find this particular soundfile on my system. Can't imagine this file is in the extra-sounds category ?! Al the other voicemail-related sound files are present. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open vm-INBOXs
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, September 22, 2010 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to open vm-INBOXs This is what happens : [Sep 22 14:22:42] -- SIP/test6-0008 Playing 'vm-INBOXs.slin' (language 'nl') [Sep 22 14:22:42] == Spawn extension (from-TEST, 1001, 5) exited non-zero on 'SIP/test6-0008' Asterisk ends the conversation because the file 'vm-INBOXs' does not exist. But the file is present : [r...@asterisk16 asterisk-1.6.2.10]# locate vm-INBOXs /var/lib/asterisk/sounds/nl/vm-INBOXs.wav Well this is completely different from what you originally posted... Anyway, what is the output of 'core show file formats'? It sounds like you're missing a format_XXX.so (perhaps unselected in menuselect?) and so the channel is falling back to trying to find a signed linear file (which, at least in name, you don't have). - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open vm-INBOXs
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, September 22, 2010 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to open vm-INBOXs On 09/22/2010 02:45 PM, Philipp von Klitzing wrote: .slin is not .wav Other files that are also in wav format play without any problem : [Sep 22 15:02:35] -- SIP/testcorp6- Playing 'vm-youhave.slin' (language 'nl') [r...@asterisk16 asterisk-1.6.2.10]# ls -l /var/lib/asterisk/sounds/nl/ total 388 drwxr-xr-x 2 root root 4096 Sep 22 11:25 digits -rw-r--r-- 1 root root 66124 Sep 22 11:10 vm-helpexit.wav -rw-r--r-- 1 root root44 Sep 22 14:19 vm-INBOXs.wav -rw-r--r-- 1 root root 16844 Sep 22 12:47 vm-INBOX.wav -rw-r--r-- 1 root root 37004 Sep 22 10:58 vm-incorrect.wav -rw-r--r-- 1 root root 26764 Sep 22 12:47 vm-messages.wav -rw-r--r-- 1 root root 23564 Sep 22 12:54 vm-message.wav -rw-r--r-- 1 root root 12364 Sep 22 11:06 vm-no.wav -rw-r--r-- 1 root root 19404 Sep 22 14:19 vm-Olds.wav -rw-r--r-- 1 root root 17164 Sep 22 14:20 vm-Old.wav -rw-r--r-- 1 root root 27404 Sep 22 12:49 vm-onefor.wav -rw-r--r-- 1 root root 31884 Sep 22 10:57 vm-password.wav -rw-r--r-- 1 root root 25164 Sep 22 11:04 vm-youhave.wav Jonas. -- Well, I think I see the problem now that you've shown a directory listing. The file in question is a mere 44 bytes. That is almost certainly not right. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, August 25, 2010 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s): That's what I understood too from this one and probably only related google search result, but even if I have just 3-4 lines of code, the error is still there. It is all English characters, so UTF-8 compatibility issue should not be there. I am sure there is some small little config change is required somewhere related to AEL, but where, I don't know. Zeeshan A Zakaria Is there any chance that these files were edited on a Windows machine and then copied back to the Asterisk boxes? That is, are there some nefarious ^m characters hiding in there? Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clustering concept
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of unsero...@aol.com Sent: Thursday, July 29, 2010 3:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Clustering concept Hi all, I am wondering if the Clustering concept described in Leif Madsens presentation http://comunidad.asterisk-es.org/documentos/astricon2008/why-cluster-an- introduction-to-asterisk-clustering-and-database-integration-astricon-20 08.pdf is still up to date or if there are newer or improved features available with 1.6 (or 1.8) to build an easy scalable and highly available Asterisk infrastructure? Maybe someone knows of configuration examples or howtos for building a HA cluster with the most actual features? Probably the biggest difference I can think of is that all of the features in the future section at the end are now in released versions of Asterisk. The other thing that is in 1.8 betas that may be of interest is distributed device states and MWI over XMPP PubSub. There are some interesting use-cases there that can provide some nifty unified communications integrations that just doing distributed device states over OpenAIS can't. Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Firmware
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, July 21, 2010 9:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco Firmware IMO, Polycom's got the right idea. If you want the latest and greatest software and support, break out your checkbook. If you're content to be a version behind and can take care of yourself, go for it. Actually, Polycom no longer makes you be a release behind for either SIP firmware or the BootROM. You can now download the latest from their website without any contract of any kind. Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on nortel sip connection
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, June 18, 2010 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on nortel sip connection I am using asterisk 1.4.32 and wish to connect using SIP to a nortel 1000 switch with the ability to have 90 calls at a one time outgoing or incoming. the nortel reseller is asking me what to do. I dont know nortel at all. I thought I just needed a SIP trunk and IP address of the their server and an account name, and provide her my IP address. They didn't know what to do with that. What do I tell them? I've successfully set up SIP connectivity to a Nortel CS1000, but it required a SIP proxy in between. The major issue I came across is that Nortel (at least in Succession 4.0 and 4.5, not sure about later versions) uses the maddr URI parameter in an RFC-compliant but otherwise unseen (at least insofar as I've come across) way that Asterisk does not handle gracefully. In order for this to be successful, you'll definitely need to determine what version of Succession they're using and, if it's 5.0 or later, if they are using the newer COTS-based servers with the SIP proxy functionality. You'll probably still need your own proxy, but but some initial testing I did when I had the time indicated that some features (transfers, in particular) may work a lot better in the never version(s). You'll also need to figure out exactly what will be handled by the Asterisk system, because call routing can kind of weird with these boxes. At least in the older versions of Succession, they tended to treat SIP trunks as second-class citizens. As a result, you may end up needing to configure the Nortel to think of the Asterisk box as a trunk of last resort. One other thing: were you planning on using voicemail on the Nortel (i.e., CallPilot)? That *can* work if you want it to, but it's yet another can of worms in setting this up. Also, when I've done it in the past I have had precisely ZERO assistance from any Nortel reseller. So expect to end up learning far more about that side of this setup that you had wanted to. Feel free to ask questions about the particulars, but that's the quick lay of the land. Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Productivity Suite on Polycom IP7000
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Tuesday, May 04, 2010 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Productivity Suite on Polycom IP7000 Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes' setting that I need to toggle? Also, does anybody know any good therapists with a side-specialty of torn-out hair replacement? :-) According to the release notes (I'm looking at 3.2.3), 4-way conferencing is not possible on the IP7000s. In fact, any of the features that are supported that would otherwise require a Productivity License (LDAP, Conference Management) are available without any license. Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continuous bothering message -- Remote UNIXconnection disconnected
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Monday, April 05, 2010 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Continuous bothering message -- Remote UNIXconnection disconnected Hi Guys, i have a small issue but bothering me, after restarting asterisk (version 1.4 running on centos) i have the following message that comes repeatedly when i am connected to the CLI: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected does any one know how to stop this or if it's a sign of a more serious issue? i would appreciate any help, thanks! I would make sure that you don't have an extra copy of safe_asterisk running. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Thursday, March 18, 2010 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning I know for a fact that you can provision a Polycom via ftp. I've included much of my dhcpd.conf file below. Pick out what you need. Let me know if I can confirm that using option 66 will work with FTP (and HTTP, for that matter) with newer BootROM versions. I don't know the exact version it changed, unfortunately, as I just noticed it in passing when I was running some tests one day. As for why we ended up choosing option 129 originally rather than 160? I wish I had a clever technical explation, but it's just a random unassigned option number. That's it. :) - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kate AEL syntax ?
I have a basic config for AEL syntax highlighting for Kate if you would like it. - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Friday, July 10, 2009 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Kate AEL syntax ? Hi, Is there something available to add AEL2 syntax highlighting support to Kate ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nortel pbx dtmf issues
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernest Byaruhanga Sent: Thursday, July 02, 2009 4:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nortel pbx dtmf issues folk, I see from the archives that the issue of nortel handsets not sending dtmf tones to asterisk has been discussed a couple of times, but there is something I quite havent seen answered yet. Is this dtmf issue a problem with the nortel handsets or the PBX itself? If the handset were changed to another one, would this issue be solved? Or must the PBX be changed completely? The only issue that I've experienced with respect to DTMF and a Nortel PBX is when using SIP trunking. Nortel used/uses (newer versions of CS1000 software, 5.0 and newer IIRC, can use RFC2833-based DTMF) a proprietary SIP INFO-based DTMF. As it stands, Asterisk is capable of receiving and interpreting those messages properly. But sending DTMF to the Nortel is an issue without code modification. If you have a specific need for that, the changes themselves are fairly trivial, and I can detail those for you if you'd like. I might even have a patch laying around that would work or could be made to work, since we needed to integrate into an older version of the CS1000 software (since upgraded, so our current system passes DTMF via RFC2833). - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why would you think that more bug fixes would be in it? Security release are only supposed to have the fix for the issue that caused the release to take place. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing LUA
Hello, all. The little bit of reading I've done on lua makes me eager to give it a try. However, when I try to install it (Asterisk 1.6.1.1 on CentOS 5.3), it is not available in menuselect. I have installed lua and lua-devel. I've seen very little about it in my Internet searches. What else must I do so that it installs? Thanks - John What do you see in the config.log in the asterisk source directory with respect to lua (say, the output of 'grep -i lua config.log)? - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing LUA
Wow! Definitely a non-trivial patch. Alas, it does not work but the errors are different: [compu...@pbx01 asterisk-1.6.1.1]$ grep -i lua config.log configure:42697: checking for luaL_newstate in -llua5.1 configure:42732: gcc -o conftest -g -O2 conftest.c -llua5.15 /usr/bin/ld: cannot find -llua5.1 | char luaL_newstate (); | return luaL_newstate (); configure:42960: checking for luaL_newstate in -llua-5.1 configure:42995: gcc -o conftest -g -O2 conftest.c -llua-5.15 /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `sqrt' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `floor' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `ceil' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `cosh' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `tan' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `tanh' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `asin' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `log' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `atan' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `sinh' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `fmod' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `acos' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `exp' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `sin' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `pow' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `atan2' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `cos' /usr/lib/gcc/x86_64-redhat-linux/4.1.2/../../../../lib64/liblu a-5.1.so: undefined reference to `log10' | char luaL_newstate (); | return luaL_newstate (); ac_cv_lib_lua5_1_luaL_newstate=no ac_cv_lib_lua_5_1_luaL_newstate=no LUA_DIR='' LUA_INCLUDE='' LUA_LIB='' PBX_LUA='0' I'm guessing there are differences in the API between what CentOS has installed (well actually the testing RPM I found to upgrade to 5.1 from 5.0) and what * expects. Given that, I would imagine it is not safe to manually edit makeopts. Thoughts? Comments? Insults? Thanks - John -- My guess is that when running the compile test ( This line: 'configure:42995: gcc -o conftest -g -O2 conftest.c -llua-5.15' ) it is necessary to add '-lm' in order to link in the standard math library. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing LUA
My guess is that when running the compile test ( This line: 'configure:42995: gcc -o conftest -g -O2 conftest.c -llua-5.15' ) it is necessary to add '-lm' in order to link in the standard math library. - Brad One more bit of magic necessary here, as pbx/pbx_lua.c has includes for: #include lua5.1/lua.h #include lua5.1/lauxlib.h #include lua5.1/lualib.h On Redhat-based systems, it needs to be: #include lua.h #include lauxlib.h #include lualib.h ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing LUA
That worked. The system is still in enough of a test phase that I can destroy it again and rebuild it if you'd like to send me a new version of the patch. Thanks - John ARGH Not so good. Asterisk now segfaults on start up :((( - John Now that is a behavior I'm not seeing, although to be fair I'm using Fedora 9 to compile/test and not CentOS. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write custom functions in AEL2 ,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, May 11, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to write custom functions in AEL2
,
Hi,
I'm using asterisk 1.6.1 and AEL2.
I'm trying to find the best way to write my own custom functions
?
At the moment, I'm using this pattern (extensions.ael) :
context foo {
123 = {
myfunc(123456);
NoOp(${GOSUB_RETVAL});
};
macro myfunc (arg) {
Return (${arg});
}
1. First, I keep getting warnings like
Warning: file /etc/asterisk/extensions.ael, line 446-446:
application call to Return affects flow of control, and needs to be
re-written using AEL if, while, goto, etc. keywords instead!
and I would like to get rid of them.
Unfortunately, AEL does not support using return with a value at the
moment. There is a patch on Reviewboard that does this, as well as
*simple* direct assignment from an AEL macro return:
http://reviewboard.digium.com/r/114/
2. Secondly, I would like not to use GOSUB_RETVAL and call a
custom function just like I'm calling other functions with statements
like :
123 = {
NoOp(TOLOWER(fOo BaR));
NoOp(myfunc(123456));
};
What would you advise me to do ?
That requires rather a lot more work than the above patch, but if you
use the direct assignment at least you needn't worry about GOSUB_RETVAL.
Regards,
- Brad
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Re: [asterisk-users] Asterisk 1.6.1-rc1 with OpenAIS and differentsubnets
The method OpenAIS uses to communicate between nodes is designed for a very low latency local connection; it is not designed to work across routed connections. Russell Bryant has spent some time talking to the OpenAIS developers about this, but so far there doesn't seem to be a good solution. true, that's why i'm hoping that distributed presence via dundi comes about sooner, rather than later :) I'm not sure if this is applicable to your environment, but what would you think about distributed events (including devicestate and voicemail MWI as in the res_ais implementation) over XMPP PubSub (XEP-060)? I've really only just begun, but at least it's something that somebody is working on currently. AFAIK, that is not yet the case with events via dundi. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
No, if-then-else works fine inside a case statement. See inline
comments.
switch(${DIALSTATUS})
{
case NOANSWER:
{
This brace, and its closing-brace mate, are superfluous though not
harmful.
// if-then-else not permitted
If (${ael-var} = 1)
Your primary problem is probably right here, the if needs to be all
lower-case ( If != if ).
{
Playback(beep);
return;
}
}
Again, unnecessary.
case BUSY:
{
return;
}
default:
{
Hangup();
};
}
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Re: [asterisk-users] AEL and };
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Klaus Darilion
Sent: Thursday, January 08, 2009 8:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AEL and };
Hi!
All the AEL examples have a semicolon after the closing curly
bracket, e.g:
context test {
1 = Hangup();
};
but without ; it works fine too, e.g:
context test {
1 = Hangup();
}
So - what is the reason for the ; after the closing curly bracket?
In the original implementation of AEL, it was required to have the
semicolon after closing a block. In the new implementation (AEL2), it
is not required but is allowed for backward compatibility with dialplans
written for the earlier parser.
- Brad
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Re: [asterisk-users] AEL Variable Warning Messages
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent
Davidson
Sent: Wednesday, December 31, 2008 1:03 PM
To: m...@digium.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] AEL Variable Warning Messages
Well, before I file a bug I have another question... In AEL,
what is the correct syntax? Do all variable references still need to be
wrapped in ${} or not? If they do, then the documentation on
voip-info.org needs to be changed to reflect that.
Yes, variable references need to be wrapped in ${}. Where on the wiki
do you see an example that is otherwise? I just looked at the main
documentation for AEL, and I didn't see any instances of it. Certainly
they can and should be fixed if they are there.
Beyond that, what are the rules for putting the values assigned
to variables in quotes? In my example above, at one point I had a space
between the = and the Zap/r2 statement with no quotes. The value
assigned to TRUNK then included a leading space. I didn't test to see
whether or not putting a space after the variable name adds a space to
the variables name. I would think that any spaces after an operator
should be ignored unless the come after a single or double quote.
The rules are that there aren't any really. Neither a single- nor
double-quote have any specific meaning in the sense of signifying a
string. I'm also curious to know where you saw an example of assignment
that used quotes of any kind, since I can't find that either.
Regards,
- Brad
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Re: [asterisk-users] 1.6 upgrade issues
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Tilghman Lesher
Sent: Tuesday, December 16, 2008 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.6 upgrade issues
[outbound] has entries similar to the following:
exten = _0[123],1,Macro(outbound,${EXTEN},
provider1, provider2)
the macro outbound is defined in extensions.ael as follows:
macro outbound (number, route1, route2) {
dosomestuff;
}
This has worked fine in 1.2 and 1.4, but seems to be
choking on 1.6. I've
looked through the various changes.txt files, and have read
mention of
replacing macro calls with Gosub(), but I'm not sure that's
relevant to
this issue.
It is precisely relevant to this issue. All subroutines,
whether they're
called macros or not, in AEL (in 1.6) are Gosub routines. So
to invoke that
subroutine, you need to call out with Gosub, not with Macro.
So it probably
should be along the lines of: Gosub(outbound,s,1
(${EXTEN},provider1,provider2)).
Also, as a result of AEL using GoSub now for what it calls macros, the
contexts are indeed named the same. You will need to have one or the
other change names.
- Brad
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Re: [asterisk-users] 1.6 upgrade issues
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Atis Lezdins
Sent: Tuesday, December 16, 2008 5:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.6 upgrade issues
On Tue, Dec 16, 2008 at 8:36 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Monday 15 December 2008 22:03:37 Chris Bagnall wrote:
Greetings list,
Over the last few days I've been gearing up to replace a
couple of our
servers with 1.6 as something of a testbed, but I'm
encountering a few
problems, and wondering if anyone can help...
In extensions.conf, there are a number of contexts defined
for each group
of users, along the lines of: [groupa] [groupb] etc.
In each of those, there's a command include = outbound
[outbound] has entries similar to the following:
exten = _0[123],1,Macro(outbound,${EXTEN},
provider1, provider2)
the macro outbound is defined in extensions.ael as follows:
macro outbound (number, route1, route2) {
dosomestuff;
}
This has worked fine in 1.2 and 1.4, but seems to be
choking on 1.6. I've
looked through the various changes.txt files, and have
read mention of
replacing macro calls with Gosub(), but I'm not sure
that's relevant to
this issue.
It is precisely relevant to this issue. All subroutines,
whether they're
called macros or not, in AEL (in 1.6) are Gosub routines.
So to invoke that
subroutine, you need to call out with Gosub, not with
Macro. So it probably
should be along the lines of: Gosub(outbound,s,1
(${EXTEN},provider1,provider2)).
Actually there's ampersand operator prefixing macro name, so AEL
parser will automatically check dependencies etc:
outbound(${EXTEN},provider1,provider2);
Regards,
Atis
Yes, but in his instance he's mixing AEL-written dialplan with regular
extensions.conf-style dialplan. He's trying to call a macro where AEL
in 1.6 is not creating a macro in the extensions.conf sense. It is
creating a subroutine to be used with GoSub. Therefore, there is no
context called [macro-outbound] if you do a dialplan show.
Were he calling this macro from AEL, then yes moving from 1.4 to 1.6
would have been more transparent although he would still have the issue
of overlapping context names (his handwritten outbound context versus
the subroutine AEL is generating).
- Brad
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Re: [asterisk-users] polycom no menu
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, December 04, 2008 7:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] polycom no menu Was messing with a polycom 501 and changed the IP from dhcp to static. Working with a user remotely. Now, the user says the phone does not show anything on the LCD and does not respond to any buttons. When rebooting, there is text shown as it proceeds. ?? Is there a way to reset this to a default? Does not respond to ping on the address we set. joe a. You could try lobotomizing it by pressing and holding 468* You'll need to enter the password (456 if you haven't changed it) and then hit the leftmost softkey. Obviously, this will all be blind since it doesn't display anything. But it's worth a shot. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, December 02, 2008 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2,1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released On Tue, Dec 2, 2008 at 8:22 PM, Dave Fullerton [EMAIL PROTECTED] wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 I see a reference in the 1.6 changelog that refers to SENTINEL not existing in 1.6.0 2008-06-27 01:09 + [r125648-125684] Mark Michelson [EMAIL PROTECTED] * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 -Dave ACK [CC] manager.c - manager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 Regards, Atis Looks like branches/1.6 got the trunk version of a fix to OpenBSD compilation rather than the 1.4 version as it should have. 1.4: http://svn.digium.com/view/asterisk/branches/1.4/main/manager.c?view=dif frev=159897r1=159896r2=159897 Trunk: http://svn.digium.com/view/asterisk/trunk/main/manager.c?view=diffrev=1 59898r1=159897r2=159898 - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip forking needed for ekiga 3.0
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murrell Sent: Friday, September 26, 2008 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0 I've read enough of the thread to know the Asterisk issue you are trying to describe is loop detection and not forking. Asterisk does support forking: Dial(SIP/user1SIP/user2) is forking. Not being able to handle duplicate requests from different IPs is loop handling and you'll already find bugs open about that. I will relay your description of loop detection back on to the Ekiga guys. I'm just the messenger here. Can you provide a SIP trace of the signalling taking place? What are you getting back from the Asterisk box? - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multipleservers
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, September 10, 2008 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Write Asterisk CDR MySQL records to multipleservers On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote: Hi to all, I actually have an asterisk server configured to write CDR mysql records in the same machine (localhost), but I want to write this records to another machine also in mysql at the same time, It is possible? It means that I want save the records in both machines. You can either use MySQL replication or you can use 2 different CDR drivers at the same time, such as ODBC, with the Mysql-ODBC-Connector and the MySQL CDR driver. Also, in 1.6, cdr_adaptive_odbc allows you to specify multiple CDR backends within the same configuration file. -- Tilghman It's also likely that you could use MySQL Proxy to achieve the result you want. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Tips and Tricks: Dynamic Subroutines inAGI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Sessions
Sent: Thursday, August 28, 2008 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Tips and Tricks: Dynamic
Subroutines inAGI
...
The hurdle in doing something like this was how to
dynamically execute
a subroutine from the results of the database query which
were dumped
into a variable. The method I used with the subroutine reference
doesn’t allow for arguments to be passed (if anyone finds / knows a
way to do this, let me know), so I use global variables.
This is a simple example of dynamic subroutine execution
(without the
database query):
use strict;
use warnings;
our $called_number;
our $calling_number;
sub run_me {
$AGI-verbose(”Called Number = “.$called_number, 1);
$AGI-verbose(”Calling Number = “.$calling_number, 1);
}
sub set_variables {
$called_number = “8005551212″;
$calling_number = “300222″;
}
sub dynamic_execute {
my ($sub) = @_;
if (!$sub) {
$AGI-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
If you don't mind disabling strict refs (no strict 'refs';), you could easily
do this.
This would allow you to use something like: $sub($argument1, $argument2);
The only other way I can think of (though I have not tried it) would be to
populate a hash with subroutine refs and use the string as the index into it.
Something like this:
#!/usr/bin/perl
use strict;
use warnings;
sub print_ref { print @_; };
my %sub_hash = (print_ref, \print_ref);
sub print_stuff {
my $sub = shift;
my $string = shift;
$sub($string);
}
print_stuff($sub_hash{print_ref}, This is printed.\n);
The first idea uses the symbol table directly, and the second one essentially
is building your own symbol table.
Hope that helps,
- Brad
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Thursday, July 31, 2008 3:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1 if after you tried both straight through crossover cables and it still give you RED alarm. just tell them you can't get any clocking signal. they'll probably send someone on site and test the line. Yes, I tried all sorts of cables and ended up getting the local contact to complain to NETCOM. An engineer came and swapped the Fast Ethernet to E1 converter. Now we use a normal RJ45 cable to connect the converter to TE412P card. The lights turns green but changes to yellow and green again. dmesg shows a continuous stream of: wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 ...and I am using the following in zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 ... I have changed the timing source from 1 to 0 to 2 but it doesn't make any difference. Any thoughts? Sounds like you're making progress. I would try the above span definition without the crc4. That might do the trick. Regards, - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()
Russell Bryant wrote:
This is a slightly different approach, but have you seen the
state interface
code that is in Asterisk 1.6? There is a backport of the
code for 1.4 floating
around somewhere, I think. It allows you to specify a
different device for a
queue member that app_queue will use to determine the state
of an agent. So,
you can still list a Local channel for dialing, but Asterisk
will look at the
state of SIP/myphone, for example, to know whether the agent
is busy or not.
Alternatively, if you would like to control the usability of
an agent through
the dialplan, then you could use the DEVICE_STATE() function
to create a custom
device state. Then, you could list your custom device as
what app_queue
should look at before attempting to call the agent.
One problem with that cunning plan is that using custom device states
doesn't work. The code for handling device state changes in app_queue
is looking for a forward-slash in the device name, and returns if it
doesn't find one:
loc = strchr(technology, '/');
if (loc) {
*loc++ = '\0';
} else {
ast_free(sc);
return 0;
}
I've worked around it by modifying that particular bit of code, though
in a way I'm not sure I'd want committed to mainline Asterisk SVN (which
is why I haven't submitted it yet).
- Brad
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Re: [asterisk-users] Polycom LDAP Corporate Directory
I actually just ordered 50 licenses to give this and the other applications a try. I'll post my results to the list once I get them and have had a chance to play around. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of faraz Sent: Friday, April 18, 2008 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory I havent tried it. I have quite a few polycoms and didnt even know polycom had this feature! :) This is obviously a separate peice of software that must be purchased and installed on the phones. Looks amazing though- any idea on pricing?. On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote: Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip /applications/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Baker Sent: Thursday, March 27, 2008 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards No actually I gave it a LOT of thought and I and even asked two different techs who repair PCs and both said what the to vendors are saying is not sufficiently clear that I would make a purchasing decision based on what you have in hand from them But, nice try at the cheap shot. The two techs you spoke with are obviously not familiar enough with the technology at hand to be able to make that determination, because all of the necessary information is contained within the original e-mail and the specs available on Digium's website. To simplify, however, here is the answer: The TE420 cards will fit in the regular x8 (not low-profile) PCI-E slots, and the TE412P will fit into the PCI-X slots. Regards, - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hackensack Sent: Monday, March 24, 2008 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Passing variables over IAX2 -- IAXVAR patch? Isn't this channel specific? Why is this being added? It does not work with SIP. It does not appear to be architecturally generic. This gets added, but yet a channel specific enhancement for SIP that would be beneficial for endusers does not get added. Again, Asterisk is good at transferring calls around, but when it comes to end users, the developers just keep closing the tickets on the much needed features. Being able to pass variables around between systems is by *definition* channel-specific, since the channel driver is the module responsible for speaking a given protocol. Besdies, SIP already has (and has had for a long time) a method for doing this (SIP headers). So does ISDN, for that matter (IEs). Also, insofar as changing things in the core for this, it isn't necessary. Asterisk has had channel variables for at least as long as I've been working with it. This is really just an extension of that so that there is a way to send this kind of information to remote systems. Finally, chiming in on this issue in the way you did is fairly childish. It's not really related, even, just a whine about a feature which you (incorrectly) equate with one that got shot down. Regards, - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status unavailable
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add
it to the
end of the hint list to hint-the-hint. Something like:
exten = 234,hint,SIP/myphonePSEUDO/234
exten = *78,1,ChanAvailIs(PSEUDO/234,Busy)
exten = *791,ChanAvailIs(PSEUDO/234,Unknown)
This could be very useful for presence indication.
Huh, this hint hint would be useful for queues with local channel
state_interface too.. i think some general usage way could be added to
allow combining of device states.
Regards,
Atis
Machinations with func_devstate is the droid you're looking for.
However, there is an issue with the current use of state_interface in
app_queue where it is required to have a '/' character in it (obviously
would for Channels, but custom device states are of the form
Custom:yourdevicestate). I've worked around it, but I've been meaning
to file a bug report about it.
Anyway, have a look at that. It is being used successfully by us (in
1.4, with Russell's backported func_devstate and custom changes to fix
the aforementioned issue).
Regards,
- Brad
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Re: [asterisk-users] TE412P and Delll PowerEdge 2900
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ash Rah Sent: Wednesday, February 06, 2008 4:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TE412P and Delll PowerEdge 2900 Hello, Looking for comments if Digium TE412P (32-bit 33MHz card keyed for 3.3 volt operation) compatible with Dell PowerEdge 2900 server board (1 PCI Express X8, 3 PCI Express X4, 2 64-bit/133MHz PCI-X)? Any know issue with Digium cards for this server family? Thanks in advance. Ash. I have successfully used that card in several 2950s, and they should be pretty similar. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL includes?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore Sent: Thursday, January 17, 2008 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AEL includes? How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay That is exactly the syntax that you should be (and I am) using. I don't know why that wouldn't work, unless you're using an older version of Asterisk and are using fully-qualified paths. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom VLAN
The switch on the Polycom will pass the frames on unchanged, so if they are untagged from the PC they will remain that way. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Wednesday, January 02, 2008 12:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom VLAN Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the packets I send from my PC(on the PC port of the phone) have the same VLAN tag? THe PC is sending untagged packets. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell, HP, Digium, homebrew - what do you use
Dozens of Dell PE2950s, mostly dual Xeon 5150s with 4GB RAM and two 73GB drives. Some have TE412Ps and some have TE420Bs. Also, 14 PE2850s (dual 3.0GHz, 4GB RAM, dual 73GB drives) with a mix of TE411Ps and TE412Ps. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent Callback Login in 1.4
On a side note, does anyone have the URL to the AEL example so I can write out an extensions.conf version for the wiki? - -- Kind Regards, Matt Riddell Director It's called queues-with-callback-members.txt in the /docs directory in the source tree. - Brad ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote: I guess that's my point. I realize asterisk is open source and FREE, however, I wouldn't expect a commercial application to crash as often as I've seen asterisk go down. Windows 98. wouldn't expect != haven't experienced Actually, I personally did expect Win98 (and worse, ME) to be as terrible as it was. For once, a Microsoft product fulfilled my every expectation! :D - Brad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Horseshit. Prior art is trivial. How old is Hylafax? Cheers, -- jra It's never trivial if you're a small company. J2 has already won settlements from several smaller companies, which gives it precedence. Once precedence is established, it's almost a done deal for future lawsuits and fighting them is exponentially harder with each settlement they get. While it may boil down in the end to prior art, having the money to fight that far in the legal system is something else. A fight like that would put most small businesses under, and forget about getting external funding if you have this hanging over your head. No one wants to fund a company with a lawsuit against it. N. The really good news here is that the recent KSR vs. Teleflex ruling as decided by SCOTUS gives you a fair bit of firepower in at least getting this patent reexamined, even with the precedent. I think that J2 would be quite unlikely to try and push a case forward in light of this. Regards, - Brad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
What we really need is for someone to pay Allison and get the lyrics recorded in her voice. ;) BTW, you just wasted about 30 minutes of my time while I looked around that site at the versions written in languages I've used over the years. :) - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Thursday, August 16, 2007 7:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 99 bottles of beer On Thu, 16 Aug 2007, Diego Iastrubni wrote: DUD! THIS KICKS ASS! (I know I am getting into trouble, but hey! it's already in our PBX!) Heh... Well I updated it and added some lyrics (and the guys from the website have said they'd put it up!) So if you want to hear a (rather odd!) mix of me Allison, then dial +44 1364 698 225. I started it at 3 as you don't want to hang about all day, I'm sure :) Get the updated code from wget http://www.drogon.net/dsx/extensions.99bottles and if you want my lyrics, then wget http://www.drogon.net/dsx/bottles.tar.bz2 Now back to our scheduled programme ... :) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce=584760da used in sip packets?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: Wednesday, August 15, 2007 7:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] why is nonce=584760da used in sip packets? This causes the asterisk server to send another unauthorisation response with an additional parameter stale in WWW-Authenticate section as shown below --- Transmitting (NAT) to 208.120.167.146:80 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c;received=208.120.167.146 From: sip:[EMAIL PROTECTED];tag=as1acc7245 To: sip:[EMAIL PROTECTED];tag=as1d329593 Call-ID: [EMAIL PROTECTED] CSeq: 19710 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4f90fab4, stale=true Content-Length: 0 this stale=true field causes the asterisk server to display the following NOTICE on the cli NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from 'sip:[EMAIL PROTECTED] ' and this will continue happening unless the next register request uses the nonce field recieved in latest unauthorisation response from server, and untill then the user agent will not be able to register with the server. This will cause problems in our services. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce=584760da used in sip packets?
You have on your hands a broken UA, since it is not responding to the changing nonce value. - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: Wednesday, August 15, 2007 7:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] why is nonce=584760da used in sip packets? --- Transmitting (NAT) to 208.120.167.146:80 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK722c974c;received=208.120.167.146 From: sip:[EMAIL PROTECTED];tag=as1acc7245 To: sip:[EMAIL PROTECTED];tag=as1d329593 Call-ID: [EMAIL PROTECTED] CSeq: 19710 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4f90fab4, stale=true Content-Length: 0 this stale=true field causes the asterisk server to display the following NOTICE on the cli NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from 'sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] ' and this will continue happening unless the next register request uses the nonce field recieved in latest unauthorisation response from server, and untill then the user agent will not be able to register with the server. This will cause problems in our services. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] global variables and updates
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian Lyndon-Smith
Sent: Saturday, July 28, 2007 5:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] global variables and updates
Sorry if this appears twice - I originally sent it nearly 18
hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I
have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as
1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
The changing of variables is not atomic as would hope, but there is a
solution for you. Look the application MacroExclusive. Put your Set to
increment the global variable inside of a macro and call it using this,
and you will get the behavior you desire. One caveat, however, is that
you will want as little logic as possible inside of this macro.
MacroExclusive will block all other calls to this macro until the first
one exits. But this is not an issue if all you are doing is a quick
var++ and then leaving.
- Brad
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Re: [asterisk-users] Will the Sangoma A40003X fit in a 2950?
I'll assume you mean a Dell PowerEdge 2950. Sangoma's web site says the cards dimensions are 55mm(H) x 290mm(L). A Full-Length PCI card is 107mm(H) x 312mm(L). According to the PowerEdge 2950 Getting Started Guide Page 10: Left riser PCI-X option: two full-height, full-length 3.3-V, 64-bit, 133-MHz (slots 2 and 3) OR PCIe option: one full-height x8 lane 3.3-V (slot 2) and one full- height x4 lane 3.3-V (slot 3) Based on that, I'd wager that it will fit. You may need a molex power connector available in the server though. Make sure you do your research. The 2950 has a molex connector for its optional internal tape drive. If you get a molex extension (at least 12), then you will be able to make it to the expansion slots. I have done with with a Digium TDM2400 and a PowerEdge 2950. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network routing
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Thursday, June 28, 2007 3:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] network routing This allows me to edit the IP Address of the NIC card, but not edit my IP routing. In your instance, you're trying to add a default gateway. Therefore, in your /etc/sysconfig/network-scripts/ifcfg-ethX file: GATEWAY=XXX.XXX.XXX.XXX If you need others, create /etc/sysconfig/network-scripts/route-ethX and use this format: GATEWAY0=1.2.3.4 NETMASK0=255.255.255.0 ADDRESS0=1.2.3.0 GATEWAY1=1.2.3.5 NETMASK1=255.255.255.0 ADDRESS1=1.2.3.5 And so forth. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Thursday, June 21, 2007 7:12 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql Yes mysql installed [EMAIL PROTECTED] asterisk-1.4.5]# rpm -q mysql mysql-4.1.20-2.RHEL4.1 You need mysql-devel - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Ex-Girlfriend Logic in 1.4.4
What does the output of 'show dialplan start' look like?
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Douglas Garstang
Sent: Tuesday, June 19, 2007 3:20 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Ex-Girlfriend Logic in 1.4.4
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/5000-0a281f80,
) in new stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/5000-0a281f80,
1) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/5000-0a281f80,
19256002182) in new stack
-- Executing [EMAIL PROTECTED]:4] Playback(SIP/5000-0a281f80,
tt-monkeys) in new stack
-- SIP/5000-0a281f80 Playing 'tt-monkeys' (language 'en')
However, when I change the extension match to:
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can
see the caller id number is 19256002182 from the NoOp() when
it does work. What am I missing here?
Doug.
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RE: [asterisk-users] Gigabit SIP Phones
Today, buying extra ports for stations having extra bandwidth requirements is acceptable as 10/100 LAN access is the norm. But it could be painful to explain executives, every IP Phone you bought during 2007 will not keep up with 1GE LAN. There is one other issue - I don't think there is a commercial PoE solution for Gb Ethernet. I know a solution does exist, as I have used kit using it, (not phones), but I'm not sure what the commercial issues (ie. economic fesability) for in-office stuff is yet. How about this: http://www.cisco.com/en/US/products/ps7077/prod_models_comparison.html Or this: http://products.nortel.com/go/product_content.jsp?segId=0catId=nullpar Id=0prod_id=49760locale=en-US Or maybe this one: http://www.force10networks.com/products/s50v.asp Alternatively: http://www.extremenetworks.com/products/summit-x450e.aspx So yes, there are commercial PoE with Gbe solutions. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Write to multiple databases as redundancy scheme
UltraMonkey (www.ultramonkey.com) and MySQL Cluster (http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster.html) It works a charm. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Moore Sent: Friday, June 08, 2007 2:13 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Write to multiple databases as redundancy scheme On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What happens then when the primary box fails? CDR not written to the primary can't be replicated... -- Justin Moore aka wantmoore --- www.wantmoore.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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Reload in 1.4 clears regexten
Please post the relevant portions of your sip.conf and extensions.conf I'll bet dollars to donuts you have the same context defined as both your regcontext and as a context in extensions.conf (or an .ael, or whatever). - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, June 06, 2007 7:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Reload in 1.4 clears regexten Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] PRI Partial Re-Rounting
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Treble Sent: Thursday, June 07, 2007 10:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: RE: [asterisk-users] PRI Partial Re-Rounting Matthew, I'm not sure what you mean when you say, [u]nfortunately though, none of the switch types support this variant of this function. Could you elaborate please. TIA. I'm sure he'll correct me if I'm wrong, but I believe he means that the protocol implementations as they exist in libpri do not support these features at the moment. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Reload in 1.4 clears regexten
So it is, I was wrong. What do you get when you do a 'show dialplan sip_autoreg'? Does it show pbx_config or anything like that, or does it say SIP? In theory at least (though I'd have to peek at the code again to refresh my memory), contexts that aren't created by pbx_config should not get destroyed when you do an 'extensions reload'. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, June 07, 2007 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Reload in 1.4 clears regexten Brad, I can't post the entire contents of sip.conf and extensions.conf/extensions.ael, but as you can see below, I don't have a sip_autoreg defined anywhere in my dial plan. [EMAIL PROTECTED] asterisk]# cat sip.conf [general] context=default allowoverlap=no bindport=5060 bindaddr=xxx.yyy.34.201 srvlookup=yes regcontext=sip_autoreg [EMAIL PROTECTED] asterisk]# grep sip_autoreg extensions.conf [EMAIL PROTECTED] asterisk]# grep sip_autoreg extensions.ael [EMAIL PROTECTED] asterisk]# Douglas. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, Bradley Sent: Thursday, June 07, 2007 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Reload in 1.4 clears regexten Please post the relevant portions of your sip.conf and extensions.conf I'll bet dollars to donuts you have the same context defined as both your regcontext and as a context in extensions.conf (or an .ael, or whatever). - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, June 06, 2007 7:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Reload in 1.4 clears regexten Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 can qualify=yes trigger some external event?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Friday, June 01, 2007 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] how can qualify=yes trigger some external event? Hi all, The option qualify=yes allows Asterisk to check if it can reach the peer. If the device does not answer within the time-out period, Asterisk considers the device off-line for future calls. Is it possible to use this feature to trigger some external event, in case of failed reply from the peer that is tried to be reached? How can that be done? Regards, Ricardo. Hi, Ricardo. Currently there is no way to do this in a pure configuration-only sort of way. However, if you're even moderately adept at C a cursory glance through chan_sip.c will show that it would be quite straightforward to modify the code in order to allow this. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 can qualify=yes trigger some external event?
qualify=yes generates events that can be viewed from AMI, they are: 'Event: PeerStatus' 'PeerStatus: Lagged' 'Event: PeerStatus' 'PeerStatus: Reachable' The other fields give the peer name and like, for more details view the chan_sip.c source, the calls you are interested in there are to a function called manager_event(). And so you are right, I wasn't thinking about AMI for some reason. Yes, that's an entirely plausible way to have actions performed when the event occurs. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Blindside Web Conferencing
Thanks Stefan! I was just thinking the other day that it would be great if I could whiteboard in Spark. Back on topic, I'm definitely interested in this web conferencing app. I'll have to check it out once a .war is made available and I have a few spare moments. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter Sent: Monday, May 28, 2007 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blindside Web Conferencing -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yes, we are looking for that. Do you know of any projects that provides those? I know one written in TCL/TK. You might also want to have a look at http://www.version2software.com/v2whiteboard.html - its a plugin for the Java based Jabber client Spark (from igniterealtime.org) =Stefan -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGW0DZcVCZDfrn+pMRAq4KAJ961ZBIsSNhn7p4+SQI4RPPe1gsHwCdG4dv pQOw6ugERcCUKy7pjDHf/qs= =JI7J -END PGP 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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 or Linksys phones bootp tftp config setup
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, May 25, 2007 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom or Linksys phones bootp tftp config setup Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Can you attach the trace, or at least let me know what DHCP server you are using? The Polycoms, at least, require that DHCP option 66 use the Microsoft-style DHCP behavior and actually encode it as a DHCP option (rather than a BootP header). On certain DHCP servers (Nortel at least I can say for sure), the default behavior is RFC-copmpliant (or at least so they say). The other responder has it right, though, that at least insofar as the Polycoms are concerned FTP is the default rather than tftp. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Poor man's High Availability solution
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, April 29, 2007 1:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Poor man's High Availability solution Who resells these products in the USA or at least ships here? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB I don't know about the Junghanns product, but we use these successfully here in the States: http://dataprobe.com/products/switch/aps/t-aps/index.html - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Nagios asterisk monitoring
Allow me to register my interest in any and all things that tie Asterisk information to Cacti. We use that here, and it's been on my to-do list for a lgg time. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse Sent: Wednesday, April 11, 2007 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nagios asterisk monitoring Yes, I have actually written a resource module for asterisk and the gui to use rrdtool to make REAL pretty gradient shaded graphs based on asterisk data. So, if you want the cacti script, email me([EMAIL PROTECTED]) to get me motivated to rewrite it and make it awesome, and encouragement would be great. But, with a pbx not a pretty graph maker, I am now working on clientside graphing using svg(z) and doing httprequests to get manager information. Let me know if you are interested in that also, I didnt realize how much of a community was out there for monitoring :] -brandon The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Passing a variable from one Asterisk box to another
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Bishop
Sent: Tuesday, February 20, 2007 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Passing a variable from one Asterisk box to
another
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro.
All works great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get
passed...
Does anyone have any clever ideas?
The correct way using SIP is to add X headers before the Dial and then
pulling them in and assigning them to channel variables on the ingress
box. Here's a snippet that shows the idea:
On the box dialing out:
exten = _23XX,1,Set(Foo=1234) --- Use Set here not SetVar
exten = _23XX,2,SIPAddHeader(X-Foo: ${FOO})
exten = _23XX,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
On the ingress box:
exten = _23XX,1,Set(Foo=${SIP_HEADER(X-Foo)})
exten = _23XX,2,Answer()
...yada yada
Regards,
- Brad
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RE: [asterisk-users] Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels
Are you saying that the Nortel will not allow you to set the clock to internal? If so that's unfortunate, as it's the only reliable solution for you in this situation. You really need your clock hierarchy to start at the received clock from the telco. - Brad From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Wed 2/14/2007 11:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nortel 81C MSDL Trunking to Asterisk TE110P,Nortel Resetting PRI Channels You have the PRIs set up to recover clock from the Asterisk box, is that what you want? If so, you certainly do *not* want span=1,1,0 or 2,2,0 since that will make Asterisk think the 81C should be clock master. Are there any telco-timed PRIs somewhere? If so, set up the PRIs on the 81C to be CLOK INT and then use span=1,1,0,esf,b8zs and span=2,2,0,esf,b8zs on the Asterisk box. I'm going to assume that a big system like an existing 81C already has the master clock set, but of course that will be a necessity if using internal clocking. - Brad They type of card we are using on the Nortel 81C will not allow clocking. The clock must be supplied by the Asterisk. We do not have any other clocking running into the Asterisk. Marlon Blair DOH, Network System Analyst (850) 245-4400, Cell (850) 528-4244 Fax (850) 412-1148 Work Hours 7 AM to 3:30 PM The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels
You have the PRIs set up to recover clock from the Asterisk box, is that what you want? If so, you certainly do *not* want span=1,1,0 or 2,2,0 since that will make Asterisk think the 81C should be clock master. Are there any telco-timed PRIs somewhere? If so, set up the PRIs on the 81C to be CLOK INT and then use span=1,1,0,esf,b8zs and span=2,2,0,esf,b8zs on the Asterisk box. I'm going to assume that a big system like an existing 81C already has the master clock set, but of course that will be a necessity if using internal clocking. - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, February 13, 2007 11:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nortel 81C MSDL Trunking to Asterisk TE110P,Nortel Resetting PRI Channels We are currently working to trunk from a Nortel 81C to an Asterisk Server 1.4 running on Red Hat Linux. We have two PRI trunks which work with the exception of the clock slips, which is causing the Nortel to reset the PRIs once a hour. Thanks for any suggestions. 81C MSDL Asterisk Digium TE110P REQ prt TYPE adan dch 10Zaptel.conf loadzone= us ADAN DCH 51 defaultzone=us CTYP MSDL span=1,1,0,esf,b8zs(also tried 1,0,0,esf,b8zs) GRP 1 bchan=1-23 DNUM 7 dchan=24 PORT 1 span=2,1,0,esf,b8zs (also tried 2,0,0,esf,b8zs) DES Asterisk VOIP bchan=25-47 USR PRI dchan=48 DCHL 51 OTBF 32 zapata.conf PARM RS422 DTE [channels] DRAT 64KC language=en CLOK EXT context=default IFC ESS5 switchtype=5ess SIDE USR signalling=pri_net CNEG 1 group=1 RLS ID 1 channel = 1-23 RCAP ND2 channel = 25-47 MBGA NO usecallerid=yes OVLR NO hidecallerid=no OVLS NO callwaiting=no T200 3 threewaycalling=yes T203 10 transfer=yes N200 3 canpark=yes N201 260 cancallforward=yes K7 echocancancel=no echocancelwhenbridged=no And I have immediate=no ADAN DCH 71 callreturn=yes Built the same asrxgain=0.0 ADAN DCH 31 txgain=0.0 musiconhold=default ROUT 1 We start up Asterisk in the following Type RDB order: CUST 00 modprobe zaptel Rout 30 modprobe wcte11xp DES ASTERISK_VOIP_1 ztcfg TKTP TIE safe_asterisk NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP CDP INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 7930 TCPP NO PII NO TARG CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML
RE: [asterisk-users] Conferencing Phones ...
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, February 09, 2007 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Conferencing Phones ... Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) I have an IP 4000, and I think the quality is excellent (on par with the analogs, which I also consider quite good). Most of our deployments continue to use fxs ports on a channel bank and analog phone, but that's mostly because we have a large investment in them. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 and Interface time out
What it actually does is tell the SIP channel driver to track whether or not any given peer has a call to it. It can then subsequently inform the Queue application so that another call will not be given to that user. If you did not have the ringinuse=no in your queue definition, you would then be able to receive up to 5 simultaneous calls (after five, then the SIP channel driver would return busy and Queue wouldn't be able to dial that peer). Regards, - Brad From: [EMAIL PROTECTED] on behalf of James Fromm Sent: Fri 1/19/2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. winmail.dat___ --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 and Interface time out
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Friday, January 19, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All member interfaces are SIP with only the basics configured in sip.conf. Even with 'ringinuse=no' configured, the Queue application continues to send callers to busy members causing them to get paused when their SIP device returns that it's busy. Does the Queue application need hints for member interfaces to determine their status? Thanks, James Queue does not need hints, but it does need the channel driver (in your case SIP) to inform it whether or not the member interface is in use. That is actually why I asked about call-limit. Can you try adding a call-limit (even if it's 10 or 20 or whatever) and see if that solves your problem? Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 and Interface time out
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Thursday, January 18, 2007 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no Am I missing something obvious? What do your SIP peers look like? Are you using the call-limit feature? - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] POE draw on Aastra 480i
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allen Casteran Sent: Friday, January 05, 2007 12:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] POE draw on Aastra 480i Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Can't seem to find any info on the Aastra site. Comments? I can't for sure with the Aastras, but I know a Polycom 601 only draws about 3.5-4 watts according to the command line of the switches we use (Nortel 5520). I can't imagine a 480i uses much more than that. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] cdr_addon_mysql.so did not register itselfduringload
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Savoy, Kevin - Williston, ND Sent: Thursday, December 28, 2006 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: FW: [asterisk-users] cdr_addon_mysql.so did not register itselfduringload Ok so I'm the only one not getting this to work. Maybe I'm doing something wrong. Here is the installation I'm using. Install Fedora Core 4 and do all the updates through yum. Then I install zdlib-devel, openssl-devel, newt-devel, gcc, gcc-c++ and then mysql and perl-DBD-MySQL all using yum install. Am I missing something? Something I'm installing I shouldn't be? After doing the Asterisk-Addons with ./configure, make and then make install as it instructs, the two files below do NOT exist anywhere on my system. Can I compile these manually? If so how? Help? You will also need mysql-devel, and if you pay close attention to the output of ./configure, it likely tells you that you don't have it. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] 1.4 with a nortel call server 1000 running SIP(sdp headers)
Actually, there was recently a bug fixed regarding multipart SDP parsing in chan_sip. That should have fixed the issue with CS1000s and SIP (among other things). I haven't actually tried it yet on my CS1000, but it should work. Regards, - Brad From: [EMAIL PROTECTED] on behalf of Jerry Geis Sent: Tue 12/26/2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1.4 with a nortel call server 1000 running SIP(sdp headers) Has anyone tried to get 1.4 running with a call server 1000 and SIP? I had 1.0.X running with a call server 1000 and had to tweek the code due to multipart SDP headers. Has multipart SDP headers been enhanced in 1.4. THanks, Jerry ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Match a Numer - then continue with dialplan
Please correct me if I'm misunderstanding your requirements, but see
below (inline) for what I would do:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Douglas Garstang
Sent: Tuesday, December 19, 2006 5:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue
with dialplan
I just know someone is going to ask 'why would you ever want
to do that?'. Here's my answer.
We have two companies, each with a dialplan similar to what's
below. In the event that the number being dialled does not
match any number within our OWN company, we want to set the
caller id to be a generic one for the company, NOT one for
the user. This is a pretty normal requirement that most
companies want. So, in the event that the logic flows beyond
coo1_OnNet, we want to reset the caller id of say, 3254001
Doug, to 3254000 Widgets Inc. If there was a way to match
against a number in the dialplan, and then continue execution
after that point, we could put this statement at the end of
the coo1_OnNet context and it would all be sweet. Without
that, I don't have a clue how to do this... unless we stick
with out current 3,000 line python script.
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = coo1_OffNet
[coo1_OffNet]
exten = _X.,1,Set(CALLERID(NUM)=3254000)
exten = _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten = _X.,3,Goto(syst_OffNet,${EXTEN},1)
The rest of this can stay untouched.
[coo1_OnNet]
exten = 3254101,1,Dial(SIP/3254101,20,tr)
exten = 3254102,1,Dial(SIP/3254102,20,tr)
exten = 3254103,1,Dial(SIP/3254103,20,tr)
exten = 1000,1,Answer
exten = 1000,2,Wait(1)
exten = 1000,3,NoOp(${FOO})
[syst_OnNet]
include = coo1_OnNet
include = coo2_OnNet
[syst_OffNet]
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],180,tr)
Regards,
- Brad
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RE: [asterisk-users] regcontext, NoOp extension vanishes when extension reload and doesn't come back
Let me guess: The context in which you have the 2 thru n priorities is the same one as you're using for regcontext right? Don't do that, bad things will happen (as you've noticed). I'd have to review the code again, but I think what you're seeing is as a result of this. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Tuesday, December 05, 2006 1:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] regcontext,NoOp extension vanishes when extension reload and doesn't come back Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a reload extension is performed, those dynamically created extensions in the regcontext vanish. Now this is ok, I understand why they vanish, but the strange thing is they don't come back when the sip device registration time expires. If I set the max regiter time of the device to be 60 seconds, after 60 seconds the phone sends another registration to the server, but since the user is already cached in, the NoOp priority 1 extension does not get re-created in the regcontext. I must perform a reload chan_sip.so, wait till the new registration hits and then the NoOp priority 1 extension is created again in the regcontext. This is a problem, if anything happens to the dialplan and it has to be reloaded, we loose active registered sip devices in the regcontext, then all hell breaks loose. Has anyone else come across this and has a work around? Ultimately, I'd like to see the regcontext function ensure the NoOp priority 1 extension is re-newed each registration cycle, whatever the time parameter is set on. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] regcontext, NoOp extension vanishes when extension reload and doesn't come back
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michiel van Baak
Sent: Tuesday, December 05, 2006 2:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] regcontext,NoOp extension
vanishes when extension reload and doesn't come back
On 13:59, Tue 05 Dec 06, Watkins, Bradley wrote:
Let me guess: The context in which you have the 2 thru n
priorities
is the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're
seeing is
as a result of this.
Then how should it be done ?
I'm playing with this as well and now I'm back to 0. I just
had it all working on paper...
You should put all of the 2 thru n priorities in a separate context and
then include the regcontext into that.
For example:
Let's say regcontext = registrations
And you have a SIP peer:
[1234]
type=peer
...
regexten=1234
You actual dialplan context should look something like:
[extensions]
exten = _1XXX,2,Dial(SIP/${EXTEN})
exten = _1XXX,3,Hangup
Include = registrations
Now, when peer 1234 registers, the registrations context will look like:
[ Context 'registrations' created by 'SIP' ]
'1234' = 1. NoOp() [SIP]
And, since the 'extensions' context includes 'registrations', any calls
that originate in 'extensions' will succeeed where they will not if 1234
is not registered.
Regards,
- Brad
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RE: [asterisk-users] RE: regcontext, NoOp extension vanishes when extension reload
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JR Richardson
Sent: Tuesday, December 05, 2006 3:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RE: regcontext,NoOp extension
vanishes when extension reload
Let me guess: The context in which you have the 2 thru n
priorities
is the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're
seeing is
as a result of this.
Regards,
- Brad
No, not exactly, I have a catchall match in the regext
priority 2 that sends the call out to another context that
further processes it.
The effect is the same, even though all you're doing is a Goto. The
problem stems from the fact that the context is created by
'pbx_config'. When you do an extensions reload pbx_config removes all
contexts for which it believes it is the owner and then starts from
scratch and creates all the dialplan entries in extensions.conf (OK, any
developers reading this will tell you it's more complex and it is, but
this is a close enough approximation). So when you have a context with
the same name as your regcontext defined in extensions.conf, then any
entries in that context will be removed and only the ones configured in
extensions.conf will be added back (note that below the regexten
priorities have [SIP] as the creator). It sounds like you've figured
that out on your own empirically.
For what I believe to be the 'correct' way (or at least *a* way that
won't make you pull your hair out) of working with regexten, see my
recent e-mail response Michael van Baak.
regcontext is sipregistration
astreg1*CLI show dialplan sipregistration [ Context
'sipregistration' created by 'pbx_config' ]
'53060' =1. Noop(53060)
[SIP]
'53061' =1. Noop(53061)
[SIP]
'53062' =1. Noop(53062)
[SIP]
'53063' =1. Noop(53063)
[SIP]
'53090' =1. Noop(53090)
[SIP]
'53091' =1. Noop(53091)
[SIP]
'53092' =1. Noop(53092)
[SIP]
'i' =1. Goto(lookupdundi|${INVALID_EXTEN}|1)
[pbx_config]
'_N' = 2. Goto(localcontact|${EXTEN}|1)
[pbx_config]
astreg1*CLI
-= 9 extensions (9 priorities) in 1 context. =-
If I take the _N and the i exten out, and don't put
[sipregistration] in the extension.conf file, then i can
reload extensions and the NoOp extensions remain in the dial
plan. Thanks for pointing that out, I can find another solution now.
It makes sense that if [sipregistration] exist in the
extension.conf file and a reload extensions is performed, all
the dynamic extensions in that context will be removed,
because they are not really there in the first place,
statically that is.
I was using the chanisavail cmd to do the local server
lookups, but was getting really sporatic results, works good
in the lab but not solid in an uncontrolled environment, live
traffic. I'm wondering if I can use a GotoIf statement to
check [sipregistration] for an active extension
Good stuff, thanks for the insight Brad.
No problem. I think I might have to go update the wiki (or add an
entry, I've never actually looked to see what exists) about this. It
comes up pretty often, and there definitely appears to be some
confusion.
Regards,
- Brad
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RE: [asterisk-users] Trouble with regexten
Well, I can't pretend to know how other people use it, but perhaps an example of how I use it would be helpful. Most of the sites that I maintain have a pair of boxes that are being loadbalanced (by UltraMonkey: www.ultramonkey.org), so I have no particular way of knowing who is registered to what box beforehand. Obviously, I need to know this. My solution is to use DUNDi and regexten. The DUNDi contexts are mapped into the context where the regextens take place (actually, it's the context where the 2 thru n priorities are, but the regcontext is included) and then I can just do a DUNDILOOKUP to found out the dialing information for any given device. It's simple, it works, and it's a good way to provide redundancy. I belive you may be expecting too much from regexten. It doesn't really do *that* much, but what it does do is useful. Regards, - Brad From: [EMAIL PROTECTED] on behalf of Andrew Joakimsen Sent: Thu 11/30/2006 10:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with regexten Because the REGEXTEN would be the phone number And the Device's userid would be the macaddress, settting regexten should create that association. There used to be an example on the voip-info wiki but its not there anymore. Would someone care to explain what regexten, in its current state, can do that the dialplan can't already do? The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. winmail.dat___ --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] Trouble with regexten
Creating a context in your extensions.conf with the same name as your regcontext will cause all kinds of oddness to happen, among them this. What you need to do is have a differently-named context in extensions.conf with your 2-n priorities and include sip_autoreg in that. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Brown Sent: Thursday, November 30, 2006 4:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Trouble with regexten Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ] '98766' =1. Dial(Sip/Tim) [pbx_config] 2. Hangup() [pbx_config] Include ='sip_autoreg'[pbx_config] Include ='widgets'[pbx_config] -= 1 extension (2 priorities) in 1 context. =- asterisk*CLI and here's sip_autoreg (the regexten context): asterisk*CLI dialplan show sip_autoreg [ Context 'sip_autoreg' created by 'pbx_config' ] '114' = 2. Dial(Sip/Tim) [pbx_config] 3. Hangup() [pbx_config] [ Context 'sip_autoreg' created by 'SIP' ] '112' = 1. Noop(Russell) [SIP] '113' = 1. Noop(Richard) [SIP] '114' = 1. Noop(Tim) [SIP] -= 4 extensions (5 priorities) in 2 contexts. =- asterisk*CLI Dialing 98766 from Sip/Russell rings Sip/Tim as expected. Dialing 114 gives Not Found :-( I'm very confused any ideas why this doesn't work? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Trouble with regexten
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, November 30, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trouble with regexten
When using autoreg, is there any way to extract the userid
somehow? IE:
SIP.com
regcontext=registrations
[123]
regexten=2125551212
extensions.conf
[phones]
include = registrations
exten = _212NXX,2,Dial(SIP/${VARIABLE}))
exten = _212NXX,3,VoiceMail(u${EXTEN})
Honestly I dont see the point of autoreg unless this can be
done...
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RE: [asterisk-users] Trouble with regexten
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, November 30, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trouble with regexten
When using autoreg, is there any way to extract the userid
somehow? IE:
SIP.com
regcontext=registrations
[123]
regexten=2125551212
extensions.conf
[phones]
include = registrations
exten = _212NXX,2,Dial(SIP/${VARIABLE}))
exten = _212NXX,3,VoiceMail(u${EXTEN})
Honestly I dont see the point of autoreg unless this can be
done...
The answer is no, but I'm not sure what you're expecting. This is no
different than if you weren't using regexten. You would still need a
way to map the DID to the proper device.
Regards,
- Brad
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RE: [asterisk-users] Re: Nortel Option 11C and SIP gateway integration
I'm most familiar with the CS1000 (formerly 81C) and Succession 4.5 with respect to integration, but perhaps I can help. Are you using external signalling server(s)? If so, have you installed and configured the NRS piece of that? Also, a SIP trace will probably be very enlightening. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Heison Chak Sent: Friday, November 03, 2006 9:26 AM To: Heison Chak Cc: asterisk-biz@lists.digium.com; asterisk-users@lists.digium.com; asterisk-bsd@lists.digium.com; [EMAIL PROTECTED] Subject: [asterisk-users] Re: Nortel Option 11C and SIP gateway integration Here's the Option 11 trace... I was not able to send as attachment. DCH 20 UIPE_IMSG CC_SETUP_IND REF 0004 CH 4 23 TOD 14:32:12 CK E4EA1E52 CALLED #:1500 NUM PLAN: UNKNOWN TON: UNKNOWN CALLING #:1567 NUM PLAN: UNKNOWN TON: UNKNOWN DCH 20 UIPE_OMSG CC_REJECT_REQ REF 8004 CH 4 23 TOD 14:32:12 CK E4EA1E6E CAUSE: #21 - CALL REJECTED DCH 20 UIPE_IMSG CC_SETUP_IND REF 0005 CH 4 23 TOD 14:32:16 CK E4EA39B2 CALLED #:1500 NUM PLAN: UNKNOWN TON: UNKNOWN CALLING #:1567 NUM PLAN: UNKNOWN TON: UNKNOWN DCH 20 UIPE_OMSG CC_REJECT_REQ REF 8005 CH 4 23 TOD 14:32:16 CK E4EA39CF CAUSE: #21 - CALL REJECTED DCH 20 UIPE_IMSG CC_SETUP_IND REF 0006 CH 4 23 TOD 14:32:20 CK E4EA540A CALLED #:1500 NUM PLAN: UNKNOWN TON: UNKNOWN CALLING #:1567 NUM PLAN: UNKNOWN TON: UNKNOWN DCH 20 UIPE_OMSG CC_REJECT_REQ REF 8006 CH 4 23 TOD 14:32:20 CK E4EA5427 CAUSE: #21 - CALL REJECTED DCH 20 UIPE_OMSG CC_SETUP_REQ REF 002C CH 4 23 TOD 14:32:36 CK E4EAD197 PROGRESS: ORIGINATING END IS NOT ISDN CALLING #:4169771414 NUM PLAN: E164 TON: NATL CALLED #:1695 NUM PLAN: E164 TON: NATL DCH 20 UIPE_IMSG CC_PROCEED_IND REF 002C CH 4 23 TOD 14:32:36 CK E4EAD254 DCH 20 UIPE_IMSG CC_SETUP_CONF REF 002C CH 4 23 TOD 14:32:36 CK E4EAD280 DCH 20 UIPE_OMSG CC_DISC_REQ REF 002C CH 4 23 TOD 14:32:38 CK E4EAEC5C CAUSE: #16 - NORMAL CALL CLEARING DCH 20 UIPE_IMSG CC_RELEASE_IND REF 002C CH 4 23 TOD 14:32:38 CK E4EAECF7 DCH 20 UIPE_OMSG CC_RELEASE_RESP REF 002C CH 4 23 TOD 14:32:38 CK E4EAECF F Heison Chak wrote: Hi, We have a Nortel Option 11C (with Succession 3.0), with 3 PRI cards connected to: 1. PSTN 2. ITG network to our other 2 offices on a 4-digit dialplan 3. SIP media gateway (for Asterisk) We normally dial access code 9 for outside PSTN calls, and when the SIP media gateway was introduced, a new access code 8 was created. Inbound calls from Nortel (originating from the PSTN, from any office handset) are being delivered to the PRI trunk on the SIP media gatway then onwards to Asterisk. However, any outgoing calls made from Asterisk, into Nortel via SIP gateway is being rejected. To narrow down the possibility, we have tried 2 different SIP gateways - AudioCodes Mediant 1000 and Cisco AS5300, and they both exhibit the same behavior (incoming works fine, ALL outgoing calls are being rejected). Attached is the capture of the console message on the Nortel side while an outbound call was made. Calls from x1567 (Cisco 7960 registered to Asterisk) to x1500 (digital extension on Option 11C) is being reject with CAUSE #21. The capture also shows a successful inbound call while 4169771414 (digital handset on Opt 11C) called x1695 (Meetme on Asterisk) via the same PRI card (Ch. 4 23) was completed with release cause #16. We suspect there is some authorization code or ACL that needs to be put in place, so that calls made to the Opt 11C can be routed. We have tired talking to 3 local Nortel vendors, AudioCodes and none has been able to help us rectify this issue. We are looking for someone who can help us identify what the problem is so that we can get this working. Thanks -Heison -- Heison ChakEmail: [EMAIL PROTECTED] 14 Bartlett Rd.Phone: +1 905 887 4694 x1508 Markham, ON L6C 2Y6Toll: +1 888 887 4694 x1508 Canada Cell: +1 416 417 8893 Fax: +1 905 887 4694 UK:+44 0207 099 5883 HK:+852 3596 4261 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
RE: [asterisk-users] regexten regcontext broken for SIP?
Can either or both of you post the relevant sections of your sip.conf and extensions.conf? - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew JoakimsenSent: Thursday, November 02, 2006 1:51 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] regexten regcontext broken for SIP? I am having the same issues. Did you ever file a bug report? On 10/6/06, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi ho,is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes(1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1being created upon SIP client registration, "show dialplan xxx" reveals no change.And yes, I have also read and checked bug 7144; if I go down that routeand no SIP client is registered I get a CLI warning that my standardcontext tries to include an empty context - go figure... http://bugs.digium.com/view.php?id=7144So, do I need to file a bug report, or is it working OK for others?Cheers, PhilippP.S.: Of course I am aware of this Wiki page: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] regexten regcontext broken for SIP?
It would be helpful if either or both of you posted the relevant sections of your sip.conf and extensions.conf. - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce ReevesSent: Thursday, November 02, 2006 4:10 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] regexten regcontext broken for SIP? I am using it on 4 boxes all pretty recent SVN versions of 1.2. I seem to recall that the after adding the setting and then reloading the context did not populate, it was only after I restarted the service and the phone registered. I was kind of rushing the process so that may be why I noted that. On 11/2/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: I am having the same issues. Did you ever file a bug report? On 10/6/06, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi ho,is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes(1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change.And yes, I have also read and checked bug 7144; if I go down that routeand no SIP client is registered I get a CLI warning that my standardcontext tries to include an empty context - go figure... http://bugs.digium.com/view.php?id=7144So, do I need to file a bug report, or is it working OK for others? Cheers, PhilippP.S.: Of course I am aware of this Wiki page: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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
Recall: [asterisk-users] regexten regcontext broken for SIP?
Title: Recall: [asterisk-users] regexten regcontext broken for SIP? Watkins, Bradley would like to recall the message, [asterisk-users] regexten regcontext broken for SIP?. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] voicemail usernames can't begin with j letter?
I playing a bit with this, it seems that if you use the new syntax it
works:
exten = _[a-z].,3,VoiceMail(${EXTEN}|u)
You can, of course, also use the b, j, s, and g flags.
Even using the VoiceMail(u${EXTEN}) still elides the 'j'.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric ManxPower Wieling
Sent: Friday, October 20, 2006 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] voicemail usernames can't begin
with j letter?
Ricardo Carvalho wrote:
I'm running Asterisk version 1.2.10. I also tried with
version 1.2.4
and got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN}) exten = _[a-z].,4,Hangup
Through my testing I found that the problem is that when someone
enters for example john's voicemail, Asterisk thinks that
j letter
is jump flag to n+1 priority. How can I disable, (if possible) this
erroneous interpretation that Asterisk does?
Have you tried exten = _[a-z].,3,VoiceMail(u${EXTEN})
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RE: [asterisk-users] Function ENUMLOOKUP
Does that entry exist also in e164.arpa (the
default)? Have you tried explicitly pointing it at e164.org
instead?
FWIW, I see nothing in particular wrong about your usage,
but make sure we're talking about the right trees here.
Regards,
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
GarstangSent: Monday, October 09, 2006 5:13 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject:
[asterisk-users] Function ENUMLOOKUP
Just
playing around with Enum. What's wrong with this in
Asterisk?
exten =
555,1,Set(foo=${ENUMLOOKUP(+16049586111)})
-- Executing
Set("SIP/3254101-081e8c58", "foo=") in new stack -- Executing
NoOp("SIP/3254101-081e8c58", "") in new stack -- Executing Hangup("SIP/3254101-081e8c58",
"") in new stack
...
because dig resolves it That's the number e164.org has as a callerid
readback on their website.
dig +short
1.1.1.6.8.5.9.4.0.6.1.e164.org any100 10 "u" "E2U+ADDRESS"
"!^.*$!ADDRESS:CN=Matthew Asham\;STREET=Eduard-Bodem-Gasse
9\;L=Burnaby\;ST=BC\;C=Canada!" .100 10 "u" "E2U+SIP"
"!^\\+16049586111$!sip:[EMAIL PROTECTED]" .
Doug.
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RE: [asterisk-users] Zaptel problems
You didn't say, but my guess is you are using either a 4-port or 2-port Digium card, right? What do the contents of /etc/modprobe.d/zaptel look like? You will probably find that there isn't an entry like: install wct4xxp /sbin/modprobe --ignore-install wct4xxp $CMDLINE_OPTS /sbin/ztcfg I put in a bug for this already, though in the report it's for FC5: http://bugs.digium.com/view.php?id=8071 Of course, tell me if this doesn't apply to your situation. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shea, Matt Sent: Wednesday, October 04, 2006 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Zaptel problems Hmmm, It appears ztcfg is not being run. Any ideas why? Matt 313-667-0970 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernardo Vieira Sent: Wednesday, October 04, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel problems -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Is ztcfg running at boot after the zaptel modules have been loaded? What's the output of ztcfg? Shea, Matt wrote: I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software runs ok with one exception. Zaptel appears to load OK on bootup, but when you check it on login, zttool still shows red/nop alarms on the T1 lines. I have to manually start it again for the alarms to disappear and the T1 lines to function properly. I've updated the drivers to 1.2.9.1 and double checked my configuration files to no effect. Any suggestions will be much appreciated. Matt -- -- ___ --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 - -- What most profoundly divides two men is a different sense and degree of cleanliness. What help is all honesty and mutual utility, what help is all the good will for each other: in the end the fact remains-they can't stand each other?s smell! - - Nietzsche -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFI+PS2QVs8jsa1mQRAnRKAKCYVjy0QKxm67RMyc3/kMw+i+sDkgCePu1U zeKUkrOK4rPfnl4+HvnpEK8= =pxJ+ -END PGP 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 ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 - Tekelec T6000 (Vocaldata, voiss)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Crocker Sent: Thursday, September 28, 2006 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk - Tekelec T6000 (Vocaldata, voiss) Thanks, The Tekelec T7000 is a traditional TDM class 4/5 switch with VoIP interface cards (PIC) formerly known as the Taqua OCX. The Teklec T6000 is a VoIP softswitch (feature server) formerly known as the VocalData VOISS. I have both and I'm trying to get outbound calls from a SIP phone registering with Asterisk through the T6000 to a T7000 and out to the PSTN. Calls are working, DTMF is not. The T7000 is acting as the voice gateway to my T6000 and requires RFC2833. So the Asterisk server has a sip.conf that sends outbound calls to the T6000. The T6000 is configured to send 800# outbound to the T7000 which has connectivity to the local Access Tandem and SS7 for IXC termination. The calls work fine, just can't navigate a voice mail tree. Tekelec doesn't officially support Asterisk, I have an open ticket with them and I'm working on packet captures. They may be able to identify what is wrong with the config but they won't be able to recommend fixes on the Asterisk side. Anyone else have a T6000 working with Asterisk? SIP signaling goes like this [SIP Phone] -- [Asterisk] -- [PIX FIrewall] -- [Tekelec SBC] -- [T6000] -- [T7000 PIC] Bearer traffic RTP goes like this [SIP Phone] -- [PIX Firewall] -- [Tekelec SBC] -- [T7000 PIC] From my understanding RFC2833 means the DTMF is encoded in the RTP stream so it is originating from the SIP phone, Maybe the SIP phone is broken.. hrmm.. -Matt Are you sure the RTP isn't going through the Asterisk box? The reason I ask is because this sounds suspiciously like the lack of variable-length DTMF in pre-1.4 Asterisk (did you say what version of Asterisk you are using and I missed it?). Of course, depending on the phone, perhaps it has a similar problem. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] HINT problems with SVN-trunk-r43322
The reason is that, at least in the SIP channel in trunk, the structure that keeps track of device state for hinting only gets allocated on peer objects and then only if call-limit is configured to some value. It's been a long time since I've done any development with 1.2 (all my 1.2 systems are waiting for 1.4 to come out so we can add a bunch of features), so I forget how that works there. Rumor has it these restrictions aren't necessary, but I forget. If by '6 months' you mean trunk from that long ago, it's entirely plausible that you got a snapshot during the evolution from where it was in 1.2 to where it is today. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, September 20, 2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322 Group Looks like the type=peer call-limit=2 Works. Now the question is why? The sample I sent is working on a system build 6 months ago. Will do some more checking and will report to the list on anything I find... Thanks Bradley for this bit of info you gave!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, September 20, 2006 1:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 On Wednesday 20 September 2006 12:31, Watkins, Bradley wrote: You will need to change the type=friend to type=peer and also define call-limit to some value (it can be large if you don't care about the actual limit). That should fix hints for you. But if you have it set to 1 then busy status won't work, isn't that the case? -A. ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] HINT problems with SVN-trunk-r43322
You will need to change the type=friend to type=peer and also define call-limit to some value (it can be large if you don't care about the actual limit). That should fix hints for you. Regards, - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.Sent: Wednesday, September 20, 2006 11:39 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] HINT problems with SVN-trunk-r43322 Im unable to get HINTS working with the new SVN-Trunk State never changed when ringing or on the phone. Below is my configs (Maybe I missed something) Thanks for any help you could give!! ##sip.conf## [general] callerid=unavailable context=default ; Default context for incoming calls 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) ;allow=all allow=ulaw allow=g729 ;allow=gsm ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY videosupport=yes allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec qualify=yes notifyringing=yes [101] type=friend ; "friend" means this device takes and makes calls username=101 ; Username on device callerid=Eric 102 secret=101 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=default ; Inbound calls from this host go here [EMAIL PROTECTED]; Activate the message waiting light if this canreinvite=no ; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=default notifyringing=yes ##extensions.conf## [general] static=yes writeprotect=no autofallthrough=yes priorityjumping=yes [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 [default] exten = 101,hint,SIP/101 exten = 102,hint,SIP/102 exten = 101,1,dial(sip/101,20,tw) exten = 101,n,voicemail(101) exten = 101,n,hanup() exten = 102,1,dial(sip/102,20,tw) exten = 102,n,voicemail(102) exten = 102,n,hanup() Commands from the CLI CLI sip show peers Name/username Host Dyn Nat ACL Port Status 102/102 206.173.108.30 D N 5060 OK (5 ms) 101/101 206.173.108.25 D N 5060 OK (5 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] CLI show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : SIP/102 State:Idle Watchers 1 [EMAIL PROTECTED] : SIP/101 State:Idle Watchers 1 - 2 hints registered CLI sip show subscriptions Peer User Call ID Extension Last state Type Mailbox 206.173.108.30 102 fb84429adb2 [EMAIL PROTECTED] Idle dialog-info+xml none 206.173.108.25 101 499798bcfa4 [EMAIL PROTECTED] Idle dialog-info+xml none 2 active SIP subscriptions The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 Soundpoint Key Remap
Did you ever try to get it working on any 1.6.x releases? I hacked at it a bit and it didn't seem to be working, though I could have been doing something wrong. I was, after all, reading the manual... ;) I'm glad to hear someone successfully doing it, as it's something I've wanted to play with for awhile now. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, September 12, 2006 4:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom Soundpoint Key Remap The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it yesterday on 2.0.1 -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 12, 2006 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Soundpoint Key Remap Hi Shawn - Unfortunately, on a Polycom, you can no longer remap a speed dial to a key. You can set extra line appearances to be speed dials (I can show you that, if you want), but none of the other keys. This feature used to be available, but was quietly removed as of 1.5.x. If you want to revert to 1.4.1 you can do it with the subpoint feature (I can show you that, too), but 1.4.1 has other serious limitations. - Noah On 9/12/06, Shawn Kelley [EMAIL PROTECTED] wrote: I'm told by Adam below that I can use a Speed Dial to accomplish this. However, I don't know how to map a speed dial to the key. I know the syntax for mapping a function to it ( IP_500 key.IP_500.31.function.prim=BuddyStatus/ ) However, I don't know how to do a speed dial. Any one out there know? Thanks! --Shawn Shawn Kelley wrote: Hi, Does anyone know how to do a re-map of a key on the Polycom to make it dial a number. I know how to remap a key to a certain function, but I don't know how to make it dial a number. I'm wanting to re-map the Service key to dial *8 for a group pickup. Any help is greatly appreciated. Thanks! --Shawn AFAIK, you will need to tell it save a speed dial for *8, and then map the key to dial the speed dial number that you saved it as. Hope this helps. Regards, Adam ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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
