[Asterisk-Users] Polycom 501 logo onscreen
Anyone know how (or if it's possible) to get a logo on the screen of a Polycom 501? I've been looking around for hints on how to do it but so far nothing would indicate Polycom supports doing it. -bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zhone channel Banks
I've got a Zhone 24 port FXS to configure. The configuration is beyond stupid. The people that designed this unit should be chased down and fired. I'm going around in circles frigging with all the options. Does anyone have a config file for this unit that I can use as a starting point? -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PRI problems.
It sounds like it might be dialplan instead of PRI related. -bill On 2-Jan-06, at 10:43 AM, Kristian Larsson wrote: On Mon, Jan 02, 2006 at 03:36:57PM +0200, [EMAIL PROTECTED] wrote: On Mon, 2 Jan 2006, Kristian Larsson wrote: I have an Avaya IP Office PBX connected to an Asterisk system via a Sangoma ISDN PRI card. Dialing from the as terisk system into the avaya works just fine but when trying to call from a phone connected to the avaya syste m something goes wrong. After punching the first four digits the Avaya calls out, shouldn't it wait for all di gits and then dial out? If I try to dial a three digit number it waits for a while then dials. Is this some feature to let the CO know of which area code the calls is going ahead of time? Is there some way to circumvent this using hacks on the asterisk side? Looks like you need to enable overlapdial=yes on the Asterisk side. It will then wait for additional digits sent from the Avaya after the initial ones sent with the SETUP. I did try enabling overlapdial=yes but I saw no real change. Is there any other variable to go with it that I might need to tune? I am quite new to the whole PRI thing. What does it do when setting up a call? First a SETUP and after that it dials? Regards, Kristian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk high load high availability servers
It's more like a research project going to proof of concept. Was very interesting tho. -bill On 11-Nov-05, at 12:23 PM, Matthew Simpson wrote: anyone using a high availability server set up for Asterisk ? I saw IBM had some kind of solution at VON but was too busy exhibiting to check it out. :( ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs problem
I've found that happens when one version of asterisk is 1.2 and the other end is running 1.0.9 and you are connecting over IAX2. If you bridge the two servers with SIP it will be fine. -bill On 9-Nov-05, at 11:52 AM, Olivier Taylor wrote: That's a call to pstn Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that there is no match and give me an error :( Any idea? Kind regards, Olivier 9 headers, 11 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 82.146.123.246:38098 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Don't know what to do if second ROSE component is of type 0x6
I also have the same problem at a customer. It's PRI setup as 5ESS. It doesn;t seem to be hurting anything but it is related to the caller ID. I tried google for docs etc but I came up with nothing I could understand or figure out. Over time I've used both Sangoma and Digium card and it appears with both. I've mostly been running CVS head at this location for several months. The problem still exists in CVS head from around 2 weeks ago. The situation seems related to 800 number inbound only. I have some PRI debug traces where I can call the PRI from my house using the toll free 800 number that is routed to the PRI. I get the second rose 6 error. When I dial the local pilot line that feeds into the PRI the call comes through with no error message. The only difference is the number I use to call the PRI. 800 toll free vs local number. It also seems to affect the Caller ID name. Caller ID number comes throug in both cases. Caller ID Name doesn;t seem to show up when I get a Rose 6 error. I was hoping to put the two traces up next to each other and see what's coming down the D channel differently in the two situations but so far I havn;t had the time... -bill Jeremy Gault wrote: FWIW, we are also seeing this message each time we receive a call. I also went the Google route and found only questions, not answers. We are running a PRI from US LEC (channels 1-10 are B-channels, with channel 24 being the D-channel, and we are only running voice on the PRI.) The PRI is connected directly to our Digium TE110P card, and obviously we are using the zaptel drivers. We did not see this message when running */zaptel/libpri 1.0.9. However, after upgrading (we went to 1.2.0-beta1 and now to CVS) we started seeing the message. (I don't remember exactly if we saw it in the beta, but we do in the CVS.) In our case, it does not seem to affect the stability of our * machine. (However, bear in mind that you may be using parts of * that we do not, and the problem could lie in those parts.) We're handling all PSTN calls via the PRI, except outbound to toll-free which are handed off to an IAX gateway on the Internet. Our employees' desks are connected via the LAN (using Polycom 500/501 SIP phones.) I have a remote extension at home (also SIP) using a Sipura SPA-2000. We did have some stability issues (Asterisk would segfault) when we first moved to CVS. Of course, safe_asterisk handled this and a couple of days later we updated again from CVS and it seemed to fix the stability issue we were having. If you are using CVS (but not the latest one) you may want to try upgrading. I wouldn't worry about that message, though. However, I would also be interested in knowing what it means/what causes it. :) Jeremy Tom Rymes wrote: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 We are running a PRI through a Sangoma card that is handling the D-channel natively at this point, but we go the error when zaptel was handling the D-channel, too. I have googled, but all of the messages are like this one with no answers that I can find. It's probably a non-issue, but we have been having issues with stability of our * install and I'd like to figure this out! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 line SIP ATAs with Asterisk using RealTime
Setup the two ports completely separately. Each should have it's own entry in realtime with a unique username. -bill On 10-Oct-05, at 1:15 PM, Dave Wise wrote: I am running CVS Head i686 running Linux on 2005-06-30 22:55:14. I have SIP Buddies installed using MySQL. If I try to set up a ATA that has 2 two phone lines (resulting in 2 lines on 1 IP address), my second line can never authenticate to dial out. I ran ethereal and found that Asterisk is "looking at the IP the request came from" and then, apparently looking up the IP address in the SIP table and responding to the first match of username to the IP address (this also happens if I plug in one phone to test it and use a designated IP address and then remove that phone and test with a different phone but with the same IP address, it uses the data from the lowest row number that the IP field matches). Is there any work around to this. I know that the SIP port is different for line 1 and line 2. Like I mentioned above, ethereal shows that Asterisk is changing the responses to a different user (or that is what I interpreted it to be doing). I also tried changing insecure to try to ignore the port number with no success. I tried the following values in insecure: port port, invite invite yes I looked on the WIKI and could not find a solution either. I would appreciate any help. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
On 7-Oct-05, at 9:45 PM, Paul wrote: The thing to remember is that the digium folks are not going to spend months slaving over a new hardware product and then put the device driver source under a closed license only. The gpl code can be used in an asterisk fork like openpbx or in something written from scratch like MyStinkingPBX as long as the license is honored. That helps digium hardware sales. Dual licensing is not such a bad thing. Suppose I want to build a proprietary black box product that uses the acme XYZ99 chipset. Do you think the author of a good GPL'ed XYZ99 device driver would refuse to consider a good legal dual-license opportunity? I doubt it. Also consider that there are situations where 100% open source is never allowed. Check out visa/mastercard processor certification for a good example. Digium dual licensing availability means I could actually stand a chance of using asterisk as the basis for systems used by military and law enforcement in applications that require extremely high security. One problem with the dual license is only Digium employee's can check code into the main code base. Lawyers could be all over product like ABE if they were not vigilant on people signing the license document. In the end this does hamper third party hardware support (ie device drivers) being integrated into the main Asterisk code tree. To do this would require the new hardware manufacturer not only to release their driver under GPL, but to also give the code away to Digium. Something many of them may be unwilling to do. The other option is for the new hardware company to purchase some other type of Asterisk license from Digium. The question is: Long term is Digium a hardware or a software company? Law enforcement and many high security systems use open source in many cases. Look at the security products built from OpenBSD for example. It's the security vs obscurity debate. -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
Have you ever read the GPL? -bill On 7-Oct-05, at 10:51 AM, Brian C. Fertig wrote: Can they do this? Is this legal? ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Meredith Sent: Friday, October 07, 2005 10:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: www.openpbx.org Further info. The domain is registered to Marc Olivier Chouinard. He has posted in the dev list. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Codec
on asterisk command line do a show translations -bill On 4-Oct-05, at 12:29 PM, Crystal Stream, Incorporated wrote: Hello, How do I make sure the G.729 codec is being utilized fully and not just as a passthru? I've registered it and followed the install instructions __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic feature support recently added to CVS HEAD
I've been trying to work with the dynamic feature support.. IE adding codes like *2 to features.conf that can trigger a dialplan application to run. I've been unable to get "goto" to work properly. "AGI" also seems to not function correctly if called as a feature. Anyone else playing around with this feature might have some insight? -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup half a call?
Scenario is as follows. Caller comes in over ZAP channel connects to handset on another ZAP channel. Call is bridged. I'd like the callee to be able to hangup on the caller and then be presented with a agi application. Basically the agent that answered the call has to enter a few responses to questions asterisk asks. On some ACD phone systems this is called a "wrap code". Lets you build basic call statistics. IE the agent enters a 1 on a sale and a 2 on no sale kinda thing. You run through your log file or sql db and produce a couple of basic counts. I'm using the new features.conf applicationmap to startup whatever I want on the dialplan. Instead of having the callee just hangup his phone, press *3 and launch on an extension in the dialplan. How do I hangup only the caller and let the callee continue? Another way I though to handle it is to introduce an H extension in the dialplan. h handles one side of the hangup, let H handle the other side Can anyone think of a way to handle this? I'd prefer not to introduce new features into asterisk at this time since they won;t be considered for Asterisk 1.2, and it looks like Asterisk 1.2 release release date has slipped to infinity. The patches will end up in bug database hell. -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma and Digium same machine?
Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec routing?
You might be able to do this in CVS head Asterisk with the SIP_HEADER variables and a agi script. Need to look in the source code. -bill On 25-Sep-05, at 3:48 AM, Anders Svensson wrote: Hi! I asked this question a couple of days ago but got no answer so I try again. Is it possible to route a call in * based on used codec, meaning that if a user use G723 that call is routed to siptrunk 1 and a user using G.729 is routed to siptrunk 2? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue_log on mysql
Checkout this. http://lists.digium.com/pipermail/asterisk-users/2005-July/116881.html I borrowed the structure from somewhere else that I found it.. Since then I've moved primarily to postgres through odbc for myself. -bill On 16-Sep-05, at 2:44 PM, lenz wrote: Thanks, is there a standard schema for queue_log or can I define it myself? Thanks l. In data Fri, 16 Sep 2005 18:48:13 +0200, William Lloyd <[EMAIL PROTECTED]> ha scritto: Best to log directly to MySQL. Add in ODBC code. You are not the first to ask for it. -bill On 16-Sep-05, at 11:06 AM, lenz wrote: Hello, is there a best practice to upload queue_log file into MySQL? or - better - to have Asterisk log the queue_log straight to MySQL? Is it worth doing? Thanks l. --Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:>> http:// lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue_log on mysql
Best to log directly to MySQL. Add in ODBC code. You are not the first to ask for it. -bill On 16-Sep-05, at 11:06 AM, lenz wrote: Hello, is there a best practice to upload queue_log file into MySQL? or - better - to have Asterisk log the queue_log straight to MySQL? Is it worth doing? Thanks l. -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for failover ideas
Can you post or email your patch can I take a look at it? It's specifically the issue of NATD that I'd like to be able to know what asterisk host a phone registers to. that way I can do an asterisk to asterisk transfer and the original asterisk host that the phone is registered to can pass the call back down the NATD connection. -bill I was going to add into Asterisk Realtime an extra field that is a server identifier. For example to allow multiple asterisk machines to use the same CDR DB. You would be able to select on the data from just one of the servers. We did this and it works fine as you might expect. Perhaps we should put it up on bugs.digium.com, but it wasn't a big change. Also for example in sip.conf for registration. Have the specific asterisk server that processes the sip registration put it's identifier (IP address for example) into a field in the DB also. That way other asterisk servers sharing the DB would be able to know what asterisk server a SIP phone is registered to. Does anybody else think this might be useful? To my mind it doesn't make much difference where the phone is registered to. Registration is just a way to find out the contact info (IP address mainly) for the SIP user. Once we have that on one machine, in principle it can be used on any machine. The only issue may be that NAT is open from the machine to which the phone originally registered, bt not from others. So in our application we didn't bother to track which machine took the original registration. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for failover ideas
I'm looking at doing something very similar. I was going to add into Asterisk Realtime an extra field that is a server identifier. For example to allow multiple asterisk machines to use the same CDR DB. You would be able to select on the data from just one of the servers. Also for example in sip.conf for registration. Have the specific asterisk server that processes the sip registration put it's identifier (IP address for example) into a field in the DB also. That way other asterisk servers sharing the DB would be able to know what asterisk server a SIP phone is registered to. Does anybody else think this might be useful? Where would be the best place to have a server identifier variable? asterisk.conf? within the sip.conf that applies just to the sip SQL tables? -bill On 23-Aug-05, at 1:28 PM, Anish Basu wrote: We are building asterisk clusters using mysql replication. All the configuration and cdr data is stored using the res_mysql module. Replication creates identical servers. Then, the phones register to each server using DNS SRV records. If any server goes down, all the phones registered to that server will register to any of the remaining servers. The only downside of this is that all active calls get dropped once the server goes down, but users can make and receive new calls instantaneously. Another idea is to use Asterisk Realtime so that the mysql replication will share registration information between servers, thus eliminating the need for the phones to re-register. Has anyone successfully implemented this or something similar? --Anish Message: 8 Date: Tue, 23 Aug 2005 08:33:03 -0700 (PDT) From: "Jeremy C. Reed" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] looking for failover ideas To: Asterisk-Users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed I found many mailing list threads and one wiki webpage with ideas and questions related to failover and high availability solutions. Is there any webpage or wiki page that summarizes all these ideas? What I have found: - case 1: two identical Asterisk boxes with one acting as hot- failure backup http://www.voip-info.org/tiki-index.php?page=Asterisk+failover+case1 - low tech DPDT relay http://www.voip-info.org/tiki-index.php?page=Asterisk+failover - VRRPD (KeepAlive daemon) - Linux-HA and DRBD and heartbeat monitoring - many mailing list postings with partial ideas Should I ask the asterisk docs list instead? Jeremy C. Reed ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] any ISDN/PRI signaling experts out there?
This sounds suspiciously similar to a problem I've had at a customer location. Telco is running 5ESS switch, Asterisk is using TE410 board. Asterisk CVS HEAD as of mid july. Most incoming calls do not have the Caller Name appearing. Caller ID number always comes in. On almost every incoming call Asterisk reports an error of "Don't know what to do if second ROSE component is of type 0x6" However, local incoming calls everything works out. Caller ID number comes in, caller name comes in and asterisk doesn;t report any errors. I did a PRI intense debug on the connection and a bit of research on the output, but I've no idea what's really up with it. On a related side note, there was/is no "pri no intense debug" tun off switch. Need to stop and restart asterisk if you do that. -bill On 21-Aug-05, at 3:54 PM, Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Belanger Sent: Friday, August 19, 2005 4:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] any ISDN/PRI signaling experts out there? See comments inline! Damon Estep wrote: I have officially engaged in a pissing contest with the local Telco over PRI calling name delivery. Welcome to my world, I deal with theses guys daily! Errgiant arn't they. We have a saying around work 'The telco is always wrong!'. The telco publishes their calling name delivery over PRI feature as being bellcore gr-1367-core compliant. The gr-1367-core spec states that the calling name is to be included as a facility IE in the setup message, or sent in a subsequent facility IE message with an indicator in the setup message that the CNAM will follow. Extensive testing and ISDN/PRI protocol analysis shows that the facility IE they are sending out with the CNAM in it comes only after we have sent back PROGRESS and ALERTING in response to the SETUP. If we block the PROGRESS and ALERTING and sit and WAIT for the FACILITY we never get it, the call will time out, so we know they are actually waiting for the call to progress before sending the facility IE CNAM. This sounds a little fishy, Orgination Number is usually transmitted in the SETUP message. Your are almost correct in your messaging: Network User(Switch) Setup CALL PROCEEDING ALERTING CALL CONNECT CALL CONNECT ACKNOWLEDGE There is about a 4sec timeout allow after SETUP is initially sent, if CALL PROCEEDING is not transmitted by that time, the Network side will terminiate the call. As far as I can tell the GR-1367-CORE spec does not define a maximum delay in sending the facility IE or whether it is acceptable to wait ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quad t1 / 1U rack server combos
I use this box with no problems at all. http://www.tyan.com/products/html/gx28b2881.html -bill On 16-Aug-05, at 12:32 PM, Damon Estep wrote: It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C’mon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E&M wink start patch
I'm trying to get a patch tested for inclusion in CVS. Anyone that is running E&M on a T1 and had to fool around with emdigitwait could you please try this. This patch removes the need for the emdigitwait parameter and speeds up dialing. This situation is mostly interfacing a legacy PBX/Key system with Asterisk. I've been running a couple of systems with this patch for about 6 months without problem. Please see the bug DB for the patch file. It's a small patch at under 20 lines. http://bugs.digium.com/view.php?id=3805 Any feedback is appreciated. -bill [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
On 23-Jul-05, at 11:22 AM, Kevin Walsh wrote: On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote: Adam Goryachev [EMAIL PROTECTED] wrote: On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: For this reason, I believe that if a fork were ever necessary, it would struggle to beat a distinct path away from the Asterisk Binary Edition Correct, until the point where there is MORE features being added to the forked version of asterisk than the digium version of asterisk. That can't happen, because the ABE could, and probably would, absorb all of the advances in the fork, while forging ahead with the original. Since the fork would be GPL only, if ABE 'absorbed' the new features, then it would 'become' GPL, and therefore would need to be released as GPL, and hence would no longer by ABE :) So, that can't happen. Any other ideas? You're forgetting about the "disclaimer" documents. Anyone who signed the perpetual agreement and made changes and/or enhancements to the Asterisk code (a fork would still be using Asterisk code) would firstly be obliged to inform "the owner", and would secondly have a prior agreement with "the owner" to allow them to use and close the code. That would neatly bypass the GPL and allow the new code to be folded into the Asterisk Binary Edition. It's unlikely that the current pool of asterisk developers will remain static however. People change jobs, new people find asterisk interesting, people that have not contributed before start to contribute. Assuming a fork were to happen one day. Lots of current developers would stay with the Digium tree because they know it, are digium partners, think it's a better idea, already signed the disclaimer and don;t have an issue with it etc. Many new developers submitting smaller patches would not bother to sign a legal disclaimer and just submit the patch to the full GPL tree. The splinter GPL tree would likely integrate the changes faster and obviously don;t care about a disclaimer. The practicalities of tracking the changes between two source trees would just get more and more time consuming for Digium. They will want to make 100% legal sure that every change they bring into their tree comes from somebody with a disclaimer. Rewriting the missing bits with other programmers would just help the tree's diverge faster. Meanwhile a full GPL tree can just plow ahead without concern. Many companies successfully manage the commercial GPL gap. MySQL for example. The difference in this case is selling a binary only version instead of making money off just hardware and support services/contracts. At the end of the day Digium own the Asterisk trademark and in the world these days, brand name recognition is often more important than the product behind it. -bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Log
I'm looking for a similar script. Care to share? -bill On 7-Jun-05, at 10:43 AM, Hugo Begglo wrote: Hello everyone, This is is my first email to this group. I'm am writing a small php program to pull some info out of our Asterisk's queue_log. I'm having trouble figuring out what some of the parameters mean. Here's an example: 1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||"Ray Balbin 25" < (716)250-3405> I found a doc that tells me about everything from "ENTERQUEUE" and on but nothing on the 4 fields before it. Can anyone she some light on this ? Hugo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT (kinda): Justification for adding Asteriskto the business plan
It's all about the testing before rollout. The problem with the run of the mill IT guy (especially ones involved in web sites) they tend to think that testing something means you try a few calls and if it works it's all fine. Testing isn;t beating into them in the same way it is in the telecom guys. There are often unforeseen implications of changing the smallest little knob on the system. Unless you are willing to really think about how to make the box fail and try the permutations and combinations the end user is always going to be unhappy. At the end of the day for some incremental income you don;t want to take an otherwise happy customer and turn him into a nightmare in the search to add a little VOIP. -bill On 15-Jul-05, at 8:15 AM, /dev/null wrote: The fact that Asterisk is "soft" and you're trying to sell to an IT Company.. Just to clarify, we make up the IT company and we'd be selling it to our customers that may or may not be IT based. The company does run VoIP but does not use Asterisk (using VoIP to add additional local lines in different area codes). What I'm trying to accomplish is a convo/thread of why IT consultancies should take on Asterisk in their normal support (or a value-add service). When I look at VoIP, I see nothing different between setting up and configuring perl scripts for AGI/* Manager interfaces (system admin scripting) to plugging the BRI into a Digium card from the old T1 router. Granted there are some differences in command configurations and might have to label a wire or two "Voice" instead of "Data" but nothing earth shattering. As for sales, it would be just adding new listings under "Telephones" in the yellow pages and adding a few words to brochures/web pages. -Don ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue_log stats
I'm in search of useful ACD type statistics from the queues. Ie talk
time, ratio's, dropped calls etc.
The flat file queue_log is nice, but more useful would be the data in
Postgres or Mysql. Unfortunately the queue module does not yet
support ODBC DB logging (yet). In the meantime this quick and dirty
hack gets the job done.
Replace the flat file with a unix named pipe. Works with all
versions of asterisk. Removes the need for daily cron job to parse
and move the log file to SQL for later processing.
This script could be easily adapted for other flat file type logging.
If anybody has some PHP, perl or whatever code that does asterisk
queue stats or has a question abut this script please contact me.
-bill
[EMAIL PROTECTED]
#!/usr/bin/perl -w
#
# [EMAIL PROTECTED]
# The asterisk version indpendant way to get queue stats into Mysql,
Postgres
# or whatever is supported by Perl DBI
# It's all about named pipes
# to setup this software
# stop asterisk
# rm /var/log/asterisk/queue_log
# mkfifo /var/log/asterisk/queue_log
# make sure permissions are setup
# chmod 777 /var/log/asterisk/queue_log
# run this program as root or under another user as you see fit.
# should start BEFORE asterisk. Add to /etc/rc.d/rc.local or whatever
# restart asterisk
# requires a DB table like the following..
# CREATE TABLE csr_queue (
# qname varchar(30) default NULL,
# agent varchar(30) default NULL,
# action text,
# info1 text,
# info2 text,
# info3 text,
# timestamp int(11) NOT NULL default '0',
# id tinytext NOT NULL
#) TYPE=MyISAM;
use DBI;
use IO::File;
my $opt_debug = 0;
# if you want postgres change this to "Pg"
my $db_type = "mysql";
my $db_host = "127.0.0.1";
my $db_user_name = 'username';
my $db_password = 'password';
my $db_database = 'asteriskstat';
my $dbh = DBI->connect("DBI:$db_type:dbname=$db_database;host=
$db_host;", $db_user_name, $db_password);
open(FIFO, "< /var/log/asterisk/queue_log")or die "Can't open
queue_log : $!\n";
while (1) {
$message = ;
next unless defined $message; # interrupted or nothing logged
chomp $message;
# remove chars that will cause DB problems
$message =~ s/\"\'//g;
@data = split(/\|/,$message);
# these messages are almost useless for my purposes
next if ($data[4] eq "QUEUESTART" );
next if ($data[4] eq "CONFIGRELOAD" );
if (!defined($data[5])) {
$data[5] = '';
}
if (!defined($data[6])) {
$data[6] = '';
}
if (!defined($data[7])) {
$data[7] = '';
}
my $sql = "INSERT INTO csr_queue (timestamp, id, qname, agent,
action, info1, info2, info3) VALUES ('$data[0]', '$data[1]', '$data
[2]', '$data[3]', '$data[4]', '$data[5]', '$data[6]', '$data[7]')";
print "$sql \n\n" if ($opt_debug);
$dbh->do($sql);
# if you want an actual logfile you might want to uncomment this
#if ( open(LOG, ">> /var/log/asterisk/queue_log_real") ) {
#print LOG "$message\n";
#close(LOG);
#} else {
#warn "Couldn't log to /var/log/asterisk_queue_log: $!\n";
#}
#
}
$dbh->disconnect();
exit 0;
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Re: [Asterisk-Users] Nortel Option 11
I'm doing a similar thing with a Nortel Norstar system. The KSU has a T1 card configured as an E&M trunk and a PRI card. The E&M interface and zaptel had some issues with dialing digits manually and having lost digits. 1.0.7 and CVS HEAD have a feature called emdigitwait to help solve the problem. Unfortunately it doesn;t totally fix the issue. I submitted another patch the other day that fixes the dialing on E&M without requiring trial and error setting of emwaitdigit but it's not been adopted yet. Look at the wiki it has good information to help set this up. FYI, I've found CVS HEAD to have better echo canceling than 1.0.7. I also had some issues with the T1 going into Yellow Alarm once or twice a day in stable, it's disappeared in HEAD. -bill On 24-Mar-05, at 2:07 PM, Jeff Pratt wrote: Friend, George E. wrote: Question...I'm fairly new to Asterisk, but one location I'm looking at deploying Asterisk has an Option 11 in place already (it's actually in someone's HOME - long story). Does anyone know if it's feasible to interconnect the two and use Asterisk to interface with the other offices and lines, and merely use the Option 11 as a gateway to use the existing digital handsets (Nortel)? George I am currently using a TDM04B connnected to 4 2500 lines, and a T100P connected to a NT5D14 line side T1 card(program them like 2500 sets). I will shortly be implementing a new T1 trunk to split our office side voice off onto asterisk. The TDM04B and T100P service multiple ACD queues, some of which handle Voicemail for external employees, and some of which provide IVR services for human operated ACD queues. You can also use a 4 port T1 card as a go between for your CO T1s and your switch. Simply set up one port as CPE (this connects to the telco T1s), and one port as NET (this connects to your switch). Then set up asterisk to send any calls that you don't want asterisk to handle down the appropriate pipe (inbounds would go up the NET T1, and outbounds up the CPE T1). If you look around voip-info.org you should find where someone documented doing that with a Norstar. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P in a soekris 4801
Maybe try another type of card in the soekris to make sure the PCI slot is working properly. Maybe an ethernet card for example. I have 1 4801 and a couple of 4501 doing VPN stuff. but I've not tried asterisk on them. Soekris needs remove a couple ethernet jacks and make a box with 2 FXO and 3 FXS onboard. I wouldn;t think it would more than $50-75 to the cost of a board. Can be the Soekris PBX... -bill Matt Ryanczak wrote: Hi All, I have a Soekris 4801 running Asterisk and I tried to add a X100P FXO card only to find that the Soekris will not boot with the card installed. The LEDs flash for a second and then nothing, it does not boot. I've tried 2 different X100p cards, both of which work in a regular desktop system, and still have no luck. I tried a different Soekris 4801 with no luck. I've also tried a larger power supply with the Soekris with no luck. I tried to post to the Soekris list but it seems to be having issues right now. My question is: Has anyone seen a Soerkis 4801 run with a X100P card? I have read testimonials online that claim it works but I'm not seeing good results on my end. Can anyone help? Thanks! Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 SIP problem when calling from outside of LAN
There are some SIP errors that appeared in CVS in the last couple of days. I checked out some CVS source from last week and everything works properly. Maybe that's part of the problem. -bill On Tuesday, July 29, 2003, at 04:59 AM, Louis-David Mitterrand wrote: Hi, I am testing a 7960 in this context: [SIP] --- > VPN ---> [*] ---> [ANY] (ANY == any type of phone: isdn, SIP, IAX, etc.) the call goes through and is dropped after 5 seconds with this message in the log: "File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call for seqno 101" when calling from the LAN with the exact same phone: [SIP] ---> LAN ---> [*] ---> [ANY] it works fine, what could be wrong? I am using the asterisk packages from debian. Thanks in advance for any help, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SetLanguage application doesn;t seem to work in latest Asterisk
I've also played around with the language support but I never got it to work either. -bill - Original Message - From: "Panagidou Anna" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 25, 2003 3:15 AM Subject: [Asterisk-Users] SetLanguage application doesn;t seem to work in latest Asterisk Hello everybody, I have installed Asterisk from CVS (18/07/2003) and although everything works fine, SetLanguage application doesn't seem to work. As it used to work with previous version I wonder if I am missing something here. The relevant line in extensions.conf is: exten => 3,1,SetLanguage(gr) In the directory where Asterisk sounf files reside, I have installed files of the type file-gr.gsm. However, although the SetLanguage(gr) application is executed, only the plain file.gsm files are played. Note that I haven't setup gr as a language in any other configuration file. Any clues are welcome. Thanks Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 2 stage firmware 2 stage upgrade
I have a 7960 phone I'm trying to upgrade to SIP mode. Unfortunately there is an error in the firmware and I need to upgrade to a newer skinny first. I'm looking for somthing like version P00303020209.bin I have the latest SIP, but the phone times out trying to upgrade. According to Cisco this is a known fault with the version of firmware that my phone has on it. -bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
