Re: [asterisk-users] A few questions about bundled pjproject
Dmitriy Serov wrote: At the moment I plan to migrate from asterisk 13.7 to 13.8. Because of relatively frequent updates I am building asterisk from a directory that is updated via git switch to the desired branch. 1. Will be updated pjproject patches with "git pull"? Yes. 2. Will update himself pjproject? When a new release of pjproject is done it will be updated in Asterisk to pull it down. This change will then subsequently be in the next release, along with any changes required to compile against it. 3. And what should be done to be sure that they are the most "fresh"? If you use the 13 branch then you will always have the latest supported by Asterisk. If you use the latest 13 release then you will have the latest when that release was created. 4. What is the status of patches to pjproject? It is expected that they should eventually become part of pjproject (to be accepted by the maintainers)? Yes. Any patches are submitted upstream and if the next pjproject release has them then they are removed from Asterisk. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] A few questions regarding Asterisk 1.8.0
Good luck as with any new version there may be some bugs so if you bump up against ones report them so they can be fixed. Also don't just drop it into production with out testing it on a box for a bit. 1.8 has a lot of changes. Most appear to be for the better. The only important difference I could find while testing was that with 1.6.x you could use nat=route. This doesn't work anymore with 1.8.0 (and I didn't find it in the ChangeLog 1.8.0). Changing nat=route to nat=yes seems to work. (I mention it here so others can find it in the future). Regards, Mark -- _ -- 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] A few questions regarding Asterisk 1.8.0
From: Mark Scholten m...@streamservice.nl Hello, I have a few questions regarding Asterisk 1.8.0. If you can answer a question, please do so. Is Asterisk 1.8.0 stable enough for production environments? It appars to be so far we are testing and hoping to go production before the end of the year. Is it possible (and if yes what is the best option) to use CDR MySQL with Asterisk 1.8.0? With 1.6.x we use the add-on package for that, however we could do something with scripts to do it (but I don't like the idea). You can use the same MySQL method you are use to but if you want to use the new more extensive CEL method you will likely need to use ODBC to write to MySQL for now. You will also need to parse the new CEL format for the info you need. It is looking realy cool but it is taking a bit of work to intagrate it into our system. We will go live using the old CDR to MySQL for now. Please not that the addons are part of the main package now use menuselect to choose which ones you want to build. If it is stable and there is a good option for CDR with MySQL we will startusing it very soon. Good luck as with any new version there may be some bugs so if you bump up against ones report them so they can be fixed. Also don't just drop it into production with out testing it on a box for a bit. 1.8 has a lot of changes. Most appear to be for the better. Regards, Mark Regards Bryant -- _ -- 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 -- _ -- 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] A few questions on a conversion to *
Warren wrote: Am I best off using Hylafax? Yes. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A few questions on a conversion to *
Hi Warren - Questions: (1) Good 2xT1 card with hard echo cancellation? I'm not sure if it has onboard echo can, but Sangoma has a two port model. I've never used them (I have a TE410P), but I've always read very positive things about them, especially on the quality of the echo can they use. (2) Good 4-port analog card? Digium TDMs have always worked very well for me. (3) How can I best handle fax? Should it come through one of the POTS lines or should it come through the T1? If you have an analog fax machine and you want to use a DID number off the T1, I would recommend getting a digium FXS card to connect to the fax machine. I've used this setup for almost 2 years now, and it's very effective. If you want to use a POTS line for fax, just skip asterisk altogether. Run the POTS line directly to the fax machine. Am I best off using Hylafax? If you want individual DID fax numbers for employees, or you want to skip a paper fax machine, use Hylafax (or spandsp). (4) Can I use an existing data T-1 or does it have to be somehow reprovisioned? I have a backup T for my web servers that is hardly used and it would be a great testing utility. You'll have to talk to the provider about that. They can use the local loop, but beyond that I'm sure it will take some re-provisioning on their part. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a few questions
On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote: On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote: Overhead paging is totally possible, there are several articles available on how to do it. But you cannot have multiple zones today unless you use a sip device that has autoanswer. Why can mutilple zones not be done?, why do I need a sip device at all for the paging? any of the follwing (and I'm sure more) will do, even for multiple zones: * PC Sound Card * Digum hardware * any type of ata type gateway (SIP/h323 or whatever else that will interface with an analog port), even one without auto answer I have yet to see an example of overhead paging with multiple zones using a soundcard, digium hardware, or an ata. -Kerry Because you have never seen it, and you don't have the skill to figure it out, therefore it never happened. Nice job. Are you a politician? If you wish to pay my fee, I can give you a tour to a few buildings where I have successfully done 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] a few questions
Overhead paging is totally possible, there are several articles available on how to do it. But you cannot have multiple zones today unless you use a sip device that has autoanswer. Easiet way to remove that message is to replace the file with one that only has a split second of silence. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy Sent: Friday, December 09, 2005 8:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] a few questions we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features that basically says it is not possible with asterisk. let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] a few questions
That article is about shared call appearance. I have this working using Grandstream GXP-2000's. It's a great new feature. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy Sent: Friday, December 09, 2005 8:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] a few questions we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features that basically says it is not possible with asterisk. let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] a few questions
On 12/9/05, Stas Khromoy [EMAIL PROTECTED] wrote: we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging Overhead paging can be done with asteirsk in anyway you want, you can even do mutilple zones, all zones, or whatever you want. now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features I cound't find *anything* on that page that has to do with paging. that basically says it is not possible with asterisk. Exactly where on that page??? let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? Yeah, just rerecord that massage. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] a few questions
On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote: Overhead paging is totally possible, there are several articles available on how to do it. But you cannot have multiple zones today unless you use a sip device that has autoanswer. Why can mutilple zones not be done?, why do I need a sip device at all for the paging? any of the follwing (and I'm sure more) will do, even for multiple zones: * PC Sound Card * Digum hardware * any type of ata type gateway (SIP/h323 or whatever else that will interface with an analog port), even one without auto answer Easiet way to remove that message is to replace the file with one that only has a split second of silence. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy Sent: Friday, December 09, 2005 8:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] a few questions we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features that basically says it is not possible with asterisk. let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] a few questions
hehe I just installed * with a T1 span to and Adit600 with 2fxs and 1fxo The 8 fxo ports were for zone pageing works great should work with any fxo device and an existing page system On Dec 9, 2005, at 11:34 AM, C F wrote: Overhead paging is totally possible, there are several articles available on how to do it. But you cannot have multiple zones today unless you use a sip device that has autoanswer. Why can mutilple zones not be done?, why do I need a sip device at all for the paging? any of the follwing (and I'm sure more) will do, even for multiple zones: * PC Sound Card * Digum hardware * any type of ata type gateway (SIP/h323 or whatever else that will interface with an analog port), even one without auto answer ___ --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] a few questions
On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote: Overhead paging is totally possible, there are several articles available on how to do it. But you cannot have multiple zones today unless you use a sip device that has autoanswer. Why can mutilple zones not be done?, why do I need a sip device at all for the paging? any of the follwing (and I'm sure more) will do, even for multiple zones: * PC Sound Card * Digum hardware * any type of ata type gateway (SIP/h323 or whatever else that will interface with an analog port), even one without auto answer I have yet to see an example of overhead paging with multiple zones using a soundcard, digium hardware, or an ata. -Kerry ___ --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] a few questions
buy a grandstream bt101, cut the handpiece cables and connect this to your speakers, program auto answer, you can have as many zoes as you want. From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Fri 12/9/2005 1:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] a few questions On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote: Overhead paging is totally possible, there are several articles available on how to do it. But you cannot have multiple zones today unless you use a sip device that has autoanswer. Why can mutilple zones not be done?, why do I need a sip device at all for the paging? any of the follwing (and I'm sure more) will do, even for multiple zones: * PC Sound Card * Digum hardware * any type of ata type gateway (SIP/h323 or whatever else that will interface with an analog port), even one without auto answer I have yet to see an example of overhead paging with multiple zones using a soundcard, digium hardware, or an ata. -Kerry ___ --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 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] a few questions
my apologies if anything but as i said i am not that knowledgeable and most probably misunderstood the post. as it looks from your reply i have if you don't mind letting me know what i got wrong i would greatly appreciate it. On 12/9/05, Stas Khromoy [EMAIL PROTECTED] wrote: we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging Overhead paging can be done with asteirsk in anyway you want, you can even do mutilple zones, all zones, or whatever you want. now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features I cound't find *anything* on that page that has to do with paging. that basically says it is not possible with asterisk. Exactly where on that page??? let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? Yeah, just rerecord that massage. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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: Re: [Asterisk-Users] a few questions
Hey - best trick is not to take any of it personally. We all fell off our bikes while learning to ride! (old quote, but still valid) PaulH Stas Khromoy [EMAIL PROTECTED] wrote: my apologies if anything but as i said i am not that knowledgeable and most probably misunderstood the post. as it looks from your reply i have if you don't mind letting me know what i got wrong i would greatly appreciate it. On 12/9/05, Stas Khromoy [EMAIL PROTECTED] wrote: we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging Overhead paging can be done with asteirsk in anyway you want, you can even do mutilple zones, all zones, or whatever you want. now i came accross this post http://forums.digium.com/viewtopic.php?t=2844highlight=features I cound't find *anything* on that page that has to do with paging. that basically says it is not possible with asterisk. Exactly where on that page??? let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? Yeah, just rerecord that massage. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A few questions before final proposal...
Adam, Thanks for your help. Does anyone know or is anyone an * guru in the New Hampshire/Vermont area? how about this example. User1 sits at his desk, a call comes in.(doesn’t matter how the call gets to his phone, DID or exten) he needs to go into the warehouse to look at something. He places the call on hold, notes the line and goes to the warehouse. Once there, he picks up another handset, presses the button for the line he would like to pickup. How is this done with FOP? Everyone has access to FOP, not just the system operator? Would the user be better off transferring the call to that phone in the warehouse? How have others implemented this feature? ~kurth ___ --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] A few questions before final proposal...
Hi Kurth, I'm in NJ. I'd be happy to help you out either on the phone or in person. Gimme a call 973 828 1625 Mark Kurth Bemis wrote: Adam, Thanks for your help. Does anyone know or is anyone an * guru in the New Hampshire/Vermont area? how about this example. User1 sits at his desk, a call comes in.(doesn’t matter how the call gets to his phone, DID or exten) he needs to go into the warehouse to look at something. He places the call on hold, notes the line and goes to the warehouse. Once there, he picks up another handset, presses the button for the line he would like to pickup. How is this done with FOP? Everyone has access to FOP, not just the system operator? Would the user be better off transferring the call to that phone in the warehouse? How have others implemented this feature? ~kurth ___ --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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] A few questions before final proposal...
On Mon, 2005-09-05 at 01:31 -0400, Kurth Bemis wrote: I am attempting to assemble a proposal for a client of mine that is looking to replace their phone system. I think it's a good first installation with 4 POTS incoming and 15 extensions, with an overhead paging system. I also think that it would make a good case for OSS applications in general. Being new to * I have a few more questions, but won’t flood the list with them. I’m not new to Linux, but new to *. I am a bit hesitant to I would suggest that you find a local asterisk 'consultant' who is willing to help you get your feet wet. Maybe you can budget an extra couple of hundred dollars, but it will be worth it having someone to turn to for assistance. As far as handsets, I'd suggest the Polycom IP600 (since it has more line appearances that the IP500, and it just looks a little nicer). Otherwise, I've never used them, and I really like the polycom's, but I keep hearing that the snom phones handle shared call appearances really well, and they have more of them etc The only other issue I'd be wary of is all the echo issues associated with analog lines. Oh, and the overhead paging could get a bit tricky depending on the specific requirements (I've never done any paging at all, so I'm not so sure about that). So, from a feature point of view, asterisk can do most anything demanded of it, and usually so much more, but things like shared line appearances are easier using FOP than a led on a phone Regards, Adam ___ --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
music on hold [was: Re: [Asterisk-Users] A Few Questions]
On Sun, Jun 05, 2005 at 11:06:48PM -0400, C. Hatton Humphrey wrote: I have Asterisk running on a FreeBSD machine that is also my router/firewall and MySQL server. Asterisk is a CPU-intesive program. It will probably work fine with a router/firewall, but with another potentially-CPU-intensive program like mysql, you may have problems. Actually, the previous paragraph was over-simplistic. What asterisk really needs is fast access to the CPU: low latency is important, otherwise calls may sound jaggy. So make sure MySQL won't compete with Asterisk on the CPU. It is running fine and I've gotten it working with FWD and will be testing a direct IAX server in the next few days. I'm migrating from a Packet8 Virtual Office setup and have managed to get their DTA-310 working on my installation. Here are my questions.- Next time please give a proper subject to your message, please. 1. Does anyone have suggestions for license-friendly MOH sources? Same for reworks of the voicemail and autoattendant prompting? What exactly is your problem with the existing MOH of asterisk? It is freely distributable and can be freelly used *as MOH only*. If you want to use it for anything else you indeed have a problem, which is why Debian have removed it from the package (as it is indeed non-free). But if you just want it as music-on-hold, there shouldn't be a problem with it. For that reason we have silently re-added those files to our (Xorcm Rapid) repository. To prevent any potential license confusion the MOH files are packed in a separate package with a clear license. It also make sense from a size point of view: most of the size of the asterisk source package is the 3 MOH files that are practically guaranteed to never change. No point in including them in the distribution. Another source of MOH files is the classical music collection from Signate: http://signate.com/moh.php I'll probably package some of those in a MOH deb pretty soon. However: why do we need a high quality 44khz stereo MP3 music only to be decoded at runtime and then converted to 8khz mono? Wouldn't it cost more to save it in advance as wav with phone quality? No real quality lost, not too much disk space lost, and a lot of CPU time saved, isn't it? Not to mention avoiding a format that is widely recognized as patent-encombered. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] A Few Questions
I've got my own Asterisk tech. He's been with the company for about 2 years, finally convinced me to switch to Asterisk.Matt Riddell [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote:[SNIP] line, you will need an FXO card. We are a _Digium reseller_ so I can get [SNIP] other questions, I just started using Asterisk about _2 weeks ago_. I know [SNIP]How do you end up being a Digium reseller after using Asterisk for two weeks? Do you plan to provide your customers with support?-- Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] A Few Questions
On 6/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You should checkout [EMAIL PROTECTED], Thanks for the link - unfortunately [EMAIL PROTECTED] won't quite do the trick for me here as it is a complete OS replacement from what I can tell... I can't do that. I have too much time and money invested in the box that I'm running Asterisk on to wipe it and reload. Besides that, I already have Asterisk installed and running; maybe my next step should be to get AMP working on it (which would entail getting a webserver and whatever other requirements AMP has). ___ 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] A Few Questions
Dear Sir, Lol, with respectthat is the dumbest idea I have heard today. Your decision to not to invest time is a fallacy. [EMAIL PROTECTED] will have you up and running in 30 minutes or your money (it's free) back. Kind Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey Sent: Monday, 6 June 2005 10:09 AM Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A Few Questions On 6/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You should checkout [EMAIL PROTECTED], Thanks for the link - unfortunately [EMAIL PROTECTED] won't quite do the trick for me here as it is a complete OS replacement from what I can tell... I can't do that. I have too much time and money invested in the box that I'm running Asterisk on to wipe it and reload. Besides that, I already have Asterisk installed and running; maybe my next step should be to get AMP working on it (which would entail getting a webserver and whatever other requirements AMP has). ___ 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] A Few Questions
Perhaps he has a server that does other things besides asterisk and can't reformat it? Or perhaps he has a server in a remote location and buiness constraints make it difficult to take the time to get to it and spend a whole day doing a reinstall? Mark Dean Collins wrote: Dear Sir, Lol, with respectthat is the dumbest idea I have heard today. Your decision to not to invest time is a fallacy. [EMAIL PROTECTED] will have you up and running in 30 minutes or your money (it's free) back. Kind Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey Sent: Monday, 6 June 2005 10:09 AM Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A Few Questions On 6/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You should checkout [EMAIL PROTECTED], Thanks for the link - unfortunately [EMAIL PROTECTED] won't quite do the trick for me here as it is a complete OS replacement from what I can tell... I can't do that. I have too much time and money invested in the box that I'm running Asterisk on to wipe it and reload. Besides that, I already have Asterisk installed and running; maybe my next step should be to get AMP working on it (which would entail getting a webserver and whatever other requirements AMP has). ___ 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 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] A Few Questions
Then download the tar ball! It will install on any redhat (and maybe other) based systems as it compiles from source. On 6/6/05, Mark Benson [EMAIL PROTECTED] wrote: Perhaps he has a server that does other things besides asterisk and can't reformat it? Or perhaps he has a server in a remote location and buiness constraints make it difficult to take the time to get to it and spend a whole day doing a reinstall? Mark Dean Collins wrote: Dear Sir, Lol, with respectthat is the dumbest idea I have heard today. Your decision to not to invest time is a fallacy. [EMAIL PROTECTED] will have you up and running in 30 minutes or your money (it's free) back. Kind Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey Sent: Monday, 6 June 2005 10:09 AM Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A Few Questions On 6/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You should checkout [EMAIL PROTECTED], Thanks for the link - unfortunately [EMAIL PROTECTED] won't quite do the trick for me here as it is a complete OS replacement from what I can tell... I can't do that. I have too much time and money invested in the box that I'm running Asterisk on to wipe it and reload. Besides that, I already have Asterisk installed and running; maybe my next step should be to get AMP working on it (which would entail getting a webserver and whatever other requirements AMP has). ___ 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 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] A Few Questions
On Mon, Jun 06, 2005 at 03:38:52PM -0400, Matt wrote: Then download the tar ball! It will install on any redhat (and maybe other) based systems as it compiles from source. Great. Get the latest RedHat for FreeBSD from http://redhat.com/bsd/ . Naturally the linux zaptel code will compile cleanly there. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] A Few Questions
You should checkout [EMAIL PROTECTED], that's what we have installed here. It gives you web access to voicemail as well as a web configuration tool (Asterisk Management Portal). For the POTS line,you will need an FXO card. We are a Digium reseller so I can get you what you need as far as hardware. TheDigium TDM01B is a 1 FXO port card that runs around $130. It's about $70 for an additional port FXS or FXO. You can also try the X100P card, look on ebay. I just saw one for about $20.As I am not in anyway an advanced user, I'm not sure on your other questions, I just started using Asterisk about 2 weeks ago. I know there are others who will be able to answer your other questions. If you need anything else let me know."C. Hatton Humphrey" [EMAIL PROTECTED] wrote: I've been watching the mailing list for a few days, have done somearchive searching and still have a handfull of questions (I've lookedfor most of these on voip-info.org and a slew of other "asteriskrelated" sites). I'm going to throw the ones that are foremost inthis email and will add more as the need arises. Before I do so, letme explain my setup.I have Asterisk running on a FreeBSD machine that is also myrouter/firewall and MySQL server. It is running fine and I've gottenit working with FWD and will be testing a direct IAX server in thenext few days. I'm migrating from a Packet8 Virtual Office setup andhave managed to get their "DTA-310" working on my installation.Here are my questions.-1. Does anyone have suggestions for license-friendly MOH sources? Same for reworks of the voicemail and autoattendant prompting?< BR>2. Help! I got the MWI light on the phone (an Astra powered by theDTA-310) but now it won't go off.3. Is there any way to have asterisk take a phone back to a plaindialtone instead of a fast busy when a call ends?4. Even though I've got the basics working I keep wondering what elseis available. For example I see on the * website that things liketransferring and web access to voicemail are available but I don'teven know where to begin looking for that stuff. Where are the guidesthat I'm missing for all of the different configuration issues?5. All my other boxes are Windows machines - can someone recommend aconfig tool that I can run on Windows to help me get everythingstraightened out?6. What hardware is really needed to bring in a copper pair? I have asingle CO line that we're using for faxes and I'd like to be able toinclude it in our outgoing call system for 911 capabilities. At thesame time I don't want to throw dow n a bill for a card.Thanks!Hatton___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] A Few Questions
[EMAIL PROTECTED] wrote: [SNIP] line, you will need an FXO card. We are a _Digium reseller_ so I can get [SNIP] other questions, I just started using Asterisk about _2 weeks ago_. I know [SNIP] How do you end up being a Digium reseller after using Asterisk for two weeks? Do you plan to provide your customers with support? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] A few questions - isdn call routing
On Wed, 4 Aug 2004 13:34:02 +0200 (CEST) Peter Svensson [EMAIL PROTECTED] wrote: On Tue, 3 Aug 2004 [EMAIL PROTECTED] wrote: There is a device called a parlay made by a crowd called voxtream which will route the ISDN calls based on the DID and/or the callerid, before the call is answered. It would be nice if this feature could be done in Asterisk as well, but at this point in time, it first answers the call. Are you sure about this? When I looked at the traces on our setup it seems that CONNECT was only sent on the incoming leg after it was received from the outgoing leg. As a graph: pstn -pri- asterisk -pri- other_device pstnasteriskother device -SETUP- dial(...) -PROCEEDING- -SETUP- -ALERTING- -PROCEEDING- -ALERTING- -CONNECT -CONNECT ACK- -CONNECT- -CONNECT ACK- Peter Peter, if thats correct, then thats great! Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] A few questions - isdn call routing
Hi
There is a device called a parlay made by a crowd called
voxtream which will route the ISDN calls based on the DID
and/or the callerid, before the call is answered.
It would be nice if this feature could be done in Asterisk
as well, but at this point in time, it first answers the
call.
regards
Clive
On Tue, 03 Aug 2004 06:51:33 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
In trying to follow Marks advice and be nice to
newcomers, I'll just put
URLS below.
On Tue, 2004-08-03 at 06:10, Mark wrote:
We have several C/T servers with PRI lines that are
under utilised, in
the following configuration
eISDN - PRI - C/T Server 1
eISDN - PRI - C/T Server 2
eISDN - PRI - C/T Server 3
For our C/T applications we need the Dialed Number
passing from the PRI
to the C/T server - is this possible ?
Not exactly passing, but recreating.
exten = 123456,1,Dial(g2,${EXTEN})
This will essentially connect the 2 legs together and
introduce the
number on the other side.
If we install 2 or more of the Quad port ISDN cards,
and a call came in
on the first card, but was re-directed out of a second
card, is there a
dedicated bus between the cards (as with Dialogic
cards) or would it use
the Server's PCI bus ?
No, there is no Sbus or whatever it is called on
Dialogic. All calls
pass through the PCI bus. Probably covered on the Wiki
somewhere
http://www.voip-info.org/
Do you have any idea of the extra load this would put
on the CPU ?
There is a whitepaper on Digiums site discussing that.
http://www.digium.com/images/pdf/QuadCardCPUBenchmark.pdf
We also have a Samsung DCS phone switch that connects
to 4 BRI lines, do
you have or know of any product that will work with
asterisk and allow
us to connect this to the Asterisk server ?
Asterisk - 4xBRI - DCS
http://ns1.jnetdns.de/jn/relaunch/asterisk/page17.html
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] A few questions
On Wed, 21 Apr 2004, Ben Merrills waxed: Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). That's great news. MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems like a common feature in most hardware PBX systems we've used that support Call Conferences. Has anyone found a way of doing this? Is there an alternative to MeetMe that would support this feature (that's as good if not better?). I don't think this is currently supported, could be wrong, tho. Would take some modification to app_meetme.c, or else just have people say there name when it beeps them in. :) Sort of the flip side, but maybe it would be more helpful to have the person entering the conference hear the name of everyone already in it. That could be done via Record and Playback apps, before executing MeetMe. Every time someone enters, have Record take their name. Then, run Playback for each of the Recorded files. Queues: I'm running the 1.0 stable from the cvs server, and I've added the queue status announcement directives to the queues.conf - yet asterisk gives me the following errors: Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': monitor-format at line 9 of queue.conf I think this only works in development, not stable, CVS. :( --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] a few questions about sip implementation
Is this correct? I see the 100 Trying on REGISTER frequently, but if it's not valid, we can take it out. It serves no really effective purpose. 2. 10.3 Processing REGISTER requests. The 5th paragraph states that the registrar has to know the set of domain(s) for which it maintains bindings. How is this specified in Asterisk? Through the context? ie should the domains be specified in sip.conf via the context parameter, of the form context=domain.com? What's the practical meaning of that? Asterisk just uses the part in front of the @ sign and ignores the rest. This way you can use the domain or IP just the same. 3. I have another SIP account (sip:[EMAIL PROTECTED]) which I would like to use within asterisk both for dialing out and for receiving calls. I see that sip.conf has a line register = [EMAIL PROTECTED]/1234 where 1234 is the local asterisk extension. From chan_sip.c, line 1390 I see that I can use the form: register = user[:secret[:[EMAIL PROTECTED]:port][/localextension] However my registrar requires that I authenticate with domain.es, but use a sip proxy at ip 1.2.3.4, the two are unrelated and domain.es has no ip address. How can I get Asterisk to register with the remote prxoy? You do: register = 912345678:password:[EMAIL PROTECTED]/1234 Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a few questions about sip implementation
1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue a provisional response to non-INVITE requests. From my message yesterday * appears to be sending a SIP/2.0 100 Trying to X-Lite's REGISTER request before sending the SIP/2.0 200 OK message. Is this correct? Yes, that is what it is doing and and while it may not adherent to the exact reading of the RFC, I have seen several other proxies doing the same thing (examples: FWD's SIP proxy (Cisco?) does send 100 Trying but SER does not) so I will assume it's an awkward industry standard, though perhaps not exactly compliant to the RFC paragraph that you describe. But note that SHOULD NOT != MUST NOT. I assume this has been added as a clarification since the old RFC. They couldn't make it MUST NOT because of existing implementations that already did it. What I'm trying to say is that Asterisk doesn't fail compliance on that point (though no doubt there are places where it does!) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a few questions about sip implementation
[EMAIL PROTECTED] (Mark Spencer) writes: Is this correct? I see the 100 Trying on REGISTER frequently, but if it's not valid, we can take it out. It serves no really effective purpose. I think that it's only on REGISTER messaegs that it shouldn't be used. Perhaps previous RFCs didn't mention this. 2. 10.3 Processing REGISTER requests. The 5th paragraph states that the registrar has to know the set of domain(s) for which it maintains bindings. How is this specified in Asterisk? Through the context? ie should the domains be specified in sip.conf via the context parameter, of the form context=domain.com? What's the practical meaning of that? Asterisk just uses the part in front of the @ sign and ignores the rest. This way you can use the domain or IP just the same. No. I need ot register with the sip To: as To: [EMAIL PROTECTED] However the sip proxy IS NOT realm.com but an ip wit no name: 1.2.3.4 Currently if i use register = [EMAIL PROTECTED] ; fails as no ip to connect to register = [EMAIL PROTECTED] ; fails as realm/domain is incorrect. John's previous message appears to be a solution if I can parse the more complicated extended format. 3. I have another SIP account (sip:[EMAIL PROTECTED]) which I would like to use within asterisk both for dialing out and for receiving calls. I see that sip.conf has a line register = [EMAIL PROTECTED]/1234 where 1234 is the local asterisk extension. From chan_sip.c, line 1390 I see that I can use the form: register = user[:secret[:[EMAIL PROTECTED]:port][/localextension] However my registrar requires that I authenticate with domain.es, but use a sip proxy at ip 1.2.3.4, the two are unrelated and domain.es has no ip address. How can I get Asterisk to register with the remote prxoy? You do: register = 912345678:password:[EMAIL PROTECTED]/1234 ok. thanks I'll try that tomorrow. Thanks again. Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a few questions about sip implementation
3. I have another SIP account (sip:[EMAIL PROTECTED]) which I would like to use within asterisk both for dialing out and for receiving calls. I see that sip.conf has a line register = [EMAIL PROTECTED]/1234 where 1234 is the local asterisk extension. From chan_sip.c, line 1390 I see that I can use the form: register = user[:secret[:[EMAIL PROTECTED]:port][/localextension] However my registrar requires that I authenticate with domain.es, but use a sip proxy at ip 1.2.3.4, the two are unrelated and domain.es has no ip address. How can I get Asterisk to register with the remote prxoy? I sent a note to Mark about this. We had discussed a patch to fix this back in Feb, but apparently it didn't do quite the trick (due to my lack of testing.) Lines like this are supported: register = [EMAIL PROTECTED]:[EMAIL PROTECTED]/9993 However, the current REGISTER routines don't chop off blatz.filbert.com in the To: and From: fields. Hopefully a repair will be seen in the future. Ok. I hadn't understood the source too well in that respect as there was also the :port bit mentioned. if this format register = [EMAIL PROTECTED]:[EMAIL PROTECTED]:port/localpart is acceptable I can look at modifying the existing source to accept this format and using it appropriately. That shouldn't be too hard (I think). Actually I just did so you can do just that, thanks to John Todd clarifying what he wanted me to do :) mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
