Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
John, For incoming fax numbers, did you port existing numbers or did you get new numbers from bandwidth.com ? If the later, what if you switch for another provider ? Would you then be able to port the given number to your new provider ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to loging Agent in asterisk 1.4.13 ?
how to loging agent asterisk 1.4.13? thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
On 10/10/07, Ex Vito [EMAIL PROTECTED] wrote: Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. More than 1 PRI? All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Private network? How good? How saturated? Could be possible to just run ulaw if the quality is as good as your LAN 1. On the locations where asterisk is installed, the solution is trivial; either by connecting FAXes to FXS ports on channelbanks or by managing faxes with iaxmodem + Hylafax. Probably a combination of both... Why channel banks? 2. On the remaining locations we have a problem b) T.38 is the answer to FoIP c) asterisk 1.2 does not support T.38 d) asterisk 1.4 only does T.38 passthrough, not good enough Use a VoIP provider with t.38 for your faxes... easy solution. e) CallWeaver seems to support T.38 gatewaying, although I'd rather move on with asterisk so as to leverage current experience and knowledge and to keep installed base with the same software. I've been waiting for callwaver 1.2 final for a while. Tried some betas and T38 gateway didnt work even when we put a Sangoma card in the machine. Problem was on the SIP side. [PSTN] ---PRI--- [asterisk] ---PRI--- [PRI-to-T38 GW] ... ... --SIP/T.38--- [T.38 ATA] ---FXS--- FAX machine Too many PRI... Try: PSTN ---PRI AS5300 --SIP- Asterisk 1.2 PSTN ---PRI AS5300 --SIP- Asterisk 1.4 -SIP T.38 ATA PSTN ---PRI AS5300 --SIP- T.38 ATA 4. Of course, I could use CallWeaver as a PRI-to-T.38 gateway... But then again, how solid would it be ? With which ATAs ? The CallWeaver website shows a very small amount of ATAs confirmed to be 100% working in T.38. There's a reason why CallWeaver is beta. As much as I'd love to support their stuff. It's still in beta. 5. Would I need to have a SIP proxy between the PRI-to-T.38 gw and the T.38 ATAs or would they be able to talk to each other directly ? (I'd say this would depend on the specific equipment, but...) If that would be a requirement, which way would you go, asterisk 1.4 ? Would SER forward T.38 traffic ? SER is a SIP proxy. T.38 is irrelevant to it. I'd use 1.4, your setup seems pretty straightforward. You don't have a diverse population of SIP phones and locations to manage. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German SIP and/or IAX providers?
Hello Ken, Hi, all. My company is setting up a branch office in Germany, and I'm very interested in a VoIP provider over thataway. as I am living in Germany, let me advise you against using VoIP providers in Germany. Most of the time they do work, but they are not as reliable as a regular phone company. What I do is, use a regular phone line (ISDN BRI) for incoming traffic, and utilize the VoIP providers for dialing out. What is your reasoning for a VoIP provider? BTW: The language should not be the problem. The service is poor, no matter what language... -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 9133i autoanswer with headset
Hijacking a thread again - the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, please, please, Digium, sort the filters out!) I've added the auto-answer header in my dialplan, and it works great. However, there is a problem with a headset - If I use a headset, then no matter what the settings on the phone are, the auto-answer *always* defaults to the speakerphone rather than the headset. Has anyone else encountered this, and have a solution ? Many thanks Julian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP Users Conference Friday Oct 12 @ 12:30 PM EDT
VOIP Users Conference Friday Oct 12 @ 12:30 PM EDT Today, besides whatever else is on anyone else's minds, we'll be discussing some asterisk communication techniques for use with SMS, email and web API. Also, some typical uses for the asteriskdb, what it is and why you care about it at all. Join us: http://VoipUsersConference.org/join.php IRC: freenode.net #voip-users-conference ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use an Application from inside an Application?
Hello, I wonder if there is a way to build my own asterisk application (let us say apps/app_myappl.c), and to launch other existing applications from it (for example, doing an apps/app_dial.c, or others). Could someone highlight me on that? thx Pirlouwi. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use an Application from inside an Application?
On Friday 12 October 2007 04:28:42 Pirlouwi wrote: I wonder if there is a way to build my own asterisk application (let us say apps/app_myappl.c), and to launch other existing applications from it (for example, doing an apps/app_dial.c, or others). Both the Page app and VoicemailMain do this, respectively for MeetMe and Directory, so you can look at their source for examples. In the case of Page, the guts of it is: app = pbx_findapp(MeetMe); pbx_exec(chan, app, meetmeopts); Please check out app_page.c for the rest of the syntax. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about PSTN pickup
On 10/12/07, Yair Hakak [EMAIL PROTECTED] wrote: you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered channel an answered channel? Asterisk 1.4 has the AMD() application that uses several thresholds (which you can vary) to determine if the other end is a human or machine: -= Info about application 'AMD' =- [Synopsis] Attempts to detect answering machines [Description] AMD([initialSilence][|greeting][|afterGreetingSilence][|totalAnalysisTime] [|minimumWordLength][|betweenWordsSilence][|maximumNumberOfWords] [|silenceThreshold]) This application attempts to detect answering machines at the beginning of outbound calls. Simply call this application after the call has been answered (outbound only, of course). When loaded, AMD reads amd.conf and uses the parameters specified as default values. Those default values get overwritten when calling AMD with parameters. - 'initialSilence' is the maximum silence duration before the greeting. If exceeded then MACHINE. - 'greeting' is the maximum length of a greeting. If exceeded then MACHINE. - 'afterGreetingSilence' is the silence after detecting a greeting. If exceeded then HUMAN. - 'totalAnalysisTime' is the maximum time allowed for the algorithm to decide on a HUMAN or MACHINE. - 'minimumWordLength'is the minimum duration of Voice to considered as a word. - 'betweenWordsSilence' is the minimum duration of silence after a word to consider the audio that follows as a new word. - 'maximumNumberOfWords'is the maximum number of words in the greeting. If exceeded then MACHINE. - 'silenceThreshold' is the silence threshold. This application sets the following channel variable upon completion: AMDSTATUS - This is the status of the answering machine detection. Possible values are: MACHINE | HUMAN | NOTSURE | HANGUP AMDCAUSE - Indicates the cause that led to the conclusion. Possible values are: TOOLONG-%d total_time INITIALSILENCE-%d silenceDuration-%d initialSilence HUMAN-%d silenceDuration-%d afterGreetingSilence MAXWORDS-%d wordsCount-%d maximumNumberOfWords LONGGREETING-%d voiceDuration-%d greeting -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
No, I am not sure whether it's still an issue with the newer series because I have not had a chance to test this with one of those models. Regards, Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield Sent: Thursday, October 11, 2007 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging in Asterisk Joseph Begumisa wrote: I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. Interesting. I've seen the reboot mystery mentioned before. Some have pointed to power, but this makes sense. Do you know if this is still an issue on the newer series (330,550,650) phones as well? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to simulate TELCO using a TE210P?
Is there an example on how to use two E1 ports connected to each other to simulate connections? Since I do not have an E1 at the office I need for one port to act normally and the other to act as if it were the telephone company so I can send calls from one E1 to the other. Someone has an example config? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-gui
Hi Steve, you are totally right, but my question is because a saw that gui into SVN and not yet released, but at the same time used into AsteriskNOW. Was just a question... ;o) Thanks 2007/10/12, Steve Totaro [EMAIL PROTECTED]: FaberK wrote: Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. I may be totally wrong but at Astricon, during the What's New at Digium (SwitchVox purchase) I asked the question of what would happen to AsteriskNow to one of the Adtran/Digium guys. There was not a real direct answer, I will try to quote as best I can from memory. He simply said It will remain opensource. I take that to mean that they will not be developing it anymore and it is up to the community to further the project. Why would Digium continue to develop a GUI for free that would compete with SwitchVox (or whatever they change the name to). Maybe I am wrong. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
In 'top', you can always look at what percentage of your CPU is idle. Subtract that from 100 and you've got your load average. Cpu(s): 1.1% us, 0.6% sy, 0.0% ni, *98.1% id*, 0.1% wa, 0.1% hi, 0.0% si Erik Anderson wrote: On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I don't think there is a formula like cpu usage = loadavg / #cpus A loadavg of 3 says that there are 3 processes waiting to be executed. Anyway, I'll admit that a loadavg of 3 /might/ be ok. Here's a quote from this page: http://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation For systems with multiple CPUs, the number needs to be divided by the number of processors in order to get a percentage. - Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
Hi, I am going to setting up an asterisk setup similar to yours. I talked to the folks at bandwidth.com and they SIP trunking for $25/mo/line with 2000 minutes per line. I haven't tried their service yet though, so I can't say how the quality is. Regards, Zaheer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of D4rk F1ber Sent: Friday, October 12, 2007 1:42 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Affordable SIP Trunk for Home PBX ? So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. I have seen mentions of Skype integration with Asterisk, but does that include say Skype IN and Skype OUT ? Or is that integration component really just for being able to contact skype users? Looking for the easiest and cheapest way to reach the PSTN, and well the options out there are plenty regarding SIP trunks, but most tend to be geared towards businesses for obvious reasons. Curious what others are using, and if anyone can make some recommendations? Not sure if this has been covered already on the list, and not sure if recommending companies are allowed, so maybe I need get replies off list? Any suggestions would be appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to simulate TELCO using a TE210P?
Crossover cable and pri_net/pri_cpe. It is well documented if you search a little. Thanks, Steve Totaro Carlos Chavez wrote: Is there an example on how to use two E1 ports connected to each other to simulate connections? Since I do not have an E1 at the office I need for one port to act normally and the other to act as if it were the telephone company so I can send calls from one E1 to the other. Someone has an example config? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] lookup_user : specified user 'asterisk' unknown? installing zaptel?
Vincent wrote: 4. install ztdummy: echo ztdummy /etc/modules modprobe ztdummy === Since you are using the OpenVOX FXO card, don't you need another module? I'm guessing you'd need wctdm INSTEAD of ztdummy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-gui
If you read between my lines, my guess is it won't be released unless the community takes it out of Digium's hands and forks it. Thanks, Steve FaberK wrote: Hi Steve, you are totally right, but my question is because a saw that gui into SVN and not yet released, but at the same time used into AsteriskNOW. Was just a question... ;o) Thanks 2007/10/12, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: FaberK wrote: Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. I may be totally wrong but at Astricon, during the What's New at Digium (SwitchVox purchase) I asked the question of what would happen to AsteriskNow to one of the Adtran/Digium guys. There was not a real direct answer, I will try to quote as best I can from memory. He simply said It will remain opensource. I take that to mean that they will not be developing it anymore and it is up to the community to further the project. Why would Digium continue to develop a GUI for free that would compete with SwitchVox (or whatever they change the name to). Maybe I am wrong. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Steve Totaro wrote: I don't think that is correct. I am running worldcommunitygrid and this is what I get top - 13:18:56 up 3 days, 22:49, 1 user, load average: 4.00, 4.04, 4.02 Cpu0:0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu1:0.0%us,0.0%sy,100.0%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu2:0.0%us,0.3%sy,98.0%ni,0.0%id,0.0%wa,1.7%hi,0.0%si,0.0%st Cpu3: 0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st According to what you are saying my load average should be 100. You're correct, it IS 100 -- but 100%. Expressed in decimal format, this is of course 1.0 -- and as each cpu has this average, 4.0 indicates that no threads regularly wait for execution. This worldcommunitygrid you mentioned binds your cpu by design it sounds like. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-gui
On Friday 12 October 2007 10:45:52 Steve Totaro wrote: FaberK wrote: I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? I may be totally wrong but at Astricon, during the What's New at Digium (SwitchVox purchase) I asked the question of what would happen to AsteriskNow to one of the Adtran/Digium guys. There was not a real direct answer, I will try to quote as best I can from memory. He simply said It will remain opensource. I take that to mean that they will not be developing it anymore and it is up to the community to further the project. Why would Digium continue to develop a GUI for free that would compete with SwitchVox (or whatever they change the name to). Maybe I am wrong. I don't think a decision has been made yet either way, although I am not a party to those discussions. Contributions are always welcomed, no matter where the main development of asterisk-gui is occurring, though. So if the community wants asterisk-gui to remain alive and there's someone willing to devote development resources, then it will remain an active project for some time to come. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec by Network?
Does anyone have any tricks to allow codecs based on what network a phone is on? i.e., allow uLaw if the device is on the LAN, and only allow g729 if the device is anywhere else? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
I don't think that is correct. I am running worldcommunitygrid and this is what I get top - 13:18:56 up 3 days, 22:49, 1 user, load average: 4.00, 4.04, 4.02 Cpu0:0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu1:0.0%us,0.0%sy,100.0%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu2:0.0%us,0.3%sy,98.0%ni,0.0%id,0.0%wa,1.7%hi,0.0%si,0.0%st Cpu3: 0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st According to what you are saying my load average should be 100. Thanks, Steve Mik Cheez wrote: In 'top', you can always look at what percentage of your CPU is idle. Subtract that from 100 and you've got your load average. Cpu(s): 1.1% us, 0.6% sy, 0.0% ni, *98.1% id*, 0.1% wa, 0.1% hi, 0.0% si Erik Anderson wrote: On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I don't think there is a formula like cpu usage = loadavg / #cpus A loadavg of 3 says that there are 3 processes waiting to be executed. Anyway, I'll admit that a loadavg of 3 /might/ be ok. Here's a quote from this page: http://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation For systems with multiple CPUs, the number needs to be divided by the number of processors in order to get a percentage. - Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
On 10/12/07, D4rk F1ber [EMAIL PROTECTED] wrote: Curious what others are using, and if anyone can make some recommendations? Not sure if this has been covered already on the list, and not sure if recommending companies are allowed, so maybe I need get replies off list? There are quite literally hundreds of VoIP service providers out there: Here's a list of some of them: http://voip-info.org/wiki/view/VOIP+Service+Providers+Residential Billing schemes usually fall into one of two categories. They'll either bill you a flat monthly fee for an unlimited plan or one with a large number of minutes. Or...they'll bill you on a per-minute, usage-only basis. The only provider I've had direct experience with is Teliax. I'm on an outgoing-only plan with them and it's been perfect so far. They bill something like $0.025/minute. If you want incoming calls as well, there's a per-month DID charge. If you are just wanting to receive incoming calls, check out IPKall - they'll give you a DID and a SIP trunk to your PBX for incoming calls. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I don't think there is a formula like cpu usage = loadavg / #cpus A loadavg of 3 says that there are 3 processes waiting to be executed. Anyway, I'll admit that a loadavg of 3 /might/ be ok. Here's a quote from this page: http://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation For systems with multiple CPUs, the number needs to be divided by the number of processors in order to get a percentage. - Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk System Setup Question
Yes that is correct. I have no analog phone service here, I plan on using direct SIP Trunking. My connections to the internet are two load balanced high-speed connections. The first is a cable modem operating at 20d/5u and the second is a fiber optic connection that is operating at 20d/5u. Both lines can be upgraded and the fiber optic connection is backed by an SLA. -Original Message- Subject: Re: [asterisk-users] Asterisk System Setup Question Zaheer Master wrote: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) You wouldn't be able to add TDM cards to this scenario if you wanted to down the line, as vmware can't access the hardware installed in a box. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Erik Anderson wrote: On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I wouldn't be too happy about a system with a loadavg of 3. The system he mentioned had 8 cores, though. So a load average of 3 is less than 50% usage. I don't think there is a formula like cpu usage = loadavg / #cpus A loadavg of 3 says that there are 3 processes waiting to be executed. Anyway, I'll admit that a loadavg of 3 /might/ be ok. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (http://www.dundi.com): Service Provider
Hi List; When I readed about DUNDI, I though there is a website and we can buy and sell routes there using DUNDI, but when I tried to browse http://www.dundi.com) then I found a strange page that contains only this text It Works! So, what is the market of DUNDI really? Is there a method to buy and sell routes using DUNDI? Or DUNDI can be applied only between my distributed PBX's in different areas? From the other side, I am looking for a Collocation Center (Hosting Center) that I can collocate my servers, and route my VoIP calls via it. I need a collocation who can provide VoIP routes and Hosting for the servers, any help? Also, what DUNDI can help in this topic? Any help? Regards Bilal Ghayad Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Display channels and codecs
Is there an easy way to show all active channels AND the codecs they're using? Other than going through EACH channel individually to check the codec which is, obviously, not a very efficient process. Thanks, Scott ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk System Setup Question
Zaheer Master wrote: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) You wouldn't be able to add TDM cards to this scenario if you wanted to down the line, as vmware can't access the hardware installed in a box. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On Fri, 12 Oct 2007, Tilghman Lesher wrote: On Friday 12 October 2007 10:29:24 Philipp Kempgen wrote: Atis Lezdins wrote: I have 8-core system that has web interface + sql + java + some other stuff running, and at 30 simultenous calls i get loadavg maximum of 3. I wouldn't be too happy about a system with a loadavg of 3. I dunno, 3 wouldn't be terrible on a 4 processor or 8 processor system, which isn't getting to be nearly as rare as it once was. Don't get too hung-up on load average under Linux. It's not always indicative of the real load on the machine, it merely indicates the number of processes running, or avalable to run - so if a process is waiting on IO, it's 'running' and will get counted. I've seen servers (non asterisk) with huge load averages but ones which were still usable because the processes were waiting on IO from a slow device, (eg. remote NFS mounts) so there was plenty of CPU left for computational tasks, etc. So 3 threads reading or writing to/from a TDM card might well spend most of their time waiting for the IO to complete (clock in/out the A/D, A/D convertors for example), give a load avg. of 3, yet the CPU should be avalable for other tasks like shoveling RTP data over Ethernet for example... Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 9133i autoanswer with headset
Eric ManxPower Wieling wrote: Julian Lyndon-Smith wrote: Hijacking a thread again - the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, please, please, Digium, sort the filters out!) You seem to be subscribed to the list as [EMAIL PROTECTED]. Is that the e-mail address you are using when you are trying to start a new it is indeed ;) Julian. thread. If the mailing list allowed messages from non-subscriber e-mail address the list would be destroyed by spam. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
Olivier wrote: 2007/10/12, Jonn Taylor [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Would you then be able to port the given number to your new provider ? Yes, if you are in the US. Number port ability is one thing that ALL VIOP providers had to provide. Here too (France), number portability is mandatory but in facts, I couldn't find any pure fax service provider complying with this. I think they bet on the fact they are not telco so they don't have to comply with telco regulation. I could find fax services from telco but services are often poor or neglected. That's fine you could find something up to your expectations : it gives me hope I could find one in the future. Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Olivier, Check this out, might help. http://www.voipproviderslist.com/country/voip-france/voip-providers-france/ Jonn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
I was told yesterday (by Cantata guy) that T.38 demands a good level of QoS. That surprised me a lot as I thought the whole purpose of T.38 was to avoid SIP and ToIP latency. Another editor (Interstar) told me T.38 passthrough doesn't work. As devil lies in details and I couldn't get any, I'm not sure these words would be of any use. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Hi, Brian: Brian West wrote: Matt, I talk very openly about this issue. It was very rude of you to bring this up. This plea was total bullshit. If you want to know the whole story feel free to call me and talk about it. 918-424-9378... anyone can call me and ask me questions about it. The plea was a deal worked out between the DOJ and my attorney which was good because I signed my plea on Sept. 4th 2001. If you try to fight the DOJ you will not win. That plea was the only way to settle the issue without a trial. All I did was click edit in frontpage and alert them of anonymous publishing priv. were on their servers and they called the FBI and three days later our office was raided. This I consider mudslinging by you and wasn't very gentle man like. I'm not making excuses for anyone, but I have the following suggestion: If you treat people respectfully, without being snide or aggressive, then you will be much less likely to open yourself up to responses in kind. You throw your weight around a lot on the list. Try giving friendly, constructive input. You will be amazed at the results. Cheers, -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
2007/10/12, Jonn Taylor [EMAIL PROTECTED]: Would you then be able to port the given number to your new provider ? Yes, if you are in the US. Number port ability is one thing that ALL VIOP providers had to provide. Here too (France), number portability is mandatory but in facts, I couldn't find any pure fax service provider complying with this. I think they bet on the fact they are not telco so they don't have to comply with telco regulation. I could find fax services from telco but services are often poor or neglected. That's fine you could find something up to your expectations : it gives me hope I could find one in the future. Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Atis Lezdins wrote: I have 8-core system that has web interface + sql + java + some other stuff running, and at 30 simultenous calls i get loadavg maximum of 3. I wouldn't be too happy about a system with a loadavg of 3. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to loging Agent in asterisk 1.4.13 ?
Search for Agent at http://www.voip-info.org/ On 10/12/07, Walter Willis [EMAIL PROTECTED] wrote: how to loging agent asterisk 1.4.13? thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk System Setup Question
Hi Zaheer, Go the standalone machine. It will save you time and effort. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer Master Sent: Friday, 12 October 2007 10:05 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk System Setup Question Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones but I've heard people have had some issues with that. The Aastra 51i seems to be a good choice. I'd eventually like to link my Asterisk installation with my CRM, so the XML capabilities of the 51i should come in handy. I'm going to use SIP Trunking through Bandwidth.com with 2-3 SIP trunks. I'd like to know what more experienced users think of this setup, and if anyone has had any experience with either the 51i phones or the Bandwidth.com service. Also, has anyone had problems using a fax machine (either computer or paper) with Asterisk? Thank you in advance for your input! Regards, Zaheer K. Master President, Adamant Security Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-gui
Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German SIP and/or IAX providers?
On Friday 12 October 2007 04:38, Ken D'Ambrosio wrote: Hi, all. My company is setting up a branch office in Germany, and I'm very interested in a VoIP provider over thataway. However, I'd need a few things: - Reliability. Can't have my branch office's DID's just going down. A then you have to chosse for incomming calls ISDN and route from there to other destinations if needed. For outgoing use an sip provider, ISDN for special numbers not supported by SIP provider. And that's about -it-. I'm even willing to pay a reasonable premium, so long as it gets me a VoIP provider with the above restrictions. save your money and spend it for ISDN lines as long as the calls have more then 80% final destination in your local office. Consider also that your DSL line can go down, thats not the responsibilty of the SIP provider. ISDN is in Germany extremly reliable. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk System Setup Question - 2nd Try
I've been trying to post this message to the list, but it keeps getting bounced back to me. Sorry if duplicate messages are coming through. Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones but I've heard people have had some issues with that. The Aastra 51i seems to be a good choice. I'd eventually like to link my Asterisk installation with my CRM, so the XML capabilities of the 51i should come in handy. I'm going to use SIP Trunking through Bandwidth.com with 2-3 SIP trunks. I'd like to know what more experienced users think of this setup, and if anyone has had any experience with either the 51i phones or the Bandwidth.com service. Also, has anyone had problems using a fax machine (either computer or paper) with Asterisk? Thank you in advance for your input! Regards, Zaheer K. Master President, Adamant Security Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
Yes, this is true when using presence on the 601's. With presence disabled, you get no reboots at all. That's why when I realized that, I decided on the setup I mentioned below. However, you could let us know whether this is true for the 650's 550's and 330's. Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Friday, October 12, 2007 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging in Asterisk I use a mysql script to dynamically generate the page command and page about 70 phones, and I have never had a reboot problem. Sometimes there is a slight delay waiting for all the phones to join the page conference. I am using a mix of 650's, 550's, and 330's. It must only be an issue if you are using presence. Maybe I will setup presence on a couple phones and see if they reboot. Forrest Beck [EMAIL PROTECTED] http://www.shift8.biz/blog On Oct 11, 2007, at 4:34 PM, Jim Canfield wrote: Joseph Begumisa wrote: I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. Interesting. I've seen the reboot mystery mentioned before. Some have pointed to power, but this makes sense. Do you know if this is still an issue on the newer series (330,550,650) phones as well? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use an Application from inside an Application?
Pirlouwi wrote: Hello, I wonder if there is a way to build my own asterisk application (let us say apps/app_myappl.c), and to launch other existing applications from it (for example, doing an apps/app_dial.c, or others). Could someone highlight me on that? thx Pirlouwi. Even better question for me is if Asterisk can call libraries not written in C, but that export their routines under cdecl calling convention. This might be a better question for dev lists though. I'd really like to start writing some .so libraries for use within Asterisk without having to use AGI. -- Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk System Setup Question
Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones but I've heard people have had some issues with that. The Aastra 51i seems to be a good choice. I'd eventually like to link my Asterisk installation with my CRM, so the XML capabilities of the 51i should come in handy. I'm going to use SIP Trunking through Bandwidth.com with 2-3 SIP trunks. I'd like to know what more experienced users think of this setup, and if anyone has had any experience with either the 51i phones or the Bandwidth.com service. Also, has anyone had problems using a fax machine (either computer or paper) with Asterisk? Thank you in advance for your input! Regards, Zaheer K. Master President, Adamant Security Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Searching archives for Colinux
Newbie to the list. Are the archives searchable like Listserve archives, or do you just use Google? In another reality I am a mainframe Sysprog, so this stuff is somewhat alien to me. I would like to find previous postings on running Asterisk with CoLinux. I have the Win32 version of Asterisk installed and I have built Asterisk on Topologic Linux running under Colinux on the same WinXP system. I can dial from one Asterisk to the other - sort of a problem with the sound card support, but I am only interested in connectivity. My current area of interest is to get Proxy Authentication set up between the instances of Asterisk, but it isn't working (407 and 403). May be a NAT issue, but perhaps someone has done this before and can give me some guidance. Dave * This email is intended solely for the use of the individual to whom it is addressed and may contain confidential and/or privileged material. Any views or opinions presented are solely those of the author and do not necessarily represent those of AGCO. If you are not the intended recipient, be advised that you have received this email in error and that any use, dissemination, forwarding, printing or copying of this email is strictly prohibited. Neither AGCO nor the sender accepts any responsibility for viruses and it is your responsibility to scan and virus check the e-mail and its attachment(s) (if any). * AGCO Limited, a limited company, registered in England (registered no.509133) with its registered office at Abbey Park Stoneleigh, Kenilworth CV8 2TQ, England. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
I use a mysql script to dynamically generate the page command and page about 70 phones, and I have never had a reboot problem. Sometimes there is a slight delay waiting for all the phones to join the page conference. I am using a mix of 650's, 550's, and 330's. It must only be an issue if you are using presence. Maybe I will setup presence on a couple phones and see if they reboot. Forrest Beck [EMAIL PROTECTED] http://www.shift8.biz/blog On Oct 11, 2007, at 4:34 PM, Jim Canfield wrote: Joseph Begumisa wrote: I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. Interesting. I've seen the reboot mystery mentioned before. Some have pointed to power, but this makes sense. Do you know if this is still an issue on the newer series (330,550,650) phones as well? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On Friday 12 October 2007 11:10:02 Gordon Henderson wrote: On Fri, 12 Oct 2007, Tilghman Lesher wrote: On Friday 12 October 2007 10:29:24 Philipp Kempgen wrote: Atis Lezdins wrote: I have 8-core system that has web interface + sql + java + some other stuff running, and at 30 simultenous calls i get loadavg maximum of 3. I wouldn't be too happy about a system with a loadavg of 3. I dunno, 3 wouldn't be terrible on a 4 processor or 8 processor system, which isn't getting to be nearly as rare as it once was. Don't get too hung-up on load average under Linux. It's not always indicative of the real load on the machine, it merely indicates the number of processes running, or avalable to run - so if a process is waiting on IO, it's 'running' and will get counted. I've seen servers (non asterisk) with huge load averages but ones which were still usable because the processes were waiting on IO from a slow device, (eg. remote NFS mounts) so there was plenty of CPU left for computational tasks, etc. So 3 threads reading or writing to/from a TDM card might well spend most of their time waiting for the IO to complete (clock in/out the A/D, A/D convertors for example), give a load avg. of 3, yet the CPU should be avalable for other tasks like shoveling RTP data over Ethernet for example... Are you saying that Linux makes no differentiation between long and short wait states? That would be a fairly major abrogation of the spec. I do know of a legal way (under the definition) to drive a load average higher: simply release the processor resource prematurely with either a usleep(1) or a sched_yield(). The definition of load average depends implicitly upon a process using its entire timeslot on the processor; if a process does significantly less, the load average will rise without a corresponding increase in actual CPU work. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weatherproof Hard Phone
Stephen Bosch wrote: Philipp Kempgen wrote: Don Kelly wrote: http://www.sandman.com/autodial.html These phones look like the ones we had in Germany 20 years ago. ;-P Hey, don't knock it, Phillipp :) -- I'm as big a fan of German technology as anybody, but these phones are amazing pieces of engineering. Reliable, with excellent sound quality, and practically indestructible. There's a reason they're still in production after all these years. Also, look at http://www.ceeco.net/ for a big selection of extra-heavy-duty phones. They're rather expensive, though. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
Olivier wrote: John, For incoming fax numbers, did you port existing numbers or did you get new numbers from bandwidth.com http://bandwidth.com ? Both, we ported numbers and got new one's. If the later, what if you switch for another provider ? I did alot of research before we went with bandwidth.com. They resell Level 3 service. If I would switch, I would verify that the provider that we switch to a) has very low latency b)has mutiple backup nodes. The other key is your internet provider, so long as they pass all TCP header info your good to go. We use Comcast Business service and get 99.999% uptime. Would you then be able to port the given number to your new provider ? Yes, if you are in the US. Number port ability is one thing that ALL VIOP providers had to provide. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jonn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about PSTN pickup
hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered channel an answered channel? thanks in advance for any help, yair ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use an Application from inside an Application?
On Fri, Oct 12, 2007 at 11:28:42AM +0200, Pirlouwi wrote:
Hello,
I wonder if there is a way to build my own asterisk application (let us say
apps/app_myappl.c),
and to launch other existing applications from it (for example, doing an
apps/app_dial.c, or others).
Could someone highlight me on that?
grep pbx_exec apps/*.c
:)
For example, apps/app_queue.c calls MixMonitor in about that way
(lots of code skipped for clarity):
struct ast_app *mixmonapp = NULL;
mixmonapp = pbx_findapp(MixMonitor);
if (mixmonapp) {
ret = pbx_exec(qe-chan, mixmonapp, mixmonargs);
} else
ast_log(LOG_WARNING, Asked to run MixMonitor on this call, but cannot
find the MixMonitor app!\n);
___
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[asterisk-users] Hiding extensions from app_directory
Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated company directory. I found some information about adding the hidefromdir=yes option to the user's entry in voicemail.conf, but that doesn't seem to have any effect. I'm a little unclear as to whether or not that option is something native to Asterisk or if it comes from one of the external applications. If it is built in, is my version too old to have this feature? Am I totally on the wrong track and is there another way to accomplish this? Thanks, -Jesse ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
On 10/12/07, D4rk F1ber wrote: So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. [...] There are several providers, I use Teliax, there are no long term commitments or contracts involved, if it doesn't work for you, you cancel : https://www.teliax.com/newaccount/ Teliax gives you detailed step by step instructions for setting up your asterisk to work with their svc in their Support section. -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German SIP and/or IAX providers?
Peer Oliver Schmidt schrieb: ... as I am living in Germany, let me advise you against using VoIP providers in Germany. Most of the time they do work, but they are not as reliable as a regular phone company. Hi, on the one hand, I should ignore this thread, because it is not asterisk related, and risks to distribute commercial information, on the other hand, I think, at least one different opinion should be opposed to Peer Oliver's statement: I would agree, that there are big differences. Anyway, there are some VoIP providers who have specialiced in providing very reliable VoIP service to companies. ISDN technology is without any doubt (still) the most reliable thing in the telephony market. However, the difference to VoIP service offered by an appropriate company is imho almost negligible. Especially some small or medium sized providers offer very good customer support. Roger. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.13 build crashed
Hi, am building the latest version of Asterisk (1.4.13) on a self-build Linux host (based on LFS-6.3). Version 1.4.12 built installed and worked fine. Last night I upgraded the kernel to 2.6.23 and rebuilt the zaptel driver package 1.4.5.1 against it. That seemed to build and install O.K. too. I dl'd the 1.4.13 source tarball and tried to build that: ./configure ran O.k. 'make menuselect' was O.K. During 'make' however, gcc segfaulted. After the crash, to validate my host environment, I re-built Asterisk version 1.4.12 against the exact same system (without restarting the host) and that built fine. My host's details: glibc-2.5.1, gcc-4.2.1, binutils-2.17. Here's the last few lines of output before it crashed: [CC] res_smdi.c - res_smdi.o [LD] res_smdi.o - res_smdi.so [CC] res_speech.c - res_speech.o [LD] res_speech.o - res_speech.so [CC] chan_agent.c - chan_agent.o [LD] chan_agent.o - chan_agent.so [CC] chan_iax2.c - chan_iax2.o [CC] iax2-parser.c - iax2-parser.o [CC] iax2-provision.c - iax2-provision.o [LD] chan_iax2.o iax2-parser.o iax2-provision.o - chan_iax2.so [CC] chan_local.c - chan_local.o [LD] chan_local.o - chan_local.so [LD] gentone.c - gentone ./gentone busy 480 620 Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Wavelength 1 (in samples): 12.90323 Minimum samples (1): 400 (31.00.3 wavelengths) Need 400 samples Wrote busy.h ./gentone ringtone 440 480 Wavelength 1 (in samples): 18.18182 Minimum samples (1): 200 (11.00.3 wavelengths) Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Need 200 samples Wrote ringtone.h [CC] chan_oss.c - chan_oss.o [LD] chan_oss.o - chan_oss.so [CC] chan_phone.c - chan_phone.o [LD] chan_phone.o - chan_phone.so [CC] chan_sip.c - chan_sip.o [LD] chan_sip.o - chan_sip.so [CC] chan_zap.c - chan_zap.o chan_zap.c: In function ‘process_zap’: chan_zap.c:11149: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://gcc.gnu.org/bugs.html for instructions. make[1]: *** [chan_zap.o] Error 1 make: *** [channels] Error 2 (FYI, my configure line was ./configure --prefix=/usr --sysconfdir=/etc --localstatedir=/var. I am installing to a DESTDIR currently, as I am trying to document a sane build process where I can run Asterisk as non-root. I am also planning to submit my docs to the BLFS team, hopefully for inclusion in their book at some later date) If I can help further just let me know. For now, I will go back to 1.4.12. Cheers Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
What a waste of time... dave On Fri, 2007-10-12 at 09:35 -0600, Stephen Bosch wrote: Brian West wrote: And what was the purpose of this? So that we would realize who we were talking to. :) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Other apps checking Day/Night
On Friday 12 October 2007 10:34:39 C. Duncan Hudson wrote: I'm fairly new to Asterisk, so please bear with me if this is silly question. I'd like to run a script on my server that would take the Call now to order banner off my website automatically when I put my phone system on night. I can handle the webserver side of things, but I don't know where to begin on the Asterisk side of things - can a simple script be run to check the value of the day/night condition, or is that value written somewhere that I could check or poll? Any help / ideas are really appreciated. Thanks in advance, See func_odbc.conf -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On Friday 12 October 2007 10:29:24 Philipp Kempgen wrote: Atis Lezdins wrote: I have 8-core system that has web interface + sql + java + some other stuff running, and at 30 simultenous calls i get loadavg maximum of 3. I wouldn't be too happy about a system with a loadavg of 3. I dunno, 3 wouldn't be terrible on a 4 processor or 8 processor system, which isn't getting to be nearly as rare as it once was. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Brian West wrote: And what was the purpose of this? So that we would realize who we were talking to. :) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote voicemail in two Asterisk
Using two Asterisk connected between they, How do I can check the voicemail in a remote system but working like *97? I mean dont want ask the voicemail box, just the password and go to the voicemail of caller. If I have the same extensions in the two Asterisk it doesn't work. Thanks. -- Linux User Registered #232544 Jabber : [EMAIL PROTECTED] Ekiga : [EMAIL PROTECTED] ICQ : 337889406 GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk System Setup Question
Thanks for the quick response Dean. Can you elaborate any more? The reason I would like to use the virtual machine is that I can dynamically allocate more resources to the asterisk server as needed. Regards, Zaheer _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, October 12, 2007 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk System Setup Question Hi Zaheer, Go the standalone machine. It will save you time and effort. Regards, Dean Collins Cognation Pty Ltd mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer Master Sent: Friday, 12 October 2007 10:05 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk System Setup Question Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones but I've heard people have had some issues with that. The Aastra 51i seems to be a good choice. I'd eventually like to link my Asterisk installation with my CRM, so the XML capabilities of the 51i should come in handy. I'm going to use SIP Trunking through Bandwidth.com with 2-3 SIP trunks. I'd like to know what more experienced users think of this setup, and if anyone has had any experience with either the 51i phones or the Bandwidth.com service. Also, has anyone had problems using a fax machine (either computer or paper) with Asterisk? Thank you in advance for your input! Regards, Zaheer K. Master President, Adamant Security Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Other apps checking Day/Night
I guess it all depends on how you put your system on night mode. If you use asterisk's database, and manually put it in night mode every day, my guess is that you dial an extension which puts it in nightmode. You could include as part of this the system command to touch a file, and then on your webserver check for existance of that file. That may be opening up more holes than needed between your web server and asterisk. As Tilghman suggested, func_odbc.conf may be better. You could then set and unset by writing it directly to your store's database. On 10/12/07, C. Duncan Hudson [EMAIL PROTECTED] wrote: I'm fairly new to Asterisk, so please bear with me if this is silly question. I'd like to run a script on my server that would take the Call now to order banner off my website automatically when I put my phone system on night. I can handle the webserver side of things, but I don't know where to begin on the Asterisk side of things - can a simple script be run to check the value of the day/night condition, or is that value written somewhere that I could check or poll? Any help / ideas are really appreciated. Thanks in advance, Dunc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Well, this has become a hot topic! :p Thinking about my original post, I was reluctant of installing my PBX on a shared system, is a Dell PowerEdge 2950 with 2 Intel Xeon Dual Core CPUs @2GHz (4 totals cores) and 4GB RAM which serves as Domain Controller and File Server (Samba), central backup server (Bacula with a LTO2 external tape drive), it has dual NIC in a bonding alb mode and redundant PSU (each one connected to a different UPS). It has a PCI slots in which I can install my Sangoma Remora A400D card. But now I think the PBX will work just fine in this system, maybe breaking the channel bonding and dedicating a NIC for the PBX and the other NIC for the remaining task, what do you think? Or its better to install the PBX on a dedicated system? Let me know your opinions! Regards... Raul On 10/12/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: You're correct, it IS 100 -- but 100%. Expressed in decimal format, this is of course 1.0 -- and as each cpu has this average, 4.0 indicates that no threads regularly wait for execution. This worldcommunitygrid you mentioned binds your cpu by design it sounds like. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Combining Flags in Dial()
Hi All, I have a quick one for you. Is there a way to mask (i.e. combine) the flags in the Dial() application? In other words, a way to do something like Dial(Zap/1,10,d|t|f) to get the effects of the three flags together in one shot? I have a need to combine the effects of the o and A flags in a dialplan. Thank you. Jeng ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [1.4] lookup_user : specified user 'asterisk' unknown? installing zaptel?
Hello 1. I don't have deep knowledge of either Linux or Asterisk, but I seem to have successfully installed 1.4 with Zaptel (for support for an OpenVox PCI FXO card) on a stock Ubuntu 7.04 Server Edition: dmesg == [ 25.990943] Zapata Telephony Interface Registered on major 196 [ 25.990948] Zaptel Version: 1.4.5.1 [ 25.990950] Zaptel Echo Canceller: MG2 [ 27.523605] ztdummy: RTC rate is 1024 [ 34.720147] Zaptel Transcoder support loaded dmesg == One thing though: When I boot up, I see the following error message on the screen (no trace dmesg or /var/log/messages): === udevd : lookup_user : specified user 'asterisk' unknown === By looking at /etc/udev/rules.d/zaptel.rules, I guess there was a step missing in the instructions I read on how to compile Zaptel, and assume I have to add a user/group for asterisk? 2. More generally, I found missing or possibly outdated information on how to go and install the Zaptel module to support PCI cards, so I'm not positive I did everything right: === 1. Compile and install Zaptel: cd zaptel-1.4.5.1 ./configure make clean make make zttool make install make config = (here, says If you have any zaptel hardware it is now recommended to edit /etc/default/zaptel or /etc/sysconfig/zaptel and set there an optimal value for the variable MODULES) 2. Compile and install Asterisk: cd /usr/src/asterisk-1.4.2 ./configure make clean make make install make samples make config cd /usr/src/asterisk-addons ./configure make clean make make install make samples 3. create /etc/zaptel.conf edit /etc/asterisk/zapata.conf edit /etc/asterisk/extensions.conf 4. install ztdummy: echo ztdummy /etc/modules modprobe ztdummy === Did I do it right? Am I missing something? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Sorry...I should have been more specific in my original reply. In 'top', you can always look at what percentage of your CPU is idle. Subtract that from 100 and you've got your load average. I should have said you get your average load percentage, rather than just average load. Mik Cheez wrote: Actually, that looks right...look at your load average... Steve Totaro wrote: I don't think that is correct. I am running worldcommunitygrid and this is what I get top - 13:18:56 up 3 days, 22:49, 1 user, load average: 4.00, 4.04, 4.02 Cpu0:0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu1:0.0%us,0.0%sy,100.0%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu2:0.0%us,0.3%sy,98.0%ni,0.0%id,0.0%wa,1.7%hi,0.0%si,0.0%st Cpu3: 0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st According to what you are saying my load average should be 100. Thanks, Steve Mik Cheez wrote: In 'top', you can always look at what percentage of your CPU is idle. Subtract that from 100 and you've got your load average. Cpu(s): 1.1% us, 0.6% sy, 0.0% ni, *98.1% id*, 0.1% wa, 0.1% hi, 0.0% si Erik Anderson wrote: On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I don't think there is a formula like cpu usage = loadavg / #cpus A loadavg of 3 says that there are 3 processes waiting to be executed. Anyway, I'll admit that a loadavg of 3 /might/ be ok. Here's a quote from this page: http://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation For systems with multiple CPUs, the number needs to be divided by the number of processors in order to get a percentage. - Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My G729 problem re-visited
Is the call being dropped or is Asterisk takng a core dump? I have core dump issues with g729 and asterisk 1.4.11, but my set up is a little different than yours... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: Friday, October 12, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My G729 problem re-visited No ideas on this one from anyone? I suppose I'm going to need to pay for some Digium support because this is a really unusual problem. Does anyone else have a gateway that speaks g729 to Asterisk and works? For whatever reason, Asterisk refuses to reply back to any of my gateways using g729. Phone (g729) to phone (g729) works. Phone (g729) to Asterisk to gateway (g711) works. But attempt g729 between Asterisk and a gateway and it fails -- every time. Asterisk responds to the gateway but never includes any codecs in the packet, unless it's g711. My configurations are shown below. Thanks, Scott On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dock-N-Talk with Asterisk, Anyone?
Hello My Aster-Friends! I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. I have a couple of these boxes that I need to make work with Asterisk, connected with Digium TDM400P card. Anyone tried it before, and how did it go? Thank you. Jeng ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
I use a mysql script to dynamically generate the page command and page about 70 phones, and I have never had a reboot problem. Sometimes there is a slight delay waiting for all the phones to join the page conference. I am using a mix of 650's, 550's, and 330's. It must only be an issue if you are using presence. Maybe I will setup presence on a couple phones and see if they reboot. It is definitely the presence. My set up is as follows: IP601 (Receptionist x7110) Buddy List turned ON (Only way to update sidcar) IP501 (32 user phones)Buddy List turned OFF (Cuts down on traffic) I have about 15 of the IP501s in the Page Group. I would have all of them, but too may more than 15 will cause the 601 to reboot EVERY time. At least with 15, it only reboots about 1 out of 15-20 pages. I have to have at least the 15 extensions in my group to get the whole building covered. When there is a page, the CLI shows Extension Changed 7118 new state InUse for Notify User 7110 Extension Changed 7134 new state InUse for Notify User 7110 Extension Changed 7117 new state InUse for Notify User 7110 Extension Changed 7125 new state InUse for Notify User 7110 Extension Changed 7123 new state InUse for Notify User 7110 Extension Changed 7114 new state InUse for Notify User 7110 Extension Changed 7137 new state InUse for Notify User 7110 ... for all 15 THIS, according to Polycom is the rush of presence information that overwhelms the 601 and causes it to reboot. They suggest moving to new firmware (which for right now I can't), but I'm going to try Joseph's GREAT suggestion to set one of the other lines on the 501 to a secondary extension within the page group... Thanks Joseph!!! I'll let you know how that works. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOw to call queue ???
Search for Queue at http://www.voip-info.org/ On 10/12/07, Walter Willis [EMAIL PROTECTED] wrote: HOw to call queues in asterisk ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 9133i autoanswer with headset
Julian Lyndon-Smith wrote: Hijacking a thread again - the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, please, please, Digium, sort the filters out!) You seem to be subscribed to the list as [EMAIL PROTECTED]. Is that the e-mail address you are using when you are trying to start a new thread. If the mailing list allowed messages from non-subscriber e-mail address the list would be destroyed by spam. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Other apps checking Day/Night
I'm fairly new to Asterisk, so please bear with me if this is silly question. I'd like to run a script on my server that would take the Call now to order banner off my website automatically when I put my phone system on night. I can handle the webserver side of things, but I don't know where to begin on the Asterisk side of things - can a simple script be run to check the value of the day/night condition, or is that value written somewhere that I could check or poll? Any help / ideas are really appreciated. Thanks in advance, Dunc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-gui
FaberK wrote: Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. I may be totally wrong but at Astricon, during the What's New at Digium (SwitchVox purchase) I asked the question of what would happen to AsteriskNow to one of the Adtran/Digium guys. There was not a real direct answer, I will try to quote as best I can from memory. He simply said It will remain opensource. I take that to mean that they will not be developing it anymore and it is up to the community to further the project. Why would Digium continue to develop a GUI for free that would compete with SwitchVox (or whatever they change the name to). Maybe I am wrong. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Other apps checking Day/Night
C. Duncan Hudson wrote: I'm fairly new to Asterisk, so please bear with me if this is silly question. I'd like to run a script on my server that would take the Call now to order banner off my website automatically when I put my phone system on night. I can handle the webserver side of things, but I don't know where to begin on the Asterisk side of things - can a simple script be run to check the value of the day/night condition, or is that value written somewhere that I could check or poll? Asterisk does not have something like day/night mode built in. But you can easily do that in the dialplan. Maybe you could use some of the examples on these pages as a starting point: http://www.voip-info.org/wiki/view/Asterisk+database http://www.the-asterisk-book.com/unstable/funktionen-db.html http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Or even http://www.google.com/search?q=asterisk+night+mode (No offense intended!) Asterisk comes with a System() or TrySystem() command (application) which will run any shell command you want. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My G729 problem re-visited
No ideas on this one from anyone? I suppose I'm going to need to pay for some Digium support because this is a really unusual problem. Does anyone else have a gateway that speaks g729 to Asterisk and works? For whatever reason, Asterisk refuses to reply back to any of my gateways using g729. Phone (g729) to phone (g729) works. Phone (g729) to Asterisk to gateway (g711) works. But attempt g729 between Asterisk and a gateway and it fails -- every time. Asterisk responds to the gateway but never includes any codecs in the packet, unless it's g711. My configurations are shown below. Thanks, Scott On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Not at all! You apparently don't realize (sic) you're talking to is just a subtle way of saying look me up! Matt was just nice enough to do the leg work :-) On 10/12/07, David Boyd [EMAIL PROTECTED] wrote: What a waste of time... dave On Fri, 2007-10-12 at 09:35 -0600, Stephen Bosch wrote: Brian West wrote: And what was the purpose of this? So that we would realize who we were talking to. :) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
On Friday 12 October 2007 01:33:44 pm Baji Panchumarti wrote: On 10/12/07, D4rk F1ber wrote: So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. [...] callwithus.com is cheap. so is diamondcard.us -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Affordable SIP Trunk for Home PBX ?
So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. I have seen mentions of Skype integration with Asterisk, but does that include say Skype IN and Skype OUT ? Or is that integration component really just for being able to contact skype users? Looking for the easiest and cheapest way to reach the PSTN, and well the options out there are plenty regarding SIP trunks, but most tend to be geared towards businesses for obvious reasons. Curious what others are using, and if anyone can make some recommendations? Not sure if this has been covered already on the list, and not sure if recommending companies are allowed, so maybe I need get replies off list? Any suggestions would be appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Actually, that looks right...look at your load average... Steve Totaro wrote: I don't think that is correct. I am running worldcommunitygrid and this is what I get top - 13:18:56 up 3 days, 22:49, 1 user, load average: 4.00, 4.04, 4.02 Cpu0:0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu1:0.0%us,0.0%sy,100.0%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st Cpu2:0.0%us,0.3%sy,98.0%ni,0.0%id,0.0%wa,1.7%hi,0.0%si,0.0%st Cpu3: 0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st According to what you are saying my load average should be 100. Thanks, Steve Mik Cheez wrote: In 'top', you can always look at what percentage of your CPU is idle. Subtract that from 100 and you've got your load average. Cpu(s): 1.1% us, 0.6% sy, 0.0% ni, *98.1% id*, 0.1% wa, 0.1% hi, 0.0% si Erik Anderson wrote: On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I don't think there is a formula like cpu usage = loadavg / #cpus A loadavg of 3 says that there are 3 processes waiting to be executed. Anyway, I'll admit that a loadavg of 3 /might/ be ok. Here's a quote from this page: http://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation For systems with multiple CPUs, the number needs to be divided by the number of processors in order to get a percentage. - Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Combining Flags in Dial()
You mean like: Dial(Zap/1,10,dtf) ? On 10/12/07, Jeng Yu [EMAIL PROTECTED] wrote: Hi All, I have a quick one for you. Is there a way to mask (i.e. combine) the flags in the Dial() application? In other words, a way to do something like Dial(Zap/1,10,d|t|f) to get the effects of the three flags together in one shot? I have a need to combine the effects of the o and A flags in a dialplan. Thank you. Jeng ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hiding extensions from app_directory
Is it possible that you are adding this, and then freePBX is overwriting your file, in effect, taking out your addition? IIRC, the newer versions of freePBX have the ability to hide users. I wouldn't bet any money on that recollection, though. On 10/12/07, Jesse Scott [EMAIL PROTECTED] wrote: Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated company directory. I found some information about adding the hidefromdir=yes option to the user's entry in voicemail.conf, but that doesn't seem to have any effect. I'm a little unclear as to whether or not that option is something native to Asterisk or if it comes from one of the external applications. If it is built in, is my version too old to have this feature? Am I totally on the wrong track and is there another way to accomplish this? Thanks, -Jesse ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone?
Worked fine for me. On 10/12/07, Jeng Yu [EMAIL PROTECTED] wrote: Hello My Aster-Friends! I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. I have a couple of these boxes that I need to make work with Asterisk, connected with Digium TDM400P card. Anyone tried it before, and how did it go? Thank you. Jeng ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My G729 problem re-visited
How do you get 11ms translation time on ulaw 729 ? we have 12ms and its dual xeons 2.6.. On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I wouldn't be too happy about a system with a loadavg of 3. The system he mentioned had 8 cores, though. So a load average of 3 is less than 50% usage. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My G729 problem re-visited
Gateway sends Asterisk an INVITE (using g729) Asterisk sends Phone an INVITE (using g711 or g729) Phone sends Asterisk an OK (using g711) Asterisk sends Gateway an OK (with no RTP choice) Gateways ends the conversation I can setup the Phone to use g729 and it will reply with an OK for g729, but the OK to the Gateway will still stay empty. Only when I enable g711 on the Gateway will this work. I have experienced this on 2 different models of gateways so far. I included my config for both the Gateway and the Phone in my original message, hoping that maybe I was configuring the Gateway wrong in Asterisk? But no one has said anything so I'm assuming its okay. Phone (g729) to Phone (g729) works Phone (anything) to Gateway (g711) works Phone (anything) to Gateway (g729) does NOT work I'm licensed for g729 (although I'm told I should not need it for pass through). And it will transcode when the phone is g729 and the gateway is g711. But for whatever reason I can't use g729 on the gateway side of the calling process? Thanks, Scott On 10/12/07, Power, Paul C. [EMAIL PROTECTED] wrote: Is the call being dropped or is Asterisk takng a core dump? I have core dump issues with g729 and asterisk 1.4.11, but my set up is a little different than yours... -Original Message- From: Scott Moseman Sent: Friday, October 12, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My G729 problem re-visited No ideas on this one from anyone? I suppose I'm going to need to pay for some Digium support because this is a really unusual problem. Does anyone else have a gateway that speaks g729 to Asterisk and works? For whatever reason, Asterisk refuses to reply back to any of my gateways using g729. Phone (g729) to phone (g729) works. Phone (g729) to Asterisk to gateway (g711) works. But attempt g729 between Asterisk and a gateway and it fails -- every time. Asterisk responds to the gateway but never includes any codecs in the packet, unless it's g711. My configurations are shown below. Thanks, Scott On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --Bandwidth and Colocation Provided by
Re: [asterisk-users] Other apps checking Day/Night
If you are going the external app method why not fire a script that updates a DB? Lacy Moore wrote: I guess it all depends on how you put your system on night mode. If you use asterisk's database, and manually put it in night mode every day, my guess is that you dial an extension which puts it in nightmode. You could include as part of this the system command to touch a file, and then on your webserver check for existance of that file. That may be opening up more holes than needed between your web server and asterisk. As Tilghman suggested, func_odbc.conf may be better. You could then set and unset by writing it directly to your store's database. On 10/12/07, *C. Duncan Hudson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm fairly new to Asterisk, so please bear with me if this is silly question. I'd like to run a script on my server that would take the Call now to order banner off my website automatically when I put my phone system on night. I can handle the webserver side of things, but I don't know where to begin on the Asterisk side of things - can a simple script be run to check the value of the day/night condition, or is that value written somewhere that I could check or poll? Any help / ideas are really appreciated. Thanks in advance, Dunc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] really sorry about this - E1 vs T1
The cards ship configured for T1. If you didn?t change the jumpers, it is set for T1. If it is set for T1 and you really want an E1 and you configure your zapata.conf as you would for an E1, you will get an error around channel 25, which tells you that you forgot to change the jumpers, and you have to call the guy on site and ask him (again) to close the jumpers on the card This has never happened to me of course, but it happens regularly to this guy that I know.. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Julian Lyndon-Smith Enviado el: viernes, 12 de octubre de 2007 0:08 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] really sorry about this - E1 vs T1 I am *really* sorry about hijacking this thread, but the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, Digium, sort the filters out!) I installed my super-duper new TE412P card today, without remembering to check the settings for T1/E1. As the server is now a hundred miles away, is there a) Any way of checking what setting is in place b) Changing that setting without having to physically remove the card and see ? Julian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My G729 problem re-visited
That was actually a VM. Here's the real server (13ms). CLI show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw- 3-12 21 3 13 - 152- alaw- 31-2 21 3 13 - 152- g729- 5444 43 5- - 174- # dmesg | grep 'Xeon(TM)' CPU0: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03 CPU1: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03 Thanks, Scott On 10/12/07, Mike Lynchfield [EMAIL PROTECTED] wrote: How do you get 11ms translation time on ulaw 729 ? we have 12ms and its dual xeons 2.6.. On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My G729 problem re-visited
How many licenses you have (show g729 should give you this info) Scott Moseman wrote: Gateway sends Asterisk an INVITE (using g729) Asterisk sends Phone an INVITE (using g711 or g729) Phone sends Asterisk an OK (using g711) Asterisk sends Gateway an OK (with no RTP choice) Gateways ends the conversation I can setup the Phone to use g729 and it will reply with an OK for g729, but the OK to the Gateway will still stay empty. Only when I enable g711 on the Gateway will this work. I have experienced this on 2 different models of gateways so far. I included my config for both the Gateway and the Phone in my original message, hoping that maybe I was configuring the Gateway wrong in Asterisk? But no one has said anything so I'm assuming its okay. Phone (g729) to Phone (g729) works Phone (anything) to Gateway (g711) works Phone (anything) to Gateway (g729) does NOT work I'm licensed for g729 (although I'm told I should not need it for pass through). And it will transcode when the phone is g729 and the gateway is g711. But for whatever reason I can't use g729 on the gateway side of the calling process? Thanks, Scott On 10/12/07, Power, Paul C. [EMAIL PROTECTED] wrote: Is the call being dropped or is Asterisk takng a core dump? I have core dump issues with g729 and asterisk 1.4.11, but my set up is a little different than yours... -Original Message- From: Scott Moseman Sent: Friday, October 12, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My G729 problem re-visited No ideas on this one from anyone? I suppose I'm going to need to pay for some Digium support because this is a really unusual problem. Does anyone else have a gateway that speaks g729 to Asterisk and works? For whatever reason, Asterisk refuses to reply back to any of my gateways using g729. Phone (g729) to phone (g729) works. Phone (g729) to Asterisk to gateway (g711) works. But attempt g729 between Asterisk and a gateway and it fails -- every time. Asterisk responds to the gateway but never includes any codecs in the packet, unless it's g711. My configurations are shown below. Thanks, Scott On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Help 60Hz Hum?
Hi, Check if you have a ground loop. If yes, this is probably the cause of this hum. Open the loop. Best Regards, Francois BERGERET France ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hiding extensions from app_directory
Doesn't look like FreePBX is nuking it. I just SSH'd in and opened the voicemail.conf directly and the entry is still in there: 305 = 1234,Users Name,[EMAIL PROTECTED],,attach=yes|saycid=yes| envelope=yes|delete=no|hidefromdir=yes I tried issuing both a 'reload' and a 'restart gracefully' to the Asterisk CLI. Is there something else I would have to do to make it take effect? Maybe I'll try disappearing the entry entirely (temporarily of course) and see if it goes away from the directory then. Is there somewhere else app_directory could be pulling entries? (I don't have the MySQL business enabled.) Thanks, -Jesse On Oct 12, 2007, at 12:58 PM, Lacy Moore wrote: Is it possible that you are adding this, and then freePBX is overwriting your file, in effect, taking out your addition? IIRC, the newer versions of freePBX have the ability to hide users. I wouldn't bet any money on that recollection, though. On 10/12/07, Jesse Scott [EMAIL PROTECTED] wrote: Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated company directory. I found some information about adding the hidefromdir=yes option to the user's entry in voicemail.conf, but that doesn't seem to have any effect. I'm a little unclear as to whether or not that option is something native to Asterisk or if it comes from one of the external applications. If it is built in, is my version too old to have this feature? Am I totally on the wrong track and is there another way to accomplish this? Thanks, -Jesse ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hiding extensions from app_directory
If you are using freepbx, I think freepbx actually simulates the app_directory, so you may have to do something in the gui to fix it, or they may not have such an option. Hope this helps. on Friday 10/12/2007 Jesse Scott([EMAIL PROTECTED]) wrote Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated company directory. I found some information about adding the hidefromdir=yes option to the user's entry in voicemail.conf, but that doesn't seem to have any effect. I'm a little unclear as to whether or not that option is something native to Asterisk or if it comes from one of the external applications. If it is built in, is my version too old to have this feature? Am I totally on the wrong track and is there another way to accomplish this? Thanks, -Jesse ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hiding extensions from app_directory
Maybe I'm wrong, but don't you have to stop/start asterisk for voicemail changes to take effect on 1.2 (like zapata) Matt On 12/10/2007, Jesse Scott [EMAIL PROTECTED] wrote: Doesn't look like FreePBX is nuking it. I just SSH'd in and opened the voicemail.conf directly and the entry is still in there: 305 = 1234,Users Name,[EMAIL PROTECTED],,attach=yes|saycid=yes|envelope=yes|delete=no|hidefromdir=yes I tried issuing both a 'reload' and a 'restart gracefully' to the Asterisk CLI. Is there something else I would have to do to make it take effect? Maybe I'll try disappearing the entry entirely (temporarily of course) and see if it goes away from the directory then. Is there somewhere else app_directory could be pulling entries? (I don't have the MySQL business enabled.) Thanks, -Jesse On Oct 12, 2007, at 12:58 PM, Lacy Moore wrote: Is it possible that you are adding this, and then freePBX is overwriting your file, in effect, taking out your addition? IIRC, the newer versions of freePBX have the ability to hide users. I wouldn't bet any money on that recollection, though. On 10/12/07, Jesse Scott [EMAIL PROTECTED] wrote: Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated company directory. I found some information about adding the hidefromdir=yes option to the user's entry in voicemail.conf, but that doesn't seem to have any effect. I'm a little unclear as to whether or not that option is something native to Asterisk or if it comes from one of the external applications. If it is built in, is my version too old to have this feature? Am I totally on the wrong track and is there another way to accomplish this? Thanks, -Jesse ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
I agree with the suggestions of Teliax, and I also use http://vitelity.net/ -- you can rent 800# DIDs from them for $0.50/month plus minutes, and I think their US minutes are like $0.02 or so Moj D4rk F1ber wrote: So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. I have seen mentions of Skype integration with Asterisk, but does that include say Skype IN and Skype OUT ? Or is that integration component really just for being able to contact skype users? Looking for the easiest and cheapest way to reach the PSTN, and well the options out there are plenty regarding SIP trunks, but most tend to be geared towards businesses for obvious reasons. Curious what others are using, and if anyone can make some recommendations? Not sure if this has been covered already on the list, and not sure if recommending companies are allowed, so maybe I need get replies off list? Any suggestions would be appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 Syntax Highlighting
Hi, I am looking for a syntax highlighter for AEL2. Google is not helping, so I thought you guys could help me. I found this vim syntax highlighter for AEL but it doesn't help if you want to code in AEL2: http://vim.sourceforge.net/scripts/script.php?script_id=1900 Cheers, PLL. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone?
How about the Intellitouch XLink? www.xlinkgateway.com http://www.xlinkgateway.com/ --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore Sent: Friday, October 12, 2007 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone? Worked fine for me. On 10/12/07, Jeng Yu [EMAIL PROTECTED] wrote: Hello My Aster-Friends! I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. I have a couple of these boxes that I need to make work with Asterisk, connected with Digium TDM400P card. Anyone tried it before, and how did it go? Thank you. Jeng ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ http://uk.answers.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hiding extensions from app_directory
On Friday 12 October 2007 17:14:39 Matt Gibson wrote: Maybe I'm wrong, but don't you have to stop/start asterisk for voicemail changes to take effect on 1.2 (like zapata) Nope, reload. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Other apps checking Day/Night
On Friday 12 October 2007 15:47:41 Anthony Francis wrote: Lacy Moore wrote: I guess it all depends on how you put your system on night mode. If you use asterisk's database, and manually put it in night mode every day, my guess is that you dial an extension which puts it in nightmode. You could include as part of this the system command to touch a file, and then on your webserver check for existance of that file. That may be opening up more holes than needed between your web server and asterisk. As Tilghman suggested, func_odbc.conf may be better. You could then set and unset by writing it directly to your store's database. If you are going the external app method why not fire a script that updates a DB? That is pretty much exactly how func_odbc can be used (except without the hit of firing up a whole new process for the simple task of flicking a field in a database to on or off). -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone?
How about the Intellitouch XLink? www.xlinkgateway.com http://www.xlinkgateway.com/ --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore Sent: Friday, October 12, 2007 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone? Worked fine for me. On 10/12/07, Jeng Yu [EMAIL PROTECTED] wrote: Hello My Aster-Friends! I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. I have a couple of these boxes that I need to make work with Asterisk, connected with Digium TDM400P card. Anyone tried it before, and how did it go? Thank you. Jeng ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ http://uk.answers.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hiding extensions from app_directory
Ok, I commented out the entry from voicemail.conf and that person is gone from the directory. However, uncommented with hidefromdir=yes in the options, they show up. I just upgraded FreePBX to 2.3.0 and I couldn't find any additional options for hiding the user from the directory. This user may not actually care about having a voicemail box, so I might just solve the problem that way. I'd love to figure out why the hidefromdir option doesn't work though. Thanks, -Jesse On Oct 12, 2007, at 1:53 PM, Jesse Scott wrote: Doesn't look like FreePBX is nuking it. I just SSH'd in and opened the voicemail.conf directly and the entry is still in there: 305 = 1234,Users Name,[EMAIL PROTECTED],,attach=yes|saycid=yes| envelope=yes|delete=no|hidefromdir=yes I tried issuing both a 'reload' and a 'restart gracefully' to the Asterisk CLI. Is there something else I would have to do to make it take effect? Maybe I'll try disappearing the entry entirely (temporarily of course) and see if it goes away from the directory then. Is there somewhere else app_directory could be pulling entries? (I don't have the MySQL business enabled.) Thanks, -Jesse On Oct 12, 2007, at 12:58 PM, Lacy Moore wrote: Is it possible that you are adding this, and then freePBX is overwriting your file, in effect, taking out your addition? IIRC, the newer versions of freePBX have the ability to hide users. I wouldn't bet any money on that recollection, though. On 10/12/07, Jesse Scott [EMAIL PROTECTED] wrote: Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated company directory. I found some information about adding the hidefromdir=yes option to the user's entry in voicemail.conf, but that doesn't seem to have any effect. I'm a little unclear as to whether or not that option is something native to Asterisk or if it comes from one of the external applications. If it is built in, is my version too old to have this feature? Am I totally on the wrong track and is there another way to accomplish this? Thanks, -Jesse ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
I've used Vitelity.net for several years and am reasonably happy with them. You can get DID's from any area code you want for $1.49 a month with per minute rates between $0.011 and $0.0149. I also use Nufone.net for outbound. Their DID's cost $5.00 a month so I don't have one :) Nufone let's you set the caller id number on outbound calls which is useful for accessing your cell phone voice mail or directing return calls to your office or cell. You can use IAX or SIP with either provider. On Fri, 12 Oct 2007, Mojo with Horan Company, LLC wrote: I agree with the suggestions of Teliax, and I also use http://vitelity.net/ -- you can rent 800# DIDs from them for $0.50/month plus minutes, and I think their US minutes are like $0.02 or so Moj D4rk F1ber wrote: So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. I have seen mentions of Skype integration with Asterisk, but does that include say Skype IN and Skype OUT ? Or is that integration component really just for being able to contact skype users? Looking for the easiest and cheapest way to reach the PSTN, and well the options out there are plenty regarding SIP trunks, but most tend to be geared towards businesses for obvious reasons. Curious what others are using, and if anyone can make some recommendations? Not sure if this has been covered already on the list, and not sure if recommending companies are allowed, so maybe I need get replies off list? Any suggestions would be appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
