Re: [asterisk-users] Recomender Server specs for 250 con-current calls
Transcoding plays a major role you can find some info @ voip-info ... http://www.voip-info.org/wiki-Asterisk+dimensioning ~Vamsi On 6/6/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I looking to implement asterisk solution for 2000 sip registrations and expecting con-current call about 250. Can some one provide me guide line that what kind of server will fullfil the requirment. what is the Processor, RAM ??? -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gigabit SIP Phones
Also, are there any IP phones that run apps other than telephony, like video, which could use more than 100Mb, even if just while switching streams? Video of 100Mb/s? ;-) HDTV doesn't consume more than 20Mb/s, Gige is an overkill for IP Phone. Though it is used for switching, I assume it is a 1 in 100 use. Thanks, ~Vamsi On 6/13/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: On Tue, 2007-06-12 at 16:44 -0700, [EMAIL PROTECTED] wrote: Date: Tue, 12 Jun 2007 17:56:34 -0500 From: Darrick Hartman [EMAIL PROTECTED] Subject: Re: [asterisk-users] Gigabit SIP Phones To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=UTF-8; format=flowed Andrew Latham wrote: Oliver The thing you missed about Gigabit enabled SIP hardphones is the demand for them. Not true. I can think of several places where I have or would like to install phones where the end users currently have Gigabit ethernet feeds to workstations. Specifically if you are using a high-overhead system like Quickbooks Point of Sale and need a phone at the same location, the end users will notice a significant performance hit by dropping them down to 100Mbit. It's not so much that the phone needs Gig, it's that the pass thru connection needs gig. And if you've got GigE installed, not 10/100Mb, and your LAN doesn't have a switch that can handle a phone's lower bitrate without bringing down the whole LAN's rate. Also, are there any IP phones that run apps other than telephony, like video, which could use more than 100Mb, even if just while switching streams? Andrew On 6/12/07, Olivier [EMAIL PROTECTED] wrote: Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darrick Hartman -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and CCM 5.x SIP trunk
Hi Greg, Narrowed the problem ot be that of codec mismatch ;-) Damn CCM, doesn't provide proper debugs. I have another query with CCM and Asterisk integration. In CCM cluster Phones register to 1st CCM and they fallback to 2nd incase the first fails and 3rd CCM incase even 2nd fails. How can asterisk know on which CCM subscriber the phone is registered to? How to make sure that Asterisk tries all avaiable CCMs to check where the phone is registered. Is there any better way to handle this? Thanks, ~Vamsi On 5/23/07, Greg Oliver [EMAIL PROTECTED] wrote: On Wed, 2007-05-23 at 19:53 +0530, Vamsi Pottangi wrote: Hi, I was able to work out SIP trunk between Asterisk and CCM 4.x without any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk. Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not replying. For the same reason Asterisk is marking it as UNREACHABLE. Anybody got Asterisk and CCM 5.x intergation working. How can I fix the problem which I'm facing with CCM 5.x? Thanks, ~Vamsi You need to make a new SIP security profile for it to work with * Under System-SecurityProfile-SIP Security Profile - you can see the trunk security settings - need to be unsecure and UDP Under Device-Trunk You will set it up using the security profile. Under Device-DeviceSettings-SIP Profile You can set all the settings. Let me know if you need more info or screenshots. Email me offline if you need some. -Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and CCM 5.x SIP trunk
Hi, I was able to work out SIP trunk between Asterisk and CCM 4.x without any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk. Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not replying. For the same reason Asterisk is marking it as UNREACHABLE. Anybody got Asterisk and CCM 5.x intergation working. How can I fix the problem which I'm facing with CCM 5.x? Thanks, ~Vamsi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [Dundi] Dial Plan for Multi-Location Support Queu
Use IAX trunk routing between them.
exten = _88.,1,Dial(IAX2/USERID:[EMAIL PROTECTED]/${EXTEN:2})
Thanks,
~Vamsi
On 5/7/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Can anyone help on this.
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote:
Hi,
I am in the process of planning a dial plan, In regards to the
requirement, I am confused how to go about the dial plan.
The scenario is like below.
BRANCH - A - (COMPANY)
Line 1 -- Extension 239
Line 2 -- Extension 8239
BRANCH - B - (COMPANY)
Line 1 -- Extension 239
Line 2 -- Extension 8239
Now what I need is that if a user in Branch - A wants to dial Branch - B, he
just needs to use 88xxx(extension of Branch - B)
Similarly, if a user in Branch - B wants to dial Branch - A, he just needs
to use 89xxx(extension of Branch - A)
In this regards, I am not sure how do I achieve inter brach connection using
asterisk to fit my 88 89 prefix dial plan for multi-location.
More over, said that, we will have a support Queue in Branch - A(extension
700), users from Branch - B should be able to join the Queue(extension
700) to accept support calls vice-versa, I dont know how this is possible
what would my dial plans be.
It would be much appreciated if someone can help me resolve this dial plan
support issue.
Thannks,
Deepak
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Re: [asterisk-users] eagi-sphinx-test how and why
http://turnkey-solution.com/asterisk-sphinx.html~VamsiOn 10/16/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: HiI see eagi-sphinx-test in agi-bin, anyone know how is it supposed to beused and what version of sphinx.Any help will be appraciatedmawali___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requirements for Asterisk SER integratio
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER~Vamsi On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi Friends,I would like have Asterisk and SER implementation. I have lot of experience with Asterisk. Can anybody tell me what I have to install to integrate SER with Asterisk? Looking forwrad to your response. Thank you. Regards,Chandra.Get your email and more, right on the new Yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSL router with integrated SIP proxy?
Check for Dlink, I vaguely remember Dlink's router with inbuilt asterisk.~VamsiOn 9/22/06, Brian Candler [EMAIL PROTECTED] wrote: Does anyone here know of an ADSL router with integrated SIP proxy?My requirement is for an expandable branch office. I want to be able to sitmultiple ATAs and/or VoIP phones on the LAN, and for them all to work to a remote soft switch.--+---+---+---+--| | | | ATA ATA ATA...DSL - DSL line| | |router _=_ _=_ _=_ /O\ /O\ /O\The DSL service has only a single IP address and hence each ATA is on aprivate IP address, with NAT on the router. The branches aren't going to be VPNed together.My understanding is that in this architecture, I need a SIP proxy somewhereon the LAN, right? (Or could I get away just with multiple UDP portforwardings, with separate SIP and RTP ports for each phone?) I see there are some solutions listed athttp://www.voip-info.org/wiki/view/VOIP+Routersunder Small NAT Routers with SIP Proxy, but as far as I can tell none of them have integrated ADSL modems.Having said that, following links to Intertex suggests that their IX68 rangemay be just what I'm looking for.Anybody have any experience with this box, or can suggest alternatives I should be looking at?Or should I just install a two box solution, a bog-standard DSL router and aLinksys running OpenWrt + OpenSER, and port-forward the SIP traffic to theOpenSER box?All experience and advice welcomed :-) Many thanks,Brian Candler.P.S. Sorry if this is off-topic because I didn't mention Asterisk. Let'sjust assume that the remote soft-switch that these telephones are going totalk to is Asterisk, OK? :-) ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silence Call {very very urgent plz}
Hi Abdul,More information about your setup would be helpfultoresolveyourproblem. RBT could be a locally generated tone at the endpoint as a result of SIP provisonal messages. Are you sure that this RBT is played from the remotes Telecom switch?AnyNATisbetween?HowareyouterminatingontothePSTN,PCIcards?gateway? ~VamsiOn 9/14/06, Abdul [EMAIL PROTECTED] wrote: Hi all,I was running my asterisk from one year. today i upgrade with 1.2.12.1 once the caller is dialing the destination number caller can hear well real RBT from telecom and once called party pickup the phone the call became silence no voice both side.Please try to help me ASAP 150 calls waiting couse of this issue. Regards,Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invite someone to Conference
Instaling web-meetme is pretty easy ... did you try to install and use it? Or checking for help even before trying? If you had already tried .. let us know where you got stuck. You could find the download and installation instructions here http://areski.net/Web-MeetMe/about.php ~VamsiOn 7/7/06, Rizwan Hisham [EMAIL PROTECTED] wrote: i cant find any help about installing the web meetme tool. on www.voip-info.org a link is given for installation instructions about web meetme but i thinks its dead. http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html i cant find anyother source of info about this on google..helpOn 7/7/06, Rizwan Hisham [EMAIL PROTECTED] wrote:Thanx alot for the tips.i'll try then out and let u know about the result On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote: Yep, forgot 'bout that. Or you could use web-meetme, it has this feature. On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote: Snip, snip. Chop Chop. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- RegardsRizwan HishamSoftware Engineer -- RegardsRizwan HishamSoftware Engineer ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Coice recognition IVR?
True, installing and running Sphinx needs some real patience and time. It's not that friendly to get it working. Even after installation, what I found out is that it doesn't recognize basic words like yes, no, etc that effectively need to explore it more when time permits. I feel Sphinx needs lot's of development effort to get it going widely hope that day is very soon. ~VamsiOn 4/3/06, Cosmin Prund [EMAIL PROTECTED] wrote: Unfortunately I already gave up myself!At first glance setting up Sphinx looks like a real pain and, while mythreshold for such pain would definitively allow me to work with it, myavailable time can't support this. And I am sorry, because it would look really nice talking to your box, asking it to reboot or something. Verystar-trek -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Joshua Colp Sent: Monday, April 03, 2006 7:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Coice recognition IVR? Cosmin Prund wrote: Hello everyone. Is it possible to do some very basic voice recognition from within Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I want to dial from my mobile phone. Dialing digits on my mobile phone while driving is not all that safe... Thanks for any input, Cosmin Prund___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This has been discussed a lot before, and people usually end up giving Sphinx a go and seeing how it is. If you search the mailing list archives you might find something useful. Joshua Colp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP in India
Nope, convergence with public phone network is not yet legalized in India. You could use VoIP for your local network. This is how call centers in India work. They use VoIP to connect to outside world but not to India PSTN. Hope this is clear. ~VamsiOn 1/26/06, Code Lover [EMAIL PROTECTED] wrote: Hi all,I would like to set an VoIP Gateway in India. Could any one tell me,is VoIP is legal in India?How I can obtain the license to start my VoIP gateway?--Thank You,Code Lover___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Conference-reg
Have you checked your zaptel interface. If you don't have hardware then use ztdummy. I guess you would have. ~Vamsi On 11/6/05, nr k [EMAIL PROTECTED] wrote: Hi allI am having Asterisk 1.0.9. now i configured themeetme conference with conference number 1234 and alsoi add the extension 1234 in extension.conf.if i callto 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices.[room]; Usage is conf = confno[,pin]conf = 1234extension.conf[default]exten = 1234,1,Meetme(1234)pls do the needful..regards ramakrishnan.n__Yahoo! Mail - PC Magazine Editors' Choice 2005http://mail.yahoo.com___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme: Sending DTMF to other users in a conference
Hi, I would like to know the possibility of sending DTMF to other users in a meetme. I'm looking at inviting a participant from within the conference, here the participant is another conference bridge. So we need to send PIN to this conference bridge. How can I bypass the IVR detect menu and send DTMF to the other participants. Does careful_write in case of frametype is AST_FRAME_DTMF will work ? Final aim here is to bridge asterisk's meetme and another conference bridge. This I need to do from within the conference. Another usage, say if we are inviting some person from within the conference, if this lands in the company's IVr then there should be some way to send DTMF to that IVR to reach that person. Anybody came across such a scenario ? Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] lucent TNT h323/sip config?
Asterisk cannot act as a H.323 gatekeeper for TNT to register. We need a gatekeeper like Lucent MVAM for TNT to register to. Asterisk will register to MVAM as a gateway. ~VamsiOn 10/31/05, Armand Sulter [EMAIL PROTECTED] wrote: Does anyone have an example of a lucentTNT h323 config to work with asterisk ?I'd like to use sip but it's not supported in theTAOS we have, if anyone has TAOS 10.x or laterthat would be awsome as well, we have the examples for a sip config.thx- Armand___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] maximum concurrent ZAP channels .... max conf ports ...
Hi All, Is it possible to go beyond 250 concurrent ZAP channels with some tweaking or workaround ? Meetme uses zap channels, so we could have a max of 250 conference ports. Is it possible to higher this ? An Asterisk system can only handle a max. of 250 concurrent ZAP channels. This is due to the design limit (255) within the ZAP channel driver. Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP created extensions busy when dialed.
Hi All, I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions. But when I installed AMP and created new extensions, I'm not able to call those extensions. I get the message that the extension is busy and it is forwarded to voicemail. What am I missing here? The workaround I found is by modifying the extensions_additional.conf entries created by AMP. Original exten = ,1,Macro( Changed it to exten = ,1,Dial(IAX2/) How can I fix this problem ? I tried both AMP1.0.006 and AMP1.0.009 Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not able to access asterisk from internet via ip-forwarding
Hi All, Asterisk is Up and running. I want to access this PC over internet. So I registered at www.dyndns.com for dynamic IP-address mapping. I had enabled the IP-forwarding (HTTP port 80) on the DSL Modem to point to the PC running asterisk. When I access from internet, I see the configuration page of modem rather than the AMP page of Asterisk PC. How can I get the Ip-forwarding working ? Also, I guess I need to get the IP-forwarding working in order to register extensions from internet. I tried both Zyxel 660 and huawei MT 880 modems. In Zyxel, I configured Ip-forwarding under NAT-SUA Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Emergency Asterisk Guru Help needed EMERGENCY
outgoing, monitor and voicemail/default are good enough. Here you would loose all the voicemail stuff. So you need to re-record busy and unavialble messages for each mailbox user (if at all you had them before). ~Vamsi On 7/7/05, Jeffrey Starin [EMAIL PROTECTED] wrote: 911 Help! I accidentially deleted all directories under /var/spool/asterisk I did use the backup facility not too long ago but cannot find the process for restore. However, I don't believe a full restore is needed -- I just need to know the names of the directories under /var/spool/asterisk and re-create them (I hope!). Can some kind soul give me some direction or tell me the directory structure under /var/spool/asterisk? Thanks, B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk failover solution
Use Realtime and host the database on a separate machine. This should solve most of your problems. ~Vamsi On 6/30/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones are dead. Does anyone have Ideas of how to overcome this. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Conferencing and Meetme
Never uses AAH, but check for two things if 8200 is mentioned meetme.conf and if you have ztdummy initialised ... ~Vamsi On 6/28/05, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi, I ve installed recently AAH 1.1 And I was wondering on how to use this conferencing feature ? I have created extension 200. and when I try to call 8200, it says that this is not a valid conference number. Is there something specific to do ? Also, when entering MeetMe console, I cannot see anything. Is that allright ? meaning that if I have not started any conferencing, then, I shall see nothing in MeetMe :o) Thanks for any help ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Conferencing and Meetme
Sorry, I meant I had never used AAH. On 6/28/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Never uses AAH, but check for two things if 8200 is mentioned meetme.conf and if you have ztdummy initialised ... ~Vamsi On 6/28/05, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi, I ve installed recently AAH 1.1 And I was wondering on how to use this conferencing feature ? I have created extension 200. and when I try to call 8200, it says that this is not a valid conference number. Is there something specific to do ? Also, when entering MeetMe console, I cannot see anything. Is that allright ? meaning that if I have not started any conferencing, then, I shall see nothing in MeetMe :o) Thanks for any help ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration
The below worked for me to integrate with CCM. pwlib-v1_6_6 openh323-v1_13_5 asterisk-oh323-0.7.1 The only change I made was -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- Steps to follow: --- To enable H323 for inter-op with Cisco Call Manager (H.323) cp pwlib-v1_6_6-src.tar.gz openh323-v1_13_5-src.tar.gz asterisk-oh323-0.7.1.tar.gz /usr/src/ cd /usr/src tar zxf pwlib-v1_6_6-src.tar.gz tar zxf openh323-v1_13_5-src.tar.gz tar zxf asterisk-oh323-0.7.1.tar.gz - Set Environment variables PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib cd /usr/src/pwlib ./configure make opt cd /usr/src/openh323 ./configure -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- make opt cd /usr/src/asterisk-oh323-0.7.1 Edit makefile and set the paths/options according to your system. Type make to build the oh323wrap library and the ASTERISK OH323 channel driver. - If compiling fails, then change the makefile to reflect the below CPPFLAGS=$(OPENH323FLAGS) -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -I/usr/src/pwlib/include -DPTRACING -I/usr/src/openh323/include -DHAS_OSS -Wall -x c++ -Os --- Type make install to install the binaries. This will also install a sample configuration file, if there isn't one. Next, add to your LD_LIBRARY_PATH the path where the oh323wrap library was installed (or edit your /etc/ld.so.conf file, add the library path, and run ldconfig). Thanks, ~Vamsi On 6/26/05, Walid Azab [EMAIL PROTECTED] wrote: I have previously tried the Asterisk/OH323/PWLIB/GNUGK combination and had problems compiling OH323. I will try again from a clean installation. On the other hand, can you send me any useful links or guides that you already used. This can make our trial and error efforts much less. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Oliver Sent: Sunday, June 26, 2005 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration We have successfully connect * .9x 1.0.x with CCM 3.3.x and up using both gatekeeper and no gatekeeper.. Using SIP usually with CCM 4.0 and up.. With CCM 3.3.x, there is a limitation where the gateway H323 in your case cannot use IP addresses, so the Asterisk box has to have correct DNS entries to resolbve your asterisk ox.. Then just use regular route patterns and direct it to asterisk.. That works well. You may also want to make sure your compatibility matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities cause more issues than I care to talk about. The GNUGk web site has the best matrix to follow.. Thanks, GReg On Sat, 2005-06-25 at 10:39 -0500, [EMAIL PROTECTED] wrote: Use a gatekeeper and have both boxes register with the gatekeeper. That way you can specify what numbers go where. From everything I have tested, * will NOT register with CCM. When I added in a gatekeeper and had both sides register with it, everything works. Walid Azab wrote: Hello, I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this subject. So please if anyone has any useful information or is sure that this can now work please send me whatever you have on that. I simply want Asterisk users to get their dial tones through CCM. Thanks and I appreciate your assistance. Walid ___ Asterisk-Users
[Asterisk-Users] asterisk-oh323: Max simultaneous calls ?
Hi All, There is a parameter simultaneousMax=10 in oh323.conf. Had anybody tried out what is the maximum value that can be achieved ? What is the maximum number of simultaneous h323 calls can the oh323 driver can handle. I tried to get it only till 30 to 40 simultaneous calls. Anybody achieved better figures than this ? or have any idea how the oh323 can be tuned to get better values ? Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lucent TNT ASTERISK
Did you try out oh323 ? It worked for me. Please follow the steps required to get oh323 worki On 5/16/05, list [EMAIL PROTECTED] wrote: Anybody using asterisk to talk to a lucent tnt gatekeeper via h323? Any suggestions or recommendations about how I can get this working? Any config examples? thanks, jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lucent TNT ASTERISK
Did you try out oh323 ? It worked for me. Please follow the steps required to get oh323 working. On Lucent gateeeper, add asterisk as a H323 gateway. Cheers, ~Vamsi On 5/16/05, list [EMAIL PROTECTED] wrote: Anybody using asterisk to talk to a lucent tnt gatekeeper via h323? Any suggestions or recommendations about how I can get this working? Any config examples? thanks, jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on SMP machine with ztdummy working ?
That is nice to hear. Congrats. Wondering who could help me out with this unique zap channel problem of mine. Thanks, ~Vamsi On 5/7/05, Tim Connolly [EMAIL PROTECTED] wrote: I've got three dual Xeon's running Redhat Enterprise 4 with 2.6.9 and CVS-HEAD from about a month ago. I didn't have any problems whatsoever, other than the problems I blame on being reluctant to RTFM. No problems with the SMP side whatsoever. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vamsi Pottangi Sent: Friday, May 06, 2005 10:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk on SMP machine with ztdummy working ? Hi All, Was any Asterisk installation on SMP machine successful. Were you able to get ztdummt working on it. If so please let me know which linux favour you are using and any important steps to follow. I have a Dell Power edge 2800 and wanted to try asterisk on it and also use meetme. Which Linux flavour should I go for and the timing source. I don't have a zaptel interface so wanted to use ztdummy. Please guide me. I tried with FC3 as mentioned in below mail but loading of zap module fails saying resource busy. Thanks, ~Vamsi -- Forwarded message -- From: Vamsi Pottangi [EMAIL PROTECTED] Date: May 5, 2005 7:51 PM Subject: chan_zap.so: load_module fails: Fedora Core 3: SMP To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi, I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this something to do with SMP ? I compile zaptel using the link to smp source code only. lrwxrwxrwx 1 root root 34 May 5 21:22 linux-2.6 - /lib/modules/2.6.9-1.667smp/source May 5 21:43:55 VERBOSE[12931]: [chan_zap.so]May 5 21:43:55 VERBOSE[12931]: [chan_zap.so] = (Zapata Telephony) May 5 21:43:55 DEBUG[12931]: Parsing /etc/asterisk/zapata.conf May 5 21:43:55 WARNING[12931]: Unable to specify channel 1: Device or resource busy May 5 21:43:55 ERROR[12931]: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 May 5 21:43:55 ERROR[12931]: Unable to register channel '1' May 5 21:43:55 WARNING[12931]: chan_zap.so: load_module failed, returning -1 May 5 21:43:55 DEBUG[12931]: Unregistering channel type 'Zap' May 5 21:43:55 VERBOSE[12931]: == Unregistered channel type 'Zap' May 5 21:43:55 WARNING[12931]: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# uname -a Linux noname11 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] ~]# [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw--- 1 asterisk asterisk 196, 254 May 5 21:31 channel crw--- 1 asterisk asterisk 196, 0 May 5 21:31 ctl crw--- 1 asterisk asterisk 196, 255 May 5 21:31 pseudo crw--- 1 asterisk asterisk 196, 253 May 5 21:31 timer [EMAIL PROTECTED] ~]# Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_zap.so: load_module fails: Fedora Core 3: SMP
Bulls Eye !!! Thanks for that Tony ! It worked. Initially I thought that default conf file would work like my previous installations. Thanks, ~Vamsi On 5/7/05, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Vamsi Pottangi [EMAIL PROTECTED] wrote: I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this something to do with SMP ? I compile zaptel using the link to smp source code only. lrwxrwxrwx 1 root root 34 May 5 21:22 linux-2.6 - /lib/modules/2.6.9-1.667smp/source May 5 21:43:55 VERBOSE[12931]: [chan_zap.so]May 5 21:43:55 VERBOSE[12931]: [chan_zap.so] = (Zapata Telephony) May 5 21:43:55 DEBUG[12931]: Parsing /etc/asterisk/zapata.conf May 5 21:43:55 WARNING[12931]: Unable to specify channel 1: Device or resource busy May 5 21:43:55 ERROR[12931]: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 May 5 21:43:55 ERROR[12931]: Unable to register channel '1' May 5 21:43:55 WARNING[12931]: chan_zap.so: load_module failed, returning -1 May 5 21:43:55 DEBUG[12931]: Unregistering channel type 'Zap' May 5 21:43:55 VERBOSE[12931]: == Unregistered channel type 'Zap' May 5 21:43:55 WARNING[12931]: Loading module chan_zap.so failed! You need to edit zapata.conf. It evidently has a channel = 1 directive somewhere, but if you're using ztdummy I assume you have no zaptel hardware. There should only be channel directives for hardware that exists, and if you DO have zaptel hardware, you don't need ztdummy. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on SMP machine with ztdummy working ?
Hi All, Was any Asterisk installation on SMP machine successful. Were you able to get ztdummt working on it. If so please let me know which linux favour you are using and any important steps to follow. I have a Dell Power edge 2800 and wanted to try asterisk on it and also use meetme. Which Linux flavour should I go for and the timing source. I don't have a zaptel interface so wanted to use ztdummy. Please guide me. I tried with FC3 as mentioned in below mail but loading of zap module fails saying resource busy. Thanks, ~Vamsi -- Forwarded message -- From: Vamsi Pottangi [EMAIL PROTECTED] Date: May 5, 2005 7:51 PM Subject: chan_zap.so: load_module fails: Fedora Core 3: SMP To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi, I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this something to do with SMP ? I compile zaptel using the link to smp source code only. lrwxrwxrwx 1 root root 34 May 5 21:22 linux-2.6 - /lib/modules/2.6.9-1.667smp/source May 5 21:43:55 VERBOSE[12931]: [chan_zap.so]May 5 21:43:55 VERBOSE[12931]: [chan_zap.so] = (Zapata Telephony) May 5 21:43:55 DEBUG[12931]: Parsing /etc/asterisk/zapata.conf May 5 21:43:55 WARNING[12931]: Unable to specify channel 1: Device or resource busy May 5 21:43:55 ERROR[12931]: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 May 5 21:43:55 ERROR[12931]: Unable to register channel '1' May 5 21:43:55 WARNING[12931]: chan_zap.so: load_module failed, returning -1 May 5 21:43:55 DEBUG[12931]: Unregistering channel type 'Zap' May 5 21:43:55 VERBOSE[12931]: == Unregistered channel type 'Zap' May 5 21:43:55 WARNING[12931]: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# uname -a Linux noname11 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] ~]# [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw--- 1 asterisk asterisk 196, 254 May 5 21:31 channel crw--- 1 asterisk asterisk 196, 0 May 5 21:31 ctl crw--- 1 asterisk asterisk 196, 255 May 5 21:31 pseudo crw--- 1 asterisk asterisk 196, 253 May 5 21:31 timer [EMAIL PROTECTED] ~]# Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mpg123 zombie processes ...
Hi All, I had noticed that MOH's mpg123 processes are not killed when asterisk is killed. Eventually after many restarts I see many of these zombie processes eating up CPU. Any Idea how could I make asterisk to clean up these properly. Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_zap.so: load_module fails: Fedora Core 3: SMP
Hi, I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this something to do with SMP ? I compile zaptel using the link to smp source code only. lrwxrwxrwx 1 root root 34 May 5 21:22 linux-2.6 - /lib/modules/2.6.9-1.667smp/source May 5 21:43:55 VERBOSE[12931]: [chan_zap.so]May 5 21:43:55 VERBOSE[12931]: [chan_zap.so] = (Zapata Telephony) May 5 21:43:55 DEBUG[12931]: Parsing /etc/asterisk/zapata.conf May 5 21:43:55 WARNING[12931]: Unable to specify channel 1: Device or resource busy May 5 21:43:55 ERROR[12931]: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 May 5 21:43:55 ERROR[12931]: Unable to register channel '1' May 5 21:43:55 WARNING[12931]: chan_zap.so: load_module failed, returning -1 May 5 21:43:55 DEBUG[12931]: Unregistering channel type 'Zap' May 5 21:43:55 VERBOSE[12931]: == Unregistered channel type 'Zap' May 5 21:43:55 WARNING[12931]: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# uname -a Linux noname11 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] ~]# [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw--- 1 asterisk asterisk 196, 254 May 5 21:31 channel crw--- 1 asterisk asterisk 196, 0 May 5 21:31 ctl crw--- 1 asterisk asterisk 196, 255 May 5 21:31 pseudo crw--- 1 asterisk asterisk 196, 253 May 5 21:31 timer [EMAIL PROTECTED] ~]# Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference solution for 100+ users
Since all would be listening, it's good to have a web streaming. Users could just use the media players rather than going for new softphones. This mailing list is not the appropriate one to discuss the above. But if you want to consider the asterisk solution, we can very well have the audience to participate in conference say for QA session. You could could use IAX2 clients behind the firewalls. ~Vamsi On 4/19/05, Sergio Veltri [EMAIL PROTECTED] wrote: Hi List, I am looking for some advice. I need to come up with a conference solution that will allow users to join mainly to listen to a guy talk about a product for an hour. My main concern is the client side. I need people from within firewalls to be able to join the conference with speakers built-in their laptops or computers. All I know is that Skype works in most of the customers this guy will be addressing. I am considering the following options: 1-Skype-like softphone for *. is there any? 2-Just do audio streaming and have the customers use windows media player. (I dont know how to do this) 3-Use some kind of Softphone with VPN... 4- Do Softphone---Port 80--- SER---Asterisk w/meetme. Whatever solution I come up with MUST allow anybody to listen in assuming nobody can change firewalls. Any one has already done this? Any feedback will be much appreciated. Thanks, Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe
MeetMe is straight forward. Follow the steps for ztdummy and there you go conferencing Check out www.voip-info.org for more info Cheers, ~Vamsi On 4/18/05, Matt Schwartz [EMAIL PROTECTED] wrote: Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe application. I don't think it installed with the standard 'make install' command. If not, how do I accomplish this? Thanks, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?
It would be easier if you could get send us your sip.conf entry and confiuration made in x-lite Also, please let us know where exactly the problem is. Is it while registering the x-lite or during the call and the exact error messages. Cheers, ~Vamsi On 4/18/05, Abraham WEI [EMAIL PROTECTED] wrote: I just want to make the simplest call in which an X-Lite calls another X-Lite via asterisk. Unfortunately I failed time and time again. If someone is kind enough to show me sample config files by which asterisk works well, it will help me a lot. Best regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme2 and meetme
Yes, you could use MeetMe2 and MeetMe simultaneously. ~Vamsi On Tue, 15 Mar 2005 08:01:28 +0900, Kuniyoshi Murata [EMAIL PROTECTED] wrote: Hi, As I read http://www.areski.net/asterisk-meetme/about.php?s=0, meetme2 seems attractive to me. My question here is... Can meetme2 and existing meetme can coexist and can be used whichever I want when I want to have a conference? Thanks for your input Kuni -- Kuniyoshi Murata.iChat/AIM:macwebcaster English-Japanese Interpreter mailto:[EMAIL PROTECTED] Macintosh Webcast Specialisthttp://www.macwebcaster.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mysql and SIP real time configuration...
You need to mention the DB deatils in sip.conf file .. please check the WiKi pages [general] dbname= dbhost= dbuser= dbpass= Also you need to mention the DB details in res_mysql.conf ~Vamsi On Mon, 14 Feb 2005 06:13:10 +, Jan Aabyevester [EMAIL PROTECTED] wrote: Dear all, My question is probably very trivial but I'll try my luck anyway. Currently I have my asterisk PBX up and running but now I would like to perform real time SIP configuration. I have been reading the article Asterisk RealTime SIP, and been creating a mysql database called sipusers, also I have created a Table called sip_buddies. My database is running on a machine called dbserver. The asterisk PBX is running on a machine called PBX which is running Fedora Core 1. So in my scenario there are two machines the actual PBX and a Database server. Next I started the apache web server service on PBX and via a browser loaded a PHP script which connect to the sipusers db on dbserver, just to make sure that I can connect to the db called sipusers from PBX to dbserver. So far so good I can connect to the database. Next I configured the file Extconfig.conf with the following lines: Sipfriends=mysql,sipusers,sip_buddies Now if I have understood it correct, the above should instruct the asterisk PBX to look in the Database called sipusers and use the table sip_buddies. But how do I instruct the asterisk PBX on how to actually connect to the database, meaning how to I tell the PBX the user name, password and name of the machine where the sipusers database reside ? I guess there must be another file where this information should be added? Also I would appreciate if you could tell me if I forgot any other configuration tasks etc. to make real time sip config work ? Looking forward to your reply Jan Bestil din ferie i dag på MSN Rejser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1's and span - what questions to ask my service provider
Yes Vikram, D channel is fixed by the provider. He would give you this along with other line configuration details. ~amsi On Thu, 3 Feb 2005 19:56:01 +0100, Vikram Rangnekar [EMAIL PROTECTED] wrote: I am planning to go in for a E1 line and whould like to know what questions i need to ask my service provider so i can connect that E1 to my asterisk box using the digium E1 card. what I mean is will my service provider give me info like LBO, framing , coding etc which i need to configure the span tag in the zaptel.conf and what about B and D channels am I allowed to setup whichever channel I like as my D channel or is that preset by my E1 service provider ? Also does anyone have any experience in setting up a asterisk box to be the NET end of the E1 line which is connected to a alcatel 400 pbx? -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Marked users with meetme2 ....
Hi, Has anybody tried using marked users with meetme2 ? I found that this is not working with meetme2 exten = 100,1,Meetme(100,w) exten = 100/vamsi_pottangi,1,Meetme(100,A) exten = 200,1,MeetMe2(200,w) exten = 200/vamsi_pottangi,1,MeetMe(200,A) Users other than vamsi_pottangi, when enters the conf room would be hearing music on hold for room no 100. This is not true for room number 200. Any ideas how to get thsi working for meetme2 ? Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] regexten for realtime sip ?
Hi, sip.conf has a paramter regexten using which we can assign an extension to a registered SIP client and can use the same number to call that client. Is there any such parameter for realtime sip table sip_buddies. Why was this missed out in this table ? Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial plan problems with realtime extensions ...
Hi, Case1: - -- extensions.conf exten = 1023,1,Voicemail(101) exten = 1023/101,1,MeetMe(200) Case2: - - extensions table (using realtime extensions) ++-+--++--+-+ | id | context | exten|priority| app | appdata | ++-+--++--+-+ | 29 | default | 1023 |1 | Voicemail | 101| | 30 | default | 1023/101 |1 | MeetMe| 200| In the first case when user 101 dials 1023, it directs him to meetme room 200. But in the case of realtime extensions it directs user 101 to Voicemail of 101, like any other user. It doesn't consider 1023/101 entry. How can I achieve proper routing in case of realtime ? Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Stress Test
SIPp has no facility to originate audio/media, it can just send back the media it receives on its RTP port, more like an RTP proxy. ~Vamsi On Thu, 20 Jan 2005 14:55:20 +0100, Stojan Sljivic - Pamet [EMAIL PROTECTED] wrote: Hi, Is there a free toll for SIP stress testing that supports RTP? Can SIPp be used for such purposes (to send audio)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime Voicemail ...
Hi,
Realtime SIP and Extensions are working fine but facing some problems
with Voicemail.
Added an entry to extconfig.conf
voicemail = mysql,asterisk,voicemail_users
Created the corresponding table and an entry for mailbox 201.
This is also reflected in the CLI as shown below.
CLI realtime load voicemail mailbox 201
Column Name Column Value
uniqueid 1
customer_id 201
mailbox 201
password 201
fullname Mailbox 201
stamp 20050118164309
CLI
When I try to log into the Voicemailmain, it cribs for incorrect login
as shown below.
Where am I going wrong ?
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Retrieve SQL: SELECT * FROM
extensions_table WHERE exten = '8500' AND context = 'default' AND
priority = '1'
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Everything is fine.
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Retrieve SQL: SELECT * FROM
extensions_table WHERE exten LIKE '\_%' AND context = 'default' AND
priority = '1' ORDER BY exten
Jan 18 17:49:12 DEBUG[5502]: MySQL RealTime: Everything is fine.
Jan 18 17:49:12 VERBOSE[5502]: -- Executing VoiceMailMain
(SIP/vamsi-0c3c, ) in new stack
Jan 18 17:49:12 DEBUG[5502]: Scheduling timer at 160 sample intervals
Jan 18 17:49:12 VERBOSE[5502]: -- Playing 'vm-login' (language 'en')
Jan 18 17:49:12 DEBUG[5502]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 101:
Found
Jan 18 17:49:14 DEBUG[5502]: Manager received command 'Command'
Jan 18 17:49:14 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:14 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:16 DEBUG[5502]: MySQL RealTime: Retrieve SQL: SELECT * FROM
voicemail_users WHERE mailbox = '201' AND context = 'default'
Jan 18 17:49:16 DEBUG[5502]: MySQL RealTime: Everything is fine.
Jan 18 17:49:16 DEBUG[5502]: Scheduling timer at 160 sample intervals
Jan 18 17:49:16 VERBOSE[5502]: -- Playing 'vm-password' (language
'en')
Jan 18 17:49:17 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:17 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:19 VERBOSE[5502]: -- Incorrect password '201' for user
'201' (context = any)
Jan 18 17:49:19 DEBUG[5502]: Scheduling timer at 160 sample intervals
Jan 18 17:49:19 VERBOSE[5502]: -- Playing 'vm-incorrect-
mailbox' (language 'en')
Jan 18 17:49:22 DEBUG[5502]: Scheduling timer at 0 sample intervals
Jan 18 17:49:22 DEBUG[5502]: Scheduling timer at 0 sample intervals
Thanks,
~Vamsi
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[Asterisk-Users] Adding SIP clients using AGI ?
Hi, Is there a way of adding SIP clients using AGI ? I see that, only extensions can be added using the AGI. If not AGI, is there any other way of adding SIP clients other than editing siop.conf manually ? Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
