Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23
hey let me know when its fixed so I can upgrade mine to :) take care hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: "Bob Goddard" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, May 24, 2005 2:49 PM Subject: Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23 On Tuesday 24 May 2005 17:07, Daniel ANDRE wrote: Bob Goddard a écrit : >On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: >>Hello, >> >>I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found >>at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send >>a register staement (nothing in thertereal log). With the 1.0.3.81 >>version, the phone register properly. >> >>Is ther any know bug with the SW Version? > >There is a registration bug with the BT101s which has been around for >years which is still in 1.0.6.3. > >If the phone is is rung but not answered, allowed to reregister as >normal, >then rung again and not answered, then the phone will never regiseter >again until rebooted. > >Grandstream have known about this bug since 3rd March but have only >acknowledged it since 15th May. Thank you Bob but I have just found what was wrong: two dhcp servers on the same network. Another question, Is the 1.0.6.3 stable enough for production use? I don't know. I only check for this single bug and until it's fixed no phone gets upgraded. 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] Broadvoice delivers CID even when restricted?
I missed the numbers can some one repost? thanks hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: "Darren Nickerson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, May 24, 2005 12:58 PM Subject: Re: [Asterisk-Users] Broadvoice delivers CID even when restricted? "Matthew Crocker" <[EMAIL PROTECTED]>: I know David Epstein and Dan Geopp personally, they are good guys, posting their direct office numbers on the mailing list is extremely bad form. As someone who has been given unacceptably vague, meaningless and often blatantly dishonest replies from badly trained support "staff" over the past three weeks, I'm glad to see these numbers released. Broadvoice has no escalation procedure ..., and they simply don't have any information for the people who eventually answer the phone to give to customers. I mean if you're going to pick up the phone, you'd think someone would at least tell them what to say!? They may be nice guys but their handling of this recent crisis means that they should anticipate taking a few calls. After three weeks of "we're working on it - call us in a few days" answers, we've been told that Broadvoice doesn't have inbound 800 service any more, and that they haven't actually been looking into our issue at all, since it's a network-wide failure whose reasons they understand, but just don't generally divulge. Apparently they're working with a new carrier to bring 800 back, but at this time it's completely out of service (can anyone confirm this?). And yet I can still buy an instant-activation 800 number online, and have been able to for this entire period. I've been a loyal customer, been patient and stuck with them for weeks. All I want now is the truth, and it's good to have new numbers at which to seek that truth. If inbound 800's not coming back, I'll look elsewhere for inbound 800 service and pick through the pros/cons of the various other ITSPs who have been reported to interoperate (with varying degrees of success) with Asterisk. If it really is coming back in any meaningful way, I can wait a few more weeks. -d ___ 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] BudgeTone 101 doesn't register with FirmWare1.5.23
they ever going to fix it? email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: "Bob Goddard" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, May 24, 2005 7:05 AM Subject: Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23 On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. 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@home
and man does it kick ass to!!! :) email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 - Original Message - From: Ariel Batista To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, May 24, 2005 4:19 AM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Asterisk @ Home This CD will install everything you need to get your Software PBX going. Its a complete ISO CD that brings together the OS (CentOS 3.4) Asterisk Software version 1.0.7 stable AMP Asterisk Management Portal Web GUI FTP TFTP Plus many more items. Every pre-configured to install and run out of the box. Just put it into your CD drive it will format and setup asterisk for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of QuintinSent: Tuesday, May 24, 2005 6:41 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] [EMAIL PROTECTED] Hi Can any one tel me what is [EMAIL PROTECTED] Thx Q ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat problem
go in to the [EMAIL PROTECTED] set up http://yourip/maint inter in your user name and password go to config edit sip.conf go to the general section Note: if you are behind a NAT Firewall, you will probably need to add thefollowing lines to hte [General] section of your sip.conf file. Adjust thenumbers as needed to match your configuration: externip=66.5.21.6 replace the above with your public ip address localnet=192.168.5.0/255.255.255.0 the 192 adddress mentioned above needs to be replaced with the ip address of your asterisk server eg mine is on 192.168.15.101 so it would look like localnet=192.168.15.101/255.255.255.0 hth hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 - Original Message - From: Betül Gözlükoğlu To: asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 3:57 AM Subject: [Asterisk-Users] nat problem Hi; Using [EMAIL PROTECTED] and it working well in network but when can not logged in over internet although the server is reachable Does anybody has any idea? Thanks Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spa-1001 not getting a dial tone on my pbx
hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology IncInfoSystemSIPProvisioningRegionalLine 1User 1User Loginbasic |advanced Table with 4 columns and 75 rows Line Enable: yes Streaming Audio Server (SAS)SAS Enable: noSAS DLG Refresh Intvl:30SAS Inbound RTP Sink: NAT SettingsNAT Mapping Enable: yesNAT Keep Alive Enable: yesNAT Keep Alive Msg:$NOTIFYNAT Keep Alive Dest:$PROXY Network SettingsSIP TOS/DiffServ Value:0x68Network Jitter Level: highRTP TOS/DiffServ Value:0xb8 SIP SettingsSIP Port:5060SIP 100REL Enable: noEXT SIP Port: Auth Resync-Reboot: yesSIP Debug Option: none Call Feature SettingsBlind Attn-Xfer Enable: noMOH Server: Xfer When Hangup Conf: yes Proxy and RegistrationProxy:67.183.118.6Use Outbound Proxy: noOutbound Proxy: Use OB Proxy In Dialog: noRegister: yesMake Call Without Reg: noRegister Expires:60Ans Call Without Reg: noUse DNS SRV: noDNS SRV Auto Prefix: noProxy Fallback Intvl:3600 Subscriber InformationDisplay Name:Herbie AllenUser ID:202Password:*Use Auth ID: yesAuth ID:202 Mini Certificate: SRTP Private Key: Supplementary Service SubscriptionCall Waiting Serv: yesBlock CID Serv: yesBlock ANC Serv: yesDist Ring Serv: yesCfwd All Serv: yesCfwd Busy Serv: yesCfwd No Ans Serv: yesCfwd Sel Serv: yesCfwd Last Serv: yesBlock Last Serv: yesAccept Last Serv: yesDND Serv: yesCID Serv: yesCWCID Serv: yesCall Return Serv: yesCall Back Serv: yesThree Way Call Serv: yesThree Way Conf Serv: yesAttn Transfer Serv: yesUnattn Transfer Serv: yesMWI Serv: yesVMWI Serv: yesSpeed Dial Serv: yesSecure Call Serv: yesReferral Serv: yesFeature Dial Serv: yes Audio ConfigurationPreferred Codec: G711uSilence Supp Enable: noUse Pref Codec Only: noSilence Threshold: mediumG729a Enable: yesEcho Canc Enable: yesG723 Enable: yesEcho Canc Adapt Enable: yesG726-16 Enable: yesEcho Supp Enable: yesG726-24 Enable: yesFAX CED Detect Enable: yesG726-32 Enable: yesFAX CNG Detect Enable: yesG726-40 Enable: yesFAX Passthru Codec: G711uDTMF Tx Method: AutoFAX Codec Symmetric: yesHook Flash Tx Method: NoneFAX Passthru Method: NSERelease Unused Codec: yesFAX Process NSE: yes Dial PlanDial Plan:(xx.|*xx.|**xx.|#xx.)Enable IP Dialing: no FXS Port Polarity ConfigurationIdle Polarity: ForwardCaller Conn Polarity: ForwardCallee Conn Polarity: Forward table end Undo All Changes Submit All Changes User Loginbasic |advanced Copyright © 2003 Sipura Technology. All Rights Reserved. --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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] play gsm files in windows
there is also a winamp plugin for playing gsm files go to www.winamp.com/plugins do a search for gsm you will find it hth hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: "El Flynn" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, May 23, 2005 8:07 AM Subject: Re: [Asterisk-Users] play gsm files in windows Brett, Gary wrote: Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated You could try Audacity (http://audacity.sourceforge.net). You have to use the "Import Raw Data" feature and open it as a GSM file. Flynn ___ 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] spa-1001 with asterisk?
hello my friend is trying to get his spa-1001 (sipura) 1001 connected to my asterisk box. he reset his spa-1001 to factory defaults I emailed him the voip-info page I found on google and yes I did look on google anyways he isn't able to get the thing to connect to it eg getting a dial tone, he did install x-lite and it worked fine with that am running [EMAIL PROTECTED] 1.0 can some one please tell me the steps to get the spa-1001 working for my friend, I will be passing the instructions to him. thanks for any help you can give. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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 with vonage linksys adapter?
hello do you know if vonage unlocks there linksys adapter to use with other providers? I want to use my ixisting vonage adapter with asterisk and cancil my vonage service. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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] having asterisk play music on hold tocallerswhile phone rings?
yep I have hold music other wise looks like I am going to have to go in to the [EMAIL PROTECTED] and configure it via that method can you give me pointers on what the dial line looks like so I dont screw this thing up?? they dont recommend editing this stuff bye hand unless you know what you are doing. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 - Original Message - From: Gary Lawrence To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Saturday, May 21, 2005 2:09 PM Subject: RE: [Asterisk-Users] having asterisk play music on hold tocallerswhile phone rings? Edit the extensions.conf and put an m at the end of the dial line. Do you have hold music otherwise? Sincerely; Gary Lawrence ITcom.Net 866.4ITcom1 866.448.2661 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smithSent: Saturday, May 21, 2005 4:26 PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings? hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings?
what config is this found in? email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: "Jon Gabrielson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, May 21, 2005 2:05 PM Subject: Re: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings? use option m in the cmd dial. Cheers, Jon. On Saturday 21 May 2005 03:26 pm, hank smith wrote: hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 ___ 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] having asterisk play music on hold to callers while phone rings?
hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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: Asterisk-1.0.7 Build - Serious issues
isn't [EMAIL PROTECTED] included in 1.07? of asterisk? also I checked the asterisk.org site and saw 1.06 but not the latest when was it put up on asterisk.org? - Original Message - From: "Tony Mountifield" <[EMAIL PROTECTED]> To: Sent: Thursday, March 31, 2005 8:52 AM Subject: [Asterisk-Users] Re: Asterisk-1.0.7 Build - Serious issues In article <[EMAIL PROTECTED]>, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote: Folks! I want to let everyone know that I have been trying to migrate from 1.0.6 to 1.0.7 last few days and I have come across serious issues in the build 1.0.7. What I found are listed below. I would recommend everyone to hold off any upgrade till the next build. But many people have successfully used 1.0.7, so it's possible the problem is at your end. 1)Voicemail - No Audio. Asterisk is not able to stream the voice to the Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say extension numbers for the called user. My guess is all these .gsm files are corrupt and hence you don't hear anything. They are fine in both the tar.gz file and from CVS stable. 2)Music on hold - .MP3 files in the ../mohmp3 and other folders are corrupt. When we tried to play these files using a media player, all we hear is gibberish. So are these. 3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we configure this for RFC2833. Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been able to find a fix? Try downloading again. If using FTP, ensure you have BINARY mode enabled. Seshu NOTICE: If received in error, please destroy and notify sender. But why destroy the sender as well as notifying them? 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
Re: [Asterisk-Users] Setting Up @Home 0.8 Guide
is there a way you can write those screen shots in to text format on the user guide? I am a blind computer user and am unable to see the examples that are shown on the site. thanks hank - Original Message - From: "Kerry Garrison" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Wednesday, March 30, 2005 11:12 PM Subject: [Asterisk-Users] Setting Up @Home 0.8 Guide Because of all of the changes to AMP, we have written up a completely new How-To Guide for [EMAIL PROTECTED] v0.8. Our first example uses BroadVoice for the trunk. http://www.geekgazette.com -Kerry ___ 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] Xten-lite for linux
do you know if it is gtk2? - Original Message - From: "Bruno Hertz" <[EMAIL PROTECTED]> To: Sent: Wednesday, March 30, 2005 10:31 PM Subject: Re: [Asterisk-Users] Xten-lite for linux Kris Edwards <[EMAIL PROTECTED]> writes: This is the best linux sip phone I've used so far. Audio quality has been perfect and it seems really stable, so hopefully it will be out of beta soon. I might actually pay for the full version! (not counting console games, that would be the second piece of software I've purchaced since 1987). Sounds rather like you want to sell the full version. Myself, I don't know about recent betas since, frankly, I didn't care anymore after initial experiences being pretty much disappointing. The first beta I got produced no audio at all, and we had a tough time to convince the developer that it wasn't a driver issue. The next releases then had huge latencies, primarily due to the Xlite audio setup. Now, I admit that setting up audio for interactive/'realtime' apps on linux is a mess, but various open source projects have already done much better. So no, in contrast to your plug I'm not as enthusiastic myself, especially since audio quality resp. latency is the one major trouble I had with linux softphones. E.g. iaxcomm would be great and totally satisfying for me if latency were (significantly) less than 1 second. Regards, Bruno. ___ 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] using amp with asterisk?
hello I have asterisk 1.0 running on fedora core3 and amp version 1.06 I think is the version its the version down below the current release, I have fwd working threw iax on outbound calls fine but I can't get inbound to work, has any one successfully gotten this to work? if so can you tell me what you did to get it working? there isn't alot of documentation on amp so I am kind of lost. thanks hank ___ 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] VoIP Telephony with Asterisk by Paul Mahler
is it in ebook format at all? I am a blind computer user and have no way of getting it scanned in to my computer even if I were to purchase it. thanks hank - Original Message - From: "Sys. Concept Inc." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, September 11, 2004 10:08 PM Subject: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler Does anybody have the book: VoIP Telephony with Asterisk by Paul Mahler. Is it for beginners or advanced users? -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1117 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice
I am hooked up with broadvoice and have been having no problems that are major there voice mail system went on the blits for about 30 minutes yesterday but that was about it. what kind of problems you expierencing? - Original Message - From: "Joel Gathercole" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, September 11, 2004 9:19 PM Subject: [Asterisk-Users] Broadvoice Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1117 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD
the user you are calling is currently offline is what I get when calling fwd number hth hank - Original Message - From: "Steve Maroney" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, September 11, 2004 9:29 AM Subject: [Asterisk-Users] FWD Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave me a message if i dont answer. FWD Number is 474538. My * box is configured to ring one of my extentions. Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1104 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
I have to have access to sound on linux to use the screen reader for linux and from what I under stand colinux don't support sound. otherwise this would be the perfict sullution. - Original Message - From: "Greg Boehnlein" <[EMAIL PROTECTED]> To: "Benjamin on Asterisk Mailing Lists" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Cc: "Andy Powell" <[EMAIL PROTECTED]> Sent: Friday, September 10, 2004 9:13 PM Subject: Re: [Asterisk-Users] iaxy vs sipura On Fri, 10 Sep 2004, Benjamin on Asterisk Mailing Lists wrote: However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. Or you could use AstWind, which runs concurrently with Windows and is built entirely on Open Source software (CoLinux Kernel, Debian, Asterisk) and avoid paying for Vmware! ;) Plus, installation is a snap. See Digium's press release: http://www.digium.com/index.php?menu=astwind You can find more information on AstWind at: http://www.voip-info.org/wiki-AstWind -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1084 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
how much ram you got on the pc running the vm? also will microsoft Virtual PC run on xp home? thanks hank - Original Message - From: "Bill Seddon" <[EMAIL PROTECTED]> To: "'Benjamin on Asterisk Mailing Lists'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Friday, September 10, 2004 6:34 AM Subject: RE: [Asterisk-Users] iaxy vs sipura I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use Zaptel cards but that's OK for me. I can put it into standby any time and it takes only a few seconds to start up the VM from its saved state and at that time the Linux session (and Asterisk) is available once again. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: September 10, 2004 2:03 PM To: Andy Powell Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxy vs sipura On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell <[EMAIL PROTECTED]> wrote: At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Certainly an option, but most business folks will want to have their Outlook contacts and Excel spreadsheets in front of them when they are on the phone. Dual boot environments are not ideal in those situations. Imagine you're talking to some guy on the phone about prices and he tells you "I cant' tell you what the discounts are right now because I would have to shut down the phone system to open Excel". However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. I remember there was a guy in Romania who reported he had VMware with Windoze and Asterisk on Linux running as a home PBX on his PC and it seemed to be alright. If you'd combine such a setup with a Windoze GUI tool that will start and stop the Linux environment and Asterisk at the push of a button, then you'd have a fairly convenient and workable SIP/IAX gateway solution for travelling biz folks. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1068 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference Phone
what phone did you purchase and how much - Original Message - From: "Deon Rodden" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Friday, September 10, 2004 5:59 AM Subject: Re: [Asterisk-Users] Conference Phone We use a nice Polycom conference phone and plugged it into the Sipura and it works crystal clear. Was cheaper than Polycom's conference phone w/ built in VOIP capabilities. Joe Dennick wrote: If it were me; I'd opt for one of the Polycom Conference phones (they are just regular analog phones), and use an FXS card to connect it to Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Thursday, September 09, 2004 4:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Phone Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they don’t have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.745 / Virus Database: 497 - Release Date: 8/27/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1062 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
it works it works it works! sorry it took it so long for the info to click thanks for the help guys!!! take care hank - Original Message - From: "Greg Boehnlein" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, September 09, 2004 11:57 PM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, hank smith wrote: I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that what I put in the xml file? Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking Specifically: "If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is "NDIS 5.0 driver" and not "Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter". It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta" Deja Vu.. Is there an echo in here? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1053 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
yasr is text based but the interesting part is going to see if it works running on a windows platform with this version of linux with out that I can't do anything with this so I will have to see. take care. hank - Original Message - From: "Greg Boehnlein" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Friday, September 10, 2004 12:00 AM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, hank smith wrote: is there going to be a gui for co linux and astwind? No. AstWind is just a Debian GNU Linux distribution with a precompiled Asterisk installation running under a CoLinux kernel. I will have to see if either there is going to be a gui or if yasr a screen reader for the blind will work with this thing. I do not know. I would assume that a blind user would probably prefer a text based interface, but I have no clue. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1050 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
is there going to be a gui for co linux and astwind? I will have to see if either there is going to be a gui or if yasr a screen reader for the blind will work with this thing. thanks hank - Original Message - From: "Greg Boehnlein" <[EMAIL PROTECTED]> To: "arsal siddiqui" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, September 09, 2004 9:11 PM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, arsal siddiqui wrote: dear khurram, i need to know the price for x100p. i've emailed convergence.com.pk and never get a reply. If you could help me in this regards, i'll be greatful. I need to know the price. send me an email off the list. if you can help me in getting * hardware. Waiting for your reply Just as a side note... CoLinux CANNOT YET interface with any Digium hardware! So if you plan to run an X100P under AstWind you may be waiting a long time before it works! ;) Someone needs to port Zaptel to CoLinux! ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1047 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that what I put in the xml file? - Original Message - From: "Greg Boehnlein" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, September 09, 2004 9:03 PM Subject: RE: [Asterisk-Users] astwind has any one got this thing to work? On Wed, 8 Sep 2004, Chris HARIGA wrote: I make it work!! My Astwind is up and running! Now is 11:53 PM and I'm going to bed. Tomorrow morning I will post how I fix the Ethernet connection. I bet you followed the following directions! ;) From: http://www.colinux.org/wiki/index.php/coLinuxNetworking "If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is "NDIS 5.0 driver" and not "Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter". It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta" -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1047 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Store data from call to database
when you get this up up can you give the phone number? this sounds rather interesting, and fun!!! - Original Message - From: "bagattin jerome" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 09, 2004 7:54 PM Subject: [Asterisk-Users] Store data from call to database Hi, I use asterisk for a phone quiz game. I need to store data in a database (MySql, postgres) : telephone number, name (voice), ... and of course the answers at the quetions. What's the best way to store my data ? - script with system() command ? - AGI script - CDR - others ... Thanks Jerome Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1045 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival Speech Synthesis 1.95:beta July 2004 Eval
ï do you got a number I can call to take a listen? - Original Message - From: Steve Murphy To: [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 12:06 PM Subject: [Asterisk-Users] Festival Speech Synthesis 1.95:beta July 2004 Eval Hello--In the interests of playing around and wasting time, I've installed the latest version of theFestival stuff, 1.95beta.And, in the interests of future Asterisk-Festival connectivity, I applied the 1.4.3 patch to put in theasterisk related routines. I did it by hand, but, it looks like the patch will apply with no comment.Asterisk works with the new server...BUTthe speed of what's played over the speaker vs. what you hear over the phone is off by maybe 2x.The voice isn't shifted in frequency at all. On Asterisk, the voice just speaks twice as fast as it does coming from festival over the speakers. And, if you are having it say jokes at double speed, well, it reminds me of speed reading.There must be some lever or pulley or switch or something to modify the speed.ANDHoo boy, try putting this in your siteinit.scm file:(set! voice_default 'voice_cstr_us_awb_arctic_multisyn)and listen to this:(SayText "Hello there, kyootee pie.")(SayText "Don't you just love the sound of my voice.")(SayText "My wife, Sonya, Makes the best bread there ever was")Best synthetic voice I've ever heard.murf -- Steve Murphy Electronic Tools Company ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter1037 spam mails have been blocked so far.Download free SPAMfighter today!___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940
what is the price range in us dollars? - Original Message - From: "Jody N. Rudolph" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, September 09, 2004 11:32 AM Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1033 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
can you post the information on how you got that thing working? thanks hank - Original Message - From: "Chris HARIGA" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Wednesday, September 08, 2004 8:55 PM Subject: RE: [Asterisk-Users] astwind has any one got this thing to work? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1030 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
please do I can't get mine to work thanks hank - Original Message - From: "arsal siddiqui" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Wednesday, September 08, 2004 5:47 PM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? I am download astwind-installer-0.1.1.exe, I'll post an update if I manage to make this thing work. Regards Arsal ----- Original Message - From: hank smith <[EMAIL PROTECTED]> Date: Wed, 8 Sep 2004 00:14:37 -0700 Subject: [Asterisk-Users] astwind has any one got this thing to work? To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am running a netgear rp614 router behind nat if this helps but I have tried and tried and tried to get this sucker up with no luck any help would be greatly greatly appreciated. thanks hank My Inbox is protected by SPAMfighter 983 spam mails have been blocked so far. Download free SPAMfighter today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1010 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I get DIDs for remote areas in Canada
are you serious? that it is elegal to watch hbo? if so what is the logic behind that one? that is so stupid email me off list on this one [EMAIL PROTECTED] - Original Message - From: "Brandon Patterson (peering)" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Wednesday, September 08, 2004 1:25 PM Subject: Re: [Asterisk-Users] How do I get DIDs for remote areas in Canada Good luck. If you did you will pay through the nose. Did you know that the CRTC in Canada is holding hearings late Sept on VOIP? Decision due in Feb 2005. Can we say why waste time? 10 people decide your entire future from radio to phone to tv. Hey, its against the law to watch HBO in Canada! Don't invest any money in small towns unless you want to go broke. Contact me off the list and I will be happy to go further. Our Motto "Canada Owned and Operated by the Very Few" I want the ability to setup DIDs in a variety of different remote locations in Canada. There are various providers that have DIDs in major cities, but none that focus on the "smaller" cities. The question is how do I actually setup these DIDs? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1002 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astwind has any one got this thing to work?
hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am running a netgear rp614 router behind nat if this helps but I have tried and tried and tried to get this sucker up with no luck any help would be greatly greatly appreciated. thanks hank My Inbox is protected by SPAMfighter983 spam mails have been blocked so far.Download free SPAMfighter today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with length of voicemail
I had that problem when I was running asterisk on my linux box before it went down so you aren't the only one having that problem - Original Message - From: Marty Mastera To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, September 07, 2004 10:50 PM Subject: RE: [Asterisk-Users] Problems with length of voicemail I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: Roger, There has been very recent discussion regarding this topic exactly...specifically when using BroadVoice as a sip provider. Calls to your BroadVoice DID that end up in VM terminate after 30 seconds The current theory is that during VM recording, * doesn't send any audio packets back to BroadVoice...after 30 seconds BroadVoice thinks that the connection has been lost and terminates the call...(I'm paraphrasing the thread that recently appeared on this topic, forgive me if this isn't completely accurate) Assuming that this is correct, you could be using BroadVoice, or another provider who disconnects after not receiving audio for some period of time... Hope that helps, Marty ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter983 spam mails have been blocked so far.Download free SPAMfighter today!___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk, fwd, and grandstream?
I have all ready been there the only refference I saw was the tips and tricks for asterisk and grandstream is there some info I am missing? thanks hank - Original Message - From: "Holger Schurig" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, July 06, 2004 12:06 AM Subject: Re: [Asterisk-Users] asterisk, fwd, and grandstream? > > can this be accomplished? > > Yes. > > > You should start reading documentation before asking. A good starting > place is http://www.voip-info.org > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback over Console
what was the problem? - Original Message - From: "Chris Foster" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 05, 2004 10:02 PM Subject: Re: [Asterisk-Users] Playback over Console > Thanks for responding. I figured it out. > > On Mon, 5 Jul 2004 22:22:37 -0600 (MDT), Greg Hill > <[EMAIL PROTECTED]> wrote: > > I suspect that after Dial has happened the auto-answer connects you to the > > console, and the call doesn't reach Playback until after the console hangs > > up. > > Which is exactly right. > > > As for how to do what you're after.. I dunno! Maybe you can find a way to > > "pick up" the console as if to dial from the console out to somewhere and > > issue the Playback then. > > > > Sort of. It turns out that Dial has option that does exactly what I > want, namely, play a tone over the speakers (console) to alert people > that somebody is about to speak/ > > A(x): Play an announcement (x.gsm) to the called party. > > Put that in Dial's options and Asterisk sends it out to the caller, > which is exactly what I wanted. You can find more in the Tiki > > The only bad part about all of this is that my roommate found a > air-raid siren sound, and so now when you call you don't get a tone > but a 2 minute long warning that a tornado is approaching. I'll have > to change that. > > > Greg > > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
will do. thanks - Original Message - From: "Greg Hill" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 05, 2004 9:26 PM Subject: Re: [Asterisk-Users] Calling an outside phone number as part of a hunt > On Mon, 5 Jul 2004, hank smith wrote: > > > how would I do this but do it with broadvoice? > > I want to give people the oppsion to call my cell phone but I use a voip > > carier > > stay tuned to see how he gets the thing figured out, then change > exten => 2000,2,Dial(Zap/1/5551212,10) > to > exten => 2000,2,Dial(SIP/[EMAIL PROTECTED],10) > > or similar. > > Greg > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk, fwd, and grandstream?
hello I want to use my grandstream witch is currently configured for fwd to use asterisk, my asterisk is configured with fwd threw iax, but I want to still recieve calls on my grandstream threw fwd threw asterisk if this makes any sense is this possible? I basicly want all of my phones to use asterisk but be able to use them with all the networks, my fwd account, my broadvoice account, etc etc. can this be accomplished? thanks hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
how would I do this but do it with broadvoice? I want to give people the oppsion to call my cell phone but I use a voip carier - Original Message - From: "Hall, Eric M." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 05, 2004 7:51 PM Subject: [Asterisk-Users] Calling an outside phone number as part of a hunt > I'm trying to see if this is even possible. > > When you dial ext 2000 I want it to ring my sip phone for 20 sec then > call my cell and let it ring for 10 sec if I do not pick up the call on > my cell I would like it to go back to * and leave a voice message for > me. Here is what I have so far in my extensions.conf > > Everything works except the call will not go back to * after the 10 sec > rule has expired. > > My hardware is 2 X100P card > > > > exten => 2000,1,Dial(SIP/2000,20) > exten => 2000,2,Dial(Zap/1/5551212,10) > exten => 2000,3,Voicemail(u2000) > exten => 2000,102,Voicemail(b2000) > exten => 2000,103,Hangup > > Any ideas? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback over Console
do you got your speakers in the 2 floors of your house hooked up to the computer? am just curious. how do you got your sound system set up? email me off list. this may be off topic. email [EMAIL PROTECTED] - Original Message - From: "Chris Foster" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 05, 2004 5:48 PM Subject: [Asterisk-Users] Playback over Console > I'm trying to setup a primitive announcement-paging system in my house > using the line-out from my * box to a cheap amplifier that runs to > speakers on our first and second floors from the basement. I have a > extension that connects to Console, and console is set to auto-pickup. > I'm using alsa drivers. > > This all works great, except for one thing. I want to play a tone over > the console after the console picks up. What i'm doing right now is > calling Playback after the Dial. However, No playback sound or > background sound is being heard over the console speakers or are any > error messages appearing in the command line. > > The extensions.conf entry looks like this: > [access-internal] > include => parkedcalls > exten => 31,1,Dial(SIP/line1,30,t) > exten => 31,2,Voicemail(u1) > exten => 32,1,Dial(SIP/line2,30,t) > exten => 32,2,Voicemail(u1) > exten => 33,1,Dial(SIP/grand,30,t) > exten => 33,2,Voicemail(u1) > exten => 310,1,Dial(Console/dsp) ;; intercom > exten => 310,2,Playback(tt-weasels) > > Thanks to anybody who can help! > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for newbie resources
hello andy is your user guide updated? - Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, July 04, 2004 5:24 PM Subject: Re: [Asterisk-Users] looking for newbie resources > > On 04/07/2004 at 14:53 Steven M. Sawczyn wrote: > > >Hi, I am very interested in VOIP and telephony in general, although > >admittedly, I don't know much about the theories and protocols behind it. > >Having also an interest in Linux, I was really excited to come upon > >Asterisk. I would really like to learn more about Asterisk and VOIP in > >general and am wondering if anyone could suggest some beginner resources? > >Of course I've found that the best way to learn something is to just dive > >in > >and try it, but I don't think I'm ready to tackle installing Asterisk yet. > > > In which case, http://www.automated.it/asterisk/ You'll find a link there for > my Asterisk Live! CD (it's a test version, but feedback so far has been > favourable) > > > > > >I'm running Slackware Linux on a machine which at the moment, is just > >hosting mail. In addition, I have accounts with both Vonage and > >Broadvoice. > >My idea is to set up a mini PBX here at home using both VOIP providers as > >my > >main lines and using my LAN to connect a few extensions. Might this be a > >good way to start learning, or am I way off track? > > > >Again, I am very new to this, so any info/resources/suggestions greatly > >appreciated. > > You could also try http://www.automated.it/guidetoasterisk.htm to > get you going... > > The wiki has useful info too > > http://www.voip-info.org > > > Andy > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] music on hold question with asterisk
hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream ringtones - makering.pl usage for 1.0.50
how can you create your own ring tone? - Original Message - From: "Maron Kristófersson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, June 08, 2004 6:57 AM Subject: [Asterisk-Users] grandstream ringtones - makering.pl usage for 1.0.50 > If you wan't to create a ringtone with makering.pl for firmware 1.0.50, > be sure to create it as ring.bin and then rename it to ring1.bin / > ring2.bin or ring3.bin. This seems to be the only change between the > format from 1.0.4.68. > > Regards, > Maron > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk to broadvoice?
hello is there any info on connecting asterisk to broadvoice? thanks hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DNS SRV records
can enum be used with asterisk? if so how? - Original Message - From: "Duane" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, June 07, 2004 6:09 PM Subject: Re: [Asterisk-Users] Re: DNS SRV records > Adam Goryachev wrote: > > > In fact, I think it would be nice if all modules/apps/chans/etc were > > marked noload by default (except the bare minimum required to get > > asterisk to start with no channels...). > > Then this goes back to my original point, I will not suggest people use > SRV records if they want to receive calls as a large majority of > Asterisk users won't be able to call them. If they want a simple method > of allowing calls they should use enum, least then it's obvious that it > isn't a email address and that they would possibly need to enable a few > things to make it work. > > -- > > Best regards, > Duane > > http://www.cacert.org - Free Security Certificates > http://www.nodedb.com - Think globally, network locally > http://www.sydneywireless.com - Telecommunications Freedom > http://happysnapper.com.au - Sell your photos over the net! > http://e164.org - Using Enum.164 to interconnect asterisk servers > > "In the confrontation between the stream and the rock, the > stream always wins; not through strength, but through persistence." > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Sniffing Calls for recording
how can you record calls with asterisk? I didn't even know this was possible can some one point me to a url for info on this? - Original Message - From: "lists" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, June 07, 2004 1:30 PM Subject: [Asterisk-Users] Network Sniffing Calls for recording > > Ok assuming I don't want to record calls using * but instead want a > dedicated server that listens to a mirror port and records calls. Is there a > cheap software package out there for doing this for mgcp/sccp? I know if > evern cut over to * there is a way but I doubt I will even cut 100% over to > * so I was wonder what the list has heard of for call recording via sniffing > my gates. I know there are some out there but $100k for 40 users is to high > for my blood. > > Offlist is fine for all flames and answers since this is a bit off topic > [EMAIL PROTECTED] > > OK it's a Monday when it takes 5 tries to get a email to the right list from > the right account. > > Either that or someone switched the coffee pot to decaf again. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata?
what is hadrware? - Original Message - From: "Richard Neese" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, June 06, 2004 7:37 PM Subject: Re: [Asterisk-Users] Zapata? > as for hadrware digitalnetworks has made a clone card . but only digium has > made any majoor card changes. there have been 2 ne rev to the cards i kow of > and you can rea d on the digium site about them. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD network from Asterisk through NAT
Hi there, I'm trying to dial into the FWD network using Asterisk, though a NAT. The sources I've read say that it's unconfirmed to work through a NAT, but I'm wondering if anyone's done it anyway. So, anyone got a clue how to do this? Hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] firefly sip question
hello I am not sure where to ask this question at so please except my apologise if this is the wrong list. I need to ask if any one has got firefly sip version to work with fre world dialup? if so what info did they use to connect? once again if this is the wrong list if the person who is developing this thing email me off list or direct me to a list fore firefly it would be greatly appreciated email is [EMAIL PROTECTED] thanks hank
Re: Asterisk-Users digest, Vol 1 #3101 - 14 msgs Subject: Re: [Asterisk-Users] Asterisk on KNOPPIX, I have it working, somewhat.
let me know if you get the noppix done
I would be interested!
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, March 14, 2004 7:18 PM
Subject: Asterisk-Users digest, Vol 1 #3101 - 14 msgs
> Send Asterisk-Users mailing list submissions to
> [EMAIL PROTECTED]
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> [EMAIL PROTECTED]
>
> You can reach the person managing the list at
> [EMAIL PROTECTED]
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Asterisk-Users digest..."
>
>
> Today's Topics:
>
>1. RE: How to send CallerID trough CAPI ? (Jakob Strebel)
>2. RE: ast_rtp_raw_write errors distorting sound on G729 passthrough
(Senad Jordanovic)
>3. Re: Asterisk on KNOPPIX, I have it working, somewhat. (Greg
Boehnlein)
>4. Re: Radius (Greg Boehnlein)
>5. RE: ast_rtp_raw_write errors distorting sound on G729 passthrough
(Senad Jordanovic)
>6. RE: VXML_URL and Cisco 7960 Phones? (Low, Adam)
>7. ISDN PRI A and B, cry for help. (Matthew Branton)
>8. Re: Radius (Derek Bruce)
>9. Asterisk NAT Gateway Setup (Kevin)
> 10. Re: Cisco SIP license (Matthew Enger)
> 11. Re: ISDN PRI A and B, cry for help. (Steve Underwood)
> 12. VoYP.Net: voip directory and ENUM registry (Greg Retkowski)
> 13. Re: VoYP.Net: voip directory and ENUM registry (Matt Riddell)
> 14. EchoCan (Matt Riddell)
>
> --__--__--
>
> Message: 1
> Date: Sun, 14 Mar 2004 17:49:37 +0100
> To: [EMAIL PROTECTED]
> From: Jakob Strebel <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] How to send CallerID trough CAPI ?
> Reply-To: [EMAIL PROTECTED]
>
> Florian,
>
> Thanks. Now it works.
> jakob
>
>
> >The correct dial syntax for CAPI channels is like this:
> >
> >CAPI/12345678:b${EXTEN}
> >
> >where: 12345678 is your outgoing MSN (you would choose 0627775171) and
> >${EXTEN} is the number to dial. My mistake was I moved the 'b' too when I
> >switched the two numbers around.
> >
> >Please try Dial(CAPI/0627775171:b123456) or Dial(CAPI/627775171:b123456)
> >
> This are the relevant sections in the config:
>
> In extensions.conf
>
>
>
> [globals]
> ;
> ; globals f=FCr ISDN
> CLID=3D0627775171
>
> [outst]
> exten =3D> _0.,1,SetCIDNum(${CLID})
> exten =3D> _0.,2,Dial(CAPI/0627775171:b${EXTEN}) ; ok CID is sent correct
>
>
> In capi.conf
> msn=3D0627775171
>
>
>
> --__--__--
>
> Message: 2
> From: "Senad Jordanovic" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] ast_rtp_raw_write errors distorting sound on
G729 passthrough
> Date: Sun, 14 Mar 2004 17:07:26 -
> Reply-To: [EMAIL PROTECTED]
>
> Olle E. Johansson wrote:
> > Check out the latest CVS, Mark applied changes to the code in this
> > area tonight. The rtp.c is changed, so the old patch in
> > bugs.digium.com may not be necessary any more.
> >
> Yes, it is done..
> BUT
> Now I get MUCH higher values is the debug messages and can not
> understand a word from other party during the conversation.
>
> Here it is:
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386825128, ms is -1247094945
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386825608, ms is -1247095005
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386826088, ms is -1247095065
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386826560, ms is -1247095124
>
>
> --__--__--
>
> Message: 3
> Date: Sun, 14 Mar 2004 12:51:37 -0500 (EST)
> From: Greg Boehnlein <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Cc: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk on KNOPPIX, I have it working,
somewhat.
> Reply-To: [EMAIL PROTECTED]
>
> I've got Asterisk running on a minimal install of Debian on a P133 w/ 16
> megs of ram. I can help with the Damn Small Linux side of things, and
> perhaps get you out of Dependency hell.
>
> Do you want the system to be self hosting? I.E. the distribution where
> Asterisk lives contains the appropriate compilers, source and includes to
> build the system?
>
> Or do you want a "Development" distribution and a "Target" platform? I.E.
> you build on the developmen distro and then run a few scripts to generate
> a target ISO w/ the binaries?
>
> In either case, I've got plenty of resources to offer. I've been toying
> with the idea of creating a "Knapterisk" installation for quite some time.
>
> --
> Vice President of N2Net, a New Age Consulting Service, Inc. Company
> http://www.n2net.net Where everything clicks into place!
> KP-216-121-ST
>
>
>
>
> --__--__--
>
> Message: 4
> Date: Sun, 14 Mar 2004 13:01:26 -0500 (EST)
> From: Greg Boehnlein <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Radius
> Reply-To: [EMAIL PROTECTED]
>
> On
Re: [Asterisk-Users] zaptel on Debian
if you do a apt-get install asterisk you can get it all ready compiled and everything ondibian. I just did that last night. hth - Original Message - From: "Yury Bokhoncovich" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Cc: "Duane" <[EMAIL PROTECTED]> Sent: Thursday, March 11, 2004 11:03 PM Subject: Re: [Asterisk-Users] zaptel on Debian > Hello! > > On Sat, 6 Mar 2004, Duane wrote: > > > Hermann Wecke wrote: > > > After trying and trying to compile and make Asterisk run on a Debian > > > box, I gave up and picked another HD with RH 9 on it. No headaches. Only 1 > > > build was necessary to build and run *. > > I guess using of "/lib/modules/$(shell uname -r)/build" is the best method > for Makefile as it is recommended by kernel developers and saves against a > lot of headaches. YMMV, e.g. I have had specified > /lib/modules/2.4.22-ac2-ow1-xattr-acl-0.8.60-byg/build > directly when compile NVIDIA drivers under another kernel. > > -- > WBR, Yury Bokhoncovich, Senior System Administrator, NOC of F1 Group. > Phone: +7 (3832) 106228, ext.140, E-mail: [EMAIL PROTECTED] > Unix is like a wigwam -- no Gates, no Windows, and an Apache inside. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and text to speech?
hello do you know where I can get scripts to use with askerisk to use it with text to speech? festival, and dec-talk? thanks hank
Re: [Asterisk-Users] windows alternitives to Asterisk?
is there any documentation on porting the asterisk program to windows? if not where can I get info on how to see if I can do this? thanks hank - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 08, 2004 11:07 AM Subject: Re: [Asterisk-Users] windows alternitives to Asterisk? > hank smith wrote: > > > hello I am just curious if there is any windows alternitives to Asterisk? > > can I also use them with free world dialup? > > thanks > > hank > > No, but maybe you could port Asterisk to Windows. No, that's not a joke. > The Zaptel drivers might be tough, but Asterisk's VoIP features would > probably run under Cygwin without too high a mountain of work. > > Regards, > Steve > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] windows alternitives to Asterisk?
is there a program that I can install on my linux box so I can configure the pbx from the internet from my windows box so I don't have to work with config files? thanks hank - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 08, 2004 11:07 AM Subject: Re: [Asterisk-Users] windows alternitives to Asterisk? > hank smith wrote: > > > hello I am just curious if there is any windows alternitives to Asterisk? > > can I also use them with free world dialup? > > thanks > > hank > > No, but maybe you could port Asterisk to Windows. No, that's not a joke. > The Zaptel drivers might be tough, but Asterisk's VoIP features would > probably run under Cygwin without too high a mountain of work. > > Regards, > Steve > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] windows alternitives to Asterisk?
hello I am just curious if there is any windows alternitives to Asterisk? can I also use them with free world dialup? thanks hank
