RE: [Asterisk-Users] Daily Phreak - Daily Telecom, Asterisk and Phreaking Updates
Dude this is just an adsense Blog site. Don't use this forum to try to make money. It's not for that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 21, 2005 8:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Daily Phreak - Daily Telecom,Asterisk and Phreaking Updates Just wanted to let everyone know about our new site http://www.dailyphreak.com. We have lots of great Asterisk stuff planned to be released and strive to bring Asterisk and other telecom related news updates on a daily basis! Check us out and let us know what you think. ___ --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] unsubscribe please
I unsubscribed but I am still getting emails to this account. Please remove [EMAIL PROTECTED] I think that the moderator added my reply to address to the list. Thanks for your help. -Jason From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Sampson Sent: Monday, December 19, 2005 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can't call out on ZAP channel - need help Yeah, the zttool program shows the PRI as having No Alarms. It is an Infinity system by Amtelco. I haven't actually tried making a call from the other pbx, but I did have my vendor (Amtelco) look at it and they verified that the span was up and everything was working correctly. The asterisk system is set to signalling=pri_net which I assumed meant that the asterisk box would be handling the timing. Here is the output from pri show span 1 asterisk1*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: Q.SIG switch Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 Michael SampsonInformation Systems ManagerCustomer Contact Services[EMAIL PROTECTED]952-936-4000 O'Connor, Jonathan wrote: Michael,Does the zttool program show the PRI as working correctly?Can the PBX push calls into the Asterisk system?Also, what type of PBX is it, and is it providing the clock etc.. Forthe T1 connection?-Jonathan Jonathan O'ConnorSenior System AdministratorInoveris LLCDirect Line (614) 791-3742Fax (614) 791-3748Helpdesk 866-456-1566 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael SampsonSent: Monday, December 19, 2005 9:30 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Can't call out on ZAP channel - need helpI'm trying to connect to another PBX via an T-1 interface. I have a T100P card.On the CLI I get the error Everyone is busy/congested at this time (1:0/0/1) When I try to dial out of the T-1 line from an SIP softphone.I have posted this question a few times here and at the asterisk forum, but can't get anyone to respond. I've seen other people on forums with the same problem but no one has ever given much of a solution. Does someone at least know what the next step in debugging this problem would be.In the file /var/log/asterisk/full I get the error Unable to create channel of type 'ZAP'Here are my configs.Zapata.conf--;; Zapata telephony interface;; Configuration file[trunkgroups][channels]language=encontext=from-pstn;signalling=fxs_kssignalling=pri_net ; pri_cpe= PRI slave ; pri_net = PRI master switchtype=qsig pridialplan=local resetinterval=never;rxwink=300 ; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0group=1callgroup=1pickupgroup=1immediate=no;faxdetect=bothfaxdetect=incoming;faxdetect=outgoing;faxdetect=no;Include genzaptelconf configs#include zapata-auto.conf;Include AMP configs#include zapata_additional.confchannel = 1-23-Zaptel.conf-# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg ## It must be in the module loading order# Span 1: WCT1/0 Digium Wildcard T100P T1/PRI Card 0# channel 1, WCT1, unhandled for now# channel 2, WCT1, unhandled for now# channel 3, WCT1, unhandled for now# channel 4, WCT1, unhandled for now# channel 5, WCT1, unhandled for now# channel 6, WCT1, unhandled for now# channel 7, WCT1, unhandled for now# channel 8, WCT1, unhandled for now# channel 9, WCT1, unhandled for now# channel 10, WCT1, unhandled for now# channel 11, WCT1, unhandled for now# channel 12, WCT1, unhandled for now# channel 13, WCT1, unhandled for now# channel 14, WCT1, unhandled for now# channel 15, WCT1, unhandled for now# channel 16, WCT1, unhandled for now# channel 17, WCT1, unhandled for now# channel 18, WCT1, unhandled for now# channel 19, WCT1, unhandled for now# channel 20, WCT1, unhandled for now# channel 21, WCT1, unhandled for now# channel 22, WCT1, unhandled for now# channel 23, WCT1, unhandled for now# channel 24, WCT1, unhandled for now# Global dataspan=1,1,0,esf,b8zsbchan=1-23dchan=24#fxsks=1loadzone = usdefaultzone = us---Michael SampsonInformation Systems ManagerCustomer Contact Services[EMAIL PROTECTED]952-936-4000___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE
RE: [Asterisk-Users] Turning off hardware echo can on TE411P
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William K. Volkman Sent: Monday, December 12, 2005 9:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Turning off hardware echo can on TE411P Hello, On Mon, 2005-12-12 at 15:42 -0600, Kevin P. Fleming wrote: Eric Bishop wrote: Anyone know if Asterisk 1.2.1 supports turning off the hardware echo canceller WITHOUT recompiling the driver like I had to in 1.0.X? Add 'vpmsupport=0' to your modprobe.conf or equivalent. OK, so is there a way to have hardware echo canceling and have DTMF digits go out correctly? We bought the expensive hardware echo canceling card however it appears that we have to have vpmsupport=0 in order to get DNIS digits correctly (see my thread about ADIT and DNIS digits earlier). Clarifications about what to tweak appreciated. Thanks, William. ___ --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] Turning off hardware echo can on TE411P
I didn't write this below. I replied with a blank line by mistake. I am truly sorry if you were confused by that. -Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, December 13, 2005 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Turning off hardware echo can on TE411P Jason Brashear wrote: OK, so is there a way to have hardware echo canceling and have DTMF digits go out correctly? We bought the expensive hardware echo canceling card however it appears that we have to have vpmsupport=0 in order to get DNIS digits correctly (see my thread about ADIT and DNIS digits earlier). Clarifications about what to tweak appreciated. Thanks, William. Is your name Jason or William? Very confusing. Please take this issue up with Digium tech support. ___ --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] SNOM 190 using 2 lines
I have a Snom 190 and setup two lines one for the local Asterisk and the Other for a remote asterisk. I can see that both likes register and in the web interface say they are ok. My problem is that line 1 takes precedence. I am not sure how to use line 2. If I go to the main setup page in the web browser for the snom I can change the Outgoing Identity: Line 2 and that works but its like switching me to the other network. What I was hoping to do was to setup my Function Keys to dial out one wither line 1 or line 2. Is that Possible? Am I missing anything? P1 is set to line : sip:[EMAIL PROTECTED];user=phone P2 is set to line : sip:[EMAIL PROTECTED];user=phone But this seems to have no effect. The Function Key 2 seem to still default to line 1. Any ideas? -Jason ___ --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] trying to get SIP to work remotly.
I am working with Xten lite for now. I am able to register in but when I call out I cant hear anything. The caller on the other end can hear me just fine. Any ideas? I can get SIP to work fine internally. I also have all the ports open in the firewall including 1 20 -J ___ --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] voicemail to two emails?
Can this be done? I have a customer service que that if full go to v-mail. I would like to know how I can put two e-mail address for it to go to. Is that possible? Thanks! -J ___ --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] ring silent
I have a request to have an extension to ring silently or different When a call comes into a queue. This extension is a manager that is monitoring the queue that the customer server is taking calls in. Is this Possible? -J ___ --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] receive fax with asterisk
Receiving faxes with Asterisk. Is there a good resource for learning how to set this up? -J ___ --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] Astra 480 i doesn't re-register
I have 4 Aastra 480i CT IP SIP Phone Some time if I have to restart Asterisk the Phones don't register in. It takes a long time for them to register like about 3 hours. If I restart the phones they register in. Is there a way to change the time in witch the phones register in with the server? -Jason ___ --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] trying to make outbound calls
OK I am trying to setup outbound calling using two ZAP channels. This is a [EMAIL PROTECTED] install 1.5 works great but having trouble dialing out using the ZAP Channels. I can call in on the POT lines and it rings through. Problem is dialing out. Here is what it said: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1003-5785' -- Executing Macro(SIP/1003-3802, dialout-trunk|1|15125382214) in new stack -- Executing GotoIf(SIP/1003-3802, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/1003-3802, record-enable|1003|OUT) in new stack -- Executing GotoIf(SIP/1003-3802, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf(SIP/1003-3802, 1?5:8) in new stack -- Goto (macro-record-enable,s,5) -- Executing DBget(SIP/1003-3802, RecEnable=RECORD-OUT/1003) in new stack -- DBget: varname=RecEnable, family=RECORD-OUT, key=1003 -- DBget: Value not found in database. -- Executing SetVar(SIP/1003-3802, CALLFILENAME=OUT1003-20051108-231813-1131509893.12) in new stack -- Executing Goto(SIP/1003-3802, s|14) in new stack -- Goto (macro-record-enable,s,14) -- Executing GotoIf(SIP/1003-3802, 0?15:99) in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp(SIP/1003-3802, NO RECORDING NEEDED) in new stack -- Executing GotoIf(SIP/1003-3802, 1?7) in new stack -- Goto (macro-dialout-trunk,s,7) -- Executing GotoIf(SIP/1003-3802, 0?9) in new stack -- Executing SetCallerID(SIP/1003-3802, 6092392505) in new stack -- Executing SetGroup(SIP/1003-3802, OUT_1) in new stack -- Executing CheckGroup(SIP/1003-3802, 1) in new stack -- Executing SetVar(SIP/1003-3802, DIAL_NUMBER=15125382214) in new stack -- Executing SetVar(SIP/1003-3802, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/1003-3802, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(SIP/1003-3802, OUTNUM=15125382214) in new stack -- Executing Cut(SIP/1003-3802, custom=OUT_1|:|1) in new stack -- Executing GotoIf(SIP/1003-3802, 0?19) in new stack -- Executing Dial(SIP/1003-3802, ZAP/g1/15125382214) in new stack == Everyone is busy/congested at this time -- Executing Goto(SIP/1003-3802, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/1003-3802, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/1003-3802, outisbusy) in new stack -- Executing Playback(SIP/1003-3802, allison7/all-circuits-busy-now) in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/1003-3802, allison7/pls-try-call-later) in new stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro(SIP/1003-3802, hangupcall) in new stack -- Executing ResetCDR(SIP/1003-3802, w) in new stack -- Executing NoCDR(SIP/1003-3802, ) in new stack -- Executing Wait(SIP/1003-3802, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1003-3802' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/1003-3802' in macro 'outisbusy' == Spawn extension (from-internal, 915125382214, 2) exited non-zero on 'SIP/1003-3802' -- Executing Macro(SIP/1003-3802, hangupcall) in new stack -- Executing ResetCDR(SIP/1003-3802, w) in new stack -- Executing NoCDR(SIP/1003-3802, ) in new stack -- Executing Wait(SIP/1003-3802, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1003-3802' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1003-3802' ___ --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] trying to make outbound calls
OK I am trying to setup outbound calling using two ZAP channels. This is a [EMAIL PROTECTED] install 1.5 works great but having trouble dialing out using the ZAP Channels. I can call in on the POT lines and it rings through. Problem is dialing out. Here is what it said: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1003-5785' -- Executing Macro(SIP/1003-3802, dialout-trunk|1|15125382214) in new stack -- Executing GotoIf(SIP/1003-3802, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/1003-3802, record-enable|1003|OUT) in new stack -- Executing GotoIf(SIP/1003-3802, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf(SIP/1003-3802, 1?5:8) in new stack -- Goto (macro-record-enable,s,5) -- Executing DBget(SIP/1003-3802, RecEnable=RECORD-OUT/1003) in new stack -- DBget: varname=RecEnable, family=RECORD-OUT, key=1003 -- DBget: Value not found in database. -- Executing SetVar(SIP/1003-3802, CALLFILENAME=OUT1003-20051108-231813-1131509893.12) in new stack -- Executing Goto(SIP/1003-3802, s|14) in new stack -- Goto (macro-record-enable,s,14) -- Executing GotoIf(SIP/1003-3802, 0?15:99) in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp(SIP/1003-3802, NO RECORDING NEEDED) in new stack -- Executing GotoIf(SIP/1003-3802, 1?7) in new stack -- Goto (macro-dialout-trunk,s,7) -- Executing GotoIf(SIP/1003-3802, 0?9) in new stack -- Executing SetCallerID(SIP/1003-3802, 6092392505) in new stack -- Executing SetGroup(SIP/1003-3802, OUT_1) in new stack -- Executing CheckGroup(SIP/1003-3802, 1) in new stack -- Executing SetVar(SIP/1003-3802, DIAL_NUMBER=15125382214) in new stack -- Executing SetVar(SIP/1003-3802, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/1003-3802, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(SIP/1003-3802, OUTNUM=15125382214) in new stack -- Executing Cut(SIP/1003-3802, custom=OUT_1|:|1) in new stack -- Executing GotoIf(SIP/1003-3802, 0?19) in new stack -- Executing Dial(SIP/1003-3802, ZAP/g1/15125382214) in new stack == Everyone is busy/congested at this time -- Executing Goto(SIP/1003-3802, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/1003-3802, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/1003-3802, outisbusy) in new stack -- Executing Playback(SIP/1003-3802, allison7/all-circuits-busy-now) in new stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/1003-3802, allison7/pls-try-call-later) in new stack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro(SIP/1003-3802, hangupcall) in new stack -- Executing ResetCDR(SIP/1003-3802, w) in new stack -- Executing NoCDR(SIP/1003-3802, ) in new stack -- Executing Wait(SIP/1003-3802, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1003-3802' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/1003-3802' in macro 'outisbusy' == Spawn extension (from-internal, 915125382214, 2) exited non-zero on 'SIP/1003-3802' -- Executing Macro(SIP/1003-3802, hangupcall) in new stack -- Executing ResetCDR(SIP/1003-3802, w) in new stack -- Executing NoCDR(SIP/1003-3802, ) in new stack -- Executing Wait(SIP/1003-3802, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1003-3802' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1003-3802' ___ --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] asterisks talking to asterisks
I have a request. I have a server in Texas And one in NJ. Is it possible for the system in Texas to log into the system in NJ so that Extensions can call each other? -J ___ --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] Uninstall AMP
Wow that was mean. -J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian C. Fertig Sent: Friday, November 04, 2005 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Uninstall AMP rm rf / ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: Friday, November 04, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Uninstall AMP Hi! How do I uninstall AMP and FOP from my Asterisk? Regards Anders Svensson This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] redial needs the 1 before the prefix howto?
When I go through me call logs I want to redial. The problem is there is no 1 in the number and it won't go through. How do I set it to add the 1 in front of a 10 digit number? -J ___ --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] set local area code
How do you set it up so that you don't have to dial you area code ie 512 ? -J ___ --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] Looking por a provider to work with asterisk
I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking por a provider to work with asterisk
I am in the US. Texas. -J (= -Original Message- From: WideVOIP [mailto:[EMAIL PROTECTED] Sent: Friday, November 04, 2005 10:39 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Looking por a provider to work with asterisk If you are in europe we can provide you sip and iax for asterisk Best regards Thierry [EMAIL PROTECTED] Tel : +33 (0)3 90 40 06 75 Fax: +33 (0)3 90 40 06 76 http://www.widevoip.com -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jason Brashear Envoyé : jeudi 3 novembre 2005 17:03 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --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] How to configure Asterisk through webmin
Alex We paid some one for your product they did a install for us but later we found that it was a demo. The $295.00 Was paid directly to this person that Said that they were an affiliate of yours. Is there anything that we can do? I would love to talk to you off line about this. -Jason Austin Texas From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn Sent: Friday, November 04, 2005 10:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin Hi Seshu, I would be happy to walk you (or anyone else who may be interested) through the Thirdlane PBX Manager features, to explain that while it wont magically configure Asterisk for you, it does help quite a bit. It is all really about the expectations and the target audience what is a good tool for some is too limiting for the others, and whatever is not limiting may appear too complex and not immediately useful. Please contact me off list at [EMAIL PROTECTED], or even better, we could spend a half an hour on the phone that may change your opinion. Best regards, Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Friday, November 04, 2005 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin The Thirdlane PBX Manager solution is just a few perl scripts. This is no better than what you can do by directly modifying the Asterisk Config files or many Open Source GUIs like Phonecall etc you have out there. Infact Areski's A2Billing has a good extension configurator in the solution. So that may be something you can consider. Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pikoro Sent: Thursday, November 03, 2005 7:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to configure Asterisk through webmin I tried the third lane asterisk manager thingy for webmin and let me tell you, it did not work. Only made things harder and i had to result to making the configuration by hand in order to get asterisk to work. Going to email them today and ask for a refund. That webmin module by third lane looks like a good solution, but the thing i noticed by reading the manual was that there are quite a few references to you'll have to change that in the config file type lines. Basically, it's good for creating extensions, but nothing more. Aaron Stefan-Michael. Guenther (in-put GbR) wrote: On Thu, November 3, 2005 17:46, nr k said: Hi allI configured asterisk and webmin.i dont know how tointegrate webmin with asterisk and how to accessasteriskthrough webmin.pls do the needful.regardsramakrishnan.n Asterisk is not managed through webmin. Webmin is a tool to helpadminister the rest of the server. and Asterisk, too:Have a look at THIRD LANE ASTERISK PBX MANAGERhttp://www.thirdlane.com/opensource.htm#managerStefan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking por a provider to work with asterisk
Thank you Gleim I will look into that. -Jason From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gleim, Jason Sent: Friday, November 04, 2005 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Looking por a provider to work with asterisk Jason, Back in August there was a post of a sip.conf and extensions.conf that would setup Asterisk to work with Vonage. I havent tried it yet but the user that posted reported success. Search the archive for Asterisk and Vonage and you should be able to find it or e-mail me off-list and Ill send you a copy. J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brashear Sent: Thursday, November 03, 2005 11:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --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] How to configure Asterisk through webmin
I just wanted to let every know that My complaint is not with Thirdlane. The Webmin module that they wrote is awesome and is well worth Getting. I have had a problem with a person that took us for a loop no pun intended. Alex has been a wonderful help and I would defiantly suggest his PBX Manager to anyone that wants a solid web based Asterisk Management tool. We found him online at www.thindlane.com. Please let me know if anyone else has had trouble with Daniel McNew his Company is Netvoci www.netvoci.com Be on the lookout. Thanks all. -Jason From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brashear Sent: Thursday, November 03, 2005 10:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin Alex We paid some one for your product they did a install for us but later we found that it was a demo. The $295.00 Was paid directly to this person that Said that they were an affiliate of yours. Is there anything that we can do? I would love to talk to you off line about this. -Jason Austin Texas From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn Sent: Friday, November 04, 2005 10:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin Hi Seshu, I would be happy to walk you (or anyone else who may be interested) through the Thirdlane PBX Manager features, to explain that while it wont magically configure Asterisk for you, it does help quite a bit. It is all really about the expectations and the target audience what is a good tool for some is too limiting for the others, and whatever is not limiting may appear too complex and not immediately useful. Please contact me off list at [EMAIL PROTECTED], or even better, we could spend a half an hour on the phone that may change your opinion. Best regards, Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Friday, November 04, 2005 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin The Thirdlane PBX Manager solution is just a few perl scripts. This is no better than what you can do by directly modifying the Asterisk Config files or many Open Source GUIs like Phonecall etc you have out there. Infact Areski's A2Billing has a good extension configurator in the solution. So that may be something you can consider. Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pikoro Sent: Thursday, November 03, 2005 7:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to configure Asterisk through webmin I tried the third lane asterisk manager thingy for webmin and let me tell you, it did not work. Only made things harder and i had to result to making the configuration by hand in order to get asterisk to work. Going to email them today and ask for a refund. That webmin module by third lane looks like a good solution, but the thing i noticed by reading the manual was that there are quite a few references to you'll have to change that in the config file type lines. Basically, it's good for creating extensions, but nothing more. Aaron Stefan-Michael. Guenther (in-put GbR) wrote: On Thu, November 3, 2005 17:46, nr k said: Hi allI configured asterisk and webmin.i dont know how tointegrate webmin with asterisk and how to accessasteriskthrough webmin.pls do the needful.regardsramakrishnan.n Asterisk is not managed through webmin. Webmin is a tool to helpadminister the rest of the server. and Asterisk, too:Have a look at THIRD LANE ASTERISK PBX MANAGERhttp://www.thirdlane.com/opensource.htm#managerStefan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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] sill looking for a provider
Is there a provider that has good support and answers the phone? (= I need to get lines for my Asterisk server and want to move from broadvoice.com. So far I havent been able to get anyone on the phone. Too funny.. Anyway. I want to work with a larger company one that we know wont dry up. Thanks all! -Jason ___ --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] Caller ID How does it get setup?
OK I am exhausted. I can't seem to figure out how to send a caller ID along with a Outbound call. Can you believe that I got Vonage to reset my Cisco ATA for $15.00 I then canceled my account! Well I was with them for over two years, now I am running Asterisk like the big boys! LOL... Anyway, Outbound Caller ID Hos is this done? I now use VoicePulse as my provider. -Jason ___ --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
