[Asterisk-Users] WTB: Grandstream Budgetone
Anybody looking to sell a Grandstream Budgetone? Contact me off list if you have one you want to get rid of. Thanks -Brian (brc007) ___ 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] Fax
Klaus-Peter Junghanns wrote: Hi, make sure you have echo cancelation disabled on that zaptel channel. I tried that but no joy. I've tried the gains at 0.8 and 1.5 I managed to get one fax to go out but it wouldn't repeat this feat! Simon -- Simon Faulkner - Dedicated Programmes 01538 303 900 - 07771 845 326 http://dpnet.co.uk ___ 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] running asterisk as non-root
Due to security reasons I want to run asterisk as a non root. http://voip-info.org/tiki-index.php?page=Asterisk+non-root This HOWTO works for great for me :) F ___ 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] Music on Hold - Context
hi I have set up a * box supporting 3 different companies but have some questions regarding MOH. Can MOH support multiple context or classes. Reason I ask each company would like to have different MOH sound files. Is this possible? yes, just specify multiple moh classes in musiconhold.conf and use each moh class for each company. example: company1 = mp3:/var/lib/asterisk/somemoh1 company2 = mp3:/var/lib/asterisk/somemoh2 company3 = mp3:/var/lib/asterisk/somemoh3 and now assign each moh class on your users/ivr/channels... matteo ___ 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] Kansas SIP or IAX Provider? - area codes corrected
Sorry, It's not dyslexic being easy. The REAL area codes are 620 with a 221 prefix 316 Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arnold Cavazos Jr. Sent: Saturday, February 14, 2004 9:25 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Kansas SIP or IAX Provider? NANPA.NET says there is no arecode 221: Is this code reserved for future use: Yes Is this code assigned: No Is this code in use:N --- Arnold Cavazos, Jr. abcjr at abcjr . net On Sat, Feb 14, 2004 at 09:10:56AM -0800, Paul Mahler wrote: Does anyone know a SIP or IAX provider for Kansas-area codes 620 and 221? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 ___ 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] Get new PRI working
Adam - I had a similar problem here in the UK using a Euro-ISDN PRI from BT. The key was to add in the line pridialplan=unknown into zapata.conf. Then it leapt into life in both directions. My files are below for your information. Rgds Tim Robinson, Basingstoke UK zaptel.conf --- # Config for a UK Euro-ISDN line span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk zapata.conf --- ; Configuration file [channels] usecallerid=yes language=en pridialplan=unknown signalling=pri_cpe switchtype=euroisdn group=1 context=inboundpstn channel = 1-15 channel = 17-31 Adam Goryachev wrote: Hi all, I received my shiny new TE405P on Friday, and after much fiddling and assistance from the irc channel, I got a OK status (telco reversed the TX/RX and I wired it wrong). Anyway, currently it works for inbound calls, but I can't seem to dialout on it. Here is the config from zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 and zapata.conf switchtype = euroisdn callgroup = 1 group = 2 busydetect = no immediate = yes context = remote signalling = pri_cpe ;stripmsd = 1 callprogress = no channel = 1-10 and here is the debug from asterisk: -- Executing Dial([EMAIL PROTECTED]:4569]/3, Zap/2/93454395||rT) in new stack Feb 15 15:58:27 DEBUG[20497]: app_dial.c:400 dial_exec: SIMPLE DIAL (NO URL) -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] Display (len= 7) [ Display (len= 7) [ 1 Display (len= 7) [ 1H Display (len= 7) [ 1Ho Display (len= 7) [ 1Hom Display (len= 7) [ 1Home Display (len= 7) [ 1Home Display (len= 7) [ 1Home 2 Display (len= 7) [ 1Home 2 ] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '651' ] Called Number (len=11) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '93454395' ] Sending Complete (len= 0) -- Called 2/93454395 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: STATUS (125) Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ] Cause data 0: 01 (1) Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (Cause) -- Processing IE 20 (Call State) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (Cause) -- Processing IE 30 (Progress Indicator) -- Channel 2, span 1 got hangup Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:2185 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/2-1 Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:1715 zt_hangup: Hangup: channel: 2 index = 0, normal = 17, callwait = -1, thirdcall = -1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] Feb 15 15:58:27 DEBUG[20497]: chan_zap.c:1133 zt_disable_ec:
RE: [Asterisk-Users] TE405P and dual Athlon systems
On Sun, 2004-02-15 at 00:46, mattf wrote: I'd like to know why too. I'm using a TE410P card in a dual Athlon XP system right now and it seems to be playing nicely with the dual Athlons, should I worry about something going wrong with my TE410P since it is basically the same card as the TE405P except it runs at a different voltage? Matt What motherboard are you using ? -- Juanjo sin .sig ___ 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] Looking for Incoming # for Area Code 713 (Houston, TX)
A customer is looking to change to VOIP but he wants a local incoming # where he lives. Anyone know a provider that offers them via SIP/IAX. I'll be running Asterisk to run all the features. Sincerely, William Suffill ___ 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] Pingtel Phones?
Hello All, Does anyone here have any experience with pingtel Xpressa hard phones? I am considering buying a couple. Already have Snom200s, but want something with better CTI and full duplex speakerphone. Michael ___ 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] TE405P and dual Athlon systems
a Tyan Thunder K7X S2468, it only has one 3.3v PCI slot but it seems happy with the te410p card. MATT--- -Original Message- From: Juan J. Sierralta P. [mailto:[EMAIL PROTECTED] Sent: Sunday, February 15, 2004 10:22 AM To: Asterisk Users Subject: RE: [Asterisk-Users] TE405P and dual Athlon systems On Sun, 2004-02-15 at 00:46, mattf wrote: I'd like to know why too. I'm using a TE410P card in a dual Athlon XP system right now and it seems to be playing nicely with the dual Athlons, should I worry about something going wrong with my TE410P since it is basically the same card as the TE405P except it runs at a different voltage? Matt What motherboard are you using ? -- Juanjo sin .sig ___ 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] Digium connectivity issue?
Brian West wrote: But CVS was alive the whole time! ;) bkw Um, no it wasn't. For all practical purposes, *.digium.com was dead. Why? Because even though there is a second cvs.digium.com out there on a different network, the nameservers digium.com are both on the same network - the network that was down. So, there was no way to actually get the addresses for CVS.DIGIUM.COM. Note to Digium: As a commercial interest, you should never consider the following a workable solution. ;; AUTHORITY SECTION: digium.com. 86308 IN NS marko.marko.net. digium.com. 86308 IN NS linux-support.net. ;; ADDITIONAL SECTION: marko.marko.net.50465 IN A 216.207.245.12 linux-support.net. 86271 IN A 216.207.245.1 Both nameservers on the same /24 = bad. There are several entities out there who will do secondary DNS for free. You might want to look into that. John ___ 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] Digium connectivity issue?
John Fraizer wrote: There are several entities out there who will do secondary DNS for free. You might want to look into that. If you pull their NS entries from one of the root servers, you get: digium.com. 172800 IN NS bos.nameserver.net. digium.com. 172800 IN NS linux-support.net. digium.com. 172800 IN NS marko.net. digium.com. 172800 IN NS phl.nameserver.net. digium.com. 172800 IN NS rdu.nameserver.net. digium.com. 172800 IN NS sjc.nameserver.net. digium.com. 172800 IN NS sou.nameserver.net. ;; ADDITIONAL SECTION: bos.nameserver.net. 172800 IN A 203.20.52.5 linux-support.net. 172800 IN A 216.207.245.1 marko.net. 172800 IN A 216.207.245.12 phl.nameserver.net. 172800 IN A 203.56.139.102 rdu.nameserver.net. 172800 IN A 64.245.56.205 sjc.nameserver.net. 172800 IN A 205.158.174.201 sou.nameserver.net. 172800 IN A 194.196.163.7 Looks like they just didn't update their digium.com zone to match. David -- David Coulsonemail: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer phone: (216) 533-6967 ___ 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] Music on Hold - Context
I was thinking about that... But here is my problem. We have 6 DID lines. We have it set up that all three companies share all lines.. Based off of the DNIS states what AutoAttendant they hit. So if I were to specify what channels the played certain MOH. Then that would mean Company 1 would have to come over on Channels 1-2 and so on. Any other thoughts. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matteo Brancaleoni Posted At: Sunday, February 15, 2004 8:27 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Music on Hold - Context Subject: Re: [Asterisk-Users] Music on Hold - Context hi I have set up a * box supporting 3 different companies but have some questions regarding MOH. Can MOH support multiple context or classes. Reason I ask each company would like to have different MOH sound files. Is this possible? yes, just specify multiple moh classes in musiconhold.conf and use each moh class for each company. example: company1 = mp3:/var/lib/asterisk/somemoh1 company2 = mp3:/var/lib/asterisk/somemoh2 company3 = mp3:/var/lib/asterisk/somemoh3 and now assign each moh class on your users/ivr/channels... matteo ___ 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] Get new PRI working
Why people don't have al least some respect about regulations? Sure that pridial=unknown solved that problem, but sadly you're overwriting the main class of service indication in ISDN... Unknown let to Class 5 switch manage (as the operator wish) understand your messages. The common sense shows that the correct parameters maybe pridial=local, where Class 5 switch don't add digits to the string. The correct way to do this is calling to your operator, and ask for the Class 5 brand and model (if the switch is Lucent, you need to use local. With the rest of switches you can use all TON's). Besides, the correct way to use PRI or S7 is to send ALWAYS the correct Nature of address, not always the same... In some parameter of your db you must define what prefix you use for national calls and international calls. The switch deletes the prefix when it was detected, and sends the correct Nature Of Address for that call. This is a normal behavior for all kind of switches. As far as I know, * always sends the same nature of address. What's the difference between local and unknown? Local never add digits and the calls will be treated mainly by the prefix that you send... unknown was designed to try to match with any rule (really the first rule) present in switch database. Best regards, Gus - Original Message - From: Tim Robinson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 15, 2004 12:16 PM Subject: Re: [Asterisk-Users] Get new PRI working Adam - I had a similar problem here in the UK using a Euro-ISDN PRI from BT. The key was to add in the line pridialplan=unknown into zapata.conf. Then it leapt into life in both directions. My files are below for your information. Rgds Tim Robinson, Basingstoke UK zaptel.conf --- # Config for a UK Euro-ISDN line span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk zapata.conf --- ; Configuration file [channels] usecallerid=yes language=en pridialplan=unknown signalling=pri_cpe switchtype=euroisdn group=1 context=inboundpstn channel = 1-15 channel = 17-31 Adam Goryachev wrote: Hi all, I received my shiny new TE405P on Friday, and after much fiddling and assistance from the irc channel, I got a OK status (telco reversed the TX/RX and I wired it wrong). Anyway, currently it works for inbound calls, but I can't seem to dialout on it. Here is the config from zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 and zapata.conf switchtype = euroisdn callgroup = 1 group = 2 busydetect = no immediate = yes context = remote signalling = pri_cpe ;stripmsd = 1 callprogress = no channel = 1-10 and here is the debug from asterisk: -- Executing Dial([EMAIL PROTECTED]:4569]/3, Zap/2/93454395||rT) in new stack Feb 15 15:58:27 DEBUG[20497]: app_dial.c:400 dial_exec: SIMPLE DIAL (NO URL) -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] Display (len= 7) [ Display (len= 7) [ 1 Display (len= 7) [ 1H Display (len= 7) [ 1Ho Display (len= 7) [ 1Hom Display (len= 7) [ 1Home Display (len= 7) [ 1Home Display (len= 7) [ 1Home 2 Display (len= 7) [ 1Home 2 ] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '651' ] Called Number (len=11) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '93454395' ] Sending Complete (len= 0) -- Called 2/93454395 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: STATUS (125) Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ] Cause data 0: 01 (1) Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (Cause) -- Processing IE 20 (Call State) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext:
[Asterisk-Users] Correct cvs checkout?
For the Stable cvs checkout, the asterisk.org site suggests: To check out code from our STABLE 1.0 Branch CVS repository for Asterisk ONLY: # cd /usr/src# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0_stable asterisk I'm assuming one should also consider zaptel and other portions if needed, or does the above actually check those directories as well? Rich ___ 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] Digium connectivity issue?
And without the secondaries knowing they're authoritative for the zone, things don't work right. John David Coulson wrote: John Fraizer wrote: There are several entities out there who will do secondary DNS for free. You might want to look into that. If you pull their NS entries from one of the root servers, you get: digium.com. 172800 IN NS bos.nameserver.net. digium.com. 172800 IN NS linux-support.net. digium.com. 172800 IN NS marko.net. digium.com. 172800 IN NS phl.nameserver.net. digium.com. 172800 IN NS rdu.nameserver.net. digium.com. 172800 IN NS sjc.nameserver.net. digium.com. 172800 IN NS sou.nameserver.net. ;; ADDITIONAL SECTION: bos.nameserver.net. 172800 IN A 203.20.52.5 linux-support.net. 172800 IN A 216.207.245.1 marko.net. 172800 IN A 216.207.245.12 phl.nameserver.net. 172800 IN A 203.56.139.102 rdu.nameserver.net. 172800 IN A 64.245.56.205 sjc.nameserver.net. 172800 IN A 205.158.174.201 sou.nameserver.net. 172800 IN A 194.196.163.7 Looks like they just didn't update their digium.com zone to match. David ___ 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] Overhead Paging
I've a Valcon V-2001A paging controller connected to an Adtran 750 FXS port. The V-2001A looks like an FXS loop start extension. When I call the extension, I can hear ringing tones and CallerID through the speaker, but the paging controller doesn't answer--it continues to ring. I also hear a relay clicking with each ring in the paging controller. Does anyone have experience with configuring these devices for paging? Thank you, Michael Welter ___ 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] Wifi Phones
Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ 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] Digium connectivity issue?
On Sun, Feb 15, 2004 at 12:18:03PM -0500, John Fraizer wrote: There are several entities out there who will do secondary DNS for free. I'll do secondary DNS if they want. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] Festival patch ?
Hello,
In contrib/README.festival talks about a patch to extend the Extension
logic of Asterisk to allow put quotes and vars to the Festival commmand.
I need this since I want to use , to add a silence. Something like
this hello, how are you.
Now if I execute Festival('Hello, how are you') Asterisk takes Hello
as the first argument and how are you as the second :(
Where I can find that patch ?
--
Juanjo sin .sig
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Re: [Asterisk-Users] Digium connectivity issue?
On Sun, 15 Feb 2004, John Fraizer wrote: Brian West wrote: But CVS was alive the whole time! ;) bkw Um, no it wasn't. For all practical purposes, *.digium.com was dead. Why? Because even though there is a second cvs.digium.com out there on a different network, the nameservers digium.com are both on the same network - the network that was down. So, there was no way to actually get the addresses for CVS.DIGIUM.COM. This is true, but no longer the case since other nameservers are now setup. Both nameservers on the same /24 = bad. Not to split hairs here, but this statement isn't necessarily true. If it read Both nameservers on the same physical network then it would be true. I've worked on systems that each of the 2 NS glue records were actually /32s located on multiple servers around the country. So even if part of the network was down, multiple servers are always reachable. Now lets return to our regular programming James ___ 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 Vonage -- possible? Other DID providers with Atlanta presence?
OK, time for this 1 day old Asterisk convert to start getting his feet wet :) I have just installed Asterisk on a Redhat box -- easy installation! I am not using any analog interface hardware, instead am going to try to test using my Vonage account. S. The question is, how would I set up Asterisk to replace my Motorola VT100v voice terminal that Vonage provided me? I have been through the manual, and determined it is using SIP, so how do you set up asterisk to send and receive from a SIP provider. Next (more advanced?) question, if I get a second SIP DID how would I set this up also. I am going to be developing some apps to use with Asterisk so plan on contributing here just as soon as I can :) Thanks in advance! --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.577 / Virus Database: 366 - Release Date: 2/4/2004 ___ 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] Official word from GalaxyVoice customer service
We don't support using Asterisk on your connection, but you're allowed to use an Asterisk box if you can get it to work. -- JustThe.net Internet New Media Services, Apple Valley, CA Steven J. Sobol, Geek In Charge / 888.480.4NET (4638) / [EMAIL PROTECTED] PGP: C57E 8B25 F994 D6D0 5F6B B961 EA08 9410 E3AE 35ED ___ 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 and Vonage -- possible? Other DID providers with Atlanta presence?
Replacing the sip terminal for Vonage isn't possible. The terminal is locked and will not allow access by the user to get the user/password info, and the user/password handshaking is encrypted which prevents it from being spied upon. The only way this will work is if you plug the analog output of the SIP terminal into an asterisk FXO card such as the X100p. I use this setup, but it has the drawback exacerbating echo problems. This issue is one of my personal pet peeves, but the ITSP's refuse to allow direct access because thier paranoid about fraud in the form of distributing user/passwords to multiple users. I suppose if you were determined, you could gain access to and dissassemble the firmware in the SIP box to try and get at the configuration info, but I suspect it won't be easy unless you have lots of reverse engineering horsepower at your disposal. Steve. On Sunday 15 February 2004 13:00, Tom Knox wrote: OK, time for this 1 day old Asterisk convert to start getting his feet wet :) I have just installed Asterisk on a Redhat box -- easy installation! I am not using any analog interface hardware, instead am going to try to test using my Vonage account. S. The question is, how would I set up Asterisk to replace my Motorola VT100v voice terminal that Vonage provided me? I have been through the manual, and determined it is using SIP, so how do you set up asterisk to send and receive from a SIP provider. Next (more advanced?) question, if I get a second SIP DID how would I set this up also. I am going to be developing some apps to use with Asterisk so plan on contributing here just as soon as I can :) Thanks in advance! --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.577 / Virus Database: 366 - Release Date: 2/4/2004 ___ 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
Re: [Asterisk-Users] Asterisk and Vonage -- possible? Other DID providers with Atlanta presence?
On Sun, 15 Feb 2004 16:00:04 -0500, Tom Knox wrote: I am not using any analog interface hardware, instead am going to try to test using my Vonage account. S. The question is, how would I set up Asterisk to replace my Motorola VT100v voice terminal that Vonage provided me? I have been through the manual, and determined it is using SIP, so how do you set up asterisk to send and receive from a SIP provider. Next (more advanced?) question, if I get a second SIP DID how would I set this up also. I used Vonage for a year, up until last month in fact. However, there doesn't seem to be a way to avoind using their ATA/MTA if you use their service. You won't be able to connect directly to their servers with your * box. SI switched to VoicePulse Connect. I actually prefer their 2.9 cent/minute rates as opposed to a flat $35/month. I rarely use 1000 minutes/month so I'm saving money over Vonage. There was a down side in that VP doesn't offer DIDs in my area. Therefore I still have 2 POTS lines for incomming calls, but that was likely going to stay that way anyhow, as abackup to when the ISP or ITSP have problems. Also, in order to pass SIP through your router you're going to have open up potentially a lot of ports. I prefer to connect to VPC using IAX2, which requires that I open only one port. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] I am easily satisfied with the very best - Winston Churchill The questions arisen, is this a prison? Some say it is, but I say it isn't. - Ian Hunter ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] Wifi Phones
Those phones look good, but, only have 10 milliwatt output. Have you looked at these: http://www.spectralink.com/products/nl-wts.html 100mw output. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miguel Cavazos Sent: 15 February 2004 12:39 To: Asterisk Users Subject: [Asterisk-Users] Wifi Phones Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ 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] RE: [Asterisk-Users] Spanish indications configurationº
Title: Mensaje Hola, ahi va la sección [es] para el indications.conf [es] description = Spain ringcadence = 1500,3000 dial = 425 busy = 425/200,0/200 ring = 425/1500,0/3000 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 callwaiting = 425/175,0/175,425/175,0/3500 dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 record = 1400/500,0/15000 info = 950/330,0/1000 dialout = 500 www.avanzada7.com Sergio Serrano RevueltoRD Manager Avanzada 7 [EMAIL PROTECTED] tel: fax: mobile: (+0034) 951014947(+0034) 951010922618747717 Signature powered by Plaxo Want a signature like this? Add me to your address book... -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de dfmEnviado el: viernes, 13 de febrero de 2004 12:18Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Spanish indications configurationº Hi all We've been using * for a while here in Spain, but some people has told us that they have problems when they type an extension calling to us. I've been trying to find out what's going on, and it's an issue that only happens with some ISDN and analog calls, not from mobile calls as long as i have observe. My concern is about the indications.conf Spanish telco lines configuration, Is in the * list any Spanish user that can share this configuration with me and see if it's ok?? i would really appreciate it. Diego
Re: [Asterisk-Users] HELP!!!! Having problems Starting Asterisk
Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Try shutting down all * processes (including mpg123). Now, see if your audio works normally. If not, rmmod the zaptel/fx? modules, and see if that works. If not, you should start by getting your audio on the consloe to work normally first, then, check with the zap/etc modules loaded, then try * . One step at a time. Tim Thanks for the advice but I don't have any console audio device, I'm still working on it so any other advise would be appreciated, do you think I need to rebuild the system? Thanks Robb ___ 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] HELP!!!! Having problems Starting Asterisk
I had this problem with an old 16bit Sound Blaster Card. Threw the card away and put in a cheap ?3.50 PCI card. Works a dream now. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Boardman Sent: 15 February 2004 23:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] HELP Having problems Starting Asterisk Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Try shutting down all * processes (including mpg123). Now, see if your audio works normally. If not, rmmod the zaptel/fx? modules, and see if that works. If not, you should start by getting your audio on the consloe to work normally first, then, check with the zap/etc modules loaded, then try * . One step at a time. Tim Thanks for the advice but I don't have any console audio device, I'm still working on it so any other advise would be appreciated, do you think I need to rebuild the system? Thanks Robb ___ 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] HELP!!!! Having problems Starting Asterisk
Robert Boardman wrote: Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Try commenting out all of the entries in musiconhold.conf. Then, stop asterisk. Then, do a pstree from the shell prompt and see if you still have some mpg123 processes hanging around. If you do, do a killall -9 mpg123 from the shell prompt. Then, start Asterisk. Look at the output of pstree again and make sure that mpg123 isn't being started. Something that I noticed was that there is a new set of mpg123 processes started for every class of musiconhold that you have specified. Making sure that mpg123 won't start at all will at least isolate if that is what is preventing Asterisk from starting properly. John -- who has NEVER been at the console while working with Asterisk but does have everything I want, including meetme musiconhold working after much 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
[Asterisk-Users] About HT-286
Hi, Did HT-286 a good answer to put into my company and with both * and my old pbx use VOip and Normal telephones without change any kind of structure ? Like with HT-286 can i just plug it using the RJ-45 part in the * and the RJ11 in my old pbx and have both working with my normal telephone ? And make this . My normal old pbx has channel from 20 to 80 (Normal local phones etc) with * i can then put (90 to 100) and has the VOIP channels both with the same phone right ? Can anyone have the answer for this question ? Thanks alot. ___ 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] multiple context in sip.conf
Hi all, Is it possible to have multiple context= for user configuration in sip.conf? Regards, Antonio Rabena ___ 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] Get new PRI working
[EMAIL PROTECTED] wrote: Hi all, I received my shiny new TE405P on Friday, and after much fiddling and assistance from the irc channel, I got a OK status (telco reversed the TX/RX and I wired it wrong). Anyway, currently it works for inbound calls, but I can't seem to dialout on OK, I found the basic problem... It seems to be that I wasn't dialling the correct number. I was dialling either 8 digits (93454395) or else 10 digits (0293454395) which are the usual conventions for Australia. What I was finally told (by coppice on IRC, thank you, he helped me all the way from diagnosing the original wiring problems through to the current mostly working situation) is that I should drop the 0, so 293454395 worked perfectly for dialling local numbers, and other 10 digit numbers work if I drop the leading 0. However, I still can't dial any mobiles (all mobiles are 10 digit like 0402 xxx xxx) which I thought would have been the same as the above local/std calls. Also, I can't dial freecall numbers (1800 xxx xxx) etc. If anyone has any hints on possible number formats for calling these sorts of numbers, I would appreciate it. Of course, if I eventually work it out, I will post back to the list for the sake of those people who do search/read the archives... Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Digium connectivity issue?
Yep thats what it looks like :( bkw On Sun, 15 Feb 2004, John Fraizer wrote: And without the secondaries knowing they're authoritative for the zone, things don't work right. John David Coulson wrote: John Fraizer wrote: There are several entities out there who will do secondary DNS for free. You might want to look into that. If you pull their NS entries from one of the root servers, you get: digium.com. 172800 IN NS bos.nameserver.net. digium.com. 172800 IN NS linux-support.net. digium.com. 172800 IN NS marko.net. digium.com. 172800 IN NS phl.nameserver.net. digium.com. 172800 IN NS rdu.nameserver.net. digium.com. 172800 IN NS sjc.nameserver.net. digium.com. 172800 IN NS sou.nameserver.net. ;; ADDITIONAL SECTION: bos.nameserver.net. 172800 IN A 203.20.52.5 linux-support.net. 172800 IN A 216.207.245.1 marko.net. 172800 IN A 216.207.245.12 phl.nameserver.net. 172800 IN A 203.56.139.102 rdu.nameserver.net. 172800 IN A 64.245.56.205 sjc.nameserver.net. 172800 IN A 205.158.174.201 sou.nameserver.net. 172800 IN A 194.196.163.7 Looks like they just didn't update their digium.com zone to match. David ___ 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] multiple context in sip.conf
no but you can do this: context=specialuser in extensions.conf do this [specialuser] include=othercontext include=yetanothercontext On Mon, 16 Feb 2004, Antonio Rabena wrote: Hi all, Is it possible to have multiple context= for user configuration in sip.conf? Regards, Antonio Rabena ___ 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] Wifi Phones
I don't know if anyone else has worked with Spectralink, but I tried to get some demo units to test with a while back and I was really disapointed. At first they claimed they were SIP complient. Then they sent me a contract for the demo. They wouldn't send a demo unless I agreed to have them do an onsite install to the tune of nearly $5000. I would only have been obligated for the install fees if I decided to buy, but not being happy about the install fees and wanting to know why I learned more about how the technology works. My sales rep shared that the phones aren't actually SIP compliant and work through the SVP server to provide SIP compliance to the PBX connection. He also shared that they want provide warranty on their products unless they do the install. Really turned me off, smelled very proprietary, although SVP QOS is pretty cool. I just received a Wisip last week and it looks pretty promising, although I think my unit may be damaged. Pulver support was also very up front with me that they technically only support the Wisip with FWD. They have been good to work with me so far, even rushed me a unit when I explained I was researching for a large purchase. It definetly connects to asterisk, but I think my unit has a bad antena or transmiter. The audio drops in and out and the signal strength indicator shows only one bar even when only about 25' from the access point. Anyone else actually gotten their hands one of these to try with Asterik? I would like to buy a couple hundred of them, but they need work reliably. I would love to compare notes with someone to see if my experiences are a typical. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Craig Waddington [EMAIL PROTECTED]: Those phones look good, but, only have 10 milliwatt output. Have you looked at these: http://www.spectralink.com/products/nl-wts.html 100mw output. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miguel Cavazos Sent: 15 February 2004 12:39 To: Asterisk Users Subject: [Asterisk-Users] Wifi Phones Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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 Vonage -- no make that VoicePulse Connect :)
Thanks Michael, VoicePulse does have local number, so I just provisioned one :) Now I am setting it up, no problem so far, the next question is. If I get 2 simultaneous calls on my inbound will one ring busy or will asterisk handle this for me? I would like to be able to receive multiple simultaneous calls if possible for an application I am developing. Also, I assume to go through my router (NAT) I just need to open up the one port, 5036, right? Whopee! I can give Vonage it's device and high prices back soon :) Too bad they couldn't play nice and allow other devices on their service. I used Vonage for a year, up until last month in fact. However, there doesn't seem to be a way to avoind using their ATA/MTA if you use their service. You won't be able to connect directly to their servers with your * box. SI switched to VoicePulse Connect. I actually prefer their 2.9 cent/minute rates as opposed to a flat $35/month. I rarely use 1000 minutes/month so I'm saving money over Vonage. There was a down side in that VP doesn't offer DIDs in my area. Therefore I still have 2 POTS lines for incomming calls, but that was likely going to stay that way anyhow, as abackup to when the ISP or ITSP have problems. Also, in order to pass SIP through your router you're going to have open up potentially a lot of ports. I prefer to connect to VPC using IAX2, which requires that I open only one port. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] I am easily satisfied with the very best - Winston Churchill The questions arisen, is this a prison? Some say it is, but I say it isn't. - Ian Hunter --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.577 / Virus Database: 366 - Release Date: 2/4/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call File Troubles
Hi all,
I'm having a hard time getting my calls to complete when creating call files
and putting them in the /var/spool/asterisk/outgoing directory.
Asterisk processes the file just fine, but the digits that enter my PSTN
switch are only 2 digits, not 10. For example:
Filename: 1.call (My call file)
Channel:Zap/g2/12125551212
MaxRetries:2
RetryTime:60
WaitTime:30
Context:office
Extension:100
Priority:1
extensions.conf
[office]
exten = 100,1,Dial(SIP/100)
exten = _1NXXNXX,1,Dial,Zap/g2/${EXTEN}
When I dial a number from my sip phone (extension 100) calls go out just
fine. But when I use a call file, only the first two digits of the
destination number are outpulsed, even though asterisk is saying otherwise.
Here is a snippet from the console:
*CLI
-- Attempting call on Zap/g2/12125551212 for [EMAIL PROTECTED]:1 (Retry 1)
Channel Zap/25-1 was answered.
-- Executing Dial(Zap/25-1, SIP/100) in new stack
-- Called 100
-- SIP/100-7c33 is ringing
-- SIP/100-7c33 answered Zap/25-1
== Spawn extension (office, 100, 1) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
Feb 15 21:32:28 NOTICE[393235]: pbx_spool.c:206 attempt_thread: Call
completed to Zap/g2/12125551212
So here * is saying that it outpulsed 12125551212 but on my switch it only
recieved the digits12 (the first two digits of the number)
I am using the TE410P em wink signalling. Asterisk CVS-01/07/04
Any help is greatly appreciated.
TIA
-Seth
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RE: [Asterisk-Users] Get new PRI working
[EMAIL PROTECTED] wrote: However, I still can't dial any mobiles (all mobiles are 10 digit like 0402 xxx xxx) which I thought would have been the same as the above local/std calls. Also, I can't dial freecall numbers (1800 xxx xxx) etc. If anyone has any hints on possible number formats for calling these sorts of numbers, I would appreciate it. OK, I worked out pretty much all my dialout problems. I set pridialplan=local. One thing I missed is that while a reload picks up some changes from zapata.conf, this doesn't seem to be one of them Now I can dial local, long distance, mobile, freecall, etc numbers. Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Easy access to visual busy status and call transfer buttons
I agree, I think they would be useful too. Just don't know of anyway to currently do it with *. Would also like to have a working intercom option. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Jeff Crews [EMAIL PROTECTED]: I want to say thanks for the great posts to this list...I learn something know about every day reading this list. Anyway...I have been using * in a test environment for 10 months and really like it. I have PRI to the PSTN and SIP to 2 Snoms and 1 Cisco 7960. I have frequently used ATT/Lucent/Avaya phone systems such as Definity or Partner that provide the ability to assign LEDs on individual phones that allow you to visually see the status of specific extensions to determine if the extension is on a call, do not disturb, or idle. If I use * to speak SIP to the phones...such as the Cisco 7960...how do you provide users with this easy visual way to see the status of an extension? Further...using a button associated with these busy status indicators makes transferring calls fast. I see some people use software on a PC to get this functionality. It still seems that there should be a way to do this on a SIP phone. Am I the only person that thinks these status LEDs are valuable? Jeff ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] Switch brands, speeds, etc.
Not really switch avice, but it you are rewiring anyway, ditch the cat3. A good portion of wiring cost is the labor to pull the wire. Since you are doing it anyone I would pull enough to handle the phones too. We are shopping too, but for an enterprise wide solution (about 1000-1500 ports WAN wide). I like the HP ProCurve 48 port for its good warranty, rep, and, low cost per port. If we decide to go with power over ethernet, however, we are leaning towards Cisco. I am curious what others are using. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Bob Klepfer [EMAIL PROTECTED]: The short of it: In light of the recent Netgear posts, I'm just curious if anyone has preferences for brands of switches - we're wiring a parallel network of 10BaseT over existing cat3 for the IP phones in our office space. The long of it: --- Our setup: * Office of 10 people spread out in 2000-3000 sq.ft. * Space previously used as computer learning center, chock full of cat-3 and multiple rj-45 jacks per wall plate. * We're rewiring anyway - company growth + lack of planning has led to switches and hubs strung everywhere * I've convinced the boss to let me implement an asterisk server, replacing the unholy phone concoction we have now * No external VOIPat least not yet. * MUCH data flying back and forth from computers in labs to offices and vice versa So we were thinking of using some of the existing cat3 for just the IP phones and stringing some cat5e alongside for intranet. Buy a cheap 10BaseT switch (SmallDog has a refurbished Asante 5324 24-port cheap) for the cat3 lines and feed that to our * server's eth1. We're geeks, but not really networking geeks, so I thought I'd ask the list populace at large if they had comments/recommendations. Best, Bob Klepfer Photon-X, Inc. ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] Constant crashes with Asterisk 0.7.2
That is also when I have seen a few crashes of 0.7.2, when doing lots of edits. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting John Fraizer [EMAIL PROTECTED]: Geert Nijpels wrote: I run 0.7.2 and have no crashes. Do you have an error message? If this is reproducable, please update to latest stable CVS. Generate a core file a make a backtrace. Then post a bug at http://bugs.digium.com/ If you need help please email. Kind regards, Geert Nijpels OK. I'm now running Asterisk CVS-02/08/04-19:17:46 Here are the backtraces from the cores I have from today: #0 0x40026cf2 in pthread_mutex_lock () from /lib/i686/libpthread.so.0 #1 0x42075bea in free () from /lib/i686/libc.so.6 #2 0x08053117 in ast_verbose (fmt=0x813b7b0 \005) at logger.c:524 #3 0x08067b47 in ast_context_add_include2 (con=0x813ba00, value=0x0, registrar=0x40026ce0 U\211å\203ì\030\211}ü\213}\b\211]ô\211uø\213G\fèÿÙÿÿ\201Ã\217) at pbx.c:3227 #4 0x41f49e77 in pbx_load_module () at pbx_config.c:1655 #5 0x41f46e3f in reload () at pbx_config.c:1714 #6 0x08056062 in ast_module_reload () at loader.c:169 #7 0x0806e58a in handle_reload (fd=9, argc=1, argv=0x467c95fc) at cli.c:106 #8 0x0806e37a in ast_cli_command (fd=9, s=0x40026ce0 U\211å\203ì\030\211}ü\213}\b\211]ô\211uø\213G\fèÿÙÿÿ\201Ã\217) at cli.c:1007 #9 0x08087842 in netconsole (vconsole=0x80c7f08) at asterisk.c:214 #10 0x40026881 in pthread_detach () from /lib/i686/libpthread.so.0 #0 0x42074d8e in calloc () from /lib/i686/libc.so.6 #1 0x42075bcc in free () from /lib/i686/libc.so.6 #2 0x08053117 in ast_verbose ( fmt=0x8130168 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to\n) at logger.c:524 #3 0x41f792b8 in dial_exec (chan=0x8153b78, data=0x41f7a3de) at app_dial.c:631 #4 0x08064359 in pbx_exec (c=0x8153b78, app=0x811dea8, data=0x469c974c, newstack=1) at pbx.c:396 #5 0x0806b8d0 in pbx_extension_helper (c=0x42139c80, context=0x8153cd0 allaccess, exten=0x8153dc4 81370109, priority=2, callerid=0x80e0d58 70109, action=1106757012) at pbx.c:1171 #6 0x0806634c in ast_pbx_run (c=0x41f7c594) at pbx.c:1655 #7 0x0806bfa1 in pbx_thread (data=0x) at pbx.c:1880 #8 0x40026881 in pthread_detach () from /lib/i686/libpthread.so.0 #0 0x42074d8e in calloc () from /lib/i686/libc.so.6 #1 0x42075bcc in free () from /lib/i686/libc.so.6 #2 0x41a0d751 in attempt_transmit (data=0x814c110) at chan_iax.c:1185 #3 0x08052159 in ast_sched_runq (con=0x80ef528) at sched.c:376 #4 0x41a110e2 in network_thread (ignore=0x0) at chan_iax.c:4530 #5 0x40026881 in pthread_detach () from /lib/i686/libpthread.so.0 #0 0x42074d8e in calloc () from /lib/i686/libc.so.6 #1 0x42075bcc in free () from /lib/i686/libc.so.6 #2 0x08056a33 in ast_destroy (ast=0x81063d0) at config.c:109 #3 0x0807c341 in init_manager () at manager.c:928 #4 0x08056015 in ast_module_reload () at loader.c:158 #5 0x0806e58a in handle_reload (fd=15, argc=1, argv=0x469c95fc) at cli.c:106 #6 0x0806e37a in ast_cli_command (fd=15, s=0x Address 0x out of bounds) at cli.c:1007 #7 0x08087842 in netconsole (vconsole=0x80c7f08) at asterisk.c:214 #8 0x40026881 in pthread_detach () from /lib/i686/libpthread.so.0 #0 0x420751ce in calloc () from /lib/i686/libc.so.6 #1 0x42074827 in calloc () from /lib/i686/libc.so.6 #2 0x42073ef1 in malloc () from /lib/i686/libc.so.6 #3 0x41d93113 in new_iax (sin=0x42139c80, lockpeer=1) at chan_iax2.c:576 #4 0x41d8fa03 in find_callno (callno=0, dcallno=0, sin=0x81201d0, new=2, lockpeer=-1) at chan_iax2.c:791 #5 0x41d915f6 in iax2_do_register (reg=0x81201d0) at chan_iax2.c:4979 #6 0x41d8f658 in reload () at chan_iax2.c:5750 #7 0x08056062 in ast_module_reload () at loader.c:169 #8 0x0806e58a in handle_reload (fd=15, argc=1, argv=0x469c95fc) at cli.c:106 #9 0x0806e37a in ast_cli_command (fd=15, s=0x Address 0x out of bounds) at cli.c:1007 #10 0x08087842 in netconsole (vconsole=0x81201d0) at asterisk.c:214 #11 0x40026881 in pthread_detach () from /lib/i686/libpthread.so.0 #0 0x42074d8e in calloc () from /lib/i686/libc.so.6 #1 0x42075bcc in free () from /lib/i686/libc.so.6 #2 0x0806989b in __ast_context_destroy (con=0x0, registrar=0x41f4a9ed pbx_config, lock=1) at pbx.c:4060 #3 0x080699e0 in ast_context_destroy (con=0xb2134945, registrar=0xb2134945 Address 0xb2134945 out of bounds) at pbx.c:4132 #4 0x41f46e3a in reload () at pbx_config.c:1709 #5 0x08056062 in ast_module_reload () at loader.c:169 #6 0x0806e58a in handle_reload (fd=10, argc=1, argv=0x467c95fc) at cli.c:106 #7 0x0806e37a in ast_cli_command (fd=10, s=0xb2134945 Address 0xb2134945 out of bounds) at cli.c:1007 #8 0x08087842 in netconsole (vconsole=0x80c7f08) at asterisk.c:214 #9 0x40026881 in
Re: [Asterisk-Users] Call File Troubles
Hate to reply to my own post here, but it was a careless mistake on my part.
(My switch was giving imediate answer supervision on the inbound trunks)
Sorry for the trouble...
Thanks,
Seth
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 15, 2004 9:25 PM
Subject: [Asterisk-Users] Call File Troubles
Hi all,
I'm having a hard time getting my calls to complete when creating call
files
and putting them in the /var/spool/asterisk/outgoing directory.
Asterisk processes the file just fine, but the digits that enter my PSTN
switch are only 2 digits, not 10. For example:
Filename: 1.call (My call file)
Channel:Zap/g2/12125551212
MaxRetries:2
RetryTime:60
WaitTime:30
Context:office
Extension:100
Priority:1
extensions.conf
[office]
exten = 100,1,Dial(SIP/100)
exten = _1NXXNXX,1,Dial,Zap/g2/${EXTEN}
When I dial a number from my sip phone (extension 100) calls go out just
fine. But when I use a call file, only the first two digits of the
destination number are outpulsed, even though asterisk is saying
otherwise.
Here is a snippet from the console:
*CLI
-- Attempting call on Zap/g2/12125551212 for [EMAIL PROTECTED]:1 (Retry 1)
Channel Zap/25-1 was answered.
-- Executing Dial(Zap/25-1, SIP/100) in new stack
-- Called 100
-- SIP/100-7c33 is ringing
-- SIP/100-7c33 answered Zap/25-1
== Spawn extension (office, 100, 1) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
Feb 15 21:32:28 NOTICE[393235]: pbx_spool.c:206 attempt_thread: Call
completed to Zap/g2/12125551212
So here * is saying that it outpulsed 12125551212 but on my switch it only
recieved the digits12 (the first two digits of the number)
I am using the TE410P em wink signalling. Asterisk CVS-01/07/04
Any help is greatly appreciated.
TIA
-Seth
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Re: [Asterisk-Users] channel bank - Adit 600
I purchased the Adit 600 to fix problems similar to what you are seeing and it worked like a charm. Before investing in the Adit 600 though I would test to make sure your pbx is actually sending the call supervision info for the call supervision. If plug in a phone with a telco lit display or buttons the lites should flash if your remote end hangs. Before doing this test, I thought we didn't have call sup, but turns out it just had a 6-11 second delay. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting denzel-infotechs [EMAIL PROTECTED]: hi! I would like to check the apllicability of Adit 600 and Adtran 750 in converting FXO + FXS to E1/T1 channel to be sent to a * voip box. we r currently using pleidaes channel bank and it has the problem FXO lines hanging forever.(Don't disconnect). Bundle of FXOs are obtained from inhouse PBX. What I'm looking for is a simple Channel Bank doing, FXO + FXS(bundle) - E1/T1 with Answer/Disconnect supervision Voice Activity Detection. impedance matching, minimized echo, etc... Can I achieve the above simple task with Adit600 ? Eventhough it's been recommended by * h/w guide it looks someting different to a channel bank. Perhaps someone could pop an Idea on this. denzel. ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] merlin legend / * as ld gw
Can anyone offer adivce for connecting * to a merlin legend ? I'd like to use a t1 interface to connect the two, * will be used as a long distance voip gateway in this scenario. Is this possible using a digium t100p ? Thanks, Chris Clifton ___ 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] merlin legend / * as ld gw
On Sun, 15 Feb 2004, Chris Clifton wrote: Can anyone offer adivce for connecting * to a merlin legend ? I'd like to use a t1 interface to connect the two, * will be used as a long distance voip gateway in this scenario. Is this possible using a digium t100p ? Is it possible? Absolutely. You'll find this to be as straightforward as it sounds - if you run into trouble, it will likely be things like remembering (or not) to use a t1 crossover cable, proper timing, and other things of that nature. 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
