Re: [Asterisk-Users] Asterisk Meridian Tie Line
I'm running pretty much the same config in Belgium. Here's what I use: zaptel.conf: span=1,1,0,ccs,hdb3 # no CRC4 used here bchan=1-15,16-31 dchan=16 zapata.conf: [trunkgroups]trunkgroup = 1,16spanmap = 1,1,1 [channels]context=incoming-priswitchtype=euroisdnpridialplan=nationalprilocaldialplan=nationalsignalling=pri_cpegroup=1channel = 1-15,17-31 Works for me, hope it can help you! On 5/18/06, Steve Totaro [EMAIL PROTECTED] wrote: Andy Kirby wrote: I am new to the group but have searched the doc's FAQ's etc before posting here. We are attempting tie our asterisk server/service to the building's PBX, the building is in the UK and the local PBX is a meridian option 11 installed and mainteined by BT. BT Have installed a NTBK50AA E1-PRI card in the meridian with daughter cards NTBK51AA (D channel) and NTAK20BD (Clocking)I have asked BT to configure the card as a Master (Exchange end) E1 Euro ISDN (Just like a standard ISDN30e)They claim to have done this in line with the model they use to interface to Cisco routers etc.I have installed a Digium TE411P in our server looped back the span 2 port (Gives a green light and OK with same config as span 1) and am using a crossover cable to link the PBX to our server. (We tried a pucker BT cross over cable with exactly the same results as mine and a striahgt through gives us nothing at all, I guess as you might expect) I have configured the Zap span for 1 clocking (Primary) 1 line build out, with the framing etc as CCS, HDB3, CRC4 But they don't appear to want to synchronise/talk to each other. ZTTool claims that the span is up and down more times than a fiddlers elbowand the clocking source is internal.( Might I expect the alarm state to be constant if the framing etc was matched and the clock source to show as external ??) The alarms are cylcling from red to red/yellow and finaly to red/yellow/recover before falling back to red and starting again.I think I may be missing something that is probably blindingly obvious to someone in the know.The BT guy has been very good and is trying to help us get this going but seems rather nonplussed with the terms CRC4, CSS and HDB3... Please can somone help and point me (and I guess by extension the BT guy) in the right direction. Cheers AndyTry your end with every different combination of settings applicable toEuroISDN.I would start by removing the CRC4.Try something, if it doesnt work, put it back and change another setting. s___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Min-Expires
17 maj 2006 kl. 12.13 skrev Samuel Tardieu: I am trying to register my Asterisk server to a SIP server which doesn't accept an Expires: field smaller than 1800 seconds and indicates it correctly with a Min-Expires: in an error response when Asterisk tries to use its default of 120 seconds. Is Asterisk supposed to honor this field and retry with the proposed minimum Expires: field? It looks like it doesn't, and I had to change the default_expirey globally. That's right, Asterisk is not aware of that header. Could I please see a SIP debug trace of it? /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * European training in Stockholm, June 2006 ___ --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] [EMAIL PROTECTED] default password doesn't
I follow the advice of Alasdair, it was happening because of the multiple kernel panics. I have installed it again, and now it's working properly! Thanks a lot for your help. I'll change also all default passwords for security reasons. BR, //Laura From: Steve Jones [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] default password doesn't matchTo: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1It's possible that for security reasons, it doesn't let you log on remotely with the default passwords. From the console, change the password to something else unique, and it should work. You should probably do each of these:passwd-maint set master password for web GUIpasswd-amp set password for amp onlypasswd-meetme set password for Web MeetMe only passwd set root password for console loginpasswd admin set admin password for checking system mailbtw: I got this with the help-aah command... Lots of good starting points there!! -SteveFrom: Laura Barquín [mailto:[EMAIL PROTECTED] ]Sent: Wednesday, May 17, 2006 4:18 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] [EMAIL PROTECTED] default password doesn't matchHi all,This is my first post! I'm newbie, yesterday I installed [EMAIL PROTECTED], and I got lot of Kernel panics, after trying to reinstall it, this messages dissapeared. My problem now is that the default password for maint user in AMP is not working... I got this error message when I try to connect via another computer in the same network, after trying to log - maint/password: FORBIDDENYou don't have permissions to access /main on this serverThanks in advance!Laura-- next part --An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060517/633e4673/attachment.html -- ___ --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] Quad BRI card
On 5/17/06, Hadley Rich [EMAIL PROTECTED] wrote: They do, but it isn't released yet. Put B410P into google and you will get a couple of hits. Digium's marketing page says it is available and the distributor I use had one on show the other day so they can't be too far away. Aside from being available.. What driver does it use? Will it be needing bristuff ? (that wouldn't work I guess) Or will the near future integrate BRI ( and hfc?) drivers in asterisk? And thus, making bristuff obsolete? (wich means, BRI users will be able to use cvs easily..) Just to make clear I'm very curious on this card. And yes I'm in europe ;) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Meridian Tie Line
The BT guy should check LD 73 block LPTI and prompt AFF. If it is crc then you need crc4 as well. Best regards Hans Steve Totaro schrieb: Andy Kirby wrote: I am new to the group but have searched the doc's FAQ's etc before posting here. We are attempting tie our asterisk server/service to the building's PBX, the building is in the UK and the local PBX is a meridian option 11 installed and mainteined by BT. BT Have installed a NTBK50AA E1-PRI card in the meridian with daughter cards NTBK51AA (D channel) and NTAK20BD (Clocking) I have asked BT to configure the card as a Master (Exchange end) E1 Euro ISDN (Just like a standard ISDN30e) They claim to have done this in line with the model they use to interface to Cisco routers etc. I have installed a Digium TE411P in our server looped back the span 2 port (Gives a green light and OK with same config as span 1) and am using a crossover cable to link the PBX to our server. (We tried a pucker BT cross over cable with exactly the same results as mine and a striahgt through gives us nothing at all, I guess as you might expect) I have configured the Zap span for 1 clocking (Primary) 1 line build out, with the framing etc as CCS, HDB3, CRC4 But they don't appear to want to synchronise/talk to each other. ZTTool claims that the span is up and down more times than a fiddlers elbow and the clocking source is internal. ( Might I expect the alarm state to be constant if the framing etc was matched and the clock source to show as external ??) The alarms are cylcling from red to red/yellow and finaly to red/yellow/recover before falling back to red and starting again. I think I may be missing something that is probably blindingly obvious to someone in the know. The BT guy has been very good and is trying to help us get this going but seems rather nonplussed with the terms CRC4, CSS and HDB3... Please can somone help and point me (and I guess by extension the BT guy) in the right direction. Cheers Andy Try your end with every different combination of settings applicable to EuroISDN. I would start by removing the CRC4. Try something, if it doesnt work, put it back and change another setting. s ___ --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] Quad BRI card
On Thursday 18 May 2006 18:35, stoffell wrote: Aside from being available.. What driver does it use? Will it be needing bristuff ? (that wouldn't work I guess) Or will the near future integrate BRI ( and hfc?) drivers in asterisk? And thus, making bristuff obsolete? (wich means, BRI users will be able to use cvs easily..) Just to make clear I'm very curious on this card. And yes I'm in europe ;) I'm curious too, unfortunately I don't know anything more about it sorry. hads. -- The means-and-ends moralists, or non-doers, always end up on their ends without any means. -- Saul Alinsky ___ --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] Plan to free myself from AAH
AMP dialplan is full of garbage and perpaphs is not fully 1.2 compatible but it is anyway an Asterisk, working dialplan! I already tried to copy config files and Asterisk starts without warnings: gradually I will clean out them from fax, queues, devices, ring groups, weather reports, etc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Wednesday, May 17, 2006 6:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Plan to free myself from AAH What I did with AMP was take the best parts of it and copy/paste to a clean extensions.conf, then add my modifications onto it. Worked for me. -Original Message- From: Strom Carlson [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Plan to free myself from AAH On 5/17/06, Mimmus [EMAIL PROTECTED] wrote: I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? I doubt it. The problem I have with AAH / AMP / FreePBX is that the configuration files are absolutely full of useless garbage and are really not at all suitable for moving to a standard asterisk install. Set up a new server from scratch and start learning how to configure asterisk manually. Rebuild everything one step at a time so that the functionality remains as you'd like it to be, but that the actual configs aren't full of that FreePBX garbage :) -- Strom Carlson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme conf
quote who=Sharon Lim hi there, i am wondering can meetme.conf able to support diffferent context. Cause currently, it has [rooms] context. ] is it possible to have same conference number with different context? thanks Try it and see ;-) -- Kind Regards, Gavin Henry. Open Source. Open Solutions(tm). http://www.suretecsystems.com/ ___ --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] soekris hadware
Hi Christopher, i know the place, in fact i've been reading a lot before post here. The problem is that even if there are a lot of good documents, personallly i can't see answered my doubts, and this is the reason i wrote. If you can be a bit more explicit and give me some light over my questions i would really love it. Thanks in advance. Kind regards, Jonathan GF On 5/17/06, Christopher Snell [EMAIL PROTECTED] wrote: Google and voip-info.org will have answers to all of your questions. On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote: Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: [...] ___ --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 -- si secretum tibi sit, tege illud, vel revela ___ --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] soekris hadware
Hi Oliver, i understand you use astlinux, even if the version nunber shows the product is quite new. If you have to decide between astlinux and [EMAIL PROTECTED], thinking in use the pbx for basic thinks like MOH, IVR, an advanced dialplan, 1 FXO, 3FXS, 3 SIP and no much more, which one would you select? Any thoughts will be welcomed. Kind regards, Jonathan GF On 5/17/06, olivier.taylor [EMAIL PROTECTED] wrote: more kindly : http://www.astlinux.org/ Olivier Christopher Snell a écrit : Google and voip-info.org will have answers to all of your questions. On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote: Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: [...] ___ --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 -- si secretum tibi sit, tege illud, vel revela ___ --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] Unable to set channel to linear mode
I have a TE110P connected in euroisdn as pri-cpe. When I dial out from a sip phone to a number over the pri, I get an error Unable to set channel 1 (index 0) to linear mode On the destination phone, I only get a terrible noise when answering the call. There doesn't seem to be a speech path... Config: libpri 1.2.2 - zaptel 1.2.5 - asterisk 1.2.6 on Fedora Core 4 (kernel 2.6.11) Anybody knows what this is all about? K ___ --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] soekris hadware
Hi Michael, your document is very good. In fact this was one of the first i read. I googled looking for soekris and asterisk and you appreared. Anyway, your document do not cover the same setup i have. You point the problem of the digium cards don't using it, while i have or i think i require one of this card with a soekris. I'm thinking buy only de SBC and look for another chassis where all equipment fit fine. I say you don't cover, or this i think, because i need to use FXO-PBX-FXS/SIP and you don't use this setup. You call forward all calls over an ip-to-pstn account while i want to receive incoming calls via pstn and call from internal sip phones to sip phones and to pstn trhu asterisk (of course, paying for the call, like a normal call). Another difference is that you use CF with the SBC, while i want to use the mini-hard disk option and make a complete installation of [EMAIL PROTECTED] It seems the product is quite stable and i couldn't see this about astlinux in any place. I would appreciate your thougts about astlinux and some recomendations will be welcomed. Thanks for you tu answer and for your magnificent document. Kind regards, Jonathan GF On 5/18/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Also...an article I wrote earlier this year http://www.tomsnetworking.com/2006/01/13/how_to_asterisk_pbx/ Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713) 861-4005 o(800) 905-6412 f(713) 864-8668 c(713) 201-1262 Original Message Subject: Re: [Asterisk-Users] soekris hadware From: olivier.taylor [EMAIL PROTECTED] Date: Wed, May 17, 2006 11:12 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com more kindly : http://www.astlinux.org/ Olivier Christopher Snell a écrit : Google and voip-info.org will have answers to all of your questions. On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote: Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: [...] ___ --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 ___ --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 -- si secretum tibi sit, tege illud, vel revela ___ --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] SIP debugging
Andres wrote: Hi Klaus, The response to a CANCEL should be a 487 Request Terminated, not a 200 OK. Maybe your innovaphone Server is to blame. Hi Andres. No. The reply to the CANCEL is a 200 Ok. The reply to the cancelled INVITE is a 487. regards klaus ___ --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] Quad BRI card
On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote: Otherwise the Diva server cards are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Ouch, you weren't joking. 1453 Euro! -- Cheers Wayne ___ --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] DM/V1200-4E1 with asterisk
Hello every body. I have this PCI card : DM/V1200-4E1 spec in this site: http://www.intel.com/network/csp/products/3967web.htm Can i use it with Asterisk, is it compatible ? Thank you in advance. ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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] Plan to free myself from AAH
AMP dialplan is full of garbage and perpaphs is not fully 1.2
compatible but it is anyway an Asterisk, working dialplan!
As example:
May 17 18:35:40 WARNING[8625] app_db.c: This application has been
deprecated, please use the ${DB(family/key)} function instead.
May 17 18:35:40 WARNING[8625] app_setcidname.c: SetCIDName is deprecated,
please use Set(CALLERID(name)=value) instead.
May 17 18:35:40 WARNING[8625] pbx.c: SetVar is deprecated, please use Set
instead.
May 17 18:35:40 WARNING[8625] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN; Input:
May 17 18:35:40 WARNING[8625] ast_expr2.fl: If you have questions, please
refer to doc/README.variables in the asterisk source.
May 17 18:35:49 WARNING[8631] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN; Input:
May 17 18:35:49 WARNING[8631] ast_expr2.fl: If you have questions, please
refer to doc/README.variables in the asterisk source.
May 18 10:59:08 WARNING[9144] app_groupcount.c: The SetGroup application has
been deprecated, please use the GROUP() function.
May 18 10:59:08 WARNING[9144] app_cut.c: The application Cut is deprecated.
Please use the CUT() function instead.
DV
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Re: [Asterisk-Users] Quad BRI card
On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote: are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Ouch, you weren't joking. 1453 Euro! But worth every penny, imo. I have a few servers running Eicon Diva Server V-4BRI cards and they are easy to install, run great with Armin's chan_capi-cm and the onboard hardware echo cancellation is excellent. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] Trunk Si without autetification
Hello, I am trying to use a trunk SIP between my Asterisk server and a Biling prepaid server. Problem: I would like to disable the authentication trunk what is the command for that request. In my log server I have: proxy authentification required Regards Rabii ___ --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] Quad BRI card
On 19:16, Thu 18 May 06, Avi Miller wrote: On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote: are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Ouch, you weren't joking. 1453 Euro! But worth every penny, imo. I have a few servers running Eicon Diva Server V-4BRI cards and they are easy to install, run great with Armin's chan_capi-cm and the onboard hardware echo cancellation is excellent. We use the junghanns.net quadbri cards. They work great too, and roughly 1/3 of the price. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Please help me...Urgent
Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to implement pure VoIP solution. Am I right? I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called "Voice Finder AP 200" and the below values: Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz (These values are dummy values) Currently we are making US calls using VoIP provided by "Vebtel". Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this? I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel? Waiting for your quick response. Thank you. Regards, Chandra. New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.___ --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] SIP debugging
Kevin P. Fleming wrote: Klaus Darilion wrote: Shouldn't there be some error indication if Asterisk discards a response? Probably, although it's not clear here that Asterisk actually discarded anything. Without seeing the entire dialog, there's no way to be sure whether there were multiple Call-IDs, multiple tags, etc. involved. The problem is caused be a forked call with pedantic=yes. Asterisk --SIP-- Proxy ---SIP Sipura \ --- Cisco phone The SIPURA sends the first 180 Ringing back. Then, Asterisk ignores the responses from the Cisco phone (180+200). When setting pedantic=no, it works (I guess with pedantic=no Asterisk does not check the To tag (ugly)). Is Asterisk not able of handling multiple early dialogs with pedantic=yes? regards Klaus PS: Following the call flows pedantic=yes: -- Executing Set(Zap/50-1, [EMAIL PROTECTED]) in new stack -- Executing GotoIf(Zap/50-1, 0?103:3) in new stack -- Goto (frompbx,059966366102,3) -- Executing SetCIDNum(Zap/50-1, 00431234600265) in new stack -- Executing Dial(Zap/50-1, SIP/[EMAIL PROTECTED]|90) in new stack -- parse_srv: SRV mapped to host sip.at43.at, port 5060 We're at 213.174.230.213 port 10392 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (NAT) to 83.136.32.160:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 18 May 2006 09:31:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 293 v=0 o=root 9803 9803 IN IP4 213.174.230.213 s=session c=IN IP4 213.174.230.213 t=0 0 m=audio 10392 RTP/AVP 8 0 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called [EMAIL PROTECTED] poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 180 Ringing t: sip:[EMAIL PROTECTED];tag=f1d48eba29dc7f4i0 f: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 i: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 Record-Route: sip:[EMAIL PROTECTED]:5065,sip:83.136.32.160;ftag=as6ce265a8;lr=on Server: Sipura/SPA2000-3.1.2(NTb) Contact: sip:[EMAIL PROTECTED]:5065 Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/enum.at43.at-3323 is ringing poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED];tag=000cce3a7bf804ac335c54e3-740eb2e1 Call-ID: [EMAIL PROTECTED] Date: Thu, 18 May 2006 09:31:25 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:83.136.32.160;ftag=as6ce265a8;lr=on ontent-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED];tag=000cce3a7bf804ac335c54e3-740eb2e1 Call-ID: [EMAIL PROTECTED] Date: Thu, 18 May 2006 09:31:36 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:83.136.32.160;ftag=as6ce265a8;lr=on Content-Type: application/sdp Content-Length: 196 v=0 o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21 s=SIP Call c=IN IP4 83.136.33.21 t=0 0 m=audio 21174 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (12 headers 9 lines)--- Destroying call '[EMAIL PROTECTED]' poeast01*CLI -- SIP read from 83.136.32.160:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060 From: 00431234600265 sip:[EMAIL PROTECTED];tag=as6ce265a8 To: sip:[EMAIL PROTECTED];tag=000cce3a7bf804ac335c54e3-740eb2e1 Call-ID: [EMAIL PROTECTED] Date: Thu, 18 May 2006 09:31:36 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:83.136.32.160;ftag=as6ce265a8;lr=on Content-Type: application/sdp Content-Length: 196 v=0 o=Cisco-SIPUA 14377
[Asterisk-Users] Re: Ringing indication not working as expected
* Eric ManxPower Wieling [EMAIL PROTECTED] wrote: R is not a valid Dial option. Sure about that? My Asterisk installation lists it as a valid option. asterisk*CLI show application Dial [...] R- indicate ringing to the calling party when the called party indicates [...] r is the option you wanted. HOWEVER, if you are not hearing ringback, r will almost never fixes the issue. r leads to the ringing being indicated right from the start. So if i call my cellphone from a SIP-connected snom, ringing is indicated to me immediately, whereas the cellphone starts to ring not until about 3-5 seconds later. =/ Make sure you have a /etc/asterisk/indications.conf In some situations if you do not have that file you will not hear ringback. Thanks for the advice. indications.conf is now existent and Asterisk is reloaded but the problem still persists. /etc/asterisk/indiciations.conf: [general] country=de [de] description = Germany ringcadance = 1000,4000 dial = 425 ring = 425/1000,0/4000 busy = 425/480,0/480 congestion = 425/480,0/480 callwaiting = 425/2000,0/6000 dialrecall = 425/500,0/500,425/500,0/500,425/500,0/500,1600/100,0/900 record = 1400/500,0/15000 info = 950/330,0/200,1400/330,0/200,1800/330,0/1000 - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Ringing indication not working as expected
Thanks for the advice. indications.conf is now existent and Asterisk is reloaded but the problem still persists. Reloaded? Peraphs restarted is better... DV ___ --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, g722 and 16 kHz audio
Hi there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is g722/8000 which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to establish a 16 kHz wideband call between SNOM phones? Cheers, Philipp ___ --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] Failing SIP registration brings * down
Hi there, this is now the second time I've seen an issue like this with 1.2.7.1, the first time it was a DNS hickup, today its some Internet congestion: When one (!) or more register statements in sip.conf fail the entire Asterisk becomes very unresponsive and does not accept registrations from local phones. This is bad because I could live with being temporarily unreachable thru my SIP carrier, but the local phones should not be affected (and therefore still work with BRI and internally). There's masses of these to be seen (btw DNS and ping work) on the CLI: chan_sip.c:5322 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #24) Question: Wasn't there a new sip.conf entry introduced recently that limited registration attempts to, say, 5 tries? I can't find any reference to this, and am not sure if this made it into 1.2. Cheers, Philipp ___ --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] Re: Ringing indication not working as expected
* Sebastian Kayser [EMAIL PROTECTED] wrote: are there any caveats regarding ringing indication with Asterisk? I have got an asterisk installation with a quadBRI driven by BRIstuff. Internal phones are various snoms (320 / 360) connected via SIP and Idefisk softphones connected via IAX2. Outgoing calls are routed through the Zap interfaces. When i set up the action for an external extension as Dial(Zap/g2/number,60,R) or Dial(Zap/g2/number,60) and initiate an outgoing call, Asterisk tells me that the called party is ringing (Zap/4-1 is ringing) but there is no ringing indicated to the calling party. No matter whether the calling party is a snom hardphone or an idefisk softphone. I tried to narrow down the problem. My installation looks like this PSTN -- 3 x BRI -- POTS (NEC) -- 3 x BRI -- Asterisk ^ ^ | | POTS telephone sets snom SIP phones With no options set in the Dial command, i.e. Dial(Zap/g2/number,60), the ringing behaviour is as follows. - snom - snom - OK - snom - POTS telephone set - OK - snom - PSTN - NOK OK = ringing is signalled to the calling party as soon as Asterisk indicates it on the console (... is ringing). NOK = no ringing is signalled to the calling party _although_ Asterisk indicates it on the console. So although the Zap interface is used for both types of external calls (snom - POST, snom - PSTN) the ringing indication to my snoms fails for calls to the PSTN. Any ideas on how to further debug / troubleshoot this behaviour? - Sebastian ___ --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] DM/V1200-4E1 with asterisk
Hello every body. I have this PCI card : DM/V1200-4E1 spec in this site: http://www.intel.com/network/csp/products/3967web.htm Can i use it with Asterisk, is it compatible ? Thank you in advance. ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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] Unable to register channel
Hii all I bought te110p card. I configured zaptel.com ; span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 and zapata.conf is; switchtype=euroisdn signalling=pri_net group=1 context=nortel dchannel = 16 channel = 1-15,17-31 I receving the following error on my gentoo; May 18 13:48:44 WARNING[8755]: Ignoring canpark May 18 13:48:44 WARNING[8755]: Ignoring dchannel May 18 13:48:44 ERROR[8755]: Channel 24 is reserved for D-channel. May 18 13:48:44 ERROR[8755]: Unable to register channel '1-15' May 18 13:48:44 WARNING[8755]: chan_zap.so: load_module failed, returning -1 May 18 13:48:44 WARNING[8755]: Loading module chan_zap.so failed! lsmod output is; wcte11xp 25120 0 zaptel182820 1 wcte11xp crc_ccitt 2176 1 zaptel and red led is not light up on card __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Re: Ringing indication not working as expected
* Sebastian Kayser [EMAIL PROTECTED] wrote: * Sebastian Kayser [EMAIL PROTECTED] wrote: are there any caveats regarding ringing indication with Asterisk? PSTN -- 3 x BRI -- POTS (NEC) -- 3 x BRI -- Asterisk ^ ^ | | POTS telephone sets snom SIP phones With no options set in the Dial command, i.e. Dial(Zap/g2/number,60), the ringing behaviour is as follows. - snom - snom - OK - snom - POTS telephone set - OK - snom - PSTN - NOK OK = ringing is signalled to the calling party as soon as Asterisk indicates it on the console (... is ringing). NOK = no ringing is signalled to the calling party _although_ Asterisk indicates it on the console. I gave brig debug span 1 a try and it led to the following obvious difference (sorry for the long lines). snom - POTS telephone set: 1 Message type: ALERTING (1) 1 [1 181 011 891 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 1e1 021 811 811 ] 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] snom - PSTN telephone: 1 Message type: ALERTING (1) 1 [1 181 011 891 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 1e1 021 821 881 ] 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) 1Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] So its: Location: Private network serving the local user Call is not end-to-end ISDN; further call progress information may be available inband. which works vs. Location: Public network serving the local user Progress Description: Inband information or appropriate pattern now available. which doesn't work. What's causing Asterisk to indicate ringing to the caller in the first place but not in the second place? Is this regular behaviour? Is there any way to also indicate ringing for snom - PSTN telephone calls? - Sebastian ___ --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] tdm21B in china
Hello Gentelmen, I am in china, just ordereda tdm21B card (2 fxs and 1 fxo), still waiting for its delivery. Whether anybody already tried this kind of card here in china, will you please tell me how it works, any issue? esp. 1) caller id 2) dtmf 3) busy tone 4) hang up Thanksa lot! RegardsÄúµÄÅóÓÑkamnpapÕýÔÚÏíÓÃ21CNµÄ10GÓÊÏ䣬ÄúÒ²¸Ï½ô¼ÓÈ룬ÏíÊÜ»ý·Ö´øÀ´µÄÎÞÏÞ¾ªÏ²°É¡£Óá°Vgo ²¥°É¡± ¿´Ãâ·ÑµçÓ°¡¢µçÊÓ¡¢ÌåÓýÖ±²¥!£¨µã»÷ÏÂÔØ£© ___ --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] Please help me...Urgent
Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to implement pure VoIP solution. Am I right? I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called Voice Finder AP 200 and the below values: Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz (These values are dummy values) Currently we are making US calls using VoIP provided by Vebtel. Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this? I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel? Waiting for your quick response. Thank you. Hi, You have sent this 10 times and received at least 20 answers but there is no development in you query! Some people have email filters for Urgent. The only Urgent in your case is that you urgently need to go to the wiki and have a long reading. Please, show some efforts, benevolence or something... Benchev P.S. http://www.voip-info.org/wiki/view/Asterisk+Configurations+for+connecting+with+VOIP+providers ___ --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] just softphone
Hi,I'm trying to start with Asterisk, but I could not put 2 softphones to talk.The asterisk server rejects the connections always when I dial.May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106What is necessary to put it to work?There is no need to configure external lines.extensions.conf [internal1]exten = 311000,1,Dial(SIP/teste1)[internal2] exten = 312000,1,Dial(SIP/teste2) [internal3]exten = 313000,1,Dial(SIP/teste3) [teste1]sip.conf[teste1]type=friendusername=teste1secret=123 qualify=yesnat=no host=dynamiccanreinvite=no context=internal[teste2]type=friendusername=teste2 secret=123 qualify=yesnat=nohost=dynamiccanreinvite=no context=internal2[teste3]type=friendusername=teste3secret=123 qualify=yesnat=nohost=dynamiccanreinvite=no context=internal3-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Re: Ringing indication not working as expected
* Mimmus [EMAIL PROTECTED] wrote: Thanks for the advice. indications.conf is now existent and Asterisk is reloaded but the problem still persists. Reloaded? Peraphs restarted is better... The reload messages informed about the re-reading of indications.conf. However, even restart doesn't change anything about the ringing indication problem. See my other reply for further debug information i have gathered. - Sebastian ___ --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] Unable to register channel
Emre BALCI wrote: snip May 18 13:48:44 ERROR[8755]: Channel 24 is reserved for D-channel. did you change the jumper setting to E1 as per http://www.digium.com/en/docs/TE110P/te110p_config.php? Looks like the card thinks its a T1 card. Leo ___ --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] Unable to register channel
Yes I changed jumper settings but I receiving following error; May 18 14:43:43 WARNING[8481]: Ignoring canpark May 18 14:43:43 WARNING[8481]: Ignoring dchannel May 18 14:43:43 WARNING[8481]: Unable to specify channel 1: No such device or ad dress May 18 14:43:43 ERROR[8481]: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 May 18 14:43:43 ERROR[8481]: Unable to register channel '1-15' May 18 14:43:43 WARNING[8481]: chan_zap.so: load_module failed, returning -1 May 18 14:43:43 WARNING[8481]: Loading module chan_zap.so failed! --- Leo Ann Boon [EMAIL PROTECTED] wrote: Emre BALCI wrote: snip May 18 13:48:44 ERROR[8755]: Channel 24 is reserved for D-channel. did you change the jumper setting to E1 as per http://www.digium.com/en/docs/TE110P/te110p_config.php? Looks like the card thinks its a T1 card. Leo ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] AAH not getting IP address, likely to be network card?
Does your AAH box have a static IP address or is it a DHCP client? Run ifconfig to check the IP address on the card. On 5/17/06, Brian McCarey [EMAIL PROTECTED] wrote: Hi all, Weuse AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable back in and rebooted AAH. AAH boots up, I log in as Root and AAH does not give me an IP address. I've used different cables. Everything else can access the network. Network card in the AAH box lights up green! Before I naff around changing the network card, as anyone got any useful thoughts. I think when I pulled out the cable, the card went on the blink..! AAH has been running faultlessly before..! Should have left the bloody cable at start. Regards Brian. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems
Looks like you're on the right track with this.
I have just diffed the 'i' version with the 'p' version and found the same
line as you found below.
I've changed the 'peercallstate' line in the 'p' version with the
'ourcallstate' one and compiled.
I tested the outbound and inbound dialling over BRI, and * hangs up when it
needs to!
I will have to test some more to see if this little patch doesn't break
anything else, but so far so good.
Thanks for the tip!
Jeroen
We had the exact same problem. It started happening for us starting at
the 'k' release of bristuff (i mailed a msg on it in february i think
to junghanns).
So, the 'i' release worked fine, while 'k' has the problem as described.
A quick diff of 'i' vs. 'k' showed me this (among other things):
diff -U0 -r -x '*.o' -x '*.so'
bristuff-0.3.0-PRE-1i/libpri-1.2.2/q931.c
bristuff-0.3.0-PRE-1k/libpri-1.2.2/q931.c
--- bristuff-0.3.0-PRE-1i/libpri-1.2.2/q931.c 2006-05-17
19:54:51.0 +0200
+++ bristuff-0.3.0-PRE-1k/libpri-1.2.2/q931.c 2006-05-17
20:04:07.0 +0200
@@ -4428,3 +4428,3 @@
- if (c-ourcallstate != c-sugcallstate) {
- pri_error(pri, updating callstate, ourcallstate
%d to %d\n, c-ourcallstate, c-sugcallstate);
- c-ourcallstate = c-sugcallstate;
+ if (c-peercallstate != c-sugcallstate) {
+ pri_error(pri, updating callstate, peercallstate
%d to %d\n, c-peercallstate, c-sugcallstate);
+ c-peercallstate = c-sugcallstate;
This was such a close match, that i reversed that change in the 'k'
release and voila! problem disappeared.
Now, i have no clue what kind of side-effects this has, if any, nor if
this is the proper solution, but it made the problem disappear for us.
I haven't tried to apply the same to later bristuff releases (all
releases up to 'p' give us the same hangup problem)
Hope this helps.
marcel
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Re: [Asterisk-Users] small form factor WAS soekris hadware
Jonathan Gonzalez wrote: I'm thinking buy only de SBC and look for another chassis where all equipment fit fine. I say you don't cover, or this i think, because i need to use FXO-PBX-FXS/SIP and you don't use this setup. Don't attempt to use a Digium or other FXO and FXS card on a soekris board, especially if you plan on using a full-featured distribution like CentOS (which is what AAH uses). See more below... You call forward all calls over an ip-to-pstn account while i want to receive incoming calls via pstn and call from internal sip phones to sip phones and to pstn trhu asterisk (of course, paying for the call, like a normal call). Another difference is that you use CF with the SBC, while i want to use the mini-hard disk option and make a complete installation of [EMAIL PROTECTED] If you are going to use a hard drive AND an FXS/FXO card (assume TDM11B), you'd be happier using an EPIA board. If you need dual ethernet ports, go with the PD1. If not, the V1 would work fine as well. It seems the product is quite stable and i couldn't see this about astlinux in any place. I would appreciate your thougts about astlinux and some recomendations will be welcomed. For information about AstLinux, go to http://www.astlinux.org Note that there have been some pretty substantial changes in the past month and some of the documentation hasn't quite caught up to the 0.4.0 image. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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] AAH not getting IP address, likely to be network card?
Before I naff around changing the network card, as anyone got any useful thoughts. I think when I pulled out the cable, the card went on the blink..! Run ifconfig as root and see what that tells you about your eth connections.-- Justin BiggsOwner, Biggs Computer Consulting [EMAIL PROTECTED]740.501.4781 ___ --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] Re: DISA SPA3000 issues
Thanks for the feedback, but the route that that I'm finding doesn't work is: Asterisk - SPA3000 - ZAP/BRI - Asterisk - DISA The problem appears to be on outbound calls from the SPA3000 where the second dial tone seems to stop audio transmission, changing the DTMF method make no difference. :( Thanks Dave Hawkes Philippe Lindheimer wrote: Just tried it on mine, worked fine: Cellphone Call - POTS - SPA3000 - Asterisk - DISA - Telasip As an FYI, I have my SPA3000 setup with INFO for the DTMF. When I originally installed it, I couldn't get the DTMF digits to work coming in using AUTO, which is why I have it using INFO (needs to be set on both the SPA and in Asterisk). I'm running the SPA300 with Software Version: 2.0.13(GWg), Hardware Version 2.0.1(4e16). philippe From: Dave Hawkes [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 17 May 2006 13:44:43 -0400 Subject: [Asterisk-Users] Re: DISA SPA3000 issues I have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug? Dave Hawkes Alchaemist wrote: Hi, These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was true no RTP DTMF events (RFC2833) are sent after the SPA listens the second dialtone. I just reviewed the 87 pages PDF of the SPA3000... and didn't find anything about such feature. Now I am going to try to figure out if it has something to do with the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on something I found on this forum, without the dialtone. Did anyone find out anything about this issue before? REGARDS!!! Alchaemist ___ --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 Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone. Get Yahoo! Messenger with Voice http://us.rd.yahoo.com/mail_us/taglines/postman10/*http://us.rd.yahoo.com/evt=39663/*http://messenger.yahoo.com ___ --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] just softphone
I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 What is necessary to put it to work? There is no need to configure external lines. extensions.conf [internal1] exten = 311000,1,Dial(SIP/teste1) [internal2] exten = 312000,1,Dial(SIP/teste2) [internal3] exten = 313000,1,Dial(SIP/teste3) [teste1] sip.conf [teste1] type=friend username=teste1 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal [teste2] type=friend username=teste2 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal2 [teste3] type=friend username=teste3 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal3 Debug/verbose is too short, but probably your peers cannot meet in a mutual context. Try: extensions.conf [default] include = internal [internal] exten = 311000,1,Dial(SIP/teste1) exten = 311000,2,Hangup ; Hangup is good exten = 312000,1,Dial(SIP/teste2) exten = 312000,2,Hangup exten = 313000,1,Dial(SIP/teste3) exten = 313000,2,Hangup Put context=internal or default in all your sip friends. Hope that would do. Benchev ___ --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] multiple calls using IAX
Hi. I have a aplication for web, when u press on the link, the application log into an asterisk(user, password), and call to one extension(ex 201). How can I do to that call go to 201, if busy, go to 202, and so on? I want to implement in a call center. Best Regards Thanks Ever ___ --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] small form factor WAS soekris hadware
On Thursday 18 May 2006 08:07, Darrick Hartman wrote: Don't attempt to use a Digium or other FXO and FXS card on a soekris board, especially if you plan on using a full-featured distribution like CentOS (which is what AAH uses). See more below... I read below, but you don't explain why not to use an FXS/FXO card with a Sokeris board if I'm planning on using a full-featured distro. You only say that the EPIA systems are better. Can you elaborate? -A. ___ --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] ACD Light on Phone?
Hi, Is there anyway I can make a softkey light on a sip phone (aastra 9133i) light up when the agent is logged into a queue? Even if I have to do it via some call in a dialplan. I guess the question is more... what command do I need to send to a sip phone to turn a light on a softkey on/off? ___ --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] just softphone
I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 Try puting a permit=0.0.0.0/0.0.0.0 In the sip.conf for your two phones. BTW: your extensions.conf looks silly, you'll only be able to call test3 from test3. Busy most of the time ;-) Stefan Märkle ___ --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] Quad BRI card
stoffell wrote: Aside from being available.. What driver does it use? Will it be needing bristuff ? (that wouldn't work I guess) The Digium B410P will use the mISDN stack and chan_misdn for Asterisk. Or will the near future integrate BRI ( and hfc?) drivers in asterisk? And thus, making bristuff obsolete? (wich means, BRI users will be able to use cvs easily..) No, that will not happen, unless the authors of those drivers want to disclaim them for inclusion into Zaptel and Asterisk. Just to make clear I'm very curious on this card. And yes I'm in europe ;) As another poster mentioned, the B410P card is definitely targeted at the non-US market... not because the card would not work here, but because there is very little availability of BRI lines in the US at all. Most telcos don't even know what they are if you ask :-) ___ --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] SIP debugging
Klaus Darilion wrote: Is Asterisk not able of handling multiple early dialogs with pedantic=yes? Asterisk is not capable of handling multiple dialogs in response to an outbound INVITE at all. The code is not prepared for requests that it sends to be forked by a proxy. The next major version of chan_sip (to be worked on during the next development cycle) will probably be able to handle this, but today, it's not expected to work properly. ___ --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] Slackware 10.2
Fernando Lujan wrote: Hi guys, I'm trying to use asterisk with my slackware 10.2 box. Kernel 2.6.13 from the testing... The udevd are not creating the /dev/zap devices. Someone already have success installing asterisk over slackware? Thanks in advance. Fernando Lujan I also use asterisk on slackware with the 2.6 kernel. You need to make sure you have the udev rules and permissions files from zaptel installed and also make sure that rc.hotplug is set executable (or at the very least that /sbin/hotplug is set in /proc/sys/kernel/hotplug). I also have slackbuild scripts I made up for both asterisk and zaptel if anyone wants them. -Dave ___ --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] Default dialplan??
Hey all! I've got my Asterisk box tied into my PBX. Currently, if a call comes into my PBX, and can't find the extension, it forwards it through my Asterisk trunk to Asterisk. This works great! Is there a special dialplan function (or common usage pattern) that can do the same thing in Asterisk? i.e. If it can't find the extension, send it out Zap/g1? My dialplan works with patterns, but patterns isn't what I need here. Is anyone doing anything like this? Thanks! ~~Aaron ___ --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] just softphone
On 5/18/06, Benchev [EMAIL PROTECTED] wrote: I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 What is necessary to put it to work? There is no need to configure external lines. extensions.conf [internal1] exten = 311000,1,Dial(SIP/teste1) [internal2] exten = 312000,1,Dial(SIP/teste2) [internal3] exten = 313000,1,Dial(SIP/teste3) [teste1] sip.conf [teste1] type=friend username=teste1 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal [teste2] type=friend username=teste2 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal2 [teste3] type=friend username=teste3 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal3Debug/verbose is too short, butprobably your peers cannot meet in a mutual context.Try:extensions.conf[default]include = internal[internal]exten = 311000,1,Dial(SIP/teste1)exten = 311000,2,Hangup ; Hangup is goodexten = 312000,1,Dial(SIP/teste2) exten = 312000,2,Hangupexten = 313000,1,Dial(SIP/teste3)exten = 313000,2,HangupPut context=internal or default in all your sip friends.Hope that would do.Benchev Benchev,thanks for the attention.But didn't solve the problem. I think it is something with access.I set debug and verbose to 10 and got this.extensions.conf ( I've changed internal by from-sip) [default]include = from-sipinclude = demo[from-sip]exten = 9222,1,Dial(SIP/9222,25)exten = 9222,2,Hangupexten = 9223,1,Dial(SIP/9223,25)exten = 9223,2,Hangup exten = 31200,1,Dial(SIP/312000,25)exten = 31200,2,Hangupsip.conf[general]context=defaultport=5060 bindaddr=0.0.0.0 ;srvlookup=yes[9222]type=friendcallerid = Nome - 9222 9222username=9222secret=9222host= dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internal [9223]type=friendcallerid = Nome - 9223 9223username=9223secret=9223host= dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internal [312000]type=friendusername=312000secret=312000host=dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internalWhen I try to connect From local machine the softphone only call itself: May 18 10:07:25 DEBUG[2218]: Allocating new SIP call for [EMAIL PROTECTED].1.73May 18 10:07:25 VERBOSE[2218]: -- Registered SIP '9222' at 192.168.1.73 port 5061 expires 1800May 18 10:07:25 VERBOSE[2218]: -- Saved useragent X-Lite release 1105d for peer 9222May 18 10:07:40 DEBUG[2218]: Auto destroying call ' [EMAIL PROTECTED]'When I call From network I got the error Call ended: unknown:May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl ass: NEWMay 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569 ]May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569 ] VERSION : 2May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl ass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550 USERNAME : 312000 Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569]May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550 USERNAME : 312000 CALLED NUMBER : 9222 DNID : 9222Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00010ms SCall: 1 DCall: 07747 [ 192.168.0.106:4569] CAUSE : No authority foundMay 18 10:10:14 NOTICE[2213]: Rejected connect attempt from 192.168.0.106May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl ass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550 USERNAME :
[Asterisk-Users] Asterisk - SPA-3000, 407 error
I recently lost my setup (bad drive) and I'm now trying to get my setup back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an SPA-3000. I can call the phone extension, I can call from the phone on the SPA to other extensions and I can call out to the PSTN. What I can't do is to call from the PSTN to through the SPA to Asterisk. It rings twice then I get fast busy. I have the SPA wait for the caller ID info (the reason it rings twice). What I've got is as follows: Code: registry=pstn:[EMAIL PROTECTED]:5061 [pstn_in] username = pstn secret= pstn type = user host = spa.uucp port = 5061 context = from-pstn mailbox = 2202 nat = never dtmfmode = rfc2833 canreinvite = yes qualify = yes insecure = very disallow = all; need disallow before we can allow allow = ulaw ; [pstn_out] username = pstn secret= pstn type = peer host = spa.uucp port = 5061 context = to-pstn from_user = pstn nat = never dtmfmode = rfc2833 canreinvite = yes qualify = yes insecure = yes disallow = all; need disallow before we can allow allow = ulaw ; My dial plan on the SPA looks like this: (S0:[EMAIL PROTECTED]) What I see on the Asterisk console is : May 17 23:36:29 NOTICE[11306]: chan_sip.c:10326 handle_request_invite: Failed to authenticate user xxx sip:[EMAIL PROTECTED];tag=blahblahblah In sniffer I traces I see a 407 (Proxy Authentication Required). I've also noticed a 401 (Unauthorized) for the phone on the SPA. What am I configuring incorrectly. I'm using Asterisk 1.2.7, previously I was using 1.2.0. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --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] soekris hadware
You will have problems. Physically the card won't fit into the Soekris case. There isn't sufficient power to provide ring signalling to FXS ports, which is why the Digium TDM400 card has a dedicated power connector. Of course the Net4801 will only transcode a couple of calls at a time...which means that you should really stick with G.711 coding calls. Then there is the old problem that most of the small FXO adapters/cards are not very good. Search the mailing list archives and you will find numerous threads discussing porblems with various FXO cards. The Sangoma A200 card would be your best shot as it seems to be able to cope with various motherboards. Otherwise, you should consider a freestanding multi-port FXO/FXS adapter like those from Audio Codes or Mediatrix. Since you want a more fully featured installation based upon AAH you may be better served by more powerful hardware. Michael On Thu, 18 May 2006 10:31:29 +0200, Jonathan Gonzalez wrote: Hi Michael, your document is very good. In fact this was one of the first i read. I googled looking for soekris and asterisk and you appreared. Anyway, your document do not cover the same setup i have. You point the problem of the digium cards don't using it, while i have or i think i require one of this card with a soekris. I'm thinking buy only de SBC and look for another chassis where all equipment fit fine. I say you don't cover, or this i think, because i need to use FXO-PBX-FXS/SIP and you don't use this setup. You call forward all calls over an ip-to-pstn account while i want to receive incoming calls via pstn and call from internal sip phones to sip phones and to pstn trhu asterisk (of course, paying for the call, like a normal call). Another difference is that you use CF with the SBC, while i want to use the mini-hard disk option and make a complete installation of [EMAIL PROTECTED] It seems the product is quite stable and i couldn't see this about astlinux in any place. I would appreciate your thougts about astlinux and some recomendations will be welcomed. Thanks for you tu answer and for your magnificent document. Kind regards, Jonathan GF On 5/18/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Also...an article I wrote earlier this year http://www.tomsnetworking.com/2006/01/13/how_to_asterisk_pbx/ Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713) 861-4005 o(800) 905-6412 f(713) 864-8668 c(713) 201-1262 Original Message Subject: Re: [Asterisk-Users] soekris hadware From: olivier.taylor [EMAIL PROTECTED] Date: Wed, May 17, 2006 11:12 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com more kindly : http://www.astlinux.org/ Olivier Christopher Snell a écrit : Google and voip-info.org will have answers to all of your questions. On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote: Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: [...] ___ --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 ___ --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 -- si secretum tibi sit, tege illud, vel revela ___ --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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --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] Default dialplan??
I'll give this one a try, but don't trust me, test it yourself :-) Of course Asterisk can do it! All you need to do is set up a rule for matching ALL extensions in the PBX in it's own separate context and include that context into your normal context. In the following example, asterisk will try matching all extensions in context Normal (all extensions defined on *) and, if no match was found, start searching the context secondary_pbx. In my sample this secondary context will match any 3-digit number and send it to the other PBX. Should work... [Normal] include = secondary_pbx exten = 101,1,Dial(sip/101) [secondary_pbx] exten = _XXX,Dial(Zap/g1) Aaron Paxson wrote: Hey all! I've got my Asterisk box tied into my PBX. Currently, if a call comes into my PBX, and can't find the extension, it forwards it through my Asterisk trunk to Asterisk. This works great! Is there a special dialplan function (or common usage pattern) that can do the same thing in Asterisk? i.e. If it can't find the extension, send it out Zap/g1? My dialplan works with patterns, but patterns isn't what I need here. Is anyone doing anything like this? Thanks! ~~Aaron ___ --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] Re: Bristuffed Asterisk: Hangup problems
On 5/17/06, Marcel van der Boom [EMAIL PROTECTED] wrote: We had the exact same problem. It started happening for us starting at the 'k' release of bristuff (i mailed a msg on it in february i think to junghanns). Marcel, thanks. This does seem to work indeed! I just tested this on our bristuff 0.3.0-pre1p, works perfect. Thanks a lot! I will also forward your mail to junghanns support. cheers! ___ --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] Home asterisk system with single PSTN Line
Hi Im new to asterisk and want to setup a small system at home to play with. Can anyone advise a good card I can use so the asterisk box Im building can act as a gateway to PSTN using my single home analogue phone line. Kind Regards Kev ___ --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] SIP re-invite and billing
I know this may sound like a stupid question but I will put on my flame retardant suit and ask anyway. Is there any way to use/allow SIP reinvite and still track the length of the call? I realize that the whole idea of reinvite is that it takes the proxy out of the media path which, from what I understand also kills the proxy's ability to track the start/end time of the call for billing purposes. Are there any really smart guys out there with propeller hats that have come up with a way to get the best of both worlds? Do we lose anything else using reinvite with Asterisk? Thanks in advance for any help.. --Mojo ___ --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] SNOM, g722 and 16 kHz audio
Philipp von Klitzing wrote: Hi there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is g722/8000 which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to establish a 16 kHz wideband call between SNOM phones? Cheers, Philipp G.722 is always a 16000 samples/second codec. There is no 8000 samples/second option for it. Steve ___ --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] Home asterisk system with single PSTN Line
Low end server with a free PCI slot, Asterisk @ Home, Digium TDM11B (1FXO / 1FXS) Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barrass Kevin Sent: Thursday, May 18, 2006 9:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Home asterisk system with single PSTN Line Hi Im new to asterisk and want to setup a small system at home to play with. Can anyone advise a good card I can use so the asterisk box Im building can act as a gateway to PSTN using my single home analogue phone line. Kind Regards Kev ___ --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] Quad BRI card
On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote: Otherwise the Diva server cards are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Ouch, you weren't joking. 1453 Euro! What about the Gerdes Primux Cards. They can be used in NT and TE Mode. Price ~ 670.- EURO We have a 2S0 card running on a customer site with chan_capi-cm and all looks good. Have a look at http://www.primuxisdn.de Perhaps it helps... Regards Guido ___ --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] SIP re-invite and billing
Mojo Jojo wrote: Is there any way to use/allow SIP reinvite and still track the length of the call? This is discussed nearly every week on this list, it's well covered on the wiki, and various other places. Have you tried to research this before asking here? The simple answer: you are mistaken. SIP reinvites for the media path have no effect on billing at all. ___ --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] small form factor WAS soekris hadware
Andrew Kohlsmith wrote: On Thursday 18 May 2006 08:07, Darrick Hartman wrote: Don't attempt to use a Digium or other FXO and FXS card on a soekris board, especially if you plan on using a full-featured distribution like CentOS (which is what AAH uses). See more below... I read below, but you don't explain why not to use an FXS/FXO card with a Sokeris board if I'm planning on using a full-featured distro. You only say that the EPIA systems are better. Can you elaborate? Yes. There are several reasons, at least for his situation. First, the case provided with the Soekris is too small to house a Digium TDM card. Second, there is no easy way to provide the additional power needed for FXS modules. Third, the board is really underpowered. For about the same money you can get an EPIA PD 1, 256 or 512MB of ram, a compact case and have a system that is much more capable. I've been using the EPIA PD as my firewall and office Asterisk system for about a year. I currently have Slackware 10.2 installed with a Digium TDM11b card, Asterisk 1.2.7.1, iaxmodem and hylafax. It also has OpenVPN running. It just plain works. I use the g729 codec for my voip calls. ulaw internally, so the box does some transcoding. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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] Applet to test VoIP quality
Hello, Does anybody know of free Java/ActiveX applet which could be placed into web and configured to check ping/latency/jitter/ports to selected server? Something like http://www.testyourvoip.com Regards/Pagarbiai, Mindaugas Kezys ___ --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] New To Asterisk - Advice needed
Give http://www.asteriskguru.com/tutorials a try. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Adams Sent: Wednesday, May 17, 2006 5:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] New To Asterisk - Advice needed Hi People, Im writing to get some advice on where to start when learning asterisk? I was going to begin learning with AAH but I wanted to find out if there is a certain build to avoid or if there is a Gui/front end that is better then another. I have been working with dialogic cards for the past 5 years and with auto dialers but I want to get into providing voip service, support and eventually help people save money with their phone systems. At the moment it is strictly for education but I really get a kick out of voip and telephone functions in general. Thanks in advance - Mark Adams ___ --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] Providers using Embedded Devices
Egads. That'a shame, because from experience I can tell you that trying to make Asterisk work in a 'hosted' manner is really tricky. Asterisk wasn't designed with multiple companies in mind. -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Providers using Embedded Devices Most of the VoIP service providers I have encountered are moving in a different direction, with a goal of NOT having any customer premise equipment other than the SIP hard phones, soft phones, and ATAs, along with an IP access router with QoS. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, May 17, 2006 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Providers using Embedded Devices Just curious... Does anyone know if any companies using Asterisk on embedded hardware (out at the customer premisis), such as the Soekris Net4801, to provide VOIP service? Doug. ___ --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 ___ --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] Quad BRI card
From the picture on the web site it looks like it uses a cologne chipset. Any idea if these cards will be available in Australia? Craig - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 18, 2006 9:15 PM Subject: Re: [Asterisk-Users] Quad BRI card stoffell wrote: Aside from being available.. What driver does it use? Will it be needing bristuff ? (that wouldn't work I guess) The Digium B410P will use the mISDN stack and chan_misdn for Asterisk. Or will the near future integrate BRI ( and hfc?) drivers in asterisk? And thus, making bristuff obsolete? (wich means, BRI users will be able to use cvs easily..) No, that will not happen, unless the authors of those drivers want to disclaim them for inclusion into Zaptel and Asterisk. Just to make clear I'm very curious on this card. And yes I'm in europe ;) As another poster mentioned, the B410P card is definitely targeted at the non-US market... not because the card would not work here, but because there is very little availability of BRI lines in the US at all. Most telcos don't even know what they are if you ask :-) ___ --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] Default dialplan??
Thanks Cosmin!! I didn't realize that the dialplans ran in sequential order. I'll try that. thanks! --Aaron - Original Message - From: Cosmin Prund [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 18, 2006 9:44 AM Subject: Re: [Asterisk-Users] Default dialplan?? I'll give this one a try, but don't trust me, test it yourself :-) Of course Asterisk can do it! All you need to do is set up a rule for matching ALL extensions in the PBX in it's own separate context and include that context into your normal context. In the following example, asterisk will try matching all extensions in context Normal (all extensions defined on *) and, if no match was found, start searching the context secondary_pbx. In my sample this secondary context will match any 3-digit number and send it to the other PBX. Should work... [Normal] include = secondary_pbx exten = 101,1,Dial(sip/101) [secondary_pbx] exten = _XXX,Dial(Zap/g1) Aaron Paxson wrote: Hey all! I've got my Asterisk box tied into my PBX. Currently, if a call comes into my PBX, and can't find the extension, it forwards it through my Asterisk trunk to Asterisk. This works great! Is there a special dialplan function (or common usage pattern) that can do the same thing in Asterisk? i.e. If it can't find the extension, send it out Zap/g1? My dialplan works with patterns, but patterns isn't what I need here. Is anyone doing anything like this? Thanks! ~~Aaron ___ --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 ___ --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] Quad BRI card
On 22:32, Thu 18 May 06, Craig Guy wrote: From the picture on the web site it looks like it uses a cologne chipset. Any idea if these cards will be available in Australia? Can't you just order them from the digium website? Or is digium not shiping to Australia? -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] just softphone
On Thursday 18 May 2006 16:32, Ralph Liebessohn wrote: On 5/18/06, Benchev [EMAIL PROTECTED] wrote: I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from Try: extensions.conf [default] include = internal [internal] exten = 311000,1,Dial(SIP/teste1) exten = 311000,2,Hangup ; Hangup is good exten = 312000,1,Dial(SIP/teste2) exten = 312000,2,Hangup exten = 313000,1,Dial(SIP/teste3) exten = 313000,2,Hangup Put context=internal or default in all your sip friends. extensions.conf ( I've changed internal by from-sip) [default] include = from-sip include = demo [from-sip] exten = 9222,1,Dial(SIP/9222,25) exten = 9222,2,Hangup exten = 9223,1,Dial(SIP/9223,25) exten = 9223,2,Hangup exten = 31200,1,Dial(SIP/312000,25) exten = 31200,2,Hangup sip.conf [general] context=default port=5060 bindaddr=0.0.0.0 ;srvlookup=yes [9222] type=friend callerid = Nome - 9222 9222 username=9222 secret=9222 host= dynamic context=from-sip dtmfmode=rfc2833 nat=yes canreinvite=nocontext=internal Ralph, canreinvite=nocontext=internal Is the above a copy/paste mistake? Try context=from-sip or default as in you case that's declared in extensions.conf. Delete context=internal. On the other hand, this thing X-light, does it require Use Auth ID or something similar? Try to turn it on/off. Benchev ___ --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] Quad BRI card
Craig Guy wrote: From the picture on the web site it looks like it uses a cologne chipset. Any idea if these cards will be available in Australia? (Please to trim your replies and not reply in the middle of quoted text...) The cards will be available through all normal Digium distribution channels. Availability in specific countries will depend on certification and approval processes in those countries, though. We will even allow US users to buy them, although I have no idea what they will do with them :-) ___ --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] Polycom 601 -- programming buttons.
Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio ___ --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] FW: NuFone Update: DIDs
On 4/18/06, Wes Baehr [EMAIL PROTECTED] wrote: Well this is disappointing. Time to find somebody else... From: NuFone Operations Sent: Tuesday, April 18, 2006 3:44 PM Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier snip I received an email from Nufone today regarding a change users need to make to bring their DID back to life. I have no idea how long the process will take, but it's an encouraging sign. ___ --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] Default dialplan??
Aaron Paxson wrote: Thanks Cosmin!! I didn't realize that the dialplans ran in sequential order. I'll try that. thanks! We originally had dialplans run in random order, but people found it too confusing. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Polycom - missed calls dial back
This is not necessarily Asterisk specific but if I have Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend a 9 can I do this via the polycom config? I cant find anything in the docs. Bill ___ --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] sip and other phones
I am trying to call phone that on meridian My asterisk connected to meridian via E1.but failed I think I have routing problem between sip and E1 because I am debuging sip call then I see call going to [EMAIL PROTECTED] I just want sip client able to call phones that on meridian and phones that on meridian able to call sip clients linux*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN 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 N200 Counter: 3 linux*CLI zap show status Description Alarms IRQbpviol CRC4 Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Polycom - missed calls dial back
I have not found a way to do this via the Polycom configs. However, what I do is just ensure that the callerid of an inbound call is set so that the recorded number on the Polycoms is a valid callback number (i.e., prepend '9' or '91' depending on the inbound CallerID). Regards, - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill GibbsSent: Thursday, May 18, 2006 11:37 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Polycom - missed calls dial back This is not necessarily Asterisk specific but if I have Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend a 9 can I do this via the polycom config? I cant find anything in the docs. BillThe contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Quad BRI card
Any device to legally connect to the PSTN in Australia must be approved by the regulatory body. A process that usually costs at least $20,000 and only allows the permit holder to sell the product for conneciton to the pstn. It is a very high barrier to entry for the Australian market. There is a guy in Victoria who certified the Fritz! card and charges $400 each for them. Paralell imports are not allowed to be connected. Some manufacturers do the right thing by certifying the card themselves (Eicon for example). Other manufacturers such as AVM leave it to the distributor to certify the card for the local market. The difference is that I can buy an Eicon card off eBay from the US or Europe and legally connect it to the PSTN in Australia as the card comes from the factory carrying the regulartory approval mark. If i was to buy AVM, Digium or Sangoma from another country I'm out of luck cause it doesn't carry the approval sticker that the Australian distributor puts on it. I can understand both points of view - as a customer I want a competitive market so I get value for money. As a distributor I want an exclusive territory so I can jack up the prices to whatever the market will bear without being undercut by nasty competition. Craig - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 18, 2006 10:45 PM Subject: Re: [Asterisk-Users] Quad BRI card On 22:32, Thu 18 May 06, Craig Guy wrote: From the picture on the web site it looks like it uses a cologne chipset. Any idea if these cards will be available in Australia? Can't you just order them from the digium website? Or is digium not shiping to Australia? -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems
On 18 mei 2006, at 14:05, Jeroen Zwarts wrote: I tested the outbound and inbound dialling over BRI, and * hangs up when it needs to! I will have to test some more to see if this little patch doesn't break anything else, but so far so good. I've done a bit more testing and in our install the patch seems to cause an issue with the 'hangup' (h) extensions. We use this to convert incoming faxes to pdf and send them off through mail after the sending fax machine hangs up. The hangup extension is never reached so that bit of our dialplan didnt work anymore. Since both patched and unpatched dont work with that particular setup, there's no way (i know) to test out wether this is actually caused by the patch or not, but i thought i'd just mention it. The 'regular' dialplan seems to work fine for us too though. No other issues were seen by us at least. marcel -- Marcel van der Boom HS-Development BV -- http://www.hsdev.com So! webapplicatie framework -- http://make-it-so.info smime.p7s Description: S/MIME cryptographic signature ___ --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] OT: Aastra Powertouch 350 caller id
Hi all, is anyone using the Aastra Powertouch 350 analog (adsi) phone with asterisk? I cannot get the phone to display incoming caller id... I can see the CID if I hook up a cheap caller id enabled phone... Only the PT350 is having problems. Looking all over for docs on this phone, but what I've found doesn't have any mention of a setting to display CID... Thx as always Dan ___ --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] Default dialplan??
Aaron,There are probably plenty of ways to do this, off the top of my head, if you add a 'include = go-to-pbx' context within the context where your Asterisk patterns are, and there is no match, Asterisk will then begin to check the 'include' contexts in order. (It does not even look at them if it can find a match in the current context. So ... put such an include with a 'catch all' dial plan within 'go-to-pbx' that will handle any patterns that don't match and send them off to your alternate pbx.philippe From: "Aaron Paxson" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Thu, 18 May 2006 09:26:50 -0400Subject: [Asterisk-Users] Default dialplan??Hey all!I've got my Asterisk box tied into my PBX. Currently, if a call comes into my PBX, and can't find the extension, it forwards it through my Asterisk trunk to Asterisk.This works great!Is there a special dialplan function (or common usage pattern) that can do the same thing in Asterisk? i.e. If it can't find the extension, send it out Zap/g1?My dialplan works with patterns, but patterns isn't what I need here. Is anyone doing anything like this?Thanks! ~~Aaron Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates.___ --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] New To Asterisk - Advice needed
I'm fairly new to Asterisk myself and I also started with AAH. Unfortunately I had to remove all configuration files generated by FreePBX (the GUI of AAH) and started over using http://voip-info.org as my guide. Configuration files generated with FreePBX make use of advanced functionality available in Asterisk and that in turn makes it hard (impossible?) to read for a newby. If you've got some experience with Linux and it's kind of configuration files you might be better of without AAH. On the other hand I'm in the process of re-installing my Asterisk on a fresh Centos 4.3 installation so I can't comment on how difficult it is for a newby to install everything from sources. Hope I'll be able to manage it :) Mark Adams wrote: Hi People, I’m writing to get some advice on where to start when learning asterisk? I was going to begin learning with AAH but I wanted to find out if there is a certain build to avoid or if there is a Gui/front end that is better then another. I have been working with dialogic cards for the past 5 years and with auto dialers but I want to get into providing voip service, support and eventually help people save money with their phone systems. At the moment it is strictly for education but I really get a kick out of voip and telephone functions in general. Thanks in advance - Mark Adams ___ --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] Re: Ringing indication not working as expected
On 05/18/06 18:45 Sebastian Kayser said the following: So although the Zap interface is used for both types of external calls (snom - POST, snom - PSTN) the ringing indication to my snoms fails for calls to the PSTN. we've got the following: E1 PRI --- Asterisk ---+--- FXS Gateway --- Analog Phones | +--- FXS Gateway --- Analog Phones | +--- FXS Gateway --- Analog Phones | +--- FXS Gateway --- Analog Phones we see the same problem for /some/ of the phones on a single fxs gateway, but not for the /other/ phones on the /same/ gateway. i'd always thought that this may have been caused by a lost SIP 180 Ringing packet between asterisk and the fxs gateway, but perhaps now i may look inside asterisk itself to try to see if it's causing it somewhere. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Auto Dial Out Madness
Hi All, I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful. Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The call is initiated and that all works, my problem is that it does not wait for the line to be answered before playing the message back. It immediately after dialing the number begins playing the message and is done playing it before the person even answers the phone. Does anyone know what I can do to get this working. I want to be able to launch a script from cron that will create the .call file and mv it into the outgoing directory. That is all working currently. Here is a sample of what my .call file looks like and what my extensions.conf looks like. .call Channel: Zap/3/1234567890 Callerid: 1234567890 MaxRetries: 200 RetryTime: 30 WaitTime: 45 Context: outboundmsg1 Extension: s Priority: 1 extensions.conf - snip - [outboundmsg1] exten = s,1,AbsoluteTimeout,40 exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,Answer exten = s,5,Wait(1) exten = s,6,Playback(outboundmsgs/msg1) ; play outbound msg exten = s,7,Background(outboundmsgs/how_to_ack) ; Press 1 to replay or 2 to acknowledge receiving this message exten = 1,1,Goto(s,5) ; replay message exten = 2,1,Goto(msgack,s,1) ; acknowledge message exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup exten = T,1,Hangup Thanks in advance for any suggestions. Jon Scottorn Systems Administrator The Possibility Forge, Inc. http://www.possibilityforge.com 435.635.0591 x.1004 ___ --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] DELL PowerEdge 2850 and TE4110P and TE110P
On Wed, 17 May 2006, Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I tried using a TE110P and a TE210P. The irq hit percentage on the Dell with zttest is quite disappointing, even after tweaking. It's just above the minimum requirement Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy a dual port ethernet adapter which will use only one irq to free up an IRQ on another slot. This just totally sucks and irq sharing in a box with only 3 pci slots is totally unnecessary I'm now using the 2850 with one TE210P, system is in production use since the 1st of May and no lockups or apparent problems so far. I would have expected better from the 2850 however and would not buy one again Sangoma cards are said to be better but I preferred to support Digium and I don't want to mess around with additional drivers Just my $0.02 ___ --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] unicall dialing problem
Hi every body.I was intalled the last version of unicall, but the problem persist.This issue can be due to a grounding problem?regards.2006/5/16, acriollo [EMAIL PROTECTED]:I will try ...Regards. 2006/5/16, Moises Silva [EMAIL PROTECTED]: A recent version of Unicall has a small bug in tone generation (amissmatching use of spandsp library), please try to upgrade to latestspandsp and unicall libraries, im pretty much sure your problem willbe solved that way. RegardsOn 5/16/06, acriollo [EMAIL PROTECTED] wrote: Hi every body, im triying to resolv thi issue with unicall. The asterisk box drop calls some times inbound and outband with no aparently reason When i triying to dial to the outside , the PBX just send a couple of digits and after this the call waits for a while and then the line is released, with no call on line. this is the log for one call. Can some body helpme please. This is my zaptel and unicall file. unicall.conf loglevel=255protocolclass=mfcr2 protocolvariant=mx,10,4 protocolend=cpe supertones=mx group = 1 context=incoming channel = 1-10 zaptel.conf span=1,1,0,cas,hdb3 cas=1-10:1101 dchan=16 Regards and thanks in advanced May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Make call May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32792May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 0001-[1/ 1/Idle/Idle ] May 16 00:35:06 WARNING[1556]: chan_unicall.c:2672 handle_uc_event: Unicall/1 event Dialing May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1- 1101[1/40/Seize /Idle ] May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 0 on-[2/40/Group I /Idle ] May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Channel gains May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Channel switching May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Call control(6) May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Clearing fwd May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 1001-[2/40/Group I /DNIS ] May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 0 off -[1/40/Group I /Idle ] May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1- 1001[1/40/Clear fwd B /Idle ] May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Destroying call with CRN 32792 May 16 00:35:43 WARNING[1556]: chan_unicall.c:2672 handle_uc_event: Unicall/1 event Release call May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/1 Channel echo cancel ___ --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 --Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home asterisk system with single PSTN Line
I am new to this also. But it is hard to figure out what can be done through a single PSTN line. maybe you can buy the SDK card and then setup IVR menu for have separate voice mailboxes. Weidong On 5/18/06, Barrass Kevin [EMAIL PROTECTED] wrote: HiIm new to asterisk and want to setup a small system at home to playwith.Can anyone advise a good card I can use so the asterisk box Im buildingcan act as a gateway to PSTN using my single home analogue phone line. Kind RegardsKev___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Powertouch 350 CallID display continued
Hi All, as a followup to my previous posting (re: not getting caller id to display on a powertouch 350), I've found the following... If I hook the PT350 up to the PSTN line, caller id is diplayed properly... If I connect it to our CAS Acess Bank II, the 350 will not display incoming CID... If I hook up a cheap callerid enabled phone to the same port that the PT350 was on (on the AB II), the CID IS displayed on the cheap phone..so it seems that something is happening with the connection of the PT350 to asterisk. Anyone used one of these handsets successfully with CID? Thanks as always Dan ___ --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] Slackware 10.2
Jonathan Feally wrote: I believe I had to do the udev permissions file and also cause udevd to launch at bootup before modprobe'ing zaptel stuff. Check to make sure that udevd is launching automatically on bootup and that the udev rules and permissions are in place. Thanks all. I have this working with the 2.4.31 kernel. I hope the next version have a best version of udev. :) Fernando Lujan ___ --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] R2/MFC Configuration.
I'm trying to put asterisk working with a proprietary pbx system. I'm doing it using a T1 crossover cable. The pbx system uses the R2/MFC specification. And the don't inform if it uses cas, ccs, ami or hbd3. My digium card is flashing a red light. How can I put this working with the R2/MFC system? Thanks. Fernando Lujan ___ --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] Auto Dial Out Madness
Jon Scottorn wrote:
Hi All,
I have been struggling with the auto dial out in asterisk. I am
trying to get a call to be auto dialed and play back a message once
the line is answered. So far I have been unsuccessful.
Currently what happens is I have my .call file. I mv it into
/var/spool/asterisk/outgoing. The call is initiated and that all
works, my problem is that it does not wait for the line to be answered
before playing the message back. It immediately after dialing the
number begins playing the message and is done playing it before the
person even answers the phone.
You can't. You'll need to setup a loop that asks the end user to press
a key to play your recording. I have mine loop 6 times with a timeout
value of 6 seconds, before giving up. Code follows:
[voice-mail-callback]
;
; Set timeouts
;
exten = s,1,Set(TIMEOUT(response)=6)
exten = s,2,Set(TIMEOUT(digit)=3)
exten = s,3,Wait(5)
exten = s,4,Set(COUNT=0)
; ***
; Play, you attention is required, press 1 to
; collect voice mail
; ***
exten = s,5,Background(attention-required)
exten = s,6,Background(press-1)
exten = s,7,Background(to-collect-voicemail)
; *
; If 1 is pressed, then play transfer and
; then jump to voice-mail context.
; *
exten = 1,1,Playback(pbx-transfer)
exten = 1,2,Goto(voice-mail,s,1)
;
; Setup a variable to count the number of
; times the message has been played, when
; $COUNT reaches 5, play you've taken
; to long to dial and hangup.
;
exten = t,1,Set(COUNT=$[${COUNT} + 1])
exten = t,2,NoOP(${COUNT})
exten = t,3,GotoIf($[ ${COUNT} 5 ]?103)
exten = t,4,Goto(voice-mail-callback,s,5)
exten = t,103,Playback(local/tolong-todial)
exten = t,104,Playback(goodbye)
exten = t,105,Hangup()
;
; Invalid extension plays invalid choice and adds
; 1 to $COUNT
;
exten = i,1,Playback(local/sorry-invalid-choice)
exten = i,2,Set(COUNT=$[${COUNT} + 1])
exten = i,3,NoOP(${COUNT})
exten = i,4,Goto(voice-mail-callback,s,5)
exten = h,1,NoOP(Hungup)
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RE: [Asterisk-Users] Plan to free myself from AAH
Colin was right! I forgot that AMP dialplan makes intensive use of database keys as AMPUSER and DEVICE: I understand its logic but migration is postponed after a GREAT dialplan rewriting :-( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: Thursday, May 18, 2006 9:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Plan to free myself from AAH AMP dialplan is full of garbage and perpaphs is not fully 1.2 compatible but it is anyway an Asterisk, working dialplan! I already tried to copy config files and Asterisk starts without warnings: gradually I will clean out them from fax, queues, devices, ring groups, weather reports, etc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Wednesday, May 17, 2006 6:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Plan to free myself from AAH What I did with AMP was take the best parts of it and copy/paste to a clean extensions.conf, then add my modifications onto it. Worked for me. -Original Message- From: Strom Carlson [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Plan to free myself from AAH On 5/17/06, Mimmus [EMAIL PROTECTED] wrote: I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? I doubt it. The problem I have with AAH / AMP / FreePBX is that the configuration files are absolutely full of useless garbage and are really not at all suitable for moving to a standard asterisk install. Set up a new server from scratch and start learning how to configure asterisk manually. Rebuild everything one step at a time so that the functionality remains as you'd like it to be, but that the actual configs aren't full of that FreePBX garbage :) -- Strom Carlson ___ --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] Polycom - missed calls dial back
Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
get out on your system...
Or, add a 9 to caller id.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
GibbsSent: May 18, 2006 11:37 AMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: [Asterisk-Users] Polycom -
missed calls dial back
This is not necessarily Asterisk
specific but if I have Polycom 301/501 and 601s and want to dial a missed call
back, how do I prepend a 9 can I do this via the polycom config? I cant
find anything in the docs.
Bill
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Re: [Asterisk-Users] SPA-1001 behind NAT - Internet Asterisk box -- BOUNTY!
Stupid not-quite-an-answer - if you're willing to pay money for a fix, why not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will traverse NATs without a problem. I'm using a GNet VP168I (same as the PA168V) and it works fine even behind a NAT which is itself behind a corporate firewall... Just a thought.On 5/17/06, Eric Lyons [EMAIL PROTECTED] wrote: I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet.It works fine to my local [EMAIL PROTECTED] box.I've tried... many things.I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working.Any Sipura experts out there? Eric.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] router with qos and compatible with stun
Hello, I have a client that needs 20-26 simultaneous voip connections and I don't want to relay all this traffic. So I m looking for a router with non symmetric NAT for SDSL. (to use STUN) Thanks Laurent ___ --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] EM and Dial tone
I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to EM Wink Start because Dialogic used this as its default settings, so it just worked without fiddling. Before Asterisk: Incoming Qwest calls would wink the Dialogic card and then send DTMF to the Dialogic after it winked back. Outgoing calls from Dialogic would come off-hook, wait for wink. At this point Qwest would send dial tone. The Dialogic has call supervision and wait for dial tone enabled. With the Asterisk in the middle, Incoming from Qwest are directed to Dialogic and are answered correctly. The problem is when the dialogic card wants to dial out, we only get the wink Asterisk - no dial tone. The dialogic reports this as a failure and hangs-up If I remove the Dial Tone Detection option on the Dialogic and add pauses before the dial string, all works OK. My question is how can I emulate the Qwest functionality and provide a dial tone after the wink? TIA Bart My zaptel.conf: # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24 # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,1,0,d4,ami em=25-48 My zapata.conf ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This T1 is attached to in-house VM System ; signalling =em_w context=from-internal group = 1 channel = 1-24 ; Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 ; This T1 is attached to Qwest LD ; signalling =em_w context=from-pstn group = 2 channel = 25-48 ___ --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] Polycom - missed calls dial back
On Thursday 18 May 2006 15:01, Chad Osmond wrote:
Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
get out on your system...
Or, add a 9 to caller id.
Yuck. While technically correct, both of those solutions suck major goat nad.
-A.
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[Asterisk-Users] monitoring sangoma cards via snmp
When used in TDM mode, the sangoma cards will work under zaptel, so you will need to perform SNMP at a higher level (i.e in Asterisk). David Yat Sin Sangoma Technologies (905) 474 1990 x119 (800) 388 2475 x199 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] Wiki: http://sangoma.editme.com Message: 2 Date: Fri, 12 May 2006 09:39:55 +0200 (CEST) From: [EMAIL PROTECTED] Subject: [Asterisk-Users] monitoring sangoma cards via snmp To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hello, Digium does not provide snmp support to monitor their cards ! Anybody has tried Sangoma product A104 Quad T1/E1 or others ? Regards harry ___ --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] SPA-1001 behind NAT - Internet Asterisk box -- BOUNTY!
Stupid not-quite-an-answer - if you're willing to pay money for a fix, why not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will traverse NATs without a problem. Looks like it wasn't a NAT of configuration problem after all... the SPA devices are quite nice, IMO. If there's a need, I guess Eric can explain it further... --Luki ___ --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] Re: Auto Dial Out Madness
On Thu, May 18, 2006 at 10:38:14 -0600, Jon Scottorn [EMAIL PROTECTED] wrote: Hi All, I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful. Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The call is initiated and that all works, my problem is that it does not wait for the line to be answered before playing the message back. It immediately after dialing the number begins playing the message and is done playing it before the person even answers the phone. Does anyone know what I can do to get this working. You didn't provide information about your hardware or where you are located (different telephone systems use different signalling). I am having the same problem with a TDM400 and as best I can tell, I have to wait for improved software to detect when the phone is answered. Digital hardware should do what you want, and TDM400s may work in different circumstances than mine. Depending where you are you may see polarity reversal as a call progress indicator. There are config settings for that in zapata.conf. There is also a call progress setting that didn't work for me. Some people claim that problems can be caused by the phone company not providing a signal to you (sometimes called forward disconnect) and that if you talk to them they may be able to turn it on for you. There is also an entry on voip-info.org where someone claims to have an app named nv_linedetect that will do call progress based on inband audio. The source hasn't been published and I haven't seen any recent references for anyone using it. It seems common for people to stick in a wait(2) to give people a chance to answer the phone. That isn't a real solution, but might make things not so bad. ___ --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] Pulling the mISDN number from an incoming call
Hi all Which command do I use to pull the mISDN number from an incoming call. -- Cheers Wayne ___ --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 Groups
Has anyone seen good scripts or documentation on Voicemail groups? We are looking to have a system where you can send a voicemail to multiple mailboxes. ___ --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
