We would like to look at the feasibility of utilizing * as a network infrastructure for a unified communications platform. We would like a list of consultants that work with * and have either developed a platform which is easily usable in a true telco environment.
The system needs to have the following: Billing, voice and fax unified messaging, integration with h323, sip, aix to produce a Vonage type of service. Please forward your information to [EMAIL PROTECTED] Sincerely, Don Feuer (949) 279-5290 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, March 12, 2004 6:25 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #3083 - 14 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: SIP call to ISDN subscriber (Derek Bruce) 2. Re: PCI front mount chassis? (Stephen Davies) 3. RE: XML Phone book software. (Alexander Romanov) 4. E1 cards in Australia (Alexander Romanov) 5. UDC SYSTEMS (Michael Devenijn) 6. Fax redirection problem (Nicolas Bougues) 7. call bridge (Alessio Focardi) 8. Native Bridge and Billing (Daniel Bichara) 9. Re: E1 cards in Australia (Peter Brown) 10. Re: PCI front mount chassis? (Rich Adamson) 11. Help on two subjects (David J Carter) 12. Re: asterisk-oh323, new version 0.5.10 (Michael Manousos) 13. Re: asterisk-oh323 (Michael Manousos) 14. Re: XML Phone book software. (stan) --__--__-- Message: 1 From: "Derek Bruce" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] SIP call to ISDN subscriber Date: Fri, 12 Mar 2004 02:45:16 -0700 Organization: Calgary Telecom Reply-To: [EMAIL PROTECTED] try adding: progress_ind setup enable 3 progress_ind alert enable 8 progress_ind connect enable 8 to the dial-peer on the Cisco GW... ----- Original Message ----- From: "Manuel Goertz" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, March 12, 2004 2:26 AM Subject: [Asterisk-Users] SIP call to ISDN subscriber > > Hi all, > > I have a problem calling from a sipset to a ISDN subscriber over > a CISCO 1760 GW. > The following setup is used. > UA ---> GW ---> ISDN > The UA is a sipset, the GW is the CISCO 1760 with ISDN BRI interface > and a standard ISDN subscriber. > The UA is registered with a registrar/proxy. > All numeric userparts of the SIP URI are routed to the GW. > The GW's BRI interface is connected to the PSTN. > The call signaling seems to work as the SIP phone indicates ringing > and the ISDN phone is ringing. After picking up the hook of the ISDN > phone the UA shows "In Call". But after a second the call is > terminated. The log shows that the GW sends to both side the call > termination messages. TX -> DISCONNECT "Normal Call Clearing" to the > ISDN side and a BYE message to the SIP side. > The signaling in short: > > UA GW ISDN > INVITE -> | > <- 100 Try > | TX -> SETUP > | RX <- CALL_PROC > | RX <- ALERTING > <- 183 Sess | > | RX <- CONNECT > | TX -> CONNECT_ACK > <- 200 OK | > Milliseconds later ! > | TX -> DISCONNECT > | RX <- RELEASE > | TX -> RELEASE_COMP > <- BYE | > 200 OK -> | > > > Any hints how to solve this problem. > > Thanks > > Manuel > > > > > > > > > -- > +KOM----------------------------------------------------------------+ > |Manuel G�rtz Merckstrasse 25| > |Darmstadt University of Technology 64283 Darmstadt, Germany| > |Multimedia Communications Tel: (+49) 6151 16-5175| > |Multimedia Networking & Distribution Fax: (+49) 6151 16-6152| > +----------------------------------------------------------------KOM+ > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 2 Date: Fri, 12 Mar 2004 12:32:40 +0200 (SAST) From: Stephen Davies <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PCI front mount chassis? Reply-To: [EMAIL PROTECTED] On Fri, 12 Mar 2004, Brian Capouch wrote: > I too am running 6 cards in my system, although not in a "high traffic > capacity" load environment. > > So far my (limited) high-load simulations have shown no problems. So - is it apocryphal that the Digium cards (drivers) won't share interrupts? If there is a real issue with sharing interrupts then it seems to me to be a bug that needs fixing. PCI bus supports shared interrupts, why doesn't the hardware/driver? Yours curiously, Steve --__--__-- Message: 3 From: "Alexander Romanov" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] XML Phone book software. Date: Fri, 12 Mar 2004 21:53:10 +1100 Reply-To: [EMAIL PROTECTED] Hi All, Does anyone have Digium E1 cards in production in Australia? Are any of them certified? Any feedback would be appreciated. Thaks Alex. --__--__-- Message: 4 From: "Alexander Romanov" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Fri, 12 Mar 2004 21:57:59 +1100 Subject: [Asterisk-Users] E1 cards in Australia Reply-To: [EMAIL PROTECTED] Sorry for double post. Wrong subject :-) Hi All, Does anyone have Digium E1 cards in production in Australia? Are any of them certified? Any feedback would be appreciated. Thaks Alex. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 5 Date: Fri, 12 Mar 2004 11:57:22 +0100 From: "Michael Devenijn" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: [Asterisk-Users] UDC SYSTEMS Reply-To: [EMAIL PROTECTED] Does anybody have experience with these units ?? =20 http://www.udcsystems.com/ DISCLAIMER: The content of this e-mail message does not constitute a = commitment of DKMA bvba This e-mail and any attachments thereto may = contain information which is confidential and/or protected by = intellectual property rights and are intended for the intended recipient = only. Any use of the information contained herein ( including, but not = limited to, total or partial reproduction, communication or distribution = in any form ) by persons other than the designated recipient(s) is = prohibited.If an addressing or transmission error has misdirected this = e-mail, please notify the author, either by telephone or by e-mail and = delete the material from any computer. --__--__-- Message: 6 Date: Fri, 12 Mar 2004 12:41:35 +0100 From: Nicolas Bougues <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Organization: Axialys Interactive http://www.axialys.net Subject: [Asterisk-Users] Fax redirection problem Reply-To: [EMAIL PROTECTED] >From time to time, during a conversation, Asterisk seems to detect a fax tone. It then tries to redirect it, and prints the following message : Redirecting Zap/2-1 to fax extension According to the source, it does this only if it matches a "fax" extension in the current context. I don't have a "fax" extension, but a wildcard one (_.). I would like these detections to be simply ignored. Is there any way to do it ? -- Nicolas Bougues Axialys Interactive --__--__-- Message: 7 Date: Fri, 12 Mar 2004 12:54:46 +0100 From: Alessio Focardi <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] call bridge Reply-To: [EMAIL PROTECTED] Hi all, I would like to have Asterisk bridge 2 calls with this schema -inbound call comes in -the caller id is passed to an external script -the external script replies with a phone number -an outbound call to the number provided by the script is made -if the outgoing call is answered we have to bridge inbound/outbound calls -if there is no answer/busy call is diverted to a voicebox what would you suggest to archive such goal ? The purpose is to connect our customers to field technicians without giving them their mobile phone number ... I think that is a very common issue in our market :) Tnx for any help you can give me ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] --__--__-- Message: 8 Date: Fri, 12 Mar 2004 09:10:08 -0300 From: Daniel Bichara <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Native Bridge and Billing Reply-To: [EMAIL PROTECTED] Hi all, I am connecting two * (A and B) using a third * (C) as passthru and billing control. All connections are IAX-2. So, when A wants to call someone outside, it Dials to "C". "C" analyzes the "extension number" and redirects it to the appropriate destination at "B", billing the call: A (exten 223) calls extension 978 at C <----> "C" knows extension 978 is "B" extension 10978 and calls it <-----> "B" accepts the call to 10978 from "C" When connection between "C" and "B" is estabilished, "C" starts native bridge mode, transfering call control. For "C", call ended and it bills as it longs only few seconds. Should I disable native bridge? How? I need "C" bills the call and controls it. Thanks in advance, Daniel --__--__-- Message: 9 Date: Fri, 12 Mar 2004 23:15:05 +1100 To: [EMAIL PROTECTED] From: Peter Brown <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] E1 cards in Australia Reply-To: [EMAIL PROTECTED] Alex, With Digium's agreement, I am certifying the TE410P for use in Australia. If you want please talk to me. At 21:57 12/03/04 +1100, you wrote: >Sorry for double post. Wrong subject :-) > > >Hi All, > >Does anyone have Digium E1 cards in production in Australia? Are any of them >certified? >Any feedback would be appreciated. > >Thaks >Alex. > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Peter Brown CEO IP Telephonics Ph 02 9153-5978 --__--__-- Message: 10 Date: Fri, 12 Mar 2004 05:26:13 -0600 From: Rich Adamson <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] PCI front mount chassis? To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] > > I too am running 6 cards in my system, although not in a "high traffic > > capacity" load environment. > > > > So far my (limited) high-load simulations have shown no problems. > > > So - is it apocryphal that the Digium cards (drivers) won't share > interrupts? > > If there is a real issue with sharing interrupts then it seems to me > to be a bug that needs fixing. PCI bus supports shared interrupts, > why doesn't the hardware/driver? In most cases, sharing an interrupt is not a problem at all. There have been a few cases where _some_ issue was resolved by moving cards around, however the majority of those seem to be: a) abrupt system changes with no effort to seriously identify the root-cause, b) newbie installations where the condition of the underlying system infrastructure is totally unknown, or, c) wild recommendations that might have had some basis a long time ago but no longer apply. Example: 'cat /proc/interrupts' 9: 1854652239 XT-PIC ehci-hcd, eth0, wcfxo, Intel ICH4 works just fine, and I can't imagine a more demanding irq arrangement where the only nic shares with an x100p, etc. Obviously there are performance limits and expecting multiple quad T1 cards or some other _specific_ high-volume configuration to share one or two interrupts could create a problem. But, engineering a system for those conditions is no more difficult then understanding the requirements of whatever cards are being used and dealing with them appropriately. --__--__-- Message: 11 From: "David J Carter" <[EMAIL PROTECTED]> To: "Asterisk User Group" <[EMAIL PROTECTED]> Date: Fri, 12 Mar 2004 12:42:33 -0000 Subject: [Asterisk-Users] Help on two subjects Reply-To: [EMAIL PROTECTED] Hi All, I have now got my '*' server up and running quite good. As stated in earlier posts I am no Linux guru, so a bit of hand holding required. First Subject. I would now like to add h323 boxes to the '*' server, I have looked through the wiki and followed the instructions about what I need but I am a little thick as I can't seem to get to grips with it. Has anybody got a dummies step by step guide to installing things needed for h323. ala 1. turn on your server. 2. log onto your server. 3. make a cup of coffee because ya gonna need it. 4. ...... and so on. Second Subject. I have never used or seen a channel bank, but I think it is what I require for a project I am looking at. I have 12 Analogue (CO) lines that I would like to bring into the '*' server. I have 12 Analogue POTS that I would like to connect to the '*' server, these are along with SIP phones (Grandstream), and IAX clients. The later two I have no problems with, see First Subject for the other failings. If any one can help then please either answer on or off list. Regards & thanks in advance. Dave --__--__-- Message: 12 Date: Fri, 12 Mar 2004 14:51:01 +0200 From: Michael Manousos <[EMAIL PROTECTED]> Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10 Reply-To: [EMAIL PROTECTED] T.38 FAX is in the short-term plans for asterisk-oh323. Michael T. Chan wrote: > Dear Michael > > Do you foresee implementing these in the near future, one or the other or > both? Thanks > > Tc > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Michael > Manousos > Sent: Thursday, March 11, 2004 4:49 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10 > > > > Hi TC, > T.38 FAX and native bridging are not supported by asterisk-oh323. > > Michael. > > > T. Chan wrote: > >>Dear Michael, >> >>Does your H323 driver run T38 Fax? Also, does your H323 driver have the >>capability of just proxying signal, and NOT proxying signal and media, > > just > >>like the canrevite=yes in the sip scenario? Thanks >> >>TC >> >>-----Original Message----- >>From: [EMAIL PROTECTED] >>[mailto:[EMAIL PROTECTED] Behalf Of Michael >>Manousos >>Sent: Wednesday, March 10, 2004 7:11 AM >>To: [EMAIL PROTECTED] >>Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.10 >> >> >> >>Hello all, >> >>asterisk-oh323 has been updated. The new version 0.5.10 fixes >>the incorrect answering of H.323 channels (thanks to the people >>of the list who helped to trace the problem). Also, I have added >>support for Gnomemeeting text messages (just for fun). >>Additionally, the new version contains stability improvements. >> >>This will be the last version using the OpenH323/Pwlib v1.12.2/1.5.2. >>The next version will move on to the latest versions of these >>libraries. >> >>Regards, >>Michael. >> >> >>_______________________________________________ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >>--- >>Incoming mail is certified Virus Free. >>Checked by AVG anti-virus system (http://www.grisoft.com). >>Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 >> >>--- >>Outgoing mail is certified Virus Free. >>Checked by AVG anti-virus system (http://www.grisoft.com). >>Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 >> >>_______________________________________________ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > ./M > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- ./M --__--__-- Message: 13 Date: Fri, 12 Mar 2004 14:53:03 +0200 From: Michael Manousos <[EMAIL PROTECTED]> Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk-oh323 Reply-To: [EMAIL PROTECTED] Hi, Check the included README file for installation instructions. Michael Erick Weber V. wrote: > Hi all: > > Does someone can direct me to an asterisk-oh323 how to or installation > manual > > Thanks > > Erick > > --__--__-- Message: 14 Date: Fri, 12 Mar 2004 13:12:12 +0000 From: stan <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] XML Phone book software. Reply-To: [EMAIL PROTECTED] On Thu, Mar 11, 2004 at 04:06:41PM -0600, Brian R. Swan wrote: > I'm looking into writing a some phone book XML/PHP software for my Cisco > phones. Specifically, I'd like to be able to use a web interface (on the > computer) to maintain a contact list, and then dial from it on the phone. > Maybe using MySql on the back end or something (to be determined). Before I > start, and duplicate something else that exists, I wanted to see if anyone > has heard of software like that? Searches of Sourceforge, Freshmeat, and > Google didn't turn up much or anything. > see the cmxml software section of http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx --__--__-- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
