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 dfeuer@cox.net Sincerely, Don Feuer (949) 279-5290 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Friday, March 12, 2004 6:25 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3083 - 14 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com 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 asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com 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" <dbruce@calgarytelecom.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] SIP call to ISDN subscriber Date: Fri, 12 Mar 2004 02:45:16 -0700 Organization: Calgary Telecom Reply-To: asterisk-users@lists.digium.com 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" <mgoertz@KOM.tu-darmstadt.de> To: <asterisk-users@lists.digium.com> 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users--__--__-- Message: 2 Date: Fri, 12 Mar 2004 12:32:40 +0200 (SAST) From: Stephen Davies <steve@daviesfam.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] PCI front mount chassis? Reply-To: asterisk-users@lists.digium.com 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" <alex@rnsinternational.com.au> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] XML Phone book software. Date: Fri, 12 Mar 2004 21:53:10 +1100 Reply-To: asterisk-users@lists.digium.com 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" <alex@rnsinternational.com.au> To: <asterisk-users@lists.digium.com> Date: Fri, 12 Mar 2004 21:57:59 +1100 Subject: [Asterisk-Users] E1 cards in Australia Reply-To: asterisk-users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 5 Date: Fri, 12 Mar 2004 11:57:22 +0100 From: "Michael Devenijn" <Michael.Devenijn@dkma.be> To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] UDC SYSTEMS Reply-To: asterisk-users@lists.digium.com 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 <nbougues-listes@axialys.net> To: asterisk-users@lists.digium.com Organization: Axialys Interactive http://www.axialys.net Subject: [Asterisk-Users] Fax redirection problem Reply-To: asterisk-users@lists.digium.com
Don, I would be more than willing to speak to you regarding this inquiry. It sounds like an interesting project. Please call me, when convenient, at 617-848-8899, or let me know where I can reach you. I am both a security analyst focusing on VoIP and a consultant in the IT field. Regards, Phil Jackson -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Don Feuer Sent: Saturday, March 13, 2004 10:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Consultants 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 dfeuer@cox.net Sincerely, Don Feuer (949) 279-5290 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Friday, March 12, 2004 6:25 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3083 - 14 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com 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 asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com 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" <dbruce@calgarytelecom.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] SIP call to ISDN subscriber Date: Fri, 12 Mar 2004 02:45:16 -0700 Organization: Calgary Telecom Reply-To: asterisk-users@lists.digium.com 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" <mgoertz@KOM.tu-darmstadt.de> To: <asterisk-users@lists.digium.com> 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users--__--__-- Message: 2 Date: Fri, 12 Mar 2004 12:32:40 +0200 (SAST) From: Stephen Davies <steve@daviesfam.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] PCI front mount chassis? Reply-To: asterisk-users@lists.digium.com 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" <alex@rnsinternational.com.au> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] XML Phone book software. Date: Fri, 12 Mar 2004 21:53:10 +1100 Reply-To: asterisk-users@lists.digium.com 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" <alex@rnsinternational.com.au> To: <asterisk-users@lists.digium.com> Date: Fri, 12 Mar 2004 21:57:59 +1100 Subject: [Asterisk-Users] E1 cards in Australia Reply-To: asterisk-users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 5 Date: Fri, 12 Mar 2004 11:57:22 +0100 From: "Michael Devenijn" <Michael.Devenijn@dkma.be> To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] UDC SYSTEMS Reply-To: asterisk-users@lists.digium.com 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 <nbougues-listes@axialys.net> To: asterisk-users@lists.digium.com Organization: Axialys Interactive http://www.axialys.net Subject: [Asterisk-Users] Fax redirection problem Reply-To: asterisk-users@lists.digium.com
Better get a hardware expert, We are currently adapting asterisk to a cpci platform to get around the serious hardware limitations that digium always stops at. Anton Sphyrna Inc - We do what others dare not to- ----- Original Message ----- From: "Don Feuer" <dfeuer@cox.net> To: <asterisk-users@lists.digium.com> Sent: Saturday, March 13, 2004 10:04 PM Subject: [Asterisk-Users] Consultants 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 dfeuer@cox.net Sincerely, Don Feuer (949) 279-5290 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Friday, March 12, 2004 6:25 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3083 - 14 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com 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 asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com 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" <dbruce@calgarytelecom.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] SIP call to ISDN subscriber Date: Fri, 12 Mar 2004 02:45:16 -0700 Organization: Calgary Telecom Reply-To: asterisk-users@lists.digium.com 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" <mgoertz@KOM.tu-darmstadt.de> To: <asterisk-users@lists.digium.com> 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users--__--__-- Message: 2 Date: Fri, 12 Mar 2004 12:32:40 +0200 (SAST) From: Stephen Davies <steve@daviesfam.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] PCI front mount chassis? Reply-To: asterisk-users@lists.digium.com 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" <alex@rnsinternational.com.au> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] XML Phone book software. Date: Fri, 12 Mar 2004 21:53:10 +1100 Reply-To: asterisk-users@lists.digium.com 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" <alex@rnsinternational.com.au> To: <asterisk-users@lists.digium.com> Date: Fri, 12 Mar 2004 21:57:59 +1100 Subject: [Asterisk-Users] E1 cards in Australia Reply-To: asterisk-users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 5 Date: Fri, 12 Mar 2004 11:57:22 +0100 From: "Michael Devenijn" <Michael.Devenijn@dkma.be> To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] UDC SYSTEMS Reply-To: asterisk-users@lists.digium.com 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 <nbougues-listes@axialys.net> To: asterisk-users@lists.digium.com Organization: Axialys Interactive http://www.axialys.net Subject: [Asterisk-Users] Fax redirection problem Reply-To: asterisk-users@lists.digium.com
Anton wrote:> Better get a hardware expert, We are currently adapting asterisk to a > cpci platform to get around the serious hardware limitations that > digium always stops at. > > Anton > Sphyrna IncCould you explain what are those serious hardware limitations?
sure, we are an actual running clec so we require ds3 level interfaces, 99.9999% uptime, and easy management. ----- Original Message ----- From: "Senad Jordanovic" <senad@boltblue.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, March 14, 2004 4:04 AM Subject: RE: [Asterisk-Users] Consultants> Anton wrote: > > Better get a hardware expert, We are currently adapting asterisk to a > > cpci platform to get around the serious hardware limitations that > > digium always stops at. > > > > Anton > > Sphyrna Inc > > Could you explain what are those serious hardware limitations? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Does anyone use VoicePulse Inbound service and receive Caller ID Name? I receive caller ID number but no name. Thanks, Kevin
Don, Just mark another number 703 395 0238... we might help Feel free to call -- Regards, Vasyl Rublyov IonIdea, Inc. vasyl.rublyov@ionidea.com>-----Original Message----- >From: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Don Feuer >Sent: Saturday, March 13, 2004 10:05 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Consultants > >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. =20 > >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 dfeuer@cox.net > >Sincerely, > >Don Feuer >(949) 279-5290