Fwd: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Josue and ISDN experts,I spent 2 days doing some home works on connecting TMS2 and Sangoma A101. So far, it's just partially worked.I was not sure, what's correspond to EuroISDN in Siemens language. On Hipath3750, I see a bunch of protocol listed: - AmtPP with CRC4- AmtPP without CRC4- S0 - PRI ECMA-QSIG slave (1cr/2cr)-PRI ECMA-QSIG master I tried nearly with all of them. 1/With AmtPP with CRC4:+++zaptel.conf span = 1,0,0,ccs,hdb3 (we have to be master)bchan=1-15,17-31dchan=16+++zapata.confswitchtype=euroisdnsignalling=pri_netchannels=1-15,17-31calls from Hipath- Asterisk arrived, but I heard nothing. Log is below: -BEGIN LOG- Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 9f] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] [6c 05 01 80 31 31 33] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '113' ] [70 05 81 37 31 30 30] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '7100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) -- Making new call for cr 1-- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification)-- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: CALL PROCEEDING (2) [18 04 e9 81 83 9f] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] -- Accepting call from '113' to '7100' on channel 1/31, span 1 -- Executing Answer(Zap/31-1, ) in new stack Protocol Discriminator: Q.931 (8) len=15 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: CONNECT (7) [18 04 e9 81 83 9f] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing Wait(Zap/31-1, 1) in new stack Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: STATUS (125) [08 03 81 e4 18] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] Cause data 1: 18 (24) [14 01 01] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1)-- Processing IE 8 (cs0, Cause)-- Processing IE 20 (cs0, Call State) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: STATUS (125) [08 03 81 e2 07] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Wrong message (98), class = Protocol Error (6) ] Cause data 1: 07 (7) [14 01 01] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (cs0, Cause)-- Processing IE 20 (cs0, Call State) -- Executing Echo(Zap/31-1, ) in new stack Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 9f] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] [6c 05 01 80 31 31 33] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '113' ] [70 05 81 37 31 30 30] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) '7100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.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]Asterisk -IP- Siemens HiPath 3750
Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a "channelbank" :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours:http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.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]Asterisk -IP- Siemens HiPath 3750
Hi, As long for HiPath 4000 callerID name doesn't work, only number -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Thursday, May 25, 2006 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi, interoperability between asterisk and siemens interesting me too can you tell me, if caller id _name_ is fully working between asterisk and siemens, and what signaling do you use? currently I have Q.SIG signaling between siemens and ci$co voice gateway (with HDV-E1 module), but because ci$co can't decode caller id name from isdn/Q.SIG, I'm planning to replace ci$co gw with asterisk in near feature :-) PJ Josué Conti wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué ___ --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]Asterisk -IP- Siemens HiPath 3750
Hi Viktor,So where is the PSTN side on your schema? PSTN - Asterisk - Hicom 350? Or PSTN - Hicom 350- Asterisk?ThanksNguyenOn 6/13/06, Viktor Tatianin [EMAIL PROTECTED] wrote: Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours:http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.Benchev ___--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]Asterisk -IP- Siemens HiPath 3750
Hi,Oh, I just want to get it to work. Caller Name is something luxurious for us .NguyenOn 6/13/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi,As long for HiPath 4000 callerID name doesn't work, only number -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Pavel Jezek Sent: Thursday, May 25, 2006 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi, interoperability between asterisk and siemens interesting me too can you tell me, if caller id _name_ is fully working between asterisk and siemens, and what signaling do you use? currently I have Q.SIG signaling between siemens and ci$co voice gateway (with HDV-E1 module), but because ci$co can't decode caller id name from isdn/Q.SIG, I'm planning to replace ci$co gw with asterisk in near feature:-) PJ Josué Conti wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué ___ --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 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]Asterisk -IP- Siemens HiPath 3750
I use PSTN - Hicom 350- Asterisk Asterisk I use for voice mail, ivr and gateway for voice overip I try connect Asterisk to PSTN with EDSS1 signaling it work fine at PSTN side statioon type 5ESS What problem you have ? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of NguyenSent: Tuesday, June 13, 2006 1:37 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Viktor,So where is the PSTN side on your schema? PSTN - Asterisk - Hicom 350? Or PSTN - Hicom 350- Asterisk?ThanksNguyen On 6/13/06, Viktor Tatianin [EMAIL PROTECTED] wrote: Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a "channelbank" :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours:http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Nguyen. The Asterisk with configured 50 SIP Phones is integrated with HiPath 3750, through a board TMS2. For access the PSTN I created a context in asterisk so that asterisk has access HiPath 3750 and uses the LCR's that I configured in the HiPath. The CDR's do asterisk is registered no billing do HiPath that is the Siemens Call Report. The system of voicemail for all the set (HiPath 3750 and Asterisk) I use the Asterisk with access to the Internet, where all the 120 users receive its messages perfectly and also they receive copy from the message for email. The interconnection that we effect is ETSI or EuroISDN. I wait to have helped.If to need plus some thing is to inform. Best Regards Josué 2006/6/13, Viktor Tatianin [EMAIL PROTECTED]: I use PSTN - Hicom 350- Asterisk Asterisk I use for voice mail, ivr and gateway for voice overip I try connect Asterisk to PSTN with EDSS1 signaling it work fine at PSTN side statioon type 5ESS What problem you have ? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Nguyen Sent: Tuesday, June 13, 2006 1:37 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi Viktor,So where is the PSTN side on your schema? PSTN - Asterisk - Hicom 350? Or PSTN - Hicom 350- Asterisk?ThanksNguyen On 6/13/06, Viktor Tatianin [EMAIL PROTECTED] wrote: Hi I have next working sheme Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1 This is work fine -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ]On Behalf Of NguyenSent: Tuesday, June 13, 2006 10:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750 Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN - HIPATH 3750 (14 analog trunk lines) - TMS2 - TE110P - Asterisk All extensions of Hipath 3750 are analog (120 extensions)I know that it's maybe easier if we do other way, PSTN-ASterisk (2E-1) - TMS2 - Hipath 3750. But this is not an option, due to some political debat :((( I don't have the tech manual of Hipath yet, but here is what I want to do:1/ Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail, feature server (voice log, conference, etc), after some menu prompts, will transfer back the call to Hipath 3750, using the same TMS2-TE110P connection, to one analog extension of Hipath 3750. 2/User of exteniosns of Hipath 3750, when dial out, will be transfered into Asterisk, using the same TMS2-TE110P. Asterisk will do the check of user balance account, LCR, and if approved , will transfer the call back to Hipath 3750, for getting into Analog trunk line. Since for the Hipath, TMS2 is a trunk module, so I suspect that some DISA operation must be enabled on Hipath, so we can enable the path from analog trunk port - TMS2 - Asterisk and back?Is above configuration working? And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)Very interested in your working configuration, can you explain a bit?Thank you and best regards,Nguyen On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list that might help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. - When the call arrived from PSTN through CO line
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Josue, benchevWith your guidance, I want to get back to HiPath right now. But I am on the road, so I can get in touch with that system only on Tuesday. But that's really great new Josue, that you can work out the things from those commercial system. I will be back very soon,Thanks again,NguyenOn 5/26/06, Josué Conti [EMAIL PROTECTED] wrote:Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.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 ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- With best regards,Nguyen ___ --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 -IP- Siemens HiPath 3750
Hi Nguyen , I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list that might help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTN I had the same idea because I wanted to save on the card side(single span), and use the Hipath as a channel bank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true? Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access (if this is what they mean by DISA) Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever you want. I don't know what DISA they are talking about? Do they mean S2M or similar thing(but TMS2 is S2M)? Anyone? Sorry for not being able to help, but hope somebody else would do it. 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]Asterisk -IP- Siemens HiPath 3750
Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to businessobstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.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 ___ --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 -IP- Siemens HiPath 3750
Hi, interoperability between asterisk and siemens interesting me too can you tell me, if caller id _name_ is fully working between asterisk and siemens, and what signaling do you use? currently I have Q.SIG signaling between siemens and ci$co voice gateway (with HDV-E1 module), but because ci$co can't decode caller id name from isdn/Q.SIG, I'm planning to replace ci$co gw with asterisk in near feature :-) PJ Josué Conti wrote: Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué ___ --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
