[Asterisk-Users] Problem compiling ZAPTEL on Linux 2.6.6
Hi, I have Debian Linux with kernel 2.6.6. The all packages compiled except ZAPTEL where I have the following error: voipgw:/usr/src/zaptel# make make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.6' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1165: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1237: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: warning: type defaults to `int' in declaration of `__mptr' /usr/src/zaptel/zaptel.c:1293: warning: initialization from incompatible pointer type /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: invalid type argument of `->' /usr/src/zaptel/zaptel.c:1353: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1486: warning: passing arg 1 of `unregister_hdlc_devicevoipgw:/usr/src/zaptel# make make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.6' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1165: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1237: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: warning: type defaults to `int' in declaration of `__mptr' /usr/src/zaptel/zaptel.c:1293: warning: initialization from incompatible pointer type /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: invalid type argument of `->' /usr/src/zaptel/zaptel.c:1353: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1486: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:2950: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:2955: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c:3035: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3037: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3038: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3040: warning: assignment from incompatible pointer type /usr/src/zaptel/zaptel.c:3047: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3048: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3049: warning: passing arg 1 of `register_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `__zt_getbuf_chunk': /usr/src/zaptel/zaptel.c:4626: structure has no member named `netdev' /usr/src/zaptel/zaptel.c: In function `__zt_putbuf_chunk': /usr/src/zaptel/zaptel.c:5499: structure has no member named `netdev' make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.6' make: *** [linux26] Error 2 Could you be so kind to give me some suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Problem compiling ZAPTEL on Linux 2.6.6
Hello Fran, I try with "make linux26" but the result is the same: voipgw:/usr/src/zaptel# make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.6' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1165: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1237: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: warning: type defaults to `int' in declaration of `__mptr' /usr/src/zaptel/zaptel.c:1293: warning: initialization from incompatible pointer type /usr/src/zaptel/zaptel.c:1293: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1293: invalid type argument of `->' /usr/src/zaptel/zaptel.c:1353: warning: comparison of distinct pointer types lacks a cast /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1486: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:2950: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:2955: warning: passing arg 1 of `unregister_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c:3035: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3037: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3038: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3040: warning: assignment from incompatible pointer type /usr/src/zaptel/zaptel.c:3047: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3048: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:3049: warning: passing arg 1 of `register_hdlc_device' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `__zt_getbuf_chunk': /usr/src/zaptel/zaptel.c:4626: structure has no member named `netdev' /usr/src/zaptel/zaptel.c: In function `__zt_putbuf_chunk': /usr/src/zaptel/zaptel.c:5499: structure has no member named `netdev' make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.6' make: *** [linux26] Error 2 -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Wednesday, June 2, 2004, 6:17:12 PM, you wrote: FB> On Wed, 2004-06-02 at 14:56, Miroslav Nachev wrote: >>I have Debian Linux with kernel 2.6.6. The all packages compiled >> except ZAPTEL where I have the following error: >> voipgw:/usr/src/zaptel# make FB> make linux26 FB> F FB> ___ FB> Asterisk-Users mailing list FB> [EMAIL PROTECTED] FB> http://lists.digium.com/mailman/listinfo/asterisk-users FB> To UNSUBSCRIBE or update options visit: FB>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk & SER (www.IPTel.org)
Hi, Is it possible to use SER (www.iptel.org) toghether with Asterisk? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] DSP Coding
Hi, I would like to find some way for hardware coding instead software (using the Host CPU). Are there any PCI boards just with codecs (DSP) or other way? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] Help, Ideas and Ready for use Solutions
Hi, I would like to ask you for advice how to solve the following case: I have a client (who happened to be my friend) and I have convinced him that the IP PBX solution is much better than the conventional telephone centrals (PBX). At the beginning he wanted to buy PBX Panasonic, but at this moment he is waiting for my decision. Because at the moment we are not so deeply familiar with these technologies to be able to offer him all at once, we need your help. My client wants to have 100 internal telephones, and between 10 (existing analog lines to PSTN) and 30 (E1/PRI ISDN) external telephones. Is there any ready solution for this case we could use and how much it will cost? Thank you in advance. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Help, Ideas and Ready for use Solutions
Dear Scott, The idea is to be used new SIP phones instead legacy phones. With this the network cables (UTP/FTP-5) will be used instead one cable network for the Computers and other cable network for the Phones. Best Regards, Miroslav Nachev If there is already an existing phone system in place, you could easily migrate to an asterisk based solution if your internal phones are analog. The big question for you is not number of phone lines, but peak utilization. Here's what I have. 141 Analog Phone Lines 15 SIP IP Phones (Mix Cisco 7960 and Polycom IP500) T-1 E&M For Long Distance PRI For Local Calls with 200 DID Numbers. Max concurrent calls 15-20 (30-40 active channels) I serve all of this from a machine with the following config Dual Xeon 2.4 Ghz 2 Gig Memory 3x36 Gig SCSI Drives in Hardware Raid 5 Configuration. 2xT400P 6xAdtran TA750 Channel Banks with 6xQuadFXS cards in Loopstart Config. This is more than enough for my call volume. The CPU utilization is minimal. As you need more functionality at each phone location, you could switch to IP phones as needed. If you are starting from scratch, you will have a higher startup cost in equipment. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miroslav Nachev Sent: Friday, June 04, 2004 9:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help, Ideas and Ready for use Solutions Hi, I would like to ask you for advice how to solve the following case: I have a client (who happened to be my friend) and I have convinced him that the IP PBX solution is much better than the conventional telephone centrals (PBX). At the beginning he wanted to buy PBX Panasonic, but at this moment he is waiting for my decision. Because at the moment we are not so deeply familiar with these technologies to be able to offer him all at once, we need your help. My client wants to have 100 internal telephones, and between 10 (existing analog lines to PSTN) and 30 (E1/PRI ISDN) external telephones. Is there any ready solution for this case we could use and how much it will cost? Thank you in advance. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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@lists.digium.com
Hello, I have a question about the configuration of the SIP telephone. The situation is following: We have two SIP telephones. One of them is configured to answer the incoming calls from FXO or other directions. If there is no one available to answer to the ringing phone, I would like to take (redirect) the call and answer to it using my SIP telephone. Could you give me any ideas how to set up this configuration - how can I catch the incoming call. Thank you in advance. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] Cisco DSP Modules and Linux
Hi, I am interesting is there any way to use Cisco DSP Modules with Linux? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] Zapata.conf & Signaling for Bulgaria (PSTN: Siemens PABX)
Hi, How to configure our ZAPATA.CONF in case that the PSTN in Bulgaria is based on Siemens equipment? Now my configuration is: [channels] language=en busydetect=no when is "yes" I have problems with answering of FXO when FXS line is open callprogress=no when is "yes" I have problems with answering of FXO when FXS line is open ; interfaces for internal analog phones signalling=fxo_ks threewaycalling=yes ; interfaces for external PSTN line signalling=fxs_ks Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] License and Commercial Use
Hi, I can't find anywhere on the Asterisk web the license terms for commercial use of Asterisk software. Do I have to pay something (and how much) if I want to use the Asterisk in our IP PBX solutions? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Generator for ISDN (PRI/BRI)
Hi, I am looking for Call Generator for PRI ISDN and BRI ISDN signals. From where I can found some cheap or 2nd hand call generator (tester/analyzer)? Maybe PCI based will be cheaper than standalone solution. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Call Generator for ISDN (PRI/BRI)
Dear Michael, MD> why not use asterisk with QaudBRI and/or E100P ? Because I have to be sure that I am Euro ISDN compliant. My target is Bulgaria which is in Europe. Best Regards, Miroslav Nachev -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miroslav Nachev Sent: woensdag 23 juni 2004 16:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Generator for ISDN (PRI/BRI) Hi, I am looking for Call Generator for PRI ISDN and BRI ISDN signals. From where I can found some cheap or 2nd hand call generator (tester/analyzer)? Maybe PCI based will be cheaper than standalone solution. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Transfer - to your own number
Dear Philipp, I see that you are from Germany. I would like to ask you about the configuration for Caller ID and Zone Data (zaptel.conf/loadzone/defaultzone). I am asking you that because our standards in Bulgaria are similar to German because our equipment is Siemens. Best Regards, Miroslav Nachev ___ 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] How to transfer call in case that I am the originator
Hi, I would like to make a call and then when I am connected to the destination to transfer the call to my coleague in the office. When we receive the call it is easy using "#". But when I am the originator the "#" doesn't work. Can you give me some suggestions? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] Nightly / Daily, Weekend / Weekday and Holiday regime of the Asterisk
Hi, I would like to have different type of behaviour of our IP PBX depending on the time and the day: Weekday Nightly - 18:30 to 08:30 Daily - 08:30 to 18:30 Weekend, Holiday, etc. For example Daily the IP PBX will rings to some phones, nightly will work IVR system. How can I do that? Also is it possible the manner of dialing plan to be different depending of the caller using Caller ID? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port
Hi, We have our own algorithm handling (dial plan) the calls and different events. When we receive an external call (from FXO), probably in consequence of our algorithm, some times the FXO port remains open and we could not establish the reason why the port is not closing. We were thinking a lot what might be the problem - for example we might forget to call the "hang-up method" somewhere in the script. Unfortunately we were not able to fix the problem. We came to the conclusion that the only way to establish where the mistake is, is to ask you for information about is there any log files, which could help us tracing the actions and seeing which action is completed and which not. Seeing the actions sequence will help us to establish and solve the problem we have. We count on your help for the solution of this problem. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port
Hello Steven, The caller (originator, PSTN side) is closed the line, but the asterisk side can't understand that the caller is Hangup the line. Our PSTN is based on Siemens and Ericsson. I found some materials (documentation of Siemens PBX) where the process of negotiation is described (tones in Hz, times, etc.) but I don't know how to enter this data in Asterisk files. There is not description for this information. Also, I am looking for Caller ID detection. If you can help me will be very good. I try UK settings, but this is not working in Bulgaria. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Thursday, July 1, 2004, 6:28:52 PM, you wrote: SC> On Thu, 2004-07-01 at 11:00, Miroslav Nachev wrote: >>Hi, >> >>We have our own algorithm handling (dial plan) the calls and >> different events. When we receive an external call (from FXO), >> probably in consequence of our algorithm, some times the FXO port >> remains open and we could not establish the reason why the port is not >> closing. We were thinking a lot what might be the problem - for >> example we might forget to call the "hang-up method" somewhere in the >> script. Unfortunately we were not able to fix the problem. We came to >> the conclusion that the only way to establish where the mistake is, is >> to ask you for information about is there any log files, which could >> help us tracing the actions and seeing which action is completed and >> which not. >>Seeing the actions sequence will help us to establish and solve the >> problem we have. We count on your help for the solution of this >> problem. SC> You speak of FXO, this makes me assume you are speaking of an analog SC> POTS line. SC> If so, then your next question is which side of the call did the actual SC> hangup. If the non asterisk side did the hangup, does it provide SC> disconnect supervision? If no disconnect supervision, can you get a tone SC> pattern for busydetect or callprogress to detect those events. SC> Maybe searching around for those few new terms I just used above will SC> get you hooked up with previous threads to understand anything else you SC> need. ___ 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] ZAPTEL FXO debuging - Tones, Voltages, Ampers, etc.
Hi, Is there any program for ZAPTEL FXO with which I can debug the signals that are coming from PSTN (Tones, Voltages, Ampers, etc.)? In case that I have to do this program which is the closest entry point of the ZAPTEL software? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] One way audio when the BT-100 is behind Firewall
Hi, When we use BudgeTone-100 in our Intranet together with our Asterisk IP PBX everything is working OK. When we try to use the phone behind the Firewall we can't do the connection. When I try to use STUN Server: 128.107.250.38 there is no result. The only way in which I have audio from the one direction (BT-100 to Asterisk) is when I leave blank STUN Server and specify the IP Address in "Use NAT IP" field. Can you help me to start using BT-100 in both directions when I am behind the Firewall? This is the configuration of the phone: Software Version: Program--1.0.4.71 Bootloader--1.0.0.17 HTML--1.0.0.32 VOC--1.0.0.6 Custom Ring Tone: ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) detected NAT type is open Internet IP Address: 10.1.1.110 SIP Server: 194.12.230.167 Outbound Proxy: SIP User ID: Authenticate ID: Authenticate Password: NAT Traversal: Yes, STUN server is: Use NAT IP: 193.200.15.141 -- Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] Hangup FXO line detecting & PSTN Tone Signals Detecting
Hi, The national PSTN is built up by a Siemens & Eriksson digital PBX which, in most cases, ends up in analogue interfaces with tone dialing. It proved a hard job for me to find the most important tone signals and information messages going out of the PSTN. However, I dont know where and how to use (position) them. What I have identified so far, is as follows: * Dial Tone; * Busy (engaged) line; * Congestion; * Hook off tone; * Ring tone; * Interference Tone; * Info signal; * Warning Tone; * Special Dial Tone; * Alarm tone; * Hold on tone; * Call waiting. What I need to know is how and where I can enter this information to make Asterisk identify messages and respond in a proper way. The biggest trouble occurs when the other end caller hangs up (from the PSTN part). Asterisk cannot recognize that the line is broken because our FXO port frequently gets blocked. This can be done through the tones coming from the PTSN, which in this case, are: Frequency : 425 Hz Level : 10 dBm0 Signal (ms)/Pause (ms): 320, 320 Max. Duration : 30 s -- Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] How to calculate the price for Asterisk based Solution
Hi, We have potential client which would like to offer to him VoIP solution for 2000 subscribers (SIP based Phones) and 2 x PRI ISDN interfaces to the PSTN. In the next stage the subscribers will be increased up to 13,000. Because I am not haven't done similar big project I don't know how to calculate the price. The one way is using number of subscribers and the other is using PSTN channels (60 in this case). Also I have to calculate some price for consultants in case that we have some problems. Also I have to calculate more hardware servers for reservation, voice mail, billing, UPS and etc. Please give me some suggestions. Is there any regular price? -- Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] Connecting more Asterisk Servers in Cluster to works as one IP PBX
Hi, Is it possible to connect more than one Asterisk in cluster to works as one Asterisk IP PBX? I need of this for cases where I need of more FXO/FXS ports which can't be placed in one machine (server) and in the same time all Asterisk have to works as one with one dial plan and etc. -- Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User-Oriented Management of Asterisk
Dear Chris, We are interesting of this and would like to work together with you. Best Regards, Miroslav Nachev While I was away on vacation, buried deeply in another thread (New Asterisk bounty: SIP simultaneous), Olle E. Johansson raised a question which is close to my heart - Asterisk's management model. A management model which simply manages telephone extensions and dial plans is irrelevant to most organisations. We need a model which manages users and their interaction with the PBX. I am currently constructing a CIM model of Asterisk from a user view-point so that Asterisk could offer a WBEM management interface, thus fitting in with the other enterprise equipment in the office. There is some interest in this model within the DMTF. Is anyone else interested? Chris -- Chris Hobbs Nortel Networks Tel: +1 613 765 5386 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax over IP doesn't works
Hi, We try to send Fax through IP Network but without success. The other party use NetCentrex SoftSwitch and our communication protocol between us is H.323 (OpenH323). The error that the other party receive is: "bearer capability not imoplemented". Is it possible to send Fax using Asterisk to the other party through IP network? What T.38 and Asterisk? Regards, Miro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Fax over IP doesn't works
Hi, YC> assuming that some of these CODECS do error correction and drop YC> any information that hasn't come through instead of doing error YC> detection and request to re-transmit the lost information, is YC> somewhat expected. Are there any Fax over IP protocols? There are two ways to send a fax throuhg IP: 1. If Asterisk detect the Fax to use G.711 or 2. T.38 protocol which is exactly for that. The problem maybe is that Asterisk doesn't support both ways. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] YC> Yiannis. >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] Behalf Of Pedro Howat >> Rodrigues >> Sent: 19 October 2004 15:53 >> To: Miroslav Nachev; Asterisk Users Mailing List - Non-Commercial >> Discussion >> Subject: Re: [Asterisk-Users] Fax over IP doesn't works >> >> >> Hi , >> >> I tried this a lot, but with no sucess , even in a local network , there >> is always some loss and you receive only chunks of the original file . >> >> Pedro. >> >> Miroslav Nachev wrote: >> >> > Hi, >> > >> > We try to send Fax through IP Network but without success. The >> >other party use NetCentrex SoftSwitch and our communication protocol >> >between us is H.323 (OpenH323). The error that the other party receive >> >is: "bearer capability not imoplemented". >> > >> > Is it possible to send Fax using Asterisk to the other party >> >through IP network? What T.38 and Asterisk? >> > >> > >> > Regards, >> > Miro. >> > >> >___ >> >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 >> YC> ___ YC> Asterisk-Users mailing list YC> [EMAIL PROTECTED] YC> http://lists.digium.com/mailman/listinfo/asterisk-users YC> To UNSUBSCRIBE or update options visit: YC>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Fax over IP doesn't works
Hello Steve, There is another way to send fax if you use Fax Machine on some FXS port. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Tuesday, October 19, 2004, 7:50:52 PM, you wrote: SU> So what changes with T.38? You still need spandsp to interwork with the SU> PSTN. What was so hard about getting spandsp to work? (I'm genuinely SU> interested) SU> Regards, SU> Steve SU> Darren Sessions wrote: >> Someone should put a bounty on T38. We're using spandsp right now and >> have had success - but it was an absolute pain to get it to work. >> >> On Oct 19, 2004, at 12:38 PM, Steve Underwood wrote: >> >>> Michael Loftis wrote: >>> Just my $0.02 but seems to me the VoIP community as a whole needs to extend SIP (or IAX?) with a special 'fax data' mode wherein the gateways either act locally as the modem and queue/push bits (not audio data) for the remote end or transparently bridge them through in the case of a passthrough call. IMO faxes need to die, but business still loves them. As I said, just my $0.02. >>> >>> >>> That is such a good idea they did it several years ago. Its called >>> T.38 for H.323 and SIP. IAX doesn't yet have something similar, but >>> its high on the list of things to do. >>> >>> Regards, >>> Steve >>> >>> ___ >>> Asterisk-Users mailing list >>> [EMAIL PROTECTED] >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> ___ >> 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 >> SU> ___ SU> Asterisk-Users mailing list SU> [EMAIL PROTECTED] SU> http://lists.digium.com/mailman/listinfo/asterisk-users SU> To UNSUBSCRIBE or update options visit: SU>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Fax over IP doesn't works
Hello Steve, In the project OpenH323 the T.38 is supported. The easyest way is to correct Asterisk logic with OpenH323. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Tuesday, October 19, 2004, 7:38:19 PM, you wrote: SU> Michael Loftis wrote: >> Just my $0.02 but seems to me the VoIP community as a whole needs to >> extend SIP (or IAX?) with a special 'fax data' mode wherein the >> gateways either act locally as the modem and queue/push bits (not >> audio data) for the remote end or transparently bridge them through in >> the case of a passthrough call. >> >> IMO faxes need to die, but business still loves them. >> >> As I said, just my $0.02. SU> That is such a good idea they did it several years ago. Its called T.38 SU> for H.323 and SIP. IAX doesn't yet have something similar, but its high SU> on the list of things to do. SU> Regards, SU> Steve SU> ___ SU> Asterisk-Users mailing list SU> [EMAIL PROTECTED] SU> http://lists.digium.com/mailman/listinfo/asterisk-users SU> To UNSUBSCRIBE or update options visit: SU>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Fax over IP doesn't works
Dear Steve, SU> So how does the FAX get from the fax machine to the T.38 channel SU> with spandsp? In our case we will try to strip spandsp and will use directly OpenH323. We do tests for compatibility with one of the biggest national telecom and if they are OK, they will offer Asterisk based IP PBX to their clients instead Cisco. That's why we need of T.38 and G.711 fax capabilities. Also we have the problems with the following tests: 1. When Dialing of unallocated number the resposne must be "Invalid Number", but the result is one of the following: Hangup, Congestion or Busy. 2. CLIP/CLIR User provided verified and passed - We can't find where we can set this bits for this services. 3. Fax T38 / g711 4. Codec negotiation: when 2 codecs are possible (G.711 and G.729), the two parties can't negotiate which codec to use. Best Regards, Miroslav Nachev Miroslav Nachev wrote: >SU> and exactly how does that get the FAX into the T.38 channel? :-\ > > Using G.711 or implementing T.38 in Asterisk or adjusting Asterisk >to OpenH323 T.38. From our expirience Asterisk detect that the line is >with Fax data. The problem is what next. > > > So how does the FAX get from the fax machine to the T.38 channel with spandsp? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Fax over IP doesn't works
Dear Steve, I can't understand from your mail can I use SpanDSP or not? Today we try this fax-modem: http://www.openh323.org/t38.html The problem now is that we can't start it with HylaFAX. Best Regards, Miroslav Nachev SU> Hi Miroslav, SU> It sounds like you don't really understand what T.38 is. You need some SU> form of modem to get from a normal FAX machine to a T.38 channel. SU> spandsp can do that. A normal FAX modem can do that. You need a modem SU> somewhere, though. That is why developing the FAX modems was the first SU> step towards providing T.38. SU> Regrds, SU> Steve SU> Miroslav Nachev wrote: >> Dear Steve, >> >>SU> So how does the FAX get from the fax machine to the T.38 channel >>SU> with spandsp? >> >> In our case we will try to strip spandsp and will use directly >>OpenH323. We do tests for compatibility with one of the biggest >>national telecom and if they are OK, they will offer Asterisk based IP >>PBX to their clients instead Cisco. That's why we need of T.38 and >>G.711 fax capabilities. >> Also we have the problems with the following tests: >> 1. When Dialing of unallocated number the resposne must be "Invalid >> Number", but the result is one of the following: Hangup, >> Congestion or Busy. >> 2. CLIP/CLIR User provided verified and passed - We can't find >> where we can set this bits for this services. >> 3. Fax T38 / g711 >> 4. Codec negotiation: when 2 codecs are possible (G.711 and G.729), >> the two parties can't negotiate which codec to use. >> >> >> Best Regards, >> Miroslav Nachev >> >> >>Miroslav Nachev wrote: >> >> >> >>>SU> and exactly how does that get the FAX into the T.38 channel? :-\ >>> >>> Using G.711 or implementing T.38 in Asterisk or adjusting Asterisk >>>to OpenH323 T.38. From our expirience Asterisk detect that the line is >>>with Fax data. The problem is what next. >>> >>> >>> >>> >>> >>So how does the FAX get from the fax machine to the T.38 channel with >>spandsp? >> >>Regards, >>Steve >> >>___ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ 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] Yoda SIP Devices: IAD100, IAD200, IAD211, IAD400 and other
Hi, Is there any experience with Yoda VoIP Devices & Asterisk? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] asterisk & ipv6
Dear Olle, I can say that Emil Ivov has very good knowledge on IPv6 too. You can use it. Best Regards, Miroslav Nachev OEJ> Marc Blanchet wrote: >> - no asterisk does not work over ipv6. >> - ipv6 port won't be as easy as I would like to... >> - I'm currently working on it. Had a short discussion with Mark during >> astricon on it. >> - no release date promised... >> - will certainly be available through a different source tree. Too many >> changes in the code and many impacts to try to merge it with current >> sources. When port done, would need to discuss with developers on how to >> integrate these changes. OEJ> Marc, OEJ> Can we start backwards and write coding guidelines, like OEJ> "don't use strings like ip:port with a colon, since a colon is a significant OEJ> character in IPv6 addresses" OEJ> "size the IP address string for IPv6, size being..." OEJ> Also, we need to consider some architecture changes to be able to proxy OEJ> from IPv6 to IPv4. How will this affect IAX, SIP and H.323? OEJ> Maybe it's time to start a Wiki page named "the Asterisk IPv6 project" OEJ> so we can start collecting information, creating guidelines. OEJ> For the project: Marc has very good knowledge on IPv6 and is well-known OEJ> in IPv6 circles. I was very happy to see him at Astricon, myself being one that OEJ> has been asking about IPv6 and Asterisk many times. We need to listen to OEJ> Marc. IPv6 is big in Asia and Asterisk needs to run on those networks as OEJ> well . OEJ> /O OEJ> -- OEJ> This project will be a good addition to our project list OEJ> * The Asterisk FreeBSD project OEJ> * The Asterisk Documentation Project OEJ> * The Asterisk Solaris project OEJ> * The Asterisk SS7 project OEJ> * The Asterisk testing project OEJ> As a side note, we might to create a project page listing these. OEJ> Did I miss a project? OEJ> /O OEJ> ___ OEJ> Asterisk-Users mailing list OEJ> [EMAIL PROTECTED] OEJ> http://lists.digium.com/mailman/listinfo/asterisk-users OEJ> To UNSUBSCRIBE or update options visit: OEJ>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ACT Gateways
Hello Joseph, I tested their products and the Phones are OK. They are better than GrandStream. The Gateways schould be improved. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Monday, October 25, 2004, 3:11:44 AM, you wrote: J> Has anybody tested any gateways from ACT: J> http://www.act-tel.com.tw/Index2.htm J> They have four different configurations: J> 4xFXS - 4xFXO J> 2xFXS - 2xFXO J> 1xFXS - 1xFXO J> 4xFXS J> I emailed them but they didn't bother the respond. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ACT Gateways
Dear Benjamin, Unfortunately the Mediatrix products are very expensive. For example the price for Two-port access device with SIP protocol is $275. The price for the same of good looking Gateways from Yoda (www.yoda.com.tw) is $85. Unfortunately I haven't experience with Yoda devices. Also they required min. 10 pcs for ordering. Best Regards, Miroslav Nachev BoAML> On Sun, 24 Oct 2004 18:11:44 -0600, Joseph BoAML> <[EMAIL PROTECTED]> wrote: >> Has anybody tested any gateways from ACT BoAML> Last time those gateways came up in a conversation it was concluded BoAML> that efforts should be concentrated on the phones so as to not dilute BoAML> engineering resources with too many things. This was about 2 months BoAML> ago or so. I have come to read this as: gateways are not ACT's BoAML> priority -- phones are. I would therefore recommend to look for BoAML> gateways elsewhere, for example Mediatrix. BoAML> rdgs BoAML> benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] ACT Gateways
Dear Benjamin, I think that the combination of FXS with FXO ports is very useful for the SOHO and Medium Enterprises. Also the prices of Yoda and other Asian companies are very suitable (~$40 per FX? port) and is better than GrandStream ATA. The other future that is extra than GrandStream is WAN/LAN ports and Router/NAT possibilities. Yes, I now all prices of Yoda, and I am looking for some partners with which to combine one order for samples. The problem is that they require min. 10 pcs per order. The price for samples of VG400 is $320. The device use TI DSP for coding. The prices for regual (big) quantities is $60 per port. Best Regards, Miroslav Nachev BoAML> On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev BoAML> <[EMAIL PROTECTED]> wrote: >>Unfortunately the Mediatrix products are very expensive. BoAML> just one example. my point was that as of this moment, ACT are more BoAML> focussed on their phones and it may well be wise to look for gateways BoAML> elsewhere for the time being, whereever that elsewhere may be. >> example the price for Two-port access device with SIP protocol is >> $275. BoAML> I don't really understand the obsession with FXS devices. BoAML> The only uses I see for FXS are BoAML> - connect a FAX machine, where FAX may not be the best application for BoAML> VoIP anyway, BoAML> - connect an existing cordless phone, where you probably have only one BoAML> such device and a Grandstream HT286 will just do fine, BoAML> - connect the analog phone in a hotel to a travel adapter, IAXy would BoAML> seem to be the best choice here because you are so much more likely to BoAML> encounter NAT traversal problems and other obstacles that you may not BoAML> be able to resolve with a SIP device, BoAML> - feed some Internet based phone services into a legacy PBX that wants BoAML> to see them as CO lines, here again, depending on the number of feeds, BoAML> HT286 may be cheap and cheerful enough. BoAML> For anything else IP phones should be the default with no buts and no BoAML> ifs. I am always puzzled by how people desperately hang on to legacy BoAML> stuff they don't really need and in the process create a beast of a BoAML> kludge technology. The x86 architecture (or lack thereof) should be an BoAML> example that serves to show how not to design your stuff with legacy BoAML> support as your all-overriding number one priority. So, let's not make BoAML> the same mistake with VoIP. Let's get rid of analog phones as fast and BoAML> forcefully as we possibly can. BoAML> In other words, FXS should be the very very last resort when there is BoAML> really no other way. BoAML> Having said that, I notice that Yoda have a 4 port FXO gateway BoAML> (VG400), or at least it can be configured to be a 4 port FXO gateway. BoAML> Now, that is rather interesting. Do you have any idea how much this BoAML> device costs (ballpark figure wise) and how well it can adapt to PSTNs BoAML> in other countries? BoAML> rgds BoAML> benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] ACT Gateways
Dear Alex, From where you found this device for $165? I found that the List Price of this device is $220. Can you send me the URL or some contacts? Best Regards, Miroslav Nachev AB> Hi, AB> We choose the Mediatrix 2102 with 2 analogue and 2 ethernet ports. AB> Cost: £89.99 (roughly equiv $165). AB> We are using these to hook up Faxes and DECT phones (cordless). AB> The top of the range business DECT from from BT is £30 (if you buy a few from trade). AB> Worth mentioning that even VoiceMail indication works on the AB> BT analogue phone. Also the voice quality was actually better on AB> the top of the range business DECT phone than the top of the range AB> home BT phone which retails at around £90 (the one that includes AB> SMS / mobile sim card support). AB> What other cordless choices are there for native SIP phones??? AB> Zyxel Prestige 2000W Wireless SIP Phone = £159.99 (on sale even). AB> I think you can easily do the math and realise what the best option is. AB> HTH AB> Alex AB> -Original Message- AB> From: Benjamin on Asterisk Mailing Lists AB> [mailto:[EMAIL PROTECTED] AB> Sent: 25 October 2004 11:32 AB> To: Miroslav Nachev AB> Cc: Asterisk Users Mailing List - Non-Commercial Discussion AB> Subject: Re: Re[2]: [Asterisk-Users] ACT Gateways AB> On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev <[EMAIL PROTECTED]> wrote: >>Unfortunately the Mediatrix products are very expensive. AB> just one example. my point was that as of this moment, ACT AB> are more focussed on their phones and it may well be wise to look AB> for gateways elsewhere for the time being, whereever that AB> elsewhere may be. >> example the price for Two-port access device with SIP protocol is >> $275. AB> I don't really understand the obsession with FXS devices. AB> The only uses I see for FXS are AB> - connect a FAX machine, where FAX may not be the best application for VoIP anyway, AB> - connect an existing cordless phone, where you probably have AB> only one such device and a Grandstream HT286 will just do fine, AB> - connect the analog phone in a hotel to a travel adapter, AB> IAXy would seem to be the best choice here because you are so much AB> more likely to encounter NAT traversal problems and other AB> obstacles that you may not be able to resolve with a SIP device, AB> - feed some Internet based phone services into a legacy PBX AB> that wants to see them as CO lines, here again, depending on the AB> number of feeds, HT286 may be cheap and cheerful enough. AB> For anything else IP phones should be the default with no AB> buts and no ifs. I am always puzzled by how people desperately AB> hang on to legacy stuff they don't really need and in the process AB> create a beast of a kludge technology. The x86 architecture (or AB> lack thereof) should be an example that serves to show how not to AB> design your stuff with legacy support as your all-overriding AB> number one priority. So, let's not make the same mistake with AB> VoIP. Let's get rid of analog phones as fast and forcefully as we AB> possibly can. AB> In other words, FXS should be the very very last resort when there is really no other way. AB> Having said that, I notice that Yoda have a 4 port FXO AB> gateway (VG400), or at least it can be configured to be a 4 port AB> FXO gateway. Now, that is rather interesting. Do you have any idea AB> how much this device costs (ballpark figure wise) and how well it AB> can adapt to PSTNs in other countries? AB> rgds AB> benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ACT Gateways
Dear Alex, Which model you have? We test G2DSU but the model is with too many problems: 1. The different items of the same model and the same firmware are with different problems; 2. PSTN (FXO) to IP doesn't works for some devices - see point 1; 3. SIP TA devices can't be registered to the SIP Server; 4. For devices where the calls from PSTN to the FXO is working the ring signal is not sent to SIP Server. Only the connected Analog Phone to the FXS port rings. This is maybe because of point 3. 5. The only way to place a call to the SIP TA is using: [EMAIL PROTECTED] Maybe this is because of point 3. Best Regards, Miroslav Nachev ac> hi all ac> as we have brought sample to test in singapore and ac> have start deployment , if you need may be can email ac> to me. ac> regards ac> alex chua ac> --- Miroslav Nachev <[EMAIL PROTECTED]> wrote: >> Hello Joseph, >> >> I tested their products and the Phones are OK. They >> are better than >> GrandStream. The Gateways schould be improved. >> >> >> -- >> Best regards, >> Miroslav >> mailto:[EMAIL PROTECTED] >> >> Monday, October 25, 2004, 3:11:44 AM, you wrote: >> >> J> Has anybody tested any gateways from ACT: >> J> http://www.act-tel.com.tw/Index2.htm >> >> J> They have four different configurations: >> J> 4xFXS - 4xFXO >> J> 2xFXS - 2xFXO >> J> 1xFXS - 1xFXO >> J> 4xFXS >> >> J> I emailed them but they didn't bother the >> respond. >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> ac> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> >> ac> http://lists.digium.com/mailman/listinfo/asterisk-users >> ac> __ ac> Do You Yahoo!? ac> Download the latest ringtones, games, and more! ac> http://sg.mobile.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323-0.6.3b
Dear Morina, If you use Asterisk 1.0 stable and asterisk-oh323-0.6.3b and the last OH323 from the CVS you must compile everything without errors. We had some problems with Asterisk 1.0.1 and asterisk-oh323-0.6.3b because in the new Asterisk version the callerid variable is struct comparing with the 1.0 version where is string. When we replace callerid variable with cid.cid_num the problem was solved. Best Regards, Miroslav Nachev m> Hi all, m> I'm trying to compile asterisk-oh323-0.6.3b but I got some comiling errors m> just on start. Can someone tell me the steps and the packages required to m> compile asterisk-oh323-0.6.3b. m> I'm usig asterisk-1.0.0 on Linux gentoo 2.4.25-gentoo-r3 . m> Thank you, m> Astrit Morina m> System Operator m> Tel: 038 20304050 m> Fax: 038 20304020 m> E-mail: [EMAIL PROTECTED] m> www.ipko.net m> ___ m> Asterisk-Users mailing list m> [EMAIL PROTECTED] m> http://lists.digium.com/mailman/listinfo/asterisk-users m> To UNSUBSCRIBE or update options visit: m>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to catch events like Dial, Ringing, Transfer, Hold, Forward, Hangup, Park, UnHold, Answered, No Answer, etc.
Hi, We would like to consult from where is the right place to catch the following events: 1 - Dial 2 - Ringing 3 - Transfer 4 - Hold 5 - Forward ??? 6 - Hangup 7 - Park 8 - UnHold 9 - unpark 10 - Pickup Call // Dial Statuses 11 - Answered 12 - No Answer 13 - Busy 14 - Cancel 15 - Channel Unavailable 16 - Congestion Up to now we catch the following events as follow: 1. Dial - apps/app_dial.c -> dial_exec 2. Ring - apps/app_dial.c -> wait_for_answer 6. Hangup - channel.c -> ast_check_hangup 7. Park - res/res_features.c -> ast_park_call We need of this events for new CDR & Prepaid systems. Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk & NetCentrex CCS integration
Dear Juan, We have some success using H.323 as common protocol (OpenH323), but there are some problems with Fax (T.38) and Multiply Codecs (when you set more than 1 codec. For example G.729 and G.711). Best Regards, Miroslav Nachev JVG> Has somebody ever tried to integrate Asterisk to a NetCentrex CCS platform? JVG> Juan V. Guerrero JVG> Projects and planning engineer JVG> Telecarrier Inc. JVG> Phone: (507) 3009516 JVG> Cel: (507) 6143156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 port ISDN BRI pci card
Dear Bartosz, Try this: http://www.junghanns.net/asterisk/page17.html quadBRI PCI ISDN EUR 600,- Best Regards, Miroslav Nachev BJ> Hello, BJ> I am looking for 4 port ISDN BRI card. BJ> I have checked wiki and found one, but they do not show prices BJ> for that card. Can somebody advise me which ISDN 4 port card works good BJ> with Asterisk, BJ> Thank you in advance. BJ> Bartosz BJ> ___ BJ> Asterisk-Users mailing list BJ> [EMAIL PROTECTED] BJ> http://lists.digium.com/mailman/listinfo/asterisk-users BJ> To UNSUBSCRIBE or update options visit: BJ>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Hardware Platform - Intel x86 versus Intel RISC Xscale (ARM)
Hi, I would like to take your advice about which hardware paltform is better for Asterisk - x86 or RISC ? I have the following offers: - Mobile Celeron 733MHz $380 - Xscale 667MHz $330 x86 cost is higher than RISC-solution, but the performance is better. The technical specification for both CPU is the same: External FLASH:1 x MMC/SD VGA/LCD: 1 x VGA/LCD Audio: AC 97 USB: 6 x USB 2.0 Integrated FLASH: 128 MB Integrated RAM:256 MB External RAM: DIMM up to 1 GB Ethernet: 2 x 10/100 SATA: 1 x Serial ATA Mini PCI: 1 x Mini PCI PCI: 1 x PCI for PCI Raiser Card with 3 x PCI Slots Power Supply: Single 48 VDC Power Input (36V - 72V), Power over Ethernet specification (IEEE 802.3af). Jumper selectable to choose the source: power connector (jack) or WAN port. Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re[4]: [Asterisk-Dev] Asterisk Hardware Platform - Intel x86 versus Intel RISC Xscale (ARM)
Hello Scott, SL> Does that include FP hardware? I don't believe that any of the SL> PDA Xscales do, I assume that at least some codecs need FP for SL> compression; without floating point hardware, it's going to be SL> really slow. 1st in Xscale is integrated Micro Signal Architecture (MSA), the new design incorporates DSP and microcontroller functions - Intel and Analog Devices joint development design a digital signal processor (DSP) core architecture. All speech, audio and video codecs like G.729, G.723, G.726, G.728, GSM, MP3, MPEG-4, etc. are optimized for Xscale architecture using integrated DSP functionality. 2nd we are not care about host CPU Speech coding because we are in process of development of USB/PCI Device named Multimedia Transcoder (MMT) which will handle simultaneously from 16 up to 256 Codecs. This codec will be available Apr-May 2005. The price will be between $400 and $1000. Using this MMT you can use 50 MHz CPU for Asterisk. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Thursday, November 25, 2004, 7:14:50 PM, you wrote: SL> On Nov 25, 2004, at 7:58 AM, Miroslav Nachev wrote: >> Hi, >> >> To clarify Xscale, I mean the latest Xscale Network Processor IXP465 >> which is made on 90 nm technology: >> http://www.intel.com/design/network/products/npfamily/ixp465.htm SL> Does that include FP hardware? I don't believe that any of the PDA SL> Xscales do, I assume that at least some codecs need FP for compression; SL> without floating point hardware, it's going to be really slow. SL> Scott ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] Mini-ITX Mainboard for Asterisk IP PBX, Intel Mobile Celeron 733MHz
Hi, I would like to offer you the following specialized embedded Mini-ITX Mainboard: Samples: $390 50 pcs: $270 100 pcs: $255 The Technical Specification is: Dimension: Mini-ITX, 170x170mm System Processor:Intel Mobile Celeron 733MHz (Fanless) Chipset: Intel 830M + ICH4 BIOS:Award Flash 256K BIOS System Memory: One DIMM socket for SDRAM memory module up to 512MB Display Controller: Intel 830M integrated graphics contoller CRT: Integrated 350-MHz RAMDAC, supports progressive scan analog monitor up to a resolution of 1800 x 1440 pixels LCD: Onboard LVDS Transmitter through DVO port TV: Onboard TV-out encoder Focus FS454 through DVO port Ethernet Controller: Two PCI-bus Ethernet controllers realtek RTL8100C, one for WAN with Power over Ethernet (IEEE 802.3af) and the other one is for LAN Sound Output:AC 97 V2.3 for Line-out, line-in and Mic-in CompactFlash:One type-II compact flash socket IDE Interface: 2 x IDE ports Serial-ATA: 1 x Serial ATA port USB: 6 x USB 2.0 IR Interface:1 x IrDA Expansion slots: 1 x PCI slot for PCI Raiser Card with 3 PCI 1 x Mini-PCI Power Connector: DC-in Jack for DC +48V Power management:ACPI function RTC: LPC Super I/O including Hardware Monitor:LPC Super I/O including Operating Temp.: 0-60 degree C Humidity:5-95% RH, non-condensing It is possible to be added 3rd Ethernet port for DMZ or other purposes. Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: 12.50$ per port ???
Hi, MK> ... 2) Buy 4 Carrier Access Channel Banks for $100 ... From where can I buy Carrier Access Channel Banks for $100? Any URL or related info? Best Regards, Miroslav Nachev MK> Shoval, MK> Interesting Mention. I agree, most people don't have CO exp. And I MK> wish daily I had enough. MK> Understand that what I mean by my e-mail is consumer side FXS ports, in MK> broader terms, I mean, customer picks up a phone line, it signals a MK> channel bank which signals *. 24 of those channels. MK> Not channels equipped to Send Signal to the CO that a loop has been made.. MK> meaning FXO. MK> 24 FXS, $100. I don't deal with FXO since I deal w/ PRI.. and do not need MK> FXO ports. MK> My thought here was related to downstream customers.. which in the this MK> world implies FXS. MK> Talk was mentioned for DSPs.. Echo Can and Codec Management, this MK> interests me because of the unique hardware requirements of the 4 port MK> cards.. mainly in interrupts and CPU usage. Fit 10 cards to one system, MK> that's 40 ports.. fit double density, that's 80.. this interests me on the MK> cPCI platform. Others know what I think about this and why. MK> Beyodn that, my point here is 1) Buy a T400P for $800 and 2) Buy 4 Carrier MK> Access Channel Banks for $100 and 3) You'll have a 96 Downstream Port MK> solution for $1200, meaning $12.50 a Port NRC. MK> If you can find a cheaper *OR* easier solution, let me know. Because it'll MK> save me money and you'll be my friend. ;) MK> As for FXO, my best solution was Mainstreet Newbridge 3624s from Ebay for MK> about $150-$300 a box w/ 12 port FXOs a while back. Then I moved to PRI MK> for the capability of setting Caller ID easily (e&m was a pain in the..). MK> wherd MK> -Matt MK> - MK> "I believe there are more instances of the abridgment MK> of the freedom of the people by gradual and silent MK> encroachments of those in power than by violent and MK> sudden usurpations." - James Madison MK> On Wed, 15 Dec 2004, Shoval Tomer wrote: >> >> >> > -Original Message- >> > From: Matt Klein [mailto:[EMAIL PROTECTED] >> > Sent: Wednesday, December 15, 2004 11:19 AM >> > To: Asterisk Users Mailing List - Non-Commercial Discussion >> > Cc: [EMAIL PROTECTED] >> > Subject: Re: [Asterisk-Users] Re: Ethernet Channel Bank idea >> > >> > Ethernet Channel Bank >> > >> > Hmm. >> > >> > Caught my attention for more than 34 hours, you win. >> > >> > I'm getting 24 port carrier access channel banks for $100, Digium 4 >> Port >> > Cards for about $800 (T400P) a card.. Meaning a blended cost of >> ~$12.50 >> > per channel NRC. I can mux up 96 channels for a cost of $12.50 per >> channel >> > all day long. And easily sell it at $25 to cover the cost of the box >> per >> > port, T-1 channel per port, and channel bank per port non recurring >> costs. >> > >> >> Matt, >> >> A 24 port channel bank for $100 ??? >> Where? >> If you can get one that's working and has 8 FXO ports and 16 FXS ports, >> and is one of the brands that work well with * (support all or most >> features) for $100, I'll buy it from you for $200. >> >> Now, as to why we're interested in an Ethernet channel bank, you should >> keep in mind that some of the people on the list have no telco >> experience. >> And that's fine by me (especially since I'm one of them). >> You can manage a Cisco call manager, and other VOIP systems that >> originated from the IP side of the VOIP world without being a telco >> expert. >> >> VOIP systems that originated from the VO side, like ones from Panasonic, >> LG, etc. are cryptic at best to us non telco guys. >> >> We don't have the experience and knowledge necessary to setup channel >> banks (it's not like they have 24 RJ45 ports on them, right?) >> >> An Ethernet channel bank will probably be much easier for us to set up. >> >> I can understand and respect telco guys, who've setup channel banks for >> TDM PBXs for years not being afraid to try and hook it up with *. >> >> Me? I'm scared as hell >> >> >> >> >> Shoval Tomer, >> IT Manager, >> SofTov Advanced Systems, Ltd. >> Office: +972-3-9230686 ext. 179 >> Fax: +972-3-9216642 >> Mobile: +972-54-8000200 >> >> MK> ___ MK> Asterisk-Users mailing list MK> [EMAIL PROTECTED] MK> http://lists.digium.com/mailman/listinfo/asterisk-users MK> To UNSUBSCRIBE or update options visit: MK>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware based DSP
Dear Shahed, We are in process to done hardware DSP device for coding of G.729, G.723, GSM and other speech codecs. We will support the drivers for Asterisk, SER and OpenH323. The device will be available at middle of next year. There are 6 variants of the device: - USB with 16, 32 and 64 simultaneous channels; - PCI with 64, 128 and 256 simultaneous channels. In the future the device will support Audio and Video Codecs and Windows OS. Best Regards, Miroslav Nachev S> Hi All, S> Is it correct to say that by design, asterisk wont make use of any cards S> hardware dsp capabilities ? S> I don't think that any of the hardware cards currently supported S> have any dsp capabilities, but I wanted to know if for example, S> in the future a driver was written for a card that did have dsp S> capabilities, S> would asterisk be able to make any use of it ? S> I am only just starting out with asterisk, and have not fully understood S> the architecture yet, but it seems that in order to handle VoIP and PSTN S> seamlessly, all dsp related functionality has to be handled by software S> Thanks S> Shahed S> ___ S> Asterisk-Users mailing list S> [EMAIL PROTECTED] S> http://lists.digium.com/mailman/listinfo/asterisk-users S> To UNSUBSCRIBE or update options visit: S>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Hardware based DSP
Dear Shahed, If you use hardware DSP for encoding and decoding you will need of less power Host CPU. There are no difference for what you will use these codecs - I mean conference or single conversation, because for conference all codecs are converted to G.711, then mixed and then back to the original codec for each party of conference group. So, if you would like to to conference with software coding you will need of very power computer. Best Regards, Miroslav Nachev S> Thank you all for your explanations related to my question. S> I have one follow-up question though. S> When I said that all dsp related stuff has to be handled S> by software within asterisk, I was thinking of S> conferencing at the time. S> I mean in order to be able to conference a sip session with S> a PSTN call, it would have to be handled by software, even S> if both the channels had hardware dsp capabilities. Right ?? S> If you are dealing with just a single channel, then the driver S> may handle codec/echo cancelation stuff with hardware help (??) S> As an aside, what is the best way to go about learnig about S> the aritecture of asterisk, other than "using the source" ?? S> The Wiki pages are great, but I have not (yet) found any info S> about asterisks architecture itself (in depth that is). S> Some linked websites / blogs provide good info on some topics, S> but is there a good high level design doc available anywhere ? S> It would stop people like me asking so many basic questions in S> the -dev list. S> Thanks S> Shahed S> ___ S> Asterisk-Users mailing list S> [EMAIL PROTECTED] S> http://lists.digium.com/mailman/listinfo/asterisk-users S> To UNSUBSCRIBE or update options visit: S>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-doc] New Hardware Support
Dear Yoram, This question is not for this discussion. The right places are Users or/and Dev. Which kind of hardware PCI Card or Standalone Gateway? By the way your hardware is compliant with the following main requirements, then your hardware can work with Asterisk: 1. Registrar support - accept and handle REGISTER requests; 2. Voice Gateway Capabilities (PSTN to IP and IP to PSTN media conversion) termination of incoming calls from PSTN (FXO) and analogue phones (FXS) to the SIP Devices and support of outgoing calls from SIP Devices to the PSTN and Analogue Phones; 3. Support for REFER requests (http://www.ietf.org/rfc/rfc3515.txt). -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Tuesday, July 27, 2004, 6:54:46 PM, you wrote: YH> Hi, YH> YH> If I have some hardware that supports telephony YH> interfaces,DTMF detection, different vocoders and so on, is it YH> possible to attach thishardware to the Asterisk software? YH> In other words, does the Asterisk software have someHardware YH> Independent Layer (HAL) that defines the function prototypes YH> thatshould be implemented so that the Asterisk software will run? ___ 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] How to detect the sent status of the Fax
Hi, I start the fax capabilities of Asterisk, but I don't know how to detect that the sent fax status - complete, error, etc. Any ideas? Best Regards, Miroslav Nachev ___ 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] Help for VoIP Gateway with 2 x FXO & 2 x FXS
Hi, I have MicroNet VoIP Gateway SP5014 with 2 x FXO, 2 x FXS & Ethernet ports. One friend say that the Config Menu of this GW is very similar ot WellTech Config Menu. How to start this GW together with Asterisk? Best Regards, Miroslav Nachev ___ 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] Cisco SIP Phone 7960 & DTMF Problem
Hi, When we use BudgeTone where the DTMF is set to "via RTP (RFC2833)" all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to "remote pick-up the call" through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem
Dear Nicolas, NG> Did you try by dialing just '*8' ? I try, but the result is the same. The problems is in Cisco Phone, because the same account with BudgeTone is working well. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Wednesday, August 4, 2004, 7:23:31 PM, you wrote: NG> Hello, NG> On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote: >>Hi, >> >>When we use BudgeTone where the DTMF is set to "via RTP (RFC2833)" >> all the DTMF functionality of Asterisk is working OK. When use Cisco >> 7960 the transfer is working OK, but when I try to "remote pick-up the >> call" through '*8#' I can't do that because the Cisco Phone start busy >> signal. >>How can I start using all DTMF features using Cisco Phone? NG> Did you try by dialing just '*8' ? ___ 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] Entry point & Minimal requirements for Linux Device Driver and connection with Asterisk IP PBX
Hi, I would like to make Linux Device Driver for one PCI board and I would like to know what I have to do to can communicate with Asterisk IP PBX? Are there any requirements and writen specification? There are 2 cases but I think it is possible to combine it in one. The 1st PCI Card is with DSP channels for coding and transcoding and the 2nd board is with FXO and FXS ports. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] ZAPTEL & ZAPATA
Hi, In the old Asterisk CVS we need of ZAPTEL and ZAPATA sources. Now in the Asterisk Download instructions ZAPATA is missing: http://www.asterisk.org/index.php?menu=download Do we need more of ZAPATA? What is the role of ZAPATA? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] Which is PCI HDLC Chip used for TE405P / TE410P / E100P ?
Hi, I havn't E1 board from Digium but I would like to know which is the used IC (chip) for PCI HDLC communication? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk and SER
Hi, It is good that the SER is used for SIP Proxy. But in this case how to use the PBX capabilities of Asterisk like IVR, VoiceMail, DialPlan, and etc.? Best Regards, Miroslav Nachev --- Kurtz <[EMAIL PROTECTED]> wrote: > Why is it that the wiki indirectly recommends SER (or another proxy) out in front of > Asterisk. > If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, > I'm not > familiar with network configuration really, although I do understand most of the > basics. > Asterisk is not a SIP proxy, it is a UAS, and also a SIP Registrar. Many use SER and Asterisk together, SER as SIP proxy and Asterisk as PSTN gateway. The advantage is that this combination is highly scalable. Not sure whether Asterisk supports RADIUS authentication. AFAIK, it is not supported, but i beleive some works is in progress in this direction. Do a search on the archives, and you'll get many links on this. Regards, Girish __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMoE crash the Asterisk and the Network
Hi, We try to start DTMoE but the result is that the Asterisk and the Network are crashed. Are there some successful stories with DTMoE? Any help will be very useful. Best Regards, Miroslav Nachev ___ 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] TDMoE crash the Asterisk and the Network
Hi, We try to start TDMoE but the result is that the Asterisk and the Network are crashed. Are there some successful stories with TDMoE? Any help will be very useful. Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work
Hi, We have a case where we need of 16 x FXS, 12 x FXO and 1 x E1. To do this using Digium products I need of 8 PCI slots. This is not possible to be done in one computer and that's why I try to start using TDMoE. Unfortunately all my tries are without success. The network is crashed everytime. Can you give me some ideas/suggestions how to solve this case? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work
Hi, C18> I suggest you go the channel bank route. Can you be more detailed? Any URL? What is this and how to do it? On Wed, 18 Aug 2004 10:16:01 +0200 Miroslav Nachev <[EMAIL PROTECTED]> wrote: >Hi, > >We have a case where we need of 16 x FXS, 12 x FXO and > 1 x E1. To > do this using Digium products I need of 8 PCI slots. This > is not > possible to be done in one computer and that's why I try > to start > using TDMoE. Unfortunately all my tries are without > success. The > network is crashed everytime. >Can you give me some ideas/suggestions how to solve > this case? > > >Best Regards, >Miroslav Nachev > > > > ___ > 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 _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work
OK, but there are just FXS ports. What about FXO ports? Miroslav Nachev wrote: >Hi, > > C18> I suggest you go the channel bank route. > >Can you be more detailed? Any URL? What is this and how to do it? > You can start by looking at the WiKi pages: http://www.voip-info.org/wiki-Asterisk+Hardware (under the Channel Bank section) http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware%20channel%20bank%20check http://www.voip-info.org/tiki-index.php?page=Channelbank That should be more than enough information to get you started. You could also run a seach on the lists.digium.com archives via Google. Cheers Faiz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Network Crashed when we try to start TDMoE
Hi, We try to start TDMoE but the result is that the Asterisk and the Network are crashed. Are there some successful stories with TDMoE? Any help will be very useful. Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + ser
Dear Pavel, Go to http://www.voip-info.org/wiki-Asterisk or search in google Best Regards, Miroslav Nachev Hello guys, I'm new to asterisk and I have some problems - would you help me please. I installed following configuration: Linux server with Asterisk + Digium X100P FXO card I dial the line - and I hear the voice : Congratiolations, you succesfully installed... and so on I want to use this configuration for this situation Real Phone Line - Aterisk/SIP - Office SIP Phones(X-lite) with internal numbers I want to be able to call outside using prefix - if want to call for example 123456 to enter in sip phone 0 123456 and to call this number and the other way - somebody calling the office phone to hear instructions - to hear pavel enter 1 to redirect him/her to my internal sip number, to hear y enter 2 and so on... I read so a lot of documantation but I have no success till now. Somebdoy can help me with easy guide? The other my question is how I can use ser and asterisk together? Some tutorial ? I've runned SER and it's working fine - I want to use it with asterisk for voice mail if user is not available or redirecting ... Please help me Thanks in advance Pavel ___ 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] GSM to BRI ISDN Gateway
Hi, I am looking for GSM to BRI ISDN Gateway. Any help? -- Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ 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] GrandStream HT-486 ATA as VoIP Gateway
Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? -- Best Regards, Miroslav Nachev ___ 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] ProSLIC and measuring of PSTN parameters like Voltage, Polarity, Power (A) and Frequency (Hz)
Hi, I am interesting how can I use the capabilities of ProSLIC to measure the following PSTN parameters: - Voltage (V) & Polarity (+-); - Current (A); - Frequency (Hz). Are there any ready for use tools? If there aren't ready for use tools how can I do the above measuring? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] new ATA box for sale by Linksys
Hi, http://www.costcentral.com/searchresult.php?keyword=PAP2&searchin=1 Mfg Part # Stock Price -- -- -- PAP2 No $49.86 PAP2-NA Yes $49.76 Best regards, Miroslavmailto:[EMAIL PROTECTED] Sunday, September 19, 2004, 10:32:04 PM, you wrote: JM> Matthew Boehm wrote: >> I had 2 senior level management people at linksys corp confirm that this >> would not be possible until December. They both told me that they are >> currently in development of a 'non-locked' version but that it would not be >> in stores until December. JM> Those kind of people only know what they are told: JM> http://www.nufone.net/downloads/pap2.jpg JM> Jeremy McNamara JM> ___ JM> Asterisk-Users mailing list JM> [EMAIL PROTECTED] JM> http://lists.digium.com/mailman/listinfo/asterisk-users JM> To UNSUBSCRIBE or update options visit: JM>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax Status
Hi, How can I get the Fax Status of transmited document - complete, error, etc.? Regards, Miro. ___ 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] How to transfer a call before the called party to answer
Hi, I would like to make a simple application with address book which to dial the numbers and to transfer the call to the caller before the called party is answered. How can I do that? Regards, Miro. ___ 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] How to determine duration call when is used Attended Transfer
Hi,
I am trying to determine the actual call duration (billsec) when is
used Attended Transfer but this is very dificult because there is no
relation between channels. Are there any suggestions how can be solved
this?
I have an idea where in the CDR must be added new column where to
be stored the CDR UniqueID from another channel which is linked.
Or to have another database/table (res_features) where to store all
events like transfer, hold and conferences.
How to add some very useful patch for Attended Transfer? In the
standard source code of function builtin_atxfer(...) is writen:
newchan = ast_feature_request_and_dial(
transferer, "Local",
ast_best_codec(transferer->nativeformats),
dialstr,
15000,
&outstate, cid_num, cid_name);
We replace this line with this one:
newchan = ast_feature_request_and_dial(
transferer, "Local",
ast_best_codec(transferer->nativeformats),
dialstr,
atxfernoanswertimeout,
&outstate, cid_num, cid_name);
where "atxfernoanswertimeout" is static int which can be configured
in features.conf. the default value is 15000.
Follow the main useful patch where when the other party is busy the
channel return busy signal instead return to the caller. The function
is ast_feature_request_and_dial(...):
...
else if ((f->subclass == AST_CONTROL_BUSY) ||
(f->subclass == AST_CONTROL_CONGESTION))
{
state = f->subclass;
-> new lines:
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s is busy\n", chan->name);
ast_indicate(caller, AST_CONTROL_BUSY);
-> old lines:
/*
ast_frfree(f);
f = NULL;
break;
*/
Best Regards,
Miroslav Nachev
___
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[Asterisk-Users] No Media for Ringing Indication
Hi, When we use mISDN and Asterisk some times after Dial the PSTN doesn't send Ringing Media. The indication is coming and is displayed on Asterisk CLI but there is not sound. This is very unpleasant for the user which can't see the Asterisk CLI to know that the call is in Rining State. How can catch Ringing Event that is coming from ISDN channels and then to play rining media in case that PSTN is not sending such media. The same question for the other states like busy, etc. Any help or suggestions? Best Regards, Miroslav Nachev ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Different Ringing Tones depending of the call
Hi, How can be done similar functionality like Panasonic PBX and Phones where for incoming calls from PSTN the ringing type is different with this that is from some internal extension. For example we can configure Asterisk PBX to add some special character before or after the number in Caller ID, but the phone (GXP-2000) must understand and recognize this. Best Regards, Miroslav Nachev ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] How to program Phone "Configurable line indicators" for some PSTN lines
Hi, I would like to program some PSTN lines to be indicated on the phone (GXP-2000) when are busy and etc. How can we do that using "Speed Dial/Configurable line indicators" or/and "Line 1-4 Keys"? How Asterisk will inform the phone when the line is busy or ringing, etc.? Best Regards, Miroslav Nachev ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
