Re: [Asterisk-Users] How much to use Dialogic?
I think dealer cost is about $1800. plop sound of jaw hitting floor. Why not set up a T1 interconnect with an asterisk box? I'm sure the $500 T100P and the cost of a T1 port is smaller than the Dialogic card, the licenses for the dialogic driver, and this device you mention. Not to mention the wiring mess of that many analog ports. 1 wire is nicer. BTW, I think it has been covered here before that the D41 is a half duplex card and wouldn't be good for conferencing. Steven, I will look into it. Looks like it could be a winner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Opportunistic VoIP
This is an idea from FreeSWAN, which was implemented in the recently released version 1.0. Basically the idea is that FreeSWAN sites automatically encrypt traffic between them when possible, without having to set up the link ahead of time. How this works is: The sites publish some info in DNS. FreeSWAN gets some traffic destined for that site. - looks up the info in DNS - if the info is there: sets up an encrypted connection - if the info is missing: sets up a normal connection This is a feature which can be turned off. How does this apply to asterisk? Asterisk has a call destined for a PSTN number Looks up the number in a central location If it's there, then connect to the reported IAX/SIP/whatever connection over the internet if it's up/ping is good/hops is good/whatever. Otherwise connects through the PSTN. Points: saves money possible quality issues for VoIP over many internet hops this isn't as good as the FreeSWAN way as there is no logical mapping between PSTN and DNS -- therefore need a central location potential for abuse what would be a good spot for the central location? comments? -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How much to use Dialogic?
yeah, i got one, its called google :) site:lists.digium.com searchphrase put ^ in the google box, and voila - wasim On Sun, 8 Jun 2003, Matthew John Darnell wrote: Does anyone have an application that will parse the archives so you can search them? I was going to search the archives but it is too tedious to go month by month. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How much to use Dialogic?
Try Google with this as the query: [phrase to search for] site:lists.digium.com /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew John Darnell Sent: 09 June 2003 08:47 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How much to use Dialogic? BTW, I think it has been covered here before that the D41 is a half duplex card and wouldn't be good for conferencing. -- Steven Critchfield [EMAIL PROTECTED] Does anyone have an application that will parse the archives so you can search them? I was going to search the archives but it is too tedious to go month by month. -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Content and Virus scanned by Inflex and Mcafee _ Content and Virus scanned by Inflex and Mcafee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NewbieQ: SOHO setup with x100p
After scouring the list archive and not finding the answer I decided to post to the list. I'm sure the answer is glaringly obvious but please bear with me. Using Asterisk, I'm tasked with setting up a SOHO with 5 analogue (FXS?) lines and a number of soft-phones for internal extensions. I'm confused by the telephony hardware needed for this exercise - 1) I need the equivalent of two X400P (which is not advertised on Digiums's website), else I'll have to install 5 x100p's which is silly. Is there better hardware for this solution? Is this all the hardware I need? Secondly: 2) Do I need a separate fax line? / how can Asterisk route fax calls to an extension with a fax connected? TIA for the help, Tielman /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ _ Content and Virus scanned by Inflex and Mcafee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: NewbieQ: SOHO setup with x100p
Correction - My reference to analogue (FXS?) - Should be FXO /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ _ Content and Virus scanned by Inflex and Mcafee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How much to use Dialogic?
On Monday 09 June 2003 02:47, Matthew John Darnell wrote: BTW, I think it has been covered here before that the D41 is a half duplex card and wouldn't be good for conferencing. -- Steven Critchfield [EMAIL PROTECTED] Does anyone have an application that will parse the archives so you can search them? I was going to search the archives but it is too tedious to go month by month. http://www.asteriskpbx.org/index.php?menu=support At first I was searching archives manually until I found the Google search tool on the support page. I was using raw Google and was getting too much chaff. Enjoy. -- Mike M. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player
Hi all, Finally I found why MP3Player was not working for me. In the CVS of two weeks ago the path to mpg123 was hardcoded to /usr/bin/mpg123. I installed the latest pre0.59s because previous releases were not working for me because of my fast Pentium IV 1,7Ghz processor. This release but probably previous releases also installs in /usr/local/bin/mpg123. What are the prerequisites for MP3's? The sample-hold.mp3 plays fine but my own mp3's segfault. Regards Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Only noise in zap channel
Hi list, I have an E400P using only one span with 4 channels, using EM immediate signaling. /etc/zaptel.conf span=1,1,1,cas,hdb3,yellow em=1-4 loadzone = us defaultzone=us - /etc/asterisk/zapata.conf - [channels] group = 1 context=default signalling=em channel = 1-4 This configuration works ok, I can dial on Zap/g1. But, when the other side answer the call, I only hear a lot of noise instead of the voice. Could anybody help me? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual T400P, SMP, performance issues
Are you sure that you compiled zaptel for __SMP__ ? Edit your zaptel/Makefile. 0: 75283844 75241320 75286285 75247088IO-APIC-edge timer 1: 1 0 1 1IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 3: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 15: 1 0 0 1IO-APIC-edge ide1 16: 22134870 22120997 22135905 22122829 IO-APIC-level eth0 25: 4670 4548 4614 4518 IO-APIC-level tor2 All the four CPU's should have IRQ's like in the example above. Martin On Mon, 9 Jun 2003, Alex Zarubin wrote: Hi, We are trying to validate Asterisk as a media gateway PRI - SIP with two T400P (8 T1s) per box. The first experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was encouraging - on the load test with 3 T1s worth of calls we had on average 75% idle CPU. Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3 (Dell, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support). On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU 70% of the time. Just 3 T1s out of 8. On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0, CPU1 was at 95% idle. The process ksoftirqd_CPU0 was close to the top of the 'top', with /proc/interrupts showing tor2 related numbers growing very fast. We had 2 T1s plugged into the first T400P board, with nothing going into the second, but the number of interrupts for the both boards was growing at the same pace. Here are the interrupts (after the box reboot, so they are not that big as they were) - do they look OK? CPU0 CPU1 CPU2 CPU3 0: 122556 0 0 0IO-APIC-edge timer 1: 4 0 0 0IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 12: 20 0 0 0IO-APIC-edge PS/2 Mouse 14: 23 0 2 0IO-APIC-edge ide0 20: 516930 0 0 0 IO-APIC-level tor2 24: 516524 0 0 0 IO-APIC-level tor2 28: 10600 0 0 0 IO-APIC-level eth0 29: 4837 0 0 0 IO-APIC-level eth1 30: 24831 0 0 0 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 122430 122429 122429 122428 ERR: 0 MIS: 0 Not sure what went wrong. Any suggestions on how to work with 2 T400P in a box (without hurting performance) and how to get advantage of SMP for Asterisk would be appreciated. Any known Linux kernel related issues (2.4.20-13.7smp #1 SMP for BOX3 )? Thank you. Alex Zarubin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only noise in zap channel
On Tue, 10 Jun 2003 09:37:22 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Try in /etc/zaptel.conf to add this line: alaw=1-4 sine by default EM is used in US and the ulaw codec is being used. Martin thanks for your reply, but it still doesn't work Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only noise in zap channel
Did you do ztcfg after you added that line ? Martin On Tue, 10 Jun 2003, Eduardo Goncalves wrote: On Tue, 10 Jun 2003 09:37:22 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Try in /etc/zaptel.conf to add this line: alaw=1-4 sine by default EM is used in US and the ulaw codec is being used. Martin thanks for your reply, but it still doesn't work Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Linux traffic shaping to prioritise SIP/IAX traffic?
Hi, Has anyone done anything with the Linux advanced routing stuff to give SIP and IAX traffic priority? What I have in mind is a high-pri queue for voip traffic, all the rest in another queue that gives way to the VOIP stuff. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only noise in zap channel
On Tue, 10 Jun 2003 10:08:02 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Did you do ztcfg after you added that line ? Martin yeap :~ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritiseSIP/IAX traffic?
From the same machine running asterisk or from a linux router ? Linux kernel by default prioritizes traffic if the packet has some TOS bits set. so a standard linux router should do a basic traffic shaping. Of course, more complex rules could be made... but if the *outside* world don't do traffic shaping (or just ignore TOS bits), the priority you set works only in your net, outside is not guaranteed they will do what you want to ;) matteo Il mar, 2003-06-10 alle 17:19, Stephen Davies ha scritto: Hi, Has anyone done anything with the Linux advanced routing stuff to give SIP and IAX traffic priority? What I have in mind is a high-pri queue for voip traffic, all the rest in another queue that gives way to the VOIP stuff. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual T400P, SMP, performance issues
H, I to appear to have an odd mix of interrupts. It seems that the second CPU doesn't do much at all on my dual Xeon... CPU0 CPU1 0: 40652580 0IO-APIC-edge timer 1:926 0IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 6: 0 0 IO-APIC-level usb-ohci 8: 1 0IO-APIC-edge rtc 12:308 0IO-APIC-edge PS/2 Mouse 14: 2 0IO-APIC-edge ide0 20: 406481379 0 IO-APIC-level tor2 24: 0 0 IO-APIC-level tor2 28:4516659 0 IO-APIC-level eth0 30: 911870 0 IO-APIC-level aacraid NMI: 0 0 LOC: 40653025 40653047 ERR: 0 MIS: 0 I haven't enables the second card yet but will be enabling soon. I should probably recompile * and zaptel for SMP though I thought I had... Bill Martin Pycko wrote: Are you sure that you compiled zaptel for __SMP__ ? Edit your zaptel/Makefile. 0: 75283844 75241320 75286285 75247088IO-APIC-edge timer 1: 1 0 1 1IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 3: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 15: 1 0 0 1IO-APIC-edge ide1 16: 22134870 22120997 22135905 22122829 IO-APIC-level eth0 25: 4670 4548 4614 4518 IO-APIC-level tor2 All the four CPU's should have IRQ's like in the example above. Martin On Mon, 9 Jun 2003, Alex Zarubin wrote: Hi, We are trying to validate Asterisk as a media gateway PRI - SIP with two T400P (8 T1s) per box. The first experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was encouraging - on the load test with 3 T1s worth of calls we had on average 75% idle CPU. Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3 (Dell, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support). On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU 70% of the time. Just 3 T1s out of 8. On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0, CPU1 was at 95% idle. The process ksoftirqd_CPU0 was close to the top of the 'top', with /proc/interrupts showing tor2 related numbers growing very fast. We had 2 T1s plugged into the first T400P board, with nothing going into the second, but the number of interrupts for the both boards was growing at the same pace. Here are the interrupts (after the box reboot, so they are not that big as they were) - do they look OK? CPU0 CPU1 CPU2 CPU3 0: 122556 0 0 0IO-APIC-edge timer 1: 4 0 0 0IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 12: 20 0 0 0IO-APIC-edge PS/2 Mouse 14: 23 0 2 0IO-APIC-edge ide0 20: 516930 0 0 0 IO-APIC-level tor2 24: 516524 0 0 0 IO-APIC-level tor2 28: 10600 0 0 0 IO-APIC-level eth0 29: 4837 0 0 0 IO-APIC-level eth1 30: 24831 0 0 0 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 122430 122429 122429 122428 ERR: 0 MIS: 0 Not sure what went wrong. Any suggestions on how to work with 2 T400P in a box (without hurting performance) and how to get advantage of SMP for Asterisk would be appreciated. Any known Linux kernel related issues (2.4.20-13.7smp #1 SMP for BOX3 )? Thank you. Alex Zarubin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] s extension don't work on TDM40B
Hi all, i have read in the * whitepaper the following: s: The start extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the s extension. I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] paging system (long)
I'm back on the paging again - still can't get it working as I wish.
I have listed two attempts below where I run into basically the same problem
[pagejon] - I drop a file in the spool directory that starts that context,
I must use a device though that the outbound call is placed by. The phones
ring, and when picked up get connected to the device I specify - the
festival stuff is never played that I can tell.
problem - I don't want the call connected to a local device. I want the
called party to hear the festival text if they answer, and if not,
subsequent dials to other numbers should take place in order until one answers.
[pagejon]
exten = 1,1,Dial(${jonexts}|25)
exten = 1,2,festival,you didn't answer
exten = 1,102,festival,you answered
Second attempt - all goes well leaving the message. Then I want to hangup
on the caller in some way (which 899 does), in reality the caller stays
connected, and then the pagee's extensions start to ring. If I use a
hangup directly the script stops there.
This is what I want to happen :
1)authenticate (works)
2) leave message (works if they hit # and don't hangup)
3) then hangup on caller, but keep thread running
4) dial to first set of pagee's extensions, wait for answer
5) if unanswered dial another set of numbers and wait
6) when something is answered play a festival message
7) transfer the pagee to voicemail so they can listen to the message just
left for them
[paging]
; page jon
exten = 870,1,authenticate,/etc/asterisk/pageraccess
exten = 870,2,Voicemail,u5901 ; record their message in the paging box
exten = 870,3,transfer,899|1; hangup
exten = 870,4,goto,104 ;
exten = 870,103,wait,0 ; need this since hangup could fail as
well as vm
exten = 870,104,SetCallerid,Pager 870
exten = 870,105,Dial,${JonExts}|30 ; jon house phones
exten = 870,106,goto,107
exten = 870,206,Dial,zap/g2/${JonCell}|60
exten = 870,207,festival,you were paged.
exten = 870,208,Voicemailmain
help ?
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Re: [Asterisk-Users] Dual T400P, SMP, performance issues
My dual-proc Xeon boxes didn't share IRQs across CPUs until I installed the kernel-utils RPM and made sure the irqbalance service was running... Just a word to the wise! Jared Smith On Tue, 2003-06-10 at 09:52, [EMAIL PROTECTED] wrote: H, I to appear to have an odd mix of interrupts. It seems that the second CPU doesn't do much at all on my dual Xeon... CPU0 CPU1 0: 40652580 0IO-APIC-edge timer 1:926 0IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 6: 0 0 IO-APIC-level usb-ohci 8: 1 0IO-APIC-edge rtc 12:308 0IO-APIC-edge PS/2 Mouse 14: 2 0IO-APIC-edge ide0 20: 406481379 0 IO-APIC-level tor2 24: 0 0 IO-APIC-level tor2 28:4516659 0 IO-APIC-level eth0 30: 911870 0 IO-APIC-level aacraid NMI: 0 0 LOC: 40653025 40653047 ERR: 0 MIS: 0 I haven't enables the second card yet but will be enabling soon. I should probably recompile * and zaptel for SMP though I thought I had... Bill Martin Pycko wrote: Are you sure that you compiled zaptel for __SMP__ ? Edit your zaptel/Makefile. 0: 75283844 75241320 75286285 75247088IO-APIC-edge timer 1: 1 0 1 1IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 3: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 15: 1 0 0 1IO-APIC-edge ide1 16: 22134870 22120997 22135905 22122829 IO-APIC-level eth0 25: 4670 4548 4614 4518 IO-APIC-level tor2 All the four CPU's should have IRQ's like in the example above. Martin On Mon, 9 Jun 2003, Alex Zarubin wrote: Hi, We are trying to validate Asterisk as a media gateway PRI - SIP with two T400P (8 T1s) per box. The first experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was encouraging - on the load test with 3 T1s worth of calls we had on average 75% idle CPU. Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3 (Dell, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support). On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU 70% of the time. Just 3 T1s out of 8. On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0, CPU1 was at 95% idle. The process ksoftirqd_CPU0 was close to the top of the 'top', with /proc/interrupts showing tor2 related numbers growing very fast. We had 2 T1s plugged into the first T400P board, with nothing going into the second, but the number of interrupts for the both boards was growing at the same pace. Here are the interrupts (after the box reboot, so they are not that big as they were) - do they look OK? CPU0 CPU1 CPU2 CPU3 0: 122556 0 0 0IO-APIC-edge timer 1: 4 0 0 0IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 12: 20 0 0 0IO-APIC-edge PS/2 Mouse 14: 23 0 2 0IO-APIC-edge ide0 20: 516930 0 0 0 IO-APIC-level tor2 24: 516524 0 0 0 IO-APIC-level tor2 28: 10600 0 0 0 IO-APIC-level eth0 29: 4837 0 0 0 IO-APIC-level eth1 30: 24831 0 0 0 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 122430 122429 122429 122428 ERR: 0 MIS: 0 Not sure what went wrong. Any suggestions on how to work with 2 T400P in a box (without hurting performance) and how to get advantage of SMP for Asterisk would be appreciated. Any known Linux kernel related issues (2.4.20-13.7smp #1 SMP for BOX3 )? Thank you. Alex Zarubin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s extension don't work on TDM40B
To execute the s extension automatically when you pick up the phone, you need to put that channel in immediate mode. (I'd tell you how to do it, but I can't remember the syntax off the top of my head.) Jared Smith On Tue, 2003-06-10 at 09:57, Thomas Haeger wrote: Hi all, i have read in the * whitepaper the following: s: The start extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the s extension. I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Hger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s extension don't work on TDM40B
That's a local phone. if u what a local phone to exec 's' extensions, put immediate=yes into zapata.conf . Otherwise, you'll get a dialtone waiting for a exten input. Matteo. Il mar, 2003-06-10 alle 17:57, Thomas Haeger ha scritto: Hi all, i have read in the * whitepaper the following: s: The start extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the s extension. I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s extension don't work on TDM40B
the channel has to be in immediate mode to work as you describe with s otherwise nothing happens until you type some digits that match something in the context the phone starts in. At 05:57 PM 6/10/2003 +0200, you wrote: Hi all, i have read in the * whitepaper the following: s: The start extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the s extension. I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting local IP address for the RTP port
Title: RE: [Asterisk-Users] Setting local IP address for the RTP port Listening is not a problem. When we send RTP packets it's important to make sure we use the specific interface. For example, one interface is on internal subnet and the other one is on external. QoS etc. Do you think we'll have to change code for that? My guess it's a feature needed by many (and easy to implement). Thank you. Alex Zarubin -Original Message- From: Tilghman Lesher [mailto:[EMAIL PROTECTED]] Sent: Monday, June 09, 2003 8:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Setting local IP address for the RTP port On Monday 09 June 2003 20:09, Alex Zarubin wrote: If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? You can't. RTP will automatically listen on all interfaces. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] paging system (long)
Maybe it's me, but it looks like you need to change to AGI instead of
extension logic for parts of this below.
Write a AGI app that does the authenticate, and record of message. If
the user hangs up the phone it is fine since the AGI still is running,
and can then submit a qcall to then start the outbound process. When
recording, I'd make sure the file has a garunteed unique name. In the
qcall you can then reference this unique name to make sure to play this
newly created file.
For the qcall setup, make it a macro so you can pass parameters to it,
specifically the file you want to be played back.
On Tue, 2003-06-10 at 11:03, Jon Pounder wrote:
I'm back on the paging again - still can't get it working as I wish.
I have listed two attempts below where I run into basically the same problem
[pagejon] - I drop a file in the spool directory that starts that context,
I must use a device though that the outbound call is placed by. The phones
ring, and when picked up get connected to the device I specify - the
festival stuff is never played that I can tell.
problem - I don't want the call connected to a local device. I want the
called party to hear the festival text if they answer, and if not,
subsequent dials to other numbers should take place in order until one answers.
[pagejon]
exten = 1,1,Dial(${jonexts}|25)
exten = 1,2,festival,you didn't answer
exten = 1,102,festival,you answered
Second attempt - all goes well leaving the message. Then I want to hangup
on the caller in some way (which 899 does), in reality the caller stays
connected, and then the pagee's extensions start to ring. If I use a
hangup directly the script stops there.
This is what I want to happen :
1)authenticate (works)
2) leave message (works if they hit # and don't hangup)
3) then hangup on caller, but keep thread running
4) dial to first set of pagee's extensions, wait for answer
5) if unanswered dial another set of numbers and wait
6) when something is answered play a festival message
7) transfer the pagee to voicemail so they can listen to the message just
left for them
[paging]
; page jon
exten = 870,1,authenticate,/etc/asterisk/pageraccess
exten = 870,2,Voicemail,u5901; record their message in the paging
box
exten = 870,3,transfer,899|1 ; hangup
exten = 870,4,goto,104 ;
exten = 870,103,wait,0 ; need this since hangup could fail as
well as vm
exten = 870,104,SetCallerid,Pager 870
exten = 870,105,Dial,${JonExts}|30 ; jon house phones
exten = 870,106,goto,107
exten = 870,206,Dial,zap/g2/${JonCell}|60
exten = 870,207,festival,you were paged.
exten = 870,208,Voicemailmain
help ?
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[Asterisk-Users] chan_h323 + openh323 CVS = no go?
Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external chan_oh323. That setup seems to drop small audio snippets like VoiceMail's password prompt, though. So I'm trying to give chan_h323 another chance. However, I get: ast_h323.cpp: In function `int h323_set_capability(int, int)': ast_h323.cpp:780: `H323_G729ACapability' undeclared (first use this function) ast_h323.cpp:780: (Each undeclared identifier is reported only once ast_h323.cpp:780: for each function it appears in.) ast_h323.cpp:780: `g729aCap' undeclared (first use this function) ast_h323.cpp:781: parse error before `)' ast_h323.cpp: At top level: chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone
Ditto. I think vendor help/hints/suggestions/clarifications on this list are extremely helpful and valuable. I hate spam as much as anybody but we need to become evolved enough to know the difference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jon Pounder Sent: Monday, June 09, 2003 8:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone I think this sort of request is an appropriate question for the list, and I for one would not take offence in any way if a manufacturer answers to the list as a whole. Answering a direct question is fine, but unsolicited promotion is another matter. Even product announcements where they are relevant to the list members is fine by me, and I am sure most other people agree it is fine as well. This is why we have email subject headers - if you don't care about the price of a Budgetone 100, don't read the email - simple. At 06:44 PM 6/9/2003 -0700, you wrote: Hello Wade and Asterisk users, As we are committed to supporting Asterisk community, we will not be able to answer questions related to Grandstream product through Asterisk mailing list, this is to be fair and respectful to the Asterisk community as a whole. The previous email is to clear a pricing info regarding the product because a lot users start to use that price as reference price for the phone. Should you have any questions and issues regarding Grandstream product, please send your email to [EMAIL PROTECTED] or [EMAIL PROTECTED] Thank you for your attention and interest in Grandstream product. Best regards, Grandstram Customer Support Wow! A phone manufacturer is actually monitoring this list! Nice work Grandstream. Can you tell us which phones you currently have in stock, and pricing on all models? Can you also let us know if your 1-port FXS device is shipping? Pricing? Thanks in advance, -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Zhang Sent: Monday, June 09, 2003 4:14 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone This is to state that a recent message posted to asterisk mailing list by [EMAIL PROTECTED] regarding the pricing of our sample phones is NOT accurate. Grandstream Networks has NOT changed the list price for its products and samples. Our BudgeTone 100 series IP phones lists at $75 for model 101, NOT $60. Grandstream is committed to supporting the asterisk community and this message is posted for the sole purpose of correcting a misinformation regarding our product. Thanks for your attention to this matter. Grandstream Customer Support ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
Hello, I've been working with the chan_h323 myself, and I had several problems, but finally got it working. I had to do things in the following order: (1) build and installed asterisk as root (2)I built pwlib and openh323 into my home directory (not root) and built them there as me, I downloaded the tarballs and compiled them.(you could probably do the same thing as root) I did not yet have the os install of the libraries on the system, as this seemed to mess me up. (3) I built the chan_h323 object as myself. (4) I installed the chan_h323.so (make install) as root (5) finally, I installed the system libraries for pwlib and openh323 After all of that, it seemed to work. Good luck, Kelly. On Tue, 2003-06-10 at 12:40, Siggi Langauf wrote: Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external chan_oh323. That setup seems to drop small audio snippets like VoiceMail's password prompt, though. So I'm trying to give chan_h323 another chance. However, I get: ast_h323.cpp: In function `int h323_set_capability(int, int)': ast_h323.cpp:780: `H323_G729ACapability' undeclared (first use this function) ast_h323.cpp:780: (Each undeclared identifier is reported only once ast_h323.cpp:780: for each function it appears in.) ast_h323.cpp:780: `g729aCap' undeclared (first use this function) ast_h323.cpp:781: parse error before `)' ast_h323.cpp: At top level: chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Kelly McDonald [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
If you would have followed the build instructions laid out by the Open H.323 folks you wouldn't have had to go thru all of that. http://www.openh323.org/build.html (Notice they NEVER tell you to make install ANYTHING, there is a reason for that) Jeremy McNamara Kelly McDonald wrote: Hello, I've been working with the chan_h323 myself, and I had several problems, but finally got it working. I had to do things in the following order: (1) build and installed asterisk as root (2)I built pwlib and openh323 into my home directory (not root) and built them there as me, I downloaded the tarballs and compiled them.(you could probably do the same thing as root) I did not yet have the os install of the libraries on the system, as this seemed to mess me up. (3) I built the chan_h323 object as myself. (4) I installed the chan_h323.so (make install) as root (5) finally, I installed the system libraries for pwlib and openh323 After all of that, it seemed to work. Good luck, Kelly. On Tue, 2003-06-10 at 12:40, Siggi Langauf wrote: Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external chan_oh323. That setup seems to drop small audio snippets like VoiceMail's password prompt, though. So I'm trying to give chan_h323 another chance. However, I get: ast_h323.cpp: In function `int h323_set_capability(int, int)': ast_h323.cpp:780: `H323_G729ACapability' undeclared (first use this function) ast_h323.cpp:780: (Each undeclared identifier is reported only once ast_h323.cpp:780: for each function it appears in.) ast_h323.cpp:780: `g729aCap' undeclared (first use this function) ast_h323.cpp:781: parse error before `)' ast_h323.cpp: At top level: chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] aastra pt480 and adsi
On Fri, May 30, 2003 at 02:59:45PM +, Joe Antkowiak wrote: how do I specify a different one for the voicemail script? This is what the top of my asterisk.adsi looks like: DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00; Version of stuff FDN 0x85efd9da SECURITY 0x78921d49 And asterisk is currently occupying the SL slot, I don't have the option to specify which slot I want it to be loaded into... Luckily you won't need the SL slot for voicemail (it gets brought up when the VM system opens an ADSI session to it). You'll need to hack app_voicemail.c (or app_voicemail2.c) in apps/. Look for variables adapp and adsec. You'll want to change the value. For example, say you needed the security code set to hex 0xDEADBEEF on the first slot. You'll find the lines: static char *adapp = CoMa: static char *adsec = _AST; Change them to: static char *adapp = \x00\x00\x00\x0f: static char *adsec =\xDE\xAD\xBE\xEF; Then make clean; make; make install; restart gracefully *. Hope this helps. Jayson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slow Faxing
I currently have two fax machines on my system. Both of them seem to send and receive very slowly. My end users are complaining; saying it was faster before we moved to * (Straight Analog Lines) Any help would be great. PS: I already have the d option on the Dial line. Both fax machines are in their own context: [faxes] exten = _9NXX,1,StripMSD,1 exten = _NXX,2,Dial,Zap/g1/BYEXTENSION||d exten = _91NXXNXX,1,StripMSD,1 exten = _1NXXNXX,2,Dial,Zap/g1/BYEXTENSION||d And incoming faxes go through: exten = 1083,1,Dial,Zap/59|30|d John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone
I wish Mark would chime in on his thoughts. I also am not opposed to someone answering direct questions about a product they happen to sell even if it is in a public forum. We all gain from it in that anyone searching google may see both asterisk and Budgetone. Both may get a plus by this. On Tue, 2003-06-10 at 11:48, David Carr wrote: Ditto. I think vendor help/hints/suggestions/clarifications on this list are extremely helpful and valuable. I hate spam as much as anybody but we need to become evolved enough to know the difference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jon Pounder Sent: Monday, June 09, 2003 8:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone I think this sort of request is an appropriate question for the list, and I for one would not take offence in any way if a manufacturer answers to the list as a whole. Answering a direct question is fine, but unsolicited promotion is another matter. Even product announcements where they are relevant to the list members is fine by me, and I am sure most other people agree it is fine as well. This is why we have email subject headers - if you don't care about the price of a Budgetone 100, don't read the email - simple. At 06:44 PM 6/9/2003 -0700, you wrote: Hello Wade and Asterisk users, As we are committed to supporting Asterisk community, we will not be able to answer questions related to Grandstream product through Asterisk mailing list, this is to be fair and respectful to the Asterisk community as a whole. The previous email is to clear a pricing info regarding the product because a lot users start to use that price as reference price for the phone. Should you have any questions and issues regarding Grandstream product, please send your email to [EMAIL PROTECTED] or [EMAIL PROTECTED] Thank you for your attention and interest in Grandstream product. Best regards, Grandstram Customer Support Wow! A phone manufacturer is actually monitoring this list! Nice work Grandstream. Can you tell us which phones you currently have in stock, and pricing on all models? Can you also let us know if your 1-port FXS device is shipping? Pricing? Thanks in advance, -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Zhang Sent: Monday, June 09, 2003 4:14 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone This is to state that a recent message posted to asterisk mailing list by [EMAIL PROTECTED] regarding the pricing of our sample phones is NOT accurate. Grandstream Networks has NOT changed the list price for its products and samples. Our BudgeTone 100 series IP phones lists at $75 for model 101, NOT $60. Grandstream is committed to supporting the asterisk community and this message is posted for the sole purpose of correcting a misinformation regarding our product. Thanks for your attention to this matter. Grandstream Customer Support ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritiseSIP/IAX traffic?
On 10 Jun 2003, Emanuele Pucciarelli wrote: Il mar, 2003-06-10 alle 17:19, Stephen Davies ha scritto: Has anyone done anything with the Linux advanced routing stuff to give SIP and IAX traffic priority? What I have in mind is a high-pri queue for voip traffic, all the rest in another queue that gives way to the VOIP stuff. When the tos option is set correctly (to nodelay), the default queueing in recent kernels already does that, because the pfifo_fast queue is used (if I recall correctly). But there is never any queue on my Linux box. It all storms out of the ethernet interface and gets queued up in my cable modem which doesn't know anything about tos settings. I did find the wondershaper script on www.lartc.org which looks like it will do what I need. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NewbieQ: SOHO setup with x100p
On Tue, Jun 10, 2003 at 04:30:50PM +0600, [EMAIL PROTECTED] wrote: c) get a T400P + channel bank (expensive, but it does give you 24 ports) Small typo. Just don't want newbies to be confused. The T400P mentioned above should probably be a T100P which supports one T1 line rather than four. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only noise in zap channel
On Tue, Jun 10, 2003 at 11:21:37AM -0300, Eduardo Goncalves wrote: Hi list, This configuration works ok, I can dial on Zap/g1. But, when the other side answer the call, I only hear a lot of noise instead of the voice. Could anybody help me? Is the noise loud and sounds like you have picked up the phone in the middle of a modem call? If so, I had a similar problem with my TDM20 while it was sharing an IRQ with the unused AC97 chip. I shuffled the cards around to different PCI slots and it now works. In my problem, in addition to the noise, asterisk was not responding to DTMF tones pressed on the analog handset. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Detection and noise in TDM10B
On Tue, Jun 10, 2003 at 07:08:45PM +0200, Sergio Serrano Revuelto wrote: HI all, we get a TDM10B to probe it. I find two problems: -First, I hear a lot of noise in communication. I have tried do dd if=/dev/zero of=/dev/null but it isn't work. -Second, When I pickup phone connected to TDM10B I hear a strange dial tone and * doesn't detect DTMF. That is the same problem I had last week. The system had worked before then started with that nonsense. I swapped PCI slots for the TDM and X100P cards. That made it so the TDM card had its own IRQ and the X100P was sharing with the AC97 sound device on the motherboard, which is unused. That fixed my symptoms. These cards definately have problems with IRQ sharing, which isn't supposed to happen on a PCI bus. My laptop manages to run flawlessly with every PCI device, except graphics and IDE controller, on the same IRQ. That includes 802.11b PCMCIA card, compact flash reader, sound device, Intel EEPro, Lucent Winmodem, and some other thing. Back in the OS/2 days I was even able to share IRQs with multiple ISA based serial ports using the SIO.SYS driver. Never dropped a bit. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PDA's over SIP channels on Asterisk
Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Asterisks with old Rhetorix 4108s?
Does anyone know of drivers/software that will allow me to use the old Rhetorix 4108 T/R boards with Asterisk? Thanks! Chip __ Do you Yahoo!? Yahoo! Calendar - Free online calendar with sync to Outlook(TM). http://calendar.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PDA's over SIP channels on Asterisk
On Tue, 2003-06-10 at 13:49, flickds wrote: Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Why not look into making a small wrapper around libiax for your Zaurus and speak IAX? This setup should work better when you are roaming and don't know when you will be behind a firewall,nat or not. You won't be able to use gnophone due to not running X or gnome on the little thing. BTW, do you know if the little headphone jack on there works like a cell phone where you can have a mic and a ear bud? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Asterisks with old Rhetorix 4108s?
On Tue, 2003-06-10 at 13:59, Chip G wrote: Does anyone know of drivers/software that will allow me to use the old Rhetorix 4108 T/R boards with Asterisk? Not supported yet. First you need to see if you can get them to work in linux. Then if you are still in need, there is sample drivers around to base it on if you feel the need. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Adding an app (Steven Critchfield)
Thanks for the help. I was able to get my application to load with Asterisk, now I just need to get it to work. After reading your comment, I don't know that I fully understand what's going on as far as the channels and extensions. Are you saying that the MWI is tied to the channel? If that is the case, then for my SIP phones, would the channel be (SIP/)? If it should be, then should extensions.conf look like: exten = 811NXX,1,MessageWaitOn(SIP/EXTEN:6) Again, thanks for the help. Jesse Date: Mon, 09 Jun 2003 22:37:01 -0500 From: Steven Critchfield [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Adding an app To: [EMAIL PROTECTED] Organization: Reply-To: [EMAIL PROTECTED] On Mon, 2003-06-09 at 13:01, [EMAIL PROTECTED] wrote: I am in the process of testing out the Cisco ATA 186 to provide analog phone service via VoIP for some of our remote users. I have that working fine and well, but am struggling with another aspect. We already have a large centralized voicemail system, which I would like to use for these users. I can get the call to roll to new the centralized voicemail no problem, but I'd like to provide message waiting for them as well. I've seen where Asterisk can provide MWI via stutter dial tone for its own internal voicemail. After looking at the source code, it appears that there is a flag set that determines whether or not * needs to play stutter dial tone when the user lifts the handset. I can get my voicemail system to pass the MWI to Asterisk in this form: 810NXX == turns off MWI 811NXX == turns on MWI I would like to set up extensions.conf like this: exten = 810NXX,1,MessageWaitOff(EXTEN:6) exten = 811NXX,1,MessageWaitOn(EXTEN:6) I would like the MessageWaitOff and MessageWaitOn apps to appropriately set the flag for the extension that is passed to it. This is where my problems start. I have spent some time looking through the source code, but I haven't determined what all I need to touch to add an application and have Asterisk recognize it. I have created the apps, compiled them and created the shared object file. What other steps do I need to take? Will what I'm proposing here even work? You need to get your shared object into the asterisk lib directory, and worst case, add a load command to the modules.conf file. Comment as to how you plan on implementing the above function. The exten does not get the MWI, it is a channel. Channels and extensions are different. Extensions can point to many channels, but a channel only points to one phone interface. You will need to make a lookup to go from extension to channel, then you can go toggle the MWI. Otherwise your mailbox is going to have to be your extension, possibly in that long format, and you will have to traverse every channel to find which ones to set MWI. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directory by names in VMAIL2
If there is a way possible, would someone tell me how I can setup a dial by name feature under vmail2? Thanks, Kim Callis
[Asterisk-Users] WILDCARD TDM400P or four Wildcard X100P
Can I use a WILDCARD TDM400P to connect to four Telco circuits aka FXO? Or will I need four Wildcard X100P? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Tue, 2003-06-10 at 16:22, Johnny Witt wrote: Hi Asterisk-Users Ive been reading about the Asterisk project (all that I could get my hands on J ). It sound to good to be true. But Ive got some questions which I havent found a answer to anywhere : 1) Can I use Asterisk as a Call Manager using MGCP protocol or H.323 towards a Cisco AS54xx og Cisco 3660 ? Yes, there is 1 MGCP driver, and 2 h323 drivers to choose from 2) Can I use Asterisk as a CMS with CableModems using PacketCable MGCP/CNS on the POTS ? Not sure what you have going on here. As long as the cable modem has sustainable bandwidth to make the VoIP happy, it will be irrelavent to the question. Asterisk speaks MGCP and should work with the other end. 3) Can I use VoIP system based upon Asterisk and interconnect to another system based upon Cisco AVVID solutions using either MGCP or H.323 ? The far end should not matter as long as you are speaking the same language, ie. MGCP or h323. 4) Is there any measurement regarding the performance of Asterisk as a CMS ( How many simultaneously calls ? , Extensions and so on) This depends heavily on the codecs you use, and the system you put it on. We are talking about commodity x86 hardware thats only real limitation is that it must be 386 or higher for VoIP. The number of extensions is only a matter of RAM and swap space since extensions just take up memory in the dialplan. It is channels that will start to really eat resources. Well Im probably going to have lots of questions But this is enough for the moment at least until I see if Ive posted correctly. I hope someone out there can help me with these questions. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritiseSIP/IAX traffic?
On 10 Jun 2003, Emanuele Pucciarelli wrote: That is not entirely correct. There is an output queue, and pfifo_fast is the default (see the LARTC Howto, 9.2.1.1). But you are right when you say you need something to slow down the data;the simplest choice should be addingthe Token Bucket Filter (9.2.2.2). OK - thanks for the pointers! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PDA's over SIP channels on Asterisk
Use tkcPhone, it's a SIP agent for the Zaurus. http://www.thekompany.com/embedded/tkcphone/ flickds wrote: Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WILDCARD TDM400P or four Wildcard X100P
Please... stay away from the FXS to FXO converters The cheap white box sold direct at $18 bucks and on the net for $70-$300 bucks work.. but do you want to rely on a timer to disconnect the call after hangup? -Greg - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 10, 2003 6:18 PM Subject: Re: [Asterisk-Users] WILDCARD TDM400P or four Wildcard X100P Well normally the telco phone line is an FXS line so you need an FXO port to connect to it (e.g. X100P). However if your line is FXO then you need FXS ports and TDM400P should work Martin On Tue, 10 Jun 2003, James Sizemore wrote: Can I use a WILDCARD TDM400P to connect to four Telco circuits aka FXO? Or will I need four Wildcard X100P? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PDA's over SIP channels on Asterisk
have a look on th zaurus site, there is a pay for one On Tue, 10 Jun 2003 13:49:20 -0500, flickds wrote: Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritise SIP/IAX traffic?
On Tue, Jun 10, 2003 at 09:58:14PM +0200, Emanuele Pucciarelli wrote: Il mar, 2003-06-10 alle 20:07, Stephen Davies ha scritto: When the tos option is set correctly (to nodelay), the default queueing in recent kernels already does that, because the pfifo_fast queue is used (if I recall correctly). But there is never any queue on my Linux box. It all storms out of the ethernet interface and gets queued up in my cable modem which doesn't know anything about tos settings. That is not entirely correct. There is an output queue, and pfifo_fast is the default (see the LARTC Howto, 9.2.1.1). But you are right when you say you need something to slow down the data;the simplest choice should be addingthe Token Bucket Filter (9.2.2.2). But if the wondershaper already does it all, then it's probably better to go with it... :) You should read further into lartc.org, the linux traffic shaping capabilities are really wide and you can find lots of ways of doing what you want. In your case, I guess the logical choice would be to use HTB, with two classes, let's say if your cablemodem is 512kbps, you can save 112kbps for voice and signalling (yes it's extreme but it's an example), and 400kbps for the data. Also, you can put the former with top priority to minimize latency. Under those you can use sfq to make everything more fair, that helps a lot when saturating. This is a very short script, I'm sure there is something like this on lartc or htb's website. Something like this (completely untested, from memory so don't trust me): # delete the existing qdisc tc qdisc del dev eth1 root 21 /dev/null # init the htb qdisc tc qdisc add dev eth1 root handle 1: htb tc class add dev eth1 parent 1: classid 1:5 htb rate 512kbit tc class add dev eth1 parent 1:5 classid 1:10 htb rate 112kbit prio 0 tc class add dev eth1 parent 1:5 classid 1:20 htb rate 400kbit prio 2 # sfq for all of them tc qdisc add dev eth1 parent 1:5 handle 500: sfq perturb 10 tc qdisc add dev eth1 parent 1:10 handle 100: sfq perturb 10 tc qdisc add dev eth1 parent 1:20 handle 200: sfq perturb 10 And now you need to set the filters up, which can be based on tc or using iptables' MARK target. For instance: # everything marked 10 in iptables go to 1:10 (the 112kbit) tc filter add dev eth1 protocol ip parent 1:0 prio 1 handle 10 fw \ flowid 1:10 # everything marked 20 in iptables go to 1:20 (the 400kbit) tc filter add dev eth1 protocol ip parent 1:0 prio 1 handle 20 fw \ flowid 1:20 And then mark in iptables (i don't remember sip's port very well, and you should also add RTP stuff too): # sip, mark 10 iptables -t mangle -A PREROUTING -i eth2 -p tcp --destination-port 5060 \ -j MARK --set-mark 10 iptables -t mangle -A PREROUTING -i eth2 -p udp --destination-port 5060 \ -j MARK --set-mark 10 # default, mark 20 iptables -t mangle -A PREROUTING -i eth2 -d 0.0.0.0/0 \ -j MARK --set-mark 20 In this case you could have used tc's native filters which are much faster than iptables, but also harder to setup, so if this is your first approach to this stuff I wouldn't recommend them (and don't worry, the speed difference is _not_ noticeable for that bandwidth). I hope this helps, however this is all untested, (ie. just wrote it) so please look into the docs to find out more. Thanks, Alberto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual T400P, SMP, performance issues
On Tue, Jun 10, 2003 at 10:14:09AM -0600, Jared Smith wrote: My dual-proc Xeon boxes didn't share IRQs across CPUs until I installed the kernel-utils RPM and made sure the irqbalance service was running... Just a word to the wise! Yes, you need irqbalance and a kinda modern kernel in order to be able to balance IRQs across different CPUs. This has nothing to do with *, because it's not up to it which CPU can handle each interrupt. You may also want to try out 2.5 and see how it behaves, it has improved a lot on those areas. BTW, some NAPI-alike stuff would help here, has anyone thought/tried out anything like it? Thanks, Alberto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
On Tue, 10 Jun 2003, Jeremy McNamara wrote: trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) Open H.323 1.11.7 works perfectly in all of my installations. I babysit 15 different chan_h323 based systems. One way audio usually means you have codec problems or are trying to traverse NAT. Nope, things were more weird in this case: This installation has a few hundred Cisco 79xx phones running in Skinny mode babysitted by a Cisco CallManager (actually two CCMs, if you count the fallback machine). Asterisk is used as a voicemail box attached to the CCM as an H.323 gateway. So what happens is: the CCM builds all connections to asterisk but negotiates via H.245 that the actual voice streams should be sent directly to the phone. For some reason, OpenH323 1.11.7 would ignore this and just send packets to the CCM instead, which would just drop them. Hence silence on the phone. The codec is G.711, so no problems here. and everything's running in one big private class B net without any outside connection, so no NAT. [...] It looks like the Open H.323 folks either forgot to include the G.729 Capability stubb or were forced to pull it by their legal department. I will look into this. It's still there, but skipped during compilation. Why, I can't tell. Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? We have never tested the latest cvs -HEAD of Open H.323 and PWLib, as there have been major changes, so we are giving those guys some time to make sure everything is stable before we dive in to new, untested code. I see. So I'll have to stick to chan_oh323 for now. Thanks, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go? (fwd)
-- Forwarded message -- Date: Wed, 11 Jun 2003 01:10:16 +0200 (CEST) From: Siggi Langauf [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go? On Tue, 10 Jun 2003, Jeremy McNamara wrote: If you would have followed the build instructions laid out by the Open H.323 folks you wouldn't have had to go thru all of that. http://www.openh323.org/build.html (Notice they NEVER tell you to make install ANYTHING, there is a reason for that) Granted. But is there also a reason for having such a build system? IMHO, OpenH323 is just broken in that respect, but I guess this is the wrong list to discuss such stuff. Luckily, Asterisk does a much better job, compiling, installing and building just right out of the box. Big thanks to everybody who made that possible! (and maybe sometiime H.323 support will be a as easy to build, or even not necessary any more...) Cheers, Siggi Kelly McDonald wrote: Hello, I've been working with the chan_h323 myself, and I had several problems, but finally got it working. I had to do things in the following order: (1) build and installed asterisk as root (2)I built pwlib and openh323 into my home directory (not root) and built them there as me, I downloaded the tarballs and compiled them.(you could probably do the same thing as root) I did not yet have the os install of the libraries on the system, as this seemed to mess me up. (3) I built the chan_h323 object as myself. (4) I installed the chan_h323.so (make install) as root (5) finally, I installed the system libraries for pwlib and openh323 After all of that, it seemed to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual T400P, SMP, performance issues
On Tue, 10 Jun 2003 [EMAIL PROTECTED] wrote: H, I to appear to have an odd mix of interrupts. It seems that the second CPU doesn't do much at all on my dual Xeon... You might have 'noapic' on your kernel command line... or your bios isnt configured for MP 1.4 ... -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Screenshots of admin GUI
Aloha, Does anyone have any screen shots of the Asterisk admin GUI? I couldn't find any links in the archives or the asterisk web site. -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a web admin
Hi: Any of you know a good web admin for asterisk??? Thanks Alvaro Parres - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Opportunistic VoIP
You should investigate TRIP (RFC 3129): http://www.zvon.org/tmRFC/RFC3219/Output/ Find BSD-licensed proof-of-concept code at http://www.vovida.org/downloads/trip/trip-1.0.0.tar.gz If someone could incorporate this into Asterisk and extend the functionality, that would be pretty nice. The basic ENUM support in Asterisk already can handle specific number paths, but I think TRIP or something like TRIP would be best for handling situations where larger groups of numbers need to be advertised into a routing table behind a particular Asterisk server. Think BGP for phone numbers. JT This is an idea from FreeSWAN, which was implemented in the recently released version 1.0. Basically the idea is that FreeSWAN sites automatically encrypt traffic between them when possible, without having to set up the link ahead of time. How this works is: The sites publish some info in DNS. FreeSWAN gets some traffic destined for that site. - looks up the info in DNS - if the info is there: sets up an encrypted connection - if the info is missing: sets up a normal connection This is a feature which can be turned off. How does this apply to asterisk? Asterisk has a call destined for a PSTN number Looks up the number in a central location If it's there, then connect to the reported IAX/SIP/whatever connection over the internet if it's up/ping is good/hops is good/whatever. Otherwise connects through the PSTN. Points: saves money possible quality issues for VoIP over many internet hops this isn't as good as the FreeSWAN way as there is no logical mapping between PSTN and DNS -- therefore need a central location potential for abuse what would be a good spot for the central location? comments? -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PDA's over SIP channels on Asterisk
quote who=Scott Lambert On Tue, Jun 10, 2003 at 02:10:56PM -0500, Steven Critchfield wrote: On Tue, 2003-06-10 at 13:49, flickds wrote: Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, http://www.thekompany.com/embedded/tkcphone/ My SL-5500 manual says, I/O device stereo headphone jack (monaural audio input) The manual has no description of the electrical or physical dimensions of the jack. It is an 1/8 jack. The plug on my cell phone seems to be closer to 3/32. No high precision, nor metric, measuring tools on hand. The left headphone is also wired to be a microphone. You can actually yell into the left headphone when recording. (Though, it would work better if something a bit more sensitive were connected, like a microphone.) You can, actually, just get a 1/8 - 3/32 stereo adapter from RadioShack. Then connect a handsfree headphone to it. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PDA's over SIP channels on Asterisk
On Tue, Jun 10, 2003 at 09:59:03PM -0700, Robert Hajime Lanning wrote: quote who=Scott Lambert On Tue, Jun 10, 2003 at 02:10:56PM -0500, Steven Critchfield wrote: On Tue, 2003-06-10 at 13:49, flickds wrote: Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, http://www.thekompany.com/embedded/tkcphone/ My SL-5500 manual says, I/O device stereo headphone jack (monaural audio input) The manual has no description of the electrical or physical dimensions of the jack. It is an 1/8 jack. The plug on my cell phone seems to be closer to 3/32. No high precision, nor metric, measuring tools on hand. The left headphone is also wired to be a microphone. You can actually yell into the left headphone when recording. (Though, it would work better if something a bit more sensitive were connected, like a microphone.) You can, actually, just get a 1/8 - 3/32 stereo adapter from RadioShack. Then connect a handsfree headphone to it. Or you can get a 3.5mm stereo male to 3.5mm left + right female adapter. This will let you plug a microphone into one and headphones into the other. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Hardware - Channelbank vs SIP etc
We're doing a new * installation at a remote office soon, and I was just curious what people's opinions were on hardware these days .. I've had decent luck with T100Ps and Adtran, but I know times change .. I'm looking to do roughly 15 handsets and 15 pstn, with some room to grow. I had planned on two T100Ps and two adtran 750s, one for handsets, one for pstn. I'm thinking of going SIP on the other side, though. I've been looking at the Grandstream budgetone phones, as well as their handytone. Anyone have anything good or bad to say on these? Cisco is out of that office's budget, I'm afraid. We're replacing a cheapo key system there, so it's all about the benjamins.. :\ I was also looking at: http://clipcomm.co.kr/eng/e_product/e_product_voip_analoggateway_4.html (rumored to be D-Link's OEM?) and http://www.yoda.com.tw/SOLUTIONS/vg422r.htm Any thoughts on these? Has anyone had good luck with other low-cost channels banks? (noo, not Zhone.. :) Any tips are appreciated, you can catch me here or on irc as always .. -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
