Re: [Asterisk-Users] AEL2 and CID
Yeah, thanks, that was the way I was leaning to. Just was wanting to
know if it was a syntax I was getting wrong, or if there is no other way
of doing this.
Julian.
Mojo with Horan Company, LLC wrote:
I guess you could do this, but it would be a little cumbersome:
context incoming {
s = {
if (${CALLERID(num)} = 8005551212)
{
NoOp(Dir. Asst. calling);
}
else if (${CALLERID(num)} = 800444)
{
NoOp(ANI calling);
}
}
}
Douglas Garstang wrote:
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten = 111/666,1,PlayBack(demo-congrats)
exten = 111/666,2,Hangup()
exten = 111,1,PlayBack(demo-moreinfo)
exten = 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats,
everyone else gets demo-moreinfo.
In AEL:
111 = {
Playback(demo-moreinfo);
Hangup();
};
111/666 = {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
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[Asterisk-Users] configuration
hello I have 2 services with 2different numbers. the first is 88 and the second is 99. if a user call 88 I want to execute the script1 and if he call 99 I execute the script2. How can I do my configs files? big Thanks issam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Openion on Sipura SPA-2100
Hi,As you said, May I know the correct Digium or Sipura product model (Sipura-3102 or Digium?), which is suitable to my requirements?Thank you.Regards,ChandramouliMartin Joseph [EMAIL PROTECTED] wrote: On May 31, 2006, at 10:32 PM, Crazy Boy wrote: Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. The SPA-2100 is an FXS, which allows the connection of phone handsets to your asterisk server.If you want to hook up phone lines you need an FXO.Marty___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] configuration
Create the 2 extensions in /etc/asterisk/extension.conf exten = 8,1,Answer() . Script 1 . exten = 9,1,Answer() . Script 2 . Make sure that the channel where the calls come in route the call to the context where you defined the scripts. Hope this helps, Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: donderdag 1 juni 2006 9:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] configuration hello I have 2 services with 2different numbers. the first is 88 and the second is 99. if a user call 88 I want to execute the script1 and if he call 99 I execute the script2. How can I do my configs files? big Thanks issam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple processes
I don't have any ODBC CDR stuff. I unloaded the ODBC Asterisk modules and the problem occurred again about an hour later. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodney G. McDuff Sent: 01 June 2006 01:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple processes Temporarily turn off your ODBC CDR stuff and see if the problem is still there. Lee Archer wrote: Can someone shed any light on the following. I have 2 identical systems, 1 of which seems to spawn multiple processes which have to be killed manually. It recently kicked up 2 so I ran gdb on them and this is the thread output. I current use FreePBX with these systems. 1st extra process (gdb) info thread 1 Thread 1095261104 (LWP 14213) 0xe410 in __kernel_vsyscall () (gdb) thread apply all bt Thread 1 (Thread 1095261104 (LWP 14213)): #0 0xe410 in __kernel_vsyscall () #1 0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0 #2 0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0 #3 0x0001 in ?? () #4 0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so #5 0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so #6 0x0002 in ?? () #7 0x in ?? () #8 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592 #9 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240 #10 0x40e13978 in reload () at cdr_odbc.c:465 #11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257 #12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754 #13 signal handler called #14 0xe410 in __kernel_vsyscall () #15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6 #16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803 #17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data=value optimized out, enhanced=0, dead=0) at res_agi.c:300 #18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable, exten=0x82784f4 s, priority=4, label=0x0, callerid=0x8159f38 0163861, action=1) at pbx.c:553 #19 0x40c44851 in macro_exec (chan=0x82782b0, data=0x4147c768) at app_macro.c:210 #20 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable, exten=0x82784f4 s, priority=7, label=0x0, callerid=0x8159f38 0163861, action=1) at pbx.c:553 #21 0x40c44851 in macro_exec (chan=0x82782b0, data=0x41482fd8) at app_macro.c:210 #22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=value optimized out, context=0x8278400 macro-record-enable, exten=0x82784f4 s, priority=1, label=0x0, callerid=0x8159f38 0163861, action=1) at pbx.c:553 #23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227 #24 0x0808f6cc in pbx_thread (data=0x0) at pbx.c:2514 #25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0 #26 0x401c737e in clone () from /lib/tls/libc.so.6 #27 0x41485bb0 in ?? () #0 0xe410 in __kernel_vsyscall () 2nd extra process (gdb) info thread 1 Thread 1096059824 (LWP 14214) 0xe410 in ?? () (gdb) thread apply all bt Thread 1 (Thread 1096059824 (LWP 14214)): #0 0xe410 in ?? () #1 0x41533594 in ?? () #2 0x0002 in ?? () #3 0x in ?? () #4 0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0 #5 0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0 #6 0x0001 in ?? () #7 0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so #8 0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so #9 0x0002 in ?? () #10 0x in ?? () #11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c ODBC) at lock.h:592 #12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240 #13 0x40e13978 in reload () at cdr_odbc.c:465 #14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257 #15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754 #16 signal handler called #17 0xe410 in ?? () #0 0xe410 in ?? () Regards Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group|Ex luce ad tenebras Information Technology Services | The University of Queensland | EMAIL: [EMAIL PROTECTED] | TELEPHONE: +61 7 3365 8220 | ___ --Bandwidth and Colocation
RE: [Asterisk-Users] Converting .wav to .WAV
yes use sox. that's what am using From: Mimmus [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: [Asterisk-Users] Converting .wav to .WAV Date: Wed, 31 May 2006 19:33:11 +0200 Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with misdn and BN8S0
Hi i am experiencing some problem with asterisk and misdn i've patched and recompiled the 2.6.15.5 kernel on the server i use a BN8S0 card with alla channels in TE mode. i can load hfcmulti and mISDN_dsp i load this with: /sbin/modprobe hfcmulti layermask=0xf protocol=0x22 type=0x08 and then /sbin/modprobe mISDN_dsp dmesg returns me the following: Modular ISDN Stack core $Revision: 1.34 $ mISDNd: kernel daemon started mISDNd: test event done mISDN: HFC-multi driver Rev. 1.40 0 devices registered mISDN_dsp: Audio DSP Rev. 1.17 (debug=0x0) EchoCancellor MG2 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies. i've edited misdn.conf with basic standard configuration and when i try to start asterisk with: asterisk -c i obtain: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) mISDN_close: fid(18) isize(131072) inbuf(0x816bfe0) irp(0x816bfe0) iend(0x816bfe0) Jun 1 02:35:35 ERROR[2736]: chan_misdn.c:3715 load_module: Unable to initialize mISDN Jun 1 02:35:35 WARNING[2736]: loader.c:414 __load_resource: chan_misdn.so: load_module failed, returning -1 Jun 1 02:35:35 WARNING[2736]: loader.c:554 load_modules: Loading module chan_misdn.so failed! can you help me to guess the problem? thanks nik ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astdb entry in sip.conf
Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/731 : 192.168.6.156:5060:3600:731:sip:[EMAIL PROTECTED]:5060 /dundi/secret : RpC4PXLxtslY5OZOza7OSws60yzaA /dundi/secretexpiry : 1149149129 Do I need to have anything else configured or should I report this as a bug ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DELL PowerEdge 2850 and TE4110P and TE110P
On Wed, 31 May 2006, Steven wrote: What were the kernel parameters that you changed? (what OS, by the way?) I am running CentOS 4.3, but have not changed any kernel settings yet. Nothing exciting, just adding noapic did improve a lot on the hits: title CentOS (2.6.9-34.ELsmp) root (hd0,0) kernel /vmlinuz-2.6.9-34.ELsmp ro root=/dev/VolGroup00/LogVol00 acpi=off initrd /initrd-2.6.9-34.ELsmp.img This is also on CentOS 4.3 :) Cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I guess my server capacity is ok
Bruce,the sys stats shown above is at no call and I run the ps -auwxx, i couldn't see any process taking up the resources.for example, the maximum of cpu usage was asterisk -g -c and mysql and they are together
0.5% and 0.4% respectively.I have other servers running on Dell PowerEdge 2850 and they are okay.Thanks for your responsegoksieOn 5/31/06,
Bruce Reeves [EMAIL PROTECTED] wrote:
I have seen this same type of stats on the list a couple of times now and wanted to share a thought. I am not a linux expert by any means, but in the windows world a 100% CPU idle means the percentage of CPU that is currently ideling. Basicly if I take all the process in the task manager and add up the percentages it should equal 100 and System Idle general takes up the left overs. Reaing the CPU stats above it looks like
86.2% of the CPU is currently unused, A good thin in my book. To test my theroy I loaded up an * server and with no calls there was 100% id the more calls the lower the number got. Again just thoughts from a non-linux guru.
On 5/31/06, Goke Aruna
[EMAIL PROTECTED] wrote:
Thank you...Steve, I am using it to pass a call from sip gateway to my asterisk server and sending the call to the provider thru chan_ss7.and I have my extension.conf without any timeout and other options.
[general]static=yeswriteprotect=noautofallthrough=yesclearglobalvars=nopriorityjumping=no[globals]TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
PREFIX=444[default];;THE GSM NETWORKexten = _234XX5XXX,1,Set(CALLERID(number)=1${PREFIX}${EXTEN:-5:4})exten = _234XX5XXX,2,Dial(ss7/A/0${EXTEN:-10})exten = _234XX2XXX,1,Set(CALLERID(number)=1${PREFIX}${EXTEN:-5:4})
exten = _234XX2XXX,2,Dial(ss7/B/0${EXTEN:-10})exten = _2341XXX,1,Set(CALLERID(number)=${PREFIX}${EXTEN:-5:4})exten = _2341XXX,2,Dial(ss7/C/${EXTEN:-7})and on my sip.cong I specify only g729.
am I running in pass-thru or transcoding?from the top I can see that at no calls at my server
top - 06:15:38 up 1 day, 5:42, 1 user, load average: 0.72, 0.42, 0.31
Tasks: 50 total, 2 running, 48 sleeping, 0 stopped, 0 zombie
Cpu(s): 5.7% us, 7.0% sy, 0.0% ni, 86.2% id, 1.0% wa, 0.0% hi, 0.0% si
Mem: 4084360k total, 604348k used, 3480012k free, 128832k buffers
Swap: 2031608k total, 0k used, 2031608k free, 222688k cachedand I cannot see what is using up my cpu
thank you goksieOn 5/31/06, Steve Totaro
[EMAIL PROTECTED] wrote:
All I know is that it is very processor intensive and either not usingit or just passing it through is your best bet.I will be working alot
with G729 in the near future and will post my findings but until then Iam just relying on the dimensioning page on the wiki.Thanks,Steve TotaroGoke Aruna wrote: Steve, Can you please give me an insight on how g729 problem could solved?
goksie On 5/30/06, *Steve Totaro* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
wrote: G729 is your problem. Thanks, Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message- From: Lachek Butalek [mailto:[EMAIL PROTECTED] mailto:
[EMAIL PROTECTED]]
Sent: Tuesday, May 30, 2006 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] I guess my server capacity is ok
What process is taking up 100% CPU? Is it Asterisk processes or something else? Also, is the load spread out over multiple processes, or do you have one or two processes taking up 90% or more of your
total? You also have dual CPUs (and hyperthreading, which to FC3 should look like 4 CPUs if I'm not mistaken) - is the 100% CPU usage across all
two (or four) processors, or is it only CPU1 that peaks at 100%? Have a look at Last Used CPU in top. What load are the other CPUs at? I don't have personal experience running that large of an
installation, but I imagine your system specs would allow you to handle more simultaneous calls than 50, even though you're doing some transcoding.
On 5/30/06, Goke Aruna [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: can someone overthere help?
the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3
asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way.
however, I noticed the call hit the 51 active calls which is 102channels, I run top to check the system resources usage and i discovered that
the cpu is 100% used. asterisk, sip, ss7never crashed throughout.however, since transcoding takes alot of system resources..
how can I use g729 in passthru mode.and I guess disabling hyperthreading will save me more system resouces.
I will be glad, if you can give me more info on system management cos i think with that system, it should able to handle at least five E1's.
I say thank you for finding time to reply my mail.goksie ___
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RE: [Asterisk-Users] Converting .wav to .WAV
I think WAV is the file format and .wav is the file extension of wave file. RecordPad sounds generate an extension of .WAV this creates some kind of conflict. When files from RecordPad or WavePad dont play on asterisk simply resample it with sox in same WAV format and you'd be fine. From: Mimmus [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: [Asterisk-Users] Converting .wav to .WAV Date: Wed, 31 May 2006 19:33:11 +0200 Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] configuration
thanks for your response Make sure that the channel where the calls come in route the call to the context where you defined the scripts. How can I do this? big thanks issam - Original Message - From: Henk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, June 01, 2006 7:49 AM Subject: RE: [Asterisk-Users] configuration Create the 2 extensions in /etc/asterisk/extension.conf exten = 8,1,Answer() . Script 1 . exten = 9,1,Answer() . Script 2 . Make sure that the channel where the calls come in route the call to the context where you defined the scripts. Hope this helps, Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: donderdag 1 juni 2006 9:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] configuration hello I have 2 services with 2different numbers. the first is 88 and the second is 99. if a user call 88 I want to execute the script1 and if he call 99 I execute the script2. How can I do my configs files? big Thanks issam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I guess my server capacity is ok
Which DSP based boards does Asterisk support for G729 and are any of these more cost effective than piling on Pentiums? There are none at this time. BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode? Yes. The G.729 codec we distribute is marginally (6-7%) faster on AMD64 in 64-bit mode than in 32-bit mode. what about xeon processors in 64bit mode? as i know the 3.2GHz processors with 1M cache and above are supporting 64bit operations. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk restarting in a minute
yes, it was a typo... and the problem of working too much... crontab? I restart my asterisk nightly with cron but a simple typo could make that every minute instead of every day... shrug Probably any of you meet with the following problem: asterisk is restarting in a minute (if no active call) if active call, it says cannot receive a call due to restart in progress. even if i starting with -c, i have no disconnected, but see the stuff restarting. i've tried to recompile, older version, virgin config, etc. same results. it's happened after a power loss, on a ext3 fs, sitting on a raid1. astdb was deleted, log is not showing any interesting things. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Change g729 payload
Hi All, I have a SIP provider that tells me that my RTP stream uses a 20bytes payload in the g729 coded data. And they would like that we change this to 30bytes (3 frames). But maybe I'm wrong but isn't a certain payload just a standard for a codec ? And if I'm wrong, how can I change the payload for my g729 calls in Asterisk. Greetings, Attilla ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Explicit Dialplan Exit
In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: -=-=-=-=-=- -=-=-=-=-=- Eh, I'm thinking I don't like labels very much. They aren't all they are cracked up to be. Previously, using extensions of the format extension-function, like 2944000-open or 2944000-closed for example, I could break up an extension into logical units based on function, and it made sense. By exclusively using labels, everthing is in the one extension and it isn't as easy to read at a glance. There's also the chance that statements from one section could over-run into another. or... am I missing something? I don't think labels are intended to be a replacement for logically named extensions. I prefer your original dialplan, but with the change that consecutive sequences of priorities after 1 can be numbered n. I think the evolution went like this (I remember watching it happen): a) So that we don't have to renumber lines when adding or removing a step, let's define a priority n that means one more then the previous step. b) Now what do I do when Dial wants to jump to priority+101? I don't know what number to use for the target priority! OK, let's allow n+number. c) That's ok, but if I have, say 3 steps after the Dial, I then have to number to target line of the jump something like n+97, and that offset will change when I add or delete lines above it: back to square one. d) OK, how about adding a label to a priority, so that it can be referred to by name? Oh, but that still doesn't really help the n+100 problem. e) But it does if you allow a priority to be specified as label+number. Then you can label the Dial statement, and use label+101 as the jump destination. f) After all that, priority jumping got deprecated, and now we return DIALSTATUS instead, and do a GOTO based on that. I've probably missed a bit, but certainly I don't think there was the idea that everything should become the same extension and just use labels. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dealing with trafication tone
Hi! Any of you played with tarification tone? We are planning to insert and asterisk box in front of a panasonic with PRI, but the old pbx still needs the tarification tone. Btw, it would be nice, if we could use the tone is asterisk itself (rather than connect the cdr with a tarification system). Thanks! -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Forcing Marker bit
In article [EMAIL PROTECTED], Kevin P. Fleming [EMAIL PROTECTED] wrote: Ira wrote: I would be happy to do this. Is there something that describes how I might accomplish this. One of the things I've never quite figured out is how to save the console output and a SIP debug causes way more than one screen of data. The logger (via logger.conf) can be configured to save all this output to any file you like. If you can't figure it, find a bug marshal in the #asterisk channel on FreeNode and they will be happy to help you. Alternatively, have a look at the script command, which is useful in many contexts, e.g. # script /tmp/output.txt Script started, file is /tmp/output.txt # exec asterisk -rv ... do asterisky stuff ... host*CLI exit Script done, file is /tmp/output.txt # Then everything that went to the terminal is in /tmp/output.txt Also very useful when using gdb to gather debugging information. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Optimal Hardware
I have just finished building a prototype IVR server on a pc for demonstration purpose. My goal is to build a IVR server with the 4G memory, dual xeon processor and a 4 x E1 card. The server would strictly receive incoming calls and serve WAV files. my question is: Is this not an over kill?... has anyone done any bench marks to determine the optimal size of an asrerisk machine?? Response would be appreciated. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astdb entry in sip.conf
use DBput a DBget (http://www.voip-info.org/wiki/view/Asterisk+database) astdb=chan2ext/SIP/grandstream1=1234 is only variable turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 9:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] astdb entry in sip.conf Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/731 : 192.168.6.156:5060:3600:731:sip:[EMAIL PROTECTED]:5060 /dundi/secret : RpC4PXLxtslY5OZOza7OSws60yzaA /dundi/secretexpiry : 1149149129 Do I need to have anything else configured or should I report this as a bug ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] connecting asterisk to pstn help
Hello Masters Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients ,SER is running on port 5060 and Asterisk on 5065, here i need to forward pstn calls to asterisk and i am planning to connect asterisk to a Cisco Gateway , when sip client calls to pstn SER will recieve invite message and it forwards to asterisk 1)how the Asterisk will handle this call with rtp 2)and when pstn customer calls the call goes in to SER and it looks the 'location' database and it will reject call because it is not registerd user so, we take pstn call directly to asterisk and we forward call from asterisk to SER and i want to know is how the SER handle this call that means when SER found a sip client it invites that sip client and which mediaproxy is going to handle this call the SER's or Asterisk's Can we use only one mediaproxy for both SER and ASTERISK by loading modules in ASTERISK so that it will be easy for billing ..??? please explain me how the process will take here bcoz i am with lots of questions and confusions in this particular process hope some body will solve my headache confusion ..Thanks in advance Kindly regards, Ravi. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astdb entry in sip.conf
Err, I'm not trying to write to the db using the dialplan. In sip.conf there seems to be the ability to automatically create a db entry on startup. The line in sip.conf is astdb=chan2ext/SIP/grandstream1=1234; ensures an astDB entry exists But it doesn't ensure an astDB entry exists :) Julian. turby wrote: use DBput a DBget (http://www.voip-info.org/wiki/view/Asterisk+database) astdb=chan2ext/SIP/grandstream1=1234 is only variable turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 9:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] astdb entry in sip.conf Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/731 : 192.168.6.156:5060:3600:731:sip:[EMAIL PROTECTED]:5060 /dundi/secret : RpC4PXLxtslY5OZOza7OSws60yzaA /dundi/secretexpiry : 1149149129 Do I need to have anything else configured or should I report this as a bug ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for very basic example
Hi! Im looking for a very basic example for the following simple problem. I've been searching voip-info.org and looked in the ORA book without a clue. I have a SIP account at
sip.provider.com and my own asterisk server. What I want is the following: I. Register my phone to my asterisk server, not directly to
provider.com II. My asterisk server should ring my phone when somebody calls me on mynumber@provider.com III. Asterisk forwards my outgoing calls to
provider.com I found a lot of sample snippets but none of them really works. The two main problems are: A. When somebody calls me, he get's a user unavailable from
provider.com, but my asterisk server successfully registered at provider.com: (sip.conf) register = user:pwd@
sip.provider.com/user B. When I call a number, my asterisk server says: Failed to authenticate on INVITE. But all login informations for provider.com
are correct. (sip.conf) [user] type=friend secret=pwd username=user fromuser=user canreinvite=yes (extensions.conf
) exten = 0041321112233,1,Dial(SIP/${EXT
...@sip.provider.com,60,r) Thanks for any help!
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Re: [Asterisk-Users] Re: Forcing Marker bit
On 6/1/06, Tony Mountifield [EMAIL PROTECTED] wrote: # script /tmp/output.txt Script started, file is /tmp/output.txt # exec asterisk -rv ... do asterisky stuff ... host*CLI exit Script done, file is /tmp/output.txt # Actually you need another exit in there: # script /tmp/output.txt Script started, file is /tmp/output.txt # exec asterisk -rv ... do asterisky stuff ... host*CLI exit Executing last minute cleanups # exit Script done, file is /tmp/output.txt # Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unknown host cvs.digium.com
I have Digium IAXy 101I. To provision the IAXy, I am following the instrution to download the utility (iaxyprov package): on my linux server (asterisk) I type #cd /usr/src and #export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot then #cvs login (with password anoncvs) but after typing the password (anoncvs) I receive this messages Unknown host cvs.digium.com Can you help me thank Andrea ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] connecting asterisk to pstn help
look for SER and Asterisk on voip-info. I think, you plan to got to UA-SER-(mediaproxy)-Asterisk-PSTN if yes, ser will communicate UA (user agent) on one leg, and asterisk on other. you can use your asterisk to billing and pstn connection. on incoming call dial $phone/ip.address.of.ser Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients ,SER is running on port 5060 and Asterisk on 5065, here i need to forward pstn calls to asterisk and i am planning to connect asterisk to a Cisco Gateway , when sip client calls to pstn SER will recieve invite message and it forwards to asterisk 1)how the Asterisk will handle this call with rtp 2)and when pstn customer calls the call goes in to SER and it looks the 'location' database and it will reject call because it is not registerd user so, we take pstn call directly to asterisk and we forward call from asterisk to SER and i want to know is how the SER handle this call that means when SER found a sip client it invites that sip client and which mediaproxy is going to handle this call the SER's or Asterisk's Can we use only one mediaproxy for both SER and ASTERISK by loading modules in ASTERISK so that it will be easy for billing ..??? please explain me how the process will take here bcoz i am with lots of questions and confusions in this particular process hope some body will solve my headache confusion ..Thanks in advance Kindly regards, Ravi. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for very basic example
On Thu, 2006-06-01 at 11:18 +0200, Benjamin Stocker wrote:
At least you know to break this down into different parts, it still
amazes me how many people look at something as one big thing instead of
several smaller things that interrelate :)
you should have example config files that came with asterisk, if you
built from source you have to do 'make samples' to get them installed,
most binary packages will do this automagically.
I. Register my phone to my asterisk server, not directly to
provider.com
This has 2 parts, one set your phone to use your asterisk server.
Without any knowedge of your phone I cant say how to do this. The other
part is create an account within asterisk for that. In sip.conf you can
create sip users (examples at the end of the default file), in iax.conf
you can create iax2 users, and so on.
II. My asterisk server should ring my phone when somebody calls me
on mynumber@provider.com
You normally have to do 2 things to make your asterisk box register and
work with your provider. One is to add a register directive, ie
register = user:[EMAIL PROTECTED]/extension
the /extension is optional, if specified it will cause calls from your
provider to goto that extension, if omitted generally they goto 's'.
There are examples in at least sip.conf for this but probably iax.conf
as well. Again it depends on the protocol that your provider uses.
The other part is to create a account for your provider. This is
similar to what you would have to do with your phone. The context
declaration here will be used for inbound calls.
As for making it dial your phone, when a call comes in from your
provider. Lets say that the user account created for your provider had
context=incoming and the /extension on the register line was 123, you
could do in extensions.conf:
[incoming]
exten = 123,1,dial(SIP/25)
There are examples of this in extensions.conf.
III. Asterisk forwards my outgoing calls to provider.com
The context that you set your phone into controls what it can call. If
it has a entry like:
exten = _1NXXNXX,1,dial(SIP/myprovider/${EXTEN},90)
then anything matching that pattern (north american numbering pattern
and possibly other places too) will get sent via sip to your provider.
There are examples of this in the extensions.conf sample file as well.
A. When somebody calls me, he get's a user unavailable from
provider.com, but my asterisk server successfully registered at
provider.com:
(sip.conf)
register = user:pwd@sip.provider.com/user
does a sip show peers show that you are registered? Does the extension
at the end of the register line exist?
B. When I call a number, my asterisk server says: Failed to
authenticate on INVITE. But all login informations for provider.com
are correct.
Which leg is failing to auth? The leg from your phone to your asterisk
box or asterisk to your provider?
you only showed one entry in sip.conf, and if you think about it from
your asterisk box's perspective you have 2 people sending and receiving
calls. your phone and your provider. Think of them more or less as
equals and the rest might make sense.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US NY +1 516 687 5200 FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] unknown host cvs.digium.com
digium no longer use cvs. You need to download using subversion. Julian. Andrea Bencini wrote: I have Digium IAXy 101I. To provision the IAXy, I am following the instrution to download the utility (iaxyprov package): on my linux server (asterisk) I type #cd /usr/src and #export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot then #cvs login (with password anoncvs) but after typing the password (anoncvs) I receive this messages Unknown host cvs.digium.com Can you help me thank Andrea ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unknown host cvs.digium.com
Andrea Bencini wrote: Unknown host cvs.digium.com Quoted from the message on the 23rd. As announced when the Asterisk project converted to Subversion as our version control system late last year, it is time to decommission our CVS servers. As of some time in the next couple of days, the cvs.digium.com and related servers will disappear; DNS entries for those names will be removed. If you need to continue building Asterisk or any related projects directly from our SCM (as opposed to building from released packages), you will need to switch to using Subversion to do your checkouts. We are sorry for any inconvenience this may cause you, but the continued maintenance of the CVS servers and mirroring the Subversion repositories into them is no longer something we wish to do Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrading asterisk
Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk. For the second time now, I've had asterisk on a production machine completely freeze (with no messages in any of the log files) and eventually had to be kill -9'd. The machine has a a TDM400 with 1xFXS and 3xFXO cards in it, the most recent time this happened was the day after upgrading to Asterisk 1.2.8 (where I didn't update zaptel at the same time). TIA for any help with this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] choppy audio sip - capi
Further to my previous email, I have definitely established that the audio gets choppy only when the path includes sip and capi. PAP2 to Asterisk to MyNetFone to PSTN is fine. PAP2 to Asterisk MOH is fine. PBX (via capi) to Asterisk MOH is fine PBX (via capi) to Asterisk to PAP2 is choppy PBX (via capi) to Asterisk to MyNetFone to PSTN is choppy (PAP2 is a LinkSys FXS ATA) SIP phone, PBX, and MyNetFone are all configured to use alaw (G.711a), so transcoding should be almost irrelevant. Network bandwidth is not a problem because pure SIP calls are crystal clear - people I have called cannot tell the difference. I also have a X100 card which is providing timing. Nothing is sharing interrupts with anything. Any suggestions? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Forcing Marker bit
In article [EMAIL PROTECTED], Andrew Furey [EMAIL PROTECTED] wrote: On 6/1/06, Tony Mountifield [EMAIL PROTECTED] wrote: # script /tmp/output.txt Script started, file is /tmp/output.txt # exec asterisk -rv ... do asterisky stuff ... host*CLI exit Script done, file is /tmp/output.txt # Actually you need another exit in there: # script /tmp/output.txt Script started, file is /tmp/output.txt # exec asterisk -rv ... do asterisky stuff ... host*CLI exit Executing last minute cleanups # exit Script done, file is /tmp/output.txt # Not if you do exec asterisk -r as I did, instead of just asterisk -r. Using exec makes asterisk replace the shell that was started by script, and therefore auto-exits when you exit asterisk. That is why I use exec in that way - it saves me forgetting the second exit and filling up the script file with rubbish, or even worse, trying to edit it while it is still active! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] audio streaming points different with VRRP
Hi!I've a question: I've 2 asterisk, I want pull the ethernetwireand then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connectiongoes on the second asterisk? I want this: I call to asterisk1, then Ipull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the right run, the callers must point to theasterisk2 Is there some*.config file where I can putmy vrid IP, so in automatic the asterisk1 and asterisk2translate their IP to the vrid? The vrrp is right like I set it.Asterik1 is the master with 192.160.252.1 IP and vrid like 192.160.252.10 asterisk2 is the slave with 192.160.252.2.Via Ethreal they are ok,if .1 goes down, .2 goes up,but the audio streaming point always to 192.160.252.1. Help me please, 1 thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrading asterisk
Thomas Kenyon wrote: Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk. I do as a matter or course. Libpri, Zaptel, Asterisk, Asterisk-addons and Sounds. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: unknown host cvs.digium.com
I there any good reason that is doesn't get posted to the ftp site? People that only use stable may find it easier. -- -- Steven http://www.glimasoutheast.org Doug Lytle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Andrea Bencini wrote: Unknown host cvs.digium.com Quoted from the message on the 23rd. As announced when the Asterisk project converted to Subversion as our version control system late last year, it is time to decommission our CVS servers. As of some time in the next couple of days, the cvs.digium.com and related servers will disappear; DNS entries for those names will be removed. If you need to continue building Asterisk or any related projects directly from our SCM (as opposed to building from released packages), you will need to switch to using Subversion to do your checkouts. We are sorry for any inconvenience this may cause you, but the continued maintenance of the CVS servers and mirroring the Subversion repositories into them is no longer something we wish to do Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Presence
Forrest Beck wrote: Does anyone have a working implementation of SIP Presence? I have a new Grandstream GX-2000 phone with the supported hardware and I am not sure how to setup presence with asterisk. I've just been through this myself. It is relatively simple once you manage to figure it out but really hard until you do! 1) Upgrade to the latest beta firmware for the gpx-2000. (details here: http://www.voip-info.org/wiki/view/GXP-2000) 2) Assign a speed dial button as type Asterisk BLF in the drop down in the Basic page of the grandstream web config system and have it watch a particular extension. Lets call it extension 100 for the purposes of this example. It does not matter what name you give this speed dial entry - it is just a label. In the above step you are effectively telling the grandstream to watch a special hint extension, number 100. 3) Now here's the confusing bit. In extensions.conf you need to use a hint priority which you need to define in the same section as you have your normal extensions defined (you can do it elsewhere but for the purposes of this explanation we'll keep things simple). We are using 100 as our example in step 2. BUT extension 100 does not have to exist in your current dial plan! This is a key thing to get in your head. And if you do have 100 in your current dial-plan it doesn't matter either because adding a hint for that extension will not harm the existing extension. You do not need to have your hint priority anywhere near the lines where you define extension 100 in your dial-plan either -- you can have a block of hints anywhere (as long as they are in the same section as your normal extensions) The syntax is: exten = xxx,hint,sip/y where xxx would be 100 in our example. But what is y? Basically it is the name of the phone/device you want to monitor, as defined between the [ and ] in your sip.conf for that device. You can monitor more than one phone/device at a time by using a syntax like this: exten = xxx,hint,sip/ysip/ (add as many as you like on that line with a inbetween) But lets get specific: In extension.conf, in the same [heading] as your other sip extensions are defined, add: exten = 100,hint,sip/phone1 (where phone1 is a phone as defined in sip.conf and is what you want to watch) IMPORTANT: It is not necessary for phone1 to be configured as extension 100 in extensions.conf. hint extensions and real extensions are separate entities. This is crucial to understand. There is no link between them. As mentioned previously it is not even necessary for extension 100 to be defined previously in extensions.conf at all. Now, at this point, any device set to watch hint extension 100 will be alerted to the status of phone1. (In step 2 we set the grandstream to watch 100, so it will respond to changes in status on that hint extension). It is THAT simple. Only it isn't, because there are some gotchas. First of all, in sip.conf you need to have type=peer in your phone's definition, NOT type=friend. It just doesn't work if you have type=friend for Grandstream phones (polycom phones, on the other hand, won't work if you have type=friend -- they have to be type=peer but at least hints work with them when set to type=peer) The other thing you need for granstreams at least is call-limit=1 in your phone's definition in sip.conf. You may like to experiment with this though, as I'm not 100% sure it really is required. In any case it prevents more than one call ever going into the phone at the same time, which may not be what you want. So, having done all this, restart asterisk, then reboot your phones (an asterisk restart confuses hints/presence on grandstream phones sometimes) At the asterisk command line, enter the command: show hints You should see that extension 100 is shown, and that it was status Idle and 1 watcher. If you use the phone (you have to dial something not just lift the handset) and then use show hints again you should see status = InUse and the red light next to the speed dial button on the watching phone will light up like magic. It is wondrous when it works. Additional info: When setting up a speed dial in the grandstream gxp-2000 as asterisk BLF you can also define which account you want this to work on. If each of the four possible accounts (sip registrations) is connected to different asterisk servers (as opposed to all being configured to register on the same one), depending on which account you select in the Basic page when defining the speed dial, the grandstream will watch the extension defined on that account. This allows you to monitor presence on up to four different asterisk servers. In contrast (and VERY annoyingly), Polycom phones always use the account defined in in the first account (Line 1) - there does not seem to be any way at all to get them to watch extensions on multiple asterisk servers. Faris.
Re: [Asterisk-Users] Re: unknown host cvs.digium.com
Steven wrote: I there any good reason that is doesn't get posted to the ftp site? People that only use stable may find it easier. You mean like this? http://ftp.digium.com/pub/telephony/ -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: unknown host cvs.digium.com
Steven wrote: I there any good reason that is doesn't get posted to the ftp site? People that only use stable may find it easier. I wouldn't be able to answer that. I'm just a every day user, such as yourself, that saw the posting. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Centos cause Asterisk crash
I had exactly the same problem a couple of weeks ago, but have since moved to fc5 on that box. If I recall correctly, it was the /etc/cron.daily/prelink that caused the problem. Rich Sean Kennedy wrote: chan, Run each script seperately to determine which one causes the crash. From there, check your logs to see any error messages. There should be something. My hunch is that prelink will cause the crash. chan (Alpha Trilogies Networks) wrote: Hi, Can some one who experience that does those file necessary for the CentOS and Asterisk installation /etc/cron.daily/00-makewhatis.cron /etc/cron.daily/slocate.cron /etc/cron.daily/prelink /etc/cron.daily/rpm /etc/cron.weekly/00-makewhatis.cron I experience that those file cause my Asterisk Server crash. Can I just disable them and run the Asterisk stable? Any reply will be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astdb entry in sip.conf
I've never attempted to use this feature, so I can neither confirm nor deny whether it works/doesn't work/used to work/etc. But what I find really odd, is that the code doesn't even appear to try and parse astdb when it's loading the config, at least insofar as I can tell. A quick grep -i astdb shows only two lines: #include asterisk/astdb.h ast_verbose(VERBOSE_PREFIX_3 SIP Seeding peer from astdb: '%s' at [EMAIL PROTECTED]:%d for %d\n, Neither of which, obviously, is doing what you want it to. I get the same result both with the 1.2.8 tarball and svn trunk from this morning. I would file a bug, I guess. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] astdb entry in sip.conf Err, I'm not trying to write to the db using the dialplan. In sip.conf there seems to be the ability to automatically create a db entry on startup. The line in sip.conf is astdb=chan2ext/SIP/grandstream1=1234; ensures an astDB entry exists But it doesn't ensure an astDB entry exists :) Julian. turby wrote: use DBput a DBget (http://www.voip-info.org/wiki/view/Asterisk+database) astdb=chan2ext/SIP/grandstream1=1234 is only variable turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 9:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] astdb entry in sip.conf Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/731 : 192.168.6.156:5060:3600:731:sip:[EMAIL PROTECTED]:5060 /dundi/secret : RpC4PXLxtslY5OZOza7OSws60yzaA /dundi/secretexpiry : 1149149129 Do I need to have anything else configured or should I report this as a bug ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astdb entry in sip.conf
Yeah, just found http://bugs.digium.com/view.php?id=3359 where it seems to have been closed out, the code never making it into chan_sip.c However, the option *did* make it's way into sip.conf, so I guess that the real bug is that the option is in sip.conf. Bummer. Devels: Any chance of getting 3359 re-opened and put into asterisk ? Julian Watkins, Bradley wrote: I've never attempted to use this feature, so I can neither confirm nor deny whether it works/doesn't work/used to work/etc. But what I find really odd, is that the code doesn't even appear to try and parse astdb when it's loading the config, at least insofar as I can tell. A quick grep -i astdb shows only two lines: #include asterisk/astdb.h ast_verbose(VERBOSE_PREFIX_3 SIP Seeding peer from astdb: '%s' at [EMAIL PROTECTED]:%d for %d\n, Neither of which, obviously, is doing what you want it to. I get the same result both with the 1.2.8 tarball and svn trunk from this morning. I would file a bug, I guess. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] astdb entry in sip.conf Err, I'm not trying to write to the db using the dialplan. In sip.conf there seems to be the ability to automatically create a db entry on startup. The line in sip.conf is astdb=chan2ext/SIP/grandstream1=1234; ensures an astDB entry exists But it doesn't ensure an astDB entry exists :) Julian. turby wrote: use DBput a DBget (http://www.voip-info.org/wiki/view/Asterisk+database) astdb=chan2ext/SIP/grandstream1=1234 is only variable turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 9:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] astdb entry in sip.conf Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/731 : 192.168.6.156:5060:3600:731:sip:[EMAIL PROTECTED]:5060 /dundi/secret : RpC4PXLxtslY5OZOza7OSws60yzaA /dundi/secretexpiry : 1149149129 Do I need to have anything else configured or should I report this as a bug ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astdb entry in sip.conf
Interesting I always kind of thought is was a cool option to have, though (as I already mentioned) never needed it in my situation(s). That's pretty strange that the option exists in the sample, though. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] astdb entry in sip.conf Yeah, just found http://bugs.digium.com/view.php?id=3359 where it seems to have been closed out, the code never making it into chan_sip.c However, the option *did* make it's way into sip.conf, so I guess that the real bug is that the option is in sip.conf. Bummer. Devels: Any chance of getting 3359 re-opened and put into asterisk ? Julian Watkins, Bradley wrote: I've never attempted to use this feature, so I can neither confirm nor deny whether it works/doesn't work/used to work/etc. But what I find really odd, is that the code doesn't even appear to try and parse astdb when it's loading the config, at least insofar as I can tell. A quick grep -i astdb shows only two lines: #include asterisk/astdb.h ast_verbose(VERBOSE_PREFIX_3 SIP Seeding peer from astdb: '%s' at [EMAIL PROTECTED]:%d for %d\n, Neither of which, obviously, is doing what you want it to. I get the same result both with the 1.2.8 tarball and svn trunk from this morning. I would file a bug, I guess. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] astdb entry in sip.conf Err, I'm not trying to write to the db using the dialplan. In sip.conf there seems to be the ability to automatically create a db entry on startup. The line in sip.conf is astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists But it doesn't ensure an astDB entry exists :) Julian. turby wrote: use DBput a DBget (http://www.voip-info.org/wiki/view/Asterisk+database) astdb=chan2ext/SIP/grandstream1=1234 is only variable turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, June 01, 2006 9:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] astdb entry in sip.conf Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/731 : 192.168.6.156:5060:3600:731:sip:[EMAIL PROTECTED]:5060 /dundi/secret : RpC4PXLxtslY5OZOza7OSws60yzaA /dundi/secretexpiry : 1149149129 Do I need to have anything else configured or should I report this as a bug ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it
Re: [Asterisk-Users] astdb entry in sip.conf
Julian Lyndon-Smith wrote: However, the option *did* make it's way into sip.conf, so I guess that the real bug is that the option is in sip.conf. Yes, and I will fix that in a few minutes. Devels: Any chance of getting 3359 re-opened and put into asterisk ? No, because it's doesn't really provide any value at all. The system administrator can just as easily create the relevant astdb entry when creating the sip.conf entry; the astdb entry is not based on the SIP peer's registration or reachability or anything dynamic in nature. There is also no way to ensure that the astdb entry is _removed_ if the sip.conf entry is removed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Presence
Wonderful explanation! Just a note: So, having done all this, restart asterisk, then reboot your phones (an asterisk restart confuses hints/presence on grandstream phones sometimes) It seems that Asterisk = 1.2.7 solved this issue. Bye Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 2
Hello Guru Thanks for giving reply. so, i can use mediaproxy for both SER and as well as ASTERISK but you told me about voip-info but i dint find much docs regarding SER+asterisk cookbooks and you told me you can use Asterisk as billing for pstn of incoming calls and billing os SER on out going calls it looks good ... But here is one question may be it looks inferior to you but i dont bother how we can connect Asterisk to pstn gateway in my company we have cisco gateways for pstn ..so , how can i connect to that cisco pstn gateway Do i need any hardware or by simply configuring cisco to listen asterisk messages at some port please just give me a out line scenario Thanks in Advance. Regards, Ravi. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: unknown host cvs.digium.com
yup, like that, but with an iaxyprov folder. -- -- Steven http://www.glimasoutheast.org Matt Riddell (IT) [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Steven wrote: I there any good reason that is doesn't get posted to the ftp site? People that only use stable may find it easier. You mean like this? http://ftp.digium.com/pub/telephony/ -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio streaming points different with VRRP
On Thu, 2006-06-01 at 12:30 +0200, Shenen Shenen wrote: Hi!I've a question: I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connection goes on the second asterisk? I want this: I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the right run, the callers must point to the asterisk2 Is there some *.config file where I can put my vrid IP, so in automatic the asterisk1 and asterisk2 translate their IP to the vrid? The vrrp is right like I set it.Asterik1 is the master with 192.160.252.1 IP and vrid like 192.160.252.10 asterisk2 is the slave with 192.160.252.2.Via Ethreal they are ok,if .1 goes down, .2 goes up, but the audio streaming point always to 192.160.252.1. Help me please, 1 thanks I'm not an expert, but... Shouldn't the stream point to .10? It may not be a IP issue, but an ARP cache/MAC address problem. Please provide more details on your configuration, i.e. what's running VRRP (e.g. router, linux box, etc.) and what the network looks like around the redundant machines. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
nice to see your feedback! looks promising.however, would you like to share the *, libpri and zaptel versions you're running on these servers?cheerscurrently running asterisk 1.2.4, zaptel 1.2.5, and no libpri (we're running robbed-bit T1's, EM Wink signalling. a migration to PRI is scheduled, and we'll be running the current version at that time). i also plan on upgrading to the new asterisk and zaptel versions as soon as i've had a chance to test... What were the kernel parameters that you changed? (what OS, by the way?)I am running CentOS 4.3, but have not changed any kernel settings yet.we're running fedora core 4 with the stock kernel, but a move to either CentOS, Debian, or just running Pound Key might be in the works.i currently have the following options added: vga=normal nmi_watchdog=0 acpi=off i had also tried noapic and leaving the nmi_watchdog flag off entirely, didn't seem to make much difference.sorry this isn't in the thread - i'm still getting used to using a mailing list, it's a first for me. wes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL #include
I use the goto to jump across contexts with labels all the time. goto(context,exten,label). works for me. Jason Michael Collins wrote: Oh Crud. So, if I want to jump to another extension or context, I have to specify the full context, extension and priority? I can't specify a label? It's a bit tricky trying to jump to a specific priority in an extension when they're all called 'n' ! Why is something so simple such a mess... Doug, I believe that it has to be one or the other - either labels are unique across the entire dialplan or they are not. However, you may have uncovered a great feature request: allowing the Goto() commands to jump outside the extension and priority while still using a label. I'll post this on the wish list and see what happens. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astdb entry in sip.conf
That's a shame, as I was hoping to use it. Our sip.conf file is produced automatically by a generator and ftp'd to the * server, so there is no manual editing by the administrator . I was wanting to link an [extension] to an email address so that I can do some stuff in the dialplan. This would be a trivial and automatic thing to do (I would just add the code to the generator) Instead now I've got to make sure that the administrator is reminded to manually update the astdb everytime an email address for the extension changes or new phones / people are added or removed. As you say, it can be done manually. I just don't like manual things, as they can be forgotten easily, and for the sake of 5 lines of code it would make *my* (being selfish here) life so much easier. Just my .2p worth. Julian. Kevin P. Fleming wrote: Julian Lyndon-Smith wrote: However, the option *did* make it's way into sip.conf, so I guess that the real bug is that the option is in sip.conf. Yes, and I will fix that in a few minutes. Devels: Any chance of getting 3359 re-opened and put into asterisk ? No, because it's doesn't really provide any value at all. The system administrator can just as easily create the relevant astdb entry when creating the sip.conf entry; the astdb entry is not based on the SIP peer's registration or reachability or anything dynamic in nature. There is also no way to ensure that the astdb entry is _removed_ if the sip.conf entry is removed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Openion on Sipura SPA-2100
Sipura 3000 or the Digium TDM03B On 6/1/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi, As you said, May I know the correct Digium or Sipura product model (Sipura-3102 or Digium?), which is suitable to my requirements? Thank you. Regards, Chandramouli Martin Joseph [EMAIL PROTECTED] wrote: On May 31, 2006, at 10:32 PM, Crazy Boy wrote: Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. The SPA-2100 is an FXS, which allows the connection of phone handsets to your asterisk server. If you want to hook up phone lines you need an FXO. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to decrease answer time !
That's an issue with your IP phone. Check your configuration. I believe most phones call that digit timeout or something like that... it should be set to about 3-4 seconds. You can also try pressing # after dialing the number. On most phones, that will make it dial the number. Good Luck, bp On 6/1/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear listi am using Asterisk 1.2.5 with [EMAIL PROTECTED] .here is my problem.if i dial a number (consider 79)i have to wait around 20 seconds before my Asteisk box response.now i want to decrease this waitingtime . any idea how to do that ?thanksSalaque___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astdb entry in sip.conf
Julian Lyndon-Smith wrote: Instead now I've got to make sure that the administrator is reminded to manually update the astdb everytime an email address for the extension changes or new phones / people are added or removed. As the notes in that original bug told you, there's lots of other ways to accomplish this using existing facilities. If your sip.conf is being generated from a database, then you can easily use a database lookup function directly from the dialplan to query that same database. It's possible to add and remove astdb entries via the manager interface, if you are using that. Certainly you are welcome to apply this relatively small patch to your own systems if you find it useful; that's one of the benefits of using an open source tool, you aren't forced to accept anyone else's idea of what is 'right' for you :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connect 2 Asterisk Servers via PRI
I apreciate all the help. There is something about putting your conf file in an email that helps you see the problems. As I went over them I find small things and in the end it works. Thanks again for the advice on things to check. On 5/31/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: Is the one that's giving the errors the one with the sangoma? Can you post your zaptel.conf and zapata.conf files? ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I use features without enabling 'call parking'?
Is there a way to use 'application mapping' from features.conf without the built in features (pickup, blind transfer, etc.) nor call parking? I have been trying to comment out everything in features.conf, but my asterisk stills shows the defaults... Koen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Jitter buffer. What version of Asterisk PLEASE?
Hi, I am keen to try out the SIP jitter buffer capability. I hear this was available if HEAD. I was wondering if a version of the latest STABLE with this additional feature was available some place.. Or is it simply best to use HEAD? Would some one be kind enough to point me in the right direction, Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AEL2 and CID
From: Douglas Garstang [EMAIL PROTECTED]
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten = 111/666,1,PlayBack(demo-congrats)
exten = 111/666,2,Hangup()
exten = 111,1,PlayBack(demo-moreinfo)
exten = 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats,
everyone
else gets demo-moreinfo.
In AEL:
111 = {
Playback(demo-moreinfo);
Hangup();
};
111/666 = {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
Douglas!! Now, now!! That is not the proper attitude!! ;^) AEL is still
in the formation, we'll look to see if this little problem can be
remedied.
murf
___
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Re: [Asterisk-Users] SIP Jitter buffer. What version of Asterisk PLEASE?
AFAIK, it's only available in Head. Julian. James Gardiner wrote: Hi, I am keen to try out the SIP jitter buffer capability. I hear this was available if HEAD. I was wondering if a version of the latest STABLE with this additional feature was available some place.. Or is it simply best to use HEAD? Would some one be kind enough to point me in the right direction, Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom-Asterisk hints/presence
I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are watching other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is not available, but when it is ringing the status on the watching phone shows the same status as when the extension sis in use, in other words, you can not tell that it is ringing. My goal was to have someones assistant see that the bosss line was ringing and be able to pick it up. I assumed I would have to use the callgroup/pickupgroup to do so, although was optimistic that that the call pickup could be programmed into the line watching button during the ringing state. Show hints does in fact show the different statuses (idle, ringing, inuse). Is what I am trying to accomplish possible today with Polycom/Asterisk? If not what is the closest you can get? I have set this up in a test environment with asterisk 1.2.8 and Polycom SIP 1.6.6 If there is someone out there that has achieved the result I am looking for I will gladly pay for some training - PM me (hope I dont upset the moderatorsJ) Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Several asterisk processes starting with safe_asterisk
Hi, Im running asterisk 1.2.0 on a debian rel 2.6.13 and when I start it with safe asterisk I got instantly more then 10 processes. Until now I didnt detected any impact of this process proliferation in the system, but it is strange and Im not comfortable with this. Is this a know problem? Any ideas what is the problem here, or where to start searching? I think I also have notice this behavior in another system (different debian release) when starting asterisk in the normal way, but I dont have the system here to check. Thanks for any clue you can provide. Ricardo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem when i call to asterisk from traditional phones
Hi , when i call to asterisk from a Skype or Voipbuster phone all the extensios runs good , and i can stablish ZAP to SIP comunication, also i can do a SIP to ZAP call , but when i call from a traditional analogic phone i get these error: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) What i can do ? Asterisk Box has 192.168.1.44 My extensions file: [entrada] exten = s,1,Wait,10 exten = s,2,Answer exten = s,3,Wait,1 exten = s,4,Playback(outputfile) exten = s,5,Wait,1 exten = 1,1,Dial(SIP/[EMAIL PROTECTED],10,Tt) exten = 2,1,Dial(SIP/[EMAIL PROTECTED],10,Tt) -- http://www.tuactualidad.com IM: pollo.es.pollo en gmail.com Te lo traigo fresco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astdb entry in sip.conf
Yeah, I know. Just was hoping to have things the easy way for me. I also want not to have a custom patched box as I know *one* day I'll screw up and lose / forget the patch and wonder why things aint working. Thanks anyway. I'll stop bitching now. Julian. Kevin P. Fleming wrote: Julian Lyndon-Smith wrote: Instead now I've got to make sure that the administrator is reminded to manually update the astdb everytime an email address for the extension changes or new phones / people are added or removed. As the notes in that original bug told you, there's lots of other ways to accomplish this using existing facilities. If your sip.conf is being generated from a database, then you can easily use a database lookup function directly from the dialplan to query that same database. It's possible to add and remove astdb entries via the manager interface, if you are using that. Certainly you are welcome to apply this relatively small patch to your own systems if you find it useful; that's one of the benefits of using an open source tool, you aren't forced to accept anyone else's idea of what is 'right' for you :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2 and CID
Yes! That's the answer I was hoping ! I'm not stupid - it's a *feature* :)
Anything you need testing, let me know !
Julian
Steve Murphy wrote:
From: Douglas Garstang [EMAIL PROTECTED]
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten = 111/666,1,PlayBack(demo-congrats)
exten = 111/666,2,Hangup()
exten = 111,1,PlayBack(demo-moreinfo)
exten = 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats,
everyone
else gets demo-moreinfo.
In AEL:
111 = {
Playback(demo-moreinfo);
Hangup();
};
111/666 = {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
Douglas!! Now, now!! That is not the proper attitude!! ;^) AEL is still
in the formation, we'll look to see if this little problem can be
remedied.
murf
___
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___
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[Asterisk-Users] How to redirect an incoming call to an external phone numer
Hi, I would like to know if it is possible to redirect an incoming call to an external phone number. Can this be done easily? Thanks in advance, Javier ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Delayed Answer
I have an Asterisk system connected with a CLEC that provides SIP termination. When placing calls from phones on the Astersik system to the PSTN, the calling party hears ringing while the called party is saying hello. The problem appears to happen when calling a POTS line. The problem does not seem to occur when calling a PSTN number on a T1 circuit, nor does it occur on inter-office calls. When I listen to the Monitor recordings, I hear ringing and then the called party saying hello? hello? hello? while the calling party hears ringing. Is the ringing that the calling party hears generated by Asterisk and not the ringing in the received audio stream? Would this problem occur if the 200 (OK) message were delayed for any reason? Does Asterisk wait for the 200 message before it connects the received RTP stream with the calling party? Thanks, Asterisk v1.2.4 Polycom IP501 phones Private network between the Asterisk system and the CLEC. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Several asterisk processes starting with safe_asterisk
In case this will be of any use, here it is a list of the processes. We can see that the safe_asterisk script (PID 19368) starts the first asterisk process (PID 19389) that starts a second one (PID 19401) and this second one is responsible to start all the others. root 19368 1 0 10:58 pts/4 00:00:00 /bin/sh /usr/sbin/safe_asterisk root 19389 19368 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19401 19389 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19403 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19404 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19405 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19406 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19407 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19408 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19409 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19410 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19411 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19412 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19413 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19414 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c root 19415 19401 0 10:58 pts/4 00:00:00 /usr/sbin/asterisk -vvvg -c From: Ricardo Monteiro Sent: Thursday, June 01, 2006 2:55 PM To: 'asterisk-users@lists.digium.com' Subject: Several asterisk processes starting with safe_asterisk Hi, Im running asterisk 1.2.0 on a debian rel 2.6.13 and when I start it with safe asterisk I got instantly more then 10 processes. Until now I didnt detected any impact of this process proliferation in the system, but it is strange and Im not comfortable with this. Is this a know problem? Any ideas what is the problem here, or where to start searching? I think I also have notice this behavior in another system (different debian release) when starting asterisk in the normal way, but I dont have the system here to check. Thanks for any clue you can provide. Ricardo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with ZAP dial timeout
I've been looking for the same answer and have posted it twice. I hope someone will eventually have an answer for us both! = = = Original message = = = Hi, I'm having a problem with the timeout option when dialing a ZAP channel. The goal is to ring a number for 15 seconds, if no one picks up, go to voicemail. The dial command is: exten = s,1,Dial(ZAP/1/613555,15) exten = s,2,VoiceMail(u1) exten = s,102,VoiceMail(b1) The call will continue to ring beyond 15 seconds. What's interesting is that a SIP channels does not have this problem. exten = s,1,Dial(SIP/[EMAIL PROTECTED],15) exten = s,2,VoiceMail(u1) exten = s,102,VoiceMail(b1) I have tested in Asterisk 1.2.7.1 and 1.2.8, both have a problem with the Zap channel. Any ideas? TIA, -Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729, voicemail, no codec_g729
I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only format=g729 in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the message is supposed to be recorded, the voicemail app bombs and this is displayed on the console: -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Thanks! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 and dialin
Hi, after some corrections in my settings IAX2 dialin seems to work now. I get the incoming call, but i cannot here anything or can speak. (If I take the call the other side see that the connection is established if I close the call the other site is seeing it too) If I press hold in Idefisk the other side can hear MoH but not me. Asterisk print in the CLI interface that he starts MoH. The firewall isn't blocking any incoming or outgoing package. (I cannot find anything in the log and every blocked package will be logged) My setup is, FreeBSD6 going online with ppp, NAT is done with pf and the firewall too. Asterisk is configured to bind to 0.0.0.0, so it should bind to my tun0 interface and the external IP. netstat -an says: udp4 0 0 *.4569 *.* udp4 0 0 *.5060 *.* If I call outside everything is working fine. Is this a problem with NAT or the maybe the firewall or is it necessary to change some configoptions in asterisk? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unicall Protocol Failure
Cambiando un timer que existe en el archivo mfcr2.c La variable DEFAULT_T1 tiene el valor 5000, incrementalo a 2, compilas, instalas y listo… mas o menos en la linea de codigo 102… actual #define DEFAULT_T1 5000 despues #define DEFAULT_T1 2 Espero te sirva. On 5/30/06, Anton Krall [EMAIL PROTECTED] wrote: Steve Underwood:Steve, why do some numbers give protocol errors? Ive noticed here in Mexicothat certain numbers when dialed return protocol failure and a busy tone.Any idea why this happens and why with only certain phone numbers? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to redirect an incoming call to an external phone numer
sure, exten = 1234567890,1,dial,SIP/[EMAIL PROTECTED] Obviously change SIP for Zap, or IAX if you are using those. When someone calls 1234567890, the pstn phone 9876543210 will ring. bp On 6/1/06, Javier Rodriguez [EMAIL PROTECTED] wrote: Hi,I would like to know if it is possible to redirect an incoming call toan external phone number. Can this be done easily? Thanks in advance,Javier___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729, voicemail, no codec_g729
Kristian Kielhofner wrote: I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only format=g729 in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the message is supposed to be recorded, the voicemail app bombs and this is displayed on the console: -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Thanks! From what I understand, that is the format that Asterisk uses internally. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HDI remove a key from the Asterisk database with a null key, but a value?
Using * 1.0.9 I have a cron job that runs every night and sucks Caller ID information from our SQL server based CRM and imports it into the Asterisk database. We use the Caller ID to give enhanced information about the caller (this is his customer number, he's a pain in the ass, for example). There is one guy in our database with no phone number and he has been sucked in to the 'cidname' family with no key (or a null key, or a zero-length key, I dunno) but a value. The side effect of this, is when someone calls will a caller id of nothing (which is different than UNKNOWN) it shows up as this guy. How do I delete this guy from the database? This does not work: database delete cidname '' I could wipe the family and redo it but there are a lot of manual edits in the DB that we would have to recreate. I could also export the DB and edit it elsewhere but I'm hoping someone can tell me a quick way to do it from the command line. tia ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] skype out
Hello All, Complete newbie to asterisk (OH NO). Is it possible to use my skype out account for an outgoing trunk? If so, can the syntax be found somewhere? Thanks, Peter -- cybersource.us 115 Richfield Road Williamsville, New York 14221 716-553-8525 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to redirect an incoming call to an external phone numer
Of course... you only need to Dial to other port FXO connected to PSTN and passing the number as extension:
[redirection] ; your inconming-calls context
exten=s,1,Dial(Zap/${OTHER_FXO}/${EXTERNAL_NUMBER})
exten=s,2,Hangup
2006/6/1, Javier Rodriguez [EMAIL PROTECTED]:
Hi,I would like to know if it is possible to redirect an incoming call toan external phone number. Can this be done easily?Thanks in advance,Javier___
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[Asterisk-Users] Re: G729, voicemail, no codec_g729
The codec is not just for transcoding audio. It is required to read and write it as well. -- -- Steven http://www.glimasoutheast.org Kristian Kielhofner [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only format=g729 in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the message is supposed to be recorded, the voicemail app bombs and this is displayed on the console: -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Thanks! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFC/R2 for Voice and Data
On Wed, 2006-05-31 at 19:22 -0500, Moises Silva wrote: google zaptel hdlc So it makes no difference if you are using R2 instead of ISDN? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL2 and CID
Correct me if I'm wrong, but doing this CID stuff in AEL may not make as
much sense in terms of converting dialplans over as it seems. I say this,
because with the original usage of the CID checking in the old extension
language, you could base PRIORITIES on the CID, therefore changing only
part of the actual extension logic. With how you're looking at it, that
would effectively render an extension into two separate logical forms,
which I know would definitely confuse people when converting the
languages. Also, doing the CID checks inside the extension gives a larger
degree of control, and makes the dialplan a bit more eligible. From an
administration standpoint, you could have multiple EXTEN/CID's strewn
about, but if you strictly use the in-extension checking, you know that
*THIS* is the extension you're looking for, and *THOSE* CID's are the ones
that are going to do something different.
Just my thoughts on the matter :) Hope they help a little.
On Thu, 1 Jun 2006, Julian Lyndon-Smith wrote:
Yes! That's the answer I was hoping ! I'm not stupid - it's a *feature* :)
Anything you need testing, let me know !
Julian
Steve Murphy wrote:
From: Douglas Garstang [EMAIL PROTECTED]
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten = 111/666,1,PlayBack(demo-congrats)
exten = 111/666,2,Hangup()
exten = 111,1,PlayBack(demo-moreinfo)
exten = 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats, everyone
else gets demo-moreinfo.
In AEL:
111 = {
Playback(demo-moreinfo);
Hangup();
};
111/666 = {
Playback(demo-congrats);
Hangup();
};
does not work. It always plays demo-moreinfo.
I cannot find and docs on how to do this.
Anyone got any idea ?
Many thanks.
Julian
Douglas!! Now, now!! That is not the proper attitude!! ;^) AEL is still
in the formation, we'll look to see if this little problem can be
remedied.
murf
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] skype out
Peter,There is a bounty for someone to get this working, but there's no simple solution as of yet. There is some SIP-to-Skype software that exists, but it is currently only on Windows, and involves a very convoluted setup. AlexOn 6/1/06, Cyber Source [EMAIL PROTECTED] wrote: Hello All,Complete newbie to asterisk (OH NO). Is it possible to use my skypeout account for an outgoing trunk? If so, can the syntax be foundsomewhere? Thanks, Peter-- cybersource.us 115 Richfield Road Williamsville, New York 14221716-553-8525___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729
Steven wrote: The codec is not just for transcoding audio. It is required to read and write it as well. Not true. It's possible to do playback of compressed files without having that codec installed. It should also be possible to record them. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729, voicemail, no codec_g729
Kristian Kielhofner wrote: -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Heh... you'll like this one. Recording voicemail messages involves listening for silence to know when the message recording should stop. That means converting to SLINEAR :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change g729 payload
Attilla De Groot a écrit : Hi All, I have a SIP provider that tells me that my RTP stream uses a 20bytes payload in the g729 coded data. And they would like that we change this to 30bytes (3 frames). But maybe I'm wrong but isn't a certain payload just a standard for a codec ? You're wrong :) And if I'm wrong, how can I change the payload for my g729 calls in Asterisk. I had the same problem. Unfortunately this value is hard coded in Asterisk's code. I don't know if recent versions of Asterisk support this. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue and Real roundrobin
Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent - 2nd Agent - 3rd Agent I've actually solved that by defining penelty for the accounts, but if the 1st Agent does not hear his/her phone and did not logged off correctly, the 2nd or 3rd agent has no chance to get the incoming call on his/her phone. It would be great if there is any solution - else it would be interesting how to send feature requests to asterisk-developers. Greetings from germany, Michael Konietzny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Optimal Hardware
On Jun 1, 2006, at 1:36 AM, Akpome Akpoguma wrote: I have just finished building a prototype IVR server on a pc for demonstration purpose. My goal is to build a IVR server with the 4G memory, dual xeon processor and a 4 x E1 card. The server would strictly receive incoming calls and serve WAV files. my question is: Is this not an over kill?... has anyone done any bench marks to determine the optimal size of an asrerisk machine?? This is kind of a meaningless question. Asterisk has many capabilities. Without knowing how the machine is going to work, and who is going to be using it (and how many), it's not reasonable to answer. Avoiding mismatched codecs goes a long way (ie Asterisk doesn't need to transcode). There is a section in the wiki for estimating or sizing Asterisk servers, you might look there. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Openion on Sipura SPA-2100
Digium Wildcard TDM400P with 4FXO port --- Crazy Boy [EMAIL PROTECTED] wrote: Hi, As you said, May I know the correct Digium or Sipura product model (Sipura-3102 or Digium?), which is suitable to my requirements? Thank you. Regards, Chandramouli Martin Joseph [EMAIL PROTECTED] wrote: On May 31, 2006, at 10:32 PM, Crazy Boy wrote: Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. The SPA-2100 is an FXS, which allows the connection of phone handsets to your asterisk server. If you want to hook up phone lines you need an FXO. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ The all-new Yahoo! Mail goes wherever you go - free your email address from your Internet provider. http://uk.docs.yahoo.com/nowyoucan.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astdb entry in sip.conf
you can use set-var in sip.conf to accomplish this same thing. On 6/1/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Yeah, I know. Just was hoping to have things the easy way for me. I also want not to have a custom patched box as I know *one* day I'll screw up and lose / forget the patch and wonder why things aint working. Thanks anyway. I'll stop bitching now. Julian. Kevin P. Fleming wrote: Julian Lyndon-Smith wrote: Instead now I've got to make sure that the administrator is reminded to manually update the astdb everytime an email address for the extension changes or new phones / people are added or removed. As the notes in that original bug told you, there's lots of other ways to accomplish this using existing facilities. If your sip.conf is being generated from a database, then you can easily use a database lookup function directly from the dialplan to query that same database. It's possible to add and remove astdb entries via the manager interface, if you are using that. Certainly you are welcome to apply this relatively small patch to your own systems if you find it useful; that's one of the benefits of using an open source tool, you aren't forced to accept anyone else's idea of what is 'right' for you :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk: T1 hunt group setup
Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is (555) 222 0001, to have it roll over to the next available T1 channel. Or give busy signal to the caller if all channels are busy. To have one main number for all calls. I thought this had to be done in Central Office for our T1 connection (like DID setup), and T1 provider keeps telling me that this will be function of our phone system (Asterisk). So, do I literally have to implement some sort of logic in extensions.conf, that would allow receive call on main channel and roll it over to a next available DID? And keep main DID number free all the time? Please help!! Andrei (MPI) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for very basic example
On Jun 1, 2006, at 2:18 AM, Benjamin Stocker wrote:
Hi!
Im looking for a very basic example for the following simple problem.
I've been searching voip-info.org and looked in the ORA book without a
clue. I have a SIP account at sip.provider.com and my own asterisk
server. What I want is the following:
I. Register my phone to my asterisk server, not directly to
provider.com
II. My asterisk server should ring my phone when somebody calls me
on mynumber@provider.com
III. Asterisk forwards my outgoing calls to provider.com
I found a lot of sample snippets but none of them really works. The two
main problems are:
A. When somebody calls me, he get's a user unavailable from
provider.com, but my asterisk server successfully registered at
provider.com:
(sip.conf)
register = user:pwd@ sip.provider.com/user
B. When I call a number, my asterisk server says: Failed to
authenticate on INVITE. But all login informations for provider.com
are correct.
(sip.conf)
[user]
type=friend
secret=pwd
username=user
fromuser=user
canreinvite=yes
(extensions.conf )
exten = 0041321112233,1,Dial(SIP/${EXT [EMAIL PROTECTED],60,r)
Make sure to allow the code you want to use in the general section of
sip.conf or iax.conf as the case may be (sip.conf in your case).
Also, your provider should help you configure this... Since they want
it work so you can spend some money.
Marty
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Re: [Asterisk-Users] Upgrading asterisk
Doug Lytle wrote: Thomas Kenyon wrote: Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk. I do as a matter or course. Libpri, Zaptel, Asterisk, Asterisk-addons and Sounds. Doug The problem with zaptel is that even if you can unload the modules and reload them again, it still involves some downtime. Will look at doing it over the weekend (unless I get another crash.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 4
Kevin,since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure sounds like what the error is complaining about).philippe [EMAIL PROTECTED] From: "Kevin P. Fleming" [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comDate: Thu, 01 Jun 2006 10:30:32 -0500Subject: Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729Steven wrote: The codec is not just for transcoding audio. It is required to read and write it as well.Not true. It's possible to do playback of compressed files withouthaving that codec installed. It should also be possible to record them.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729
Sorry for the repost - forgot to put the proper subject last time.Kevin,since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure sounds like what the error is complaining about).philippe [EMAIL PROTECTED] From: "Kevin P. Fleming" [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comDate: Thu, 01 Jun 2006 10:30:32 -0500Subject: Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729Steven wrote: The codec is not just for transcoding audio. It is required to read and write it as well.Not true. It's possible to do playback of compressed files withouthaving that codec installed. It should also be possible to record them.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Ring'em or ping'em. Make PC-to-phone calls as low as 1¢/min with Yahoo! Messenger with Voice.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729, voicemail, no codec_g729
Kevin P. Fleming wrote: Kristian Kielhofner wrote: -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Heh... you'll like this one. Recording voicemail messages involves listening for silence to know when the message recording should stop. That means converting to SLINEAR :-) Kevin, You're right. I do like ;) that one. Hmmm... I wonder what can be done about this, if anything. I take it disabling silence detection could be a really stupid thing. In this case, however, I have no Zap devices (certainly no POTS interfaces without proper hangup detection). It is %100 SIP (or IAX), and I'd like to think that all of my devices and endpoints are somewhat sane enough to properly hangup the channel to end recording. Any thoughts? Thanks! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Polycom-Asterisk hints/presence
Hello, Try both asterisk and ser for IM/presence . --- Damon Estep [EMAIL PROTECTED] a écrit : I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are watching other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is not available, but when it is ringing the status on the watching phone shows the same status as when the extension sis in use, in other words, you can not tell that it is ringing. My goal was to have someone's assistant see that the boss's line was ringing and be able to pick it up. I assumed I would have to use the callgroup/pickupgroup to do so, although was optimistic that that the call pickup could be programmed into the line watching button during the ringing state. Show hints does in fact show the different statuses (idle, ringing, inuse). Is what I am trying to accomplish possible today with Polycom/Asterisk? If not what is the closest you can get? I have set this up in a test environment with asterisk 1.2.8 and Polycom SIP 1.6.6 If there is someone out there that has achieved the result I am looking for I will gladly pay for some training - PM me (hope I don't upset the moderators:-)) Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cmdMonitor distortion/crackling
I've been struggling with a distortion/crackling problem with the Monitor
command in asterisk. I've even brought the dialplan down to a very simple 3
lines...
exten = 263949,1,Answer
exten = 263949,2,Monitor(wav,${CALLERIDNUM})
exten = 263949,3,Wait(10)
The .wav files generated from the monitor are severely distorted and filled
with static. The asterisk box is in production, incoming calls still sound
fine, outgoing calls still sound fine, and even recording of voicemails
still sounds fine. Both the -in.wav and the -out.wav files generated are
distorted beyond recognition. I thought timing may be an issue, but the
same machine hosts conferences with no problem. I've additionally tried a
secondary Asterisk box with a different hardware configuration and received
the same results.
If it helps identify the problem, here's a link to the -in.wav file of me
speaking.
https://www.slashtmp.iu.edu/public/download.php?FILE=scoscmil/31970cumkPb
Any advice would be appreciated.
Thank you,
Scott Miller
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Re: [Asterisk-Users] app_queue and Real roundrobin
Michael Konietzny wrote: i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent - 2nd Agent - 3rd Agent This is not roundrobin, it's linear. We don't have a linear queue strategy at this time. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue and Real roundrobin
On 17:41, Thu 01 Jun 06, Michael Konietzny wrote: Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent - 2nd Agent - 3rd Agent I've actually solved that by defining penelty for the accounts, but if the 1st Agent does not hear his/her phone and did not logged off correctly, the 2nd or 3rd agent has no chance to get the incoming call on his/her phone. It would be great if there is any solution - else it would be interesting how to send feature requests to asterisk-developers. use rrmemory -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Optimal Hardware
Avoiding mismatched codecs goes a long way (ie Asterisk doesn't need to transcode). I use a quad Xeon 700, supporting ~200 users w/ 30 PSTN calls and ~ 40 SIP ~30 IAX calls on the box pretty much continuous, 16 hours a day, 7 days a week. Looking at 'top' right now w/ 41 PSTN calls I'm seeing ~5% on CPU 1 and the rest hovering at about 0.2%. Because of affinity tricks any CPU spikes that happen because it is using mime-construct to email a fax or me running a MySql query happen on the other 3 CPU's and leave CPU1 bound to Asterisk alone. CPU use is so low because codecs on the LAN are native to the PSTN, so there's no transcoding. This is the key. As to why I would use such an old machine? Hey, it's a NetFinity, it'll never die and has redundancy up the wazoo. Also, two independent PCI busses with good throughput. I don't have to worry much about PCI bus contention, since the TDM cards are by themselves on their own bus. NetFinity's are $400 on Ebay, so I've bought a couple more as spares and a dev machine. CPU is overrated, (and in fact, irrelevant) when your config is planned out throughly hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk: T1 hunt group setup
If you have a PRI to a telco, they probably only have a single trunk group with 23 channels in it for your connection. Any calls to any of your numbers may come to you on any channel. Channels are not dedicated to individual numbers. In other words the first call may come in on channel 1, the second on channel 2. They may or may not have dialed the same number. Or perhaps I am misunderstanding something in your setup On Jun 1, 2006, at 10:47 AM, Andrei (MPI) wrote: Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is (555) 222 0001, to have it roll over to the next available T1 channel. Or give busy signal to the caller if all channels are busy. To have one main number for all calls. I thought this had to be done in Central Office for our T1 connection (like DID setup), and T1 provider keeps telling me that this will be function of our phone system (Asterisk). So, do I literally have to implement some sort of logic in extensions.conf, that would allow receive call on main channel and roll it over to a next available DID? And keep main DID number free all the time? Please help!! Andrei (MPI) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk: T1 hunt group setup
If I understand this correctly, you want to be able to accept simultaneous calls to the single DID (555) 222 0001. There is no reason to roll over to another DID if there is already a call on that DID. You can receive as many calls to a single DID for as many channels you have on your T1. -John Andrei (MPI) wrote: Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is (555) 222 0001, to have it roll over to the next available T1 channel. Or give busy signal to the caller if all channels are busy. To have one main number for all calls. I thought this had to be done in Central Office for our T1 connection (like DID setup), and T1 provider keeps telling me that this will be function of our phone system (Asterisk). So, do I literally have to implement some sort of logic in extensions.conf, that would allow receive call on main channel and roll it over to a next available DID? And keep main DID number free all the time? Please help!! Andrei (MPI) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to redirect an incoming call to an external phone numer
Hi If you have conference or 2-way calling (or whatever is that called by telco), look for Flash application. Basically, you would need to flash the line on incoming call, dial new external number with DTMF and hangup. It will redirect the call: exten = 52,1,Wait(1) exten = 52,2,Flash exten = 52,3,Wait(1) exten = 52,4,SendDTMF(1555111) exten = 52,5,Wait(5) exten = 52,6,Flash exten = 52,7,Wait(5) exten = 52,8,Hangup Andrei Javier Rodriguez wrote: Hi, I would like to know if it is possible to redirect an incoming call to an external phone number. Can this be done easily? Thanks in advance, Javier ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
