Re: [asterisk-users] headsets for linksys/sipura phones?
Andrew Joakimsen wrote: On 4/27/07, Per Jessen [EMAIL PROTECTED] wrote: Try your local mobile phone supplier. I used a headset that came with one of my cell phones, and it worked great w/ my SPA-941. Not a bad idea - which make was this for? None of my phones (Ericsson, Nokia) have a 2.5mm socket, they're all special/proprietary. The headset for any other mobile will work. And I thought Nokia did use 2.5mm but reverse polarity I have 4-5 different Nokias, none have a 2.5mm jack. Nothing that even remotely resembles a jack. So far, what I've found is a Plantronics M175 headset. The local plantronics website itself is not very informative wrt what kind of connectors they use. /Per Jessen, Zürich ___ --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] Poor man's High Availability solution
Hi We've got a redfone here and it's working great so far, despite all the TDMoE bad press. The 4-span version is slightly more expensive than a TE410P, so in the end it's gonna be a more affordable solution as you'd need two digium cards (plus maybe the ISDN guard). The downside is that it doesn't have echo cancellation, so you'll have to do it via software while some cards do it on hardware (faster, better, less CPU-intensive...) El sáb, 28-04-2007 a las 23:22 +0200, Laurent CARON escribió: Hi, I'm wondering what the best option to obtain a high availability asterisk server is. I currently use a TE410P (4 x E1) card. I'm thinking of 2 different solutions: - 2 servers configured with Heartbeat + DRBD (drbd mainly for voicemail) and the E1 span plugged to the 2 servers (with a TE410P in each server). - 2 servers configures with Heartbeat + DRBD with the E1 span hooked to an ISDN guard connected to the main server and the backup one. Here comes the real question. Is it technically good to connect an E1 span to 2 cards at the same time (with only one accepting the calls). Since it is possible with BRI cards, i'm wondering if it could be done with PRI. Thanks Laurent ___ --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 -- Vicente Aguilar [EMAIL PROTECTED] Dpto. de Infraestructuras Tlf.: 965 98 71 92 Recursos en la Red, S.L.U. http://www.renr.es ___ --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] Priority in ACD
Hi all, Can someone help me in reslving issue with priority in ACD I am using Asterisk 1.4 and also ACD but when my agent login using priority 1 and 2 or 1 and 3 call come to both the priority which is unusual if anyone encounter this issue please let me know also help me how to comeout of this issue Mantu Jha___ --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] don't want call to get answered
In my * box I've configured two queues and incoming number and whenever any
one calls those number call comes to my *box and it sends call to my agents
in queue. but if no agent is available it still answer the call. Is there
any why when my agents are not available I don't want call to get answered.
Here is my dialplan:
exten = ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
exten = ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
exten = ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
exten = ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
exten = ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
exten = ,6,Goto(out-of-hours,5003,1)
exten = ,7,Answer()
exten = ,8,Playback(custom/next-avail-advisor)
exten =
,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID})
exten = ,10,Monitor(wav,${MONITOR_FILENAME},mb)
exten = ,11,Queue(kbsupport,t)
exten = ,12,Hangup()
thanks
arun
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[asterisk-users] TDM400P and Junghanns QuadBRI issue
Hi List, I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns QuadBRI on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, libpri-1.2.4 and asterisk-1.2.17 When it's the time for ztcfg to do its job it complains with ZT_SPANCONFIG failed on span 2: No such device or address (6) I'm out of ideas what to do to make it to work. Your help is very much appreciated. The config files and results from various commands follow. Thanks, Costa. --- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXO Loopstart (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: D-channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: D-channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: D-channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) 16 channels configured. ZT_SPANCONFIG failed on span 2: No such device or address (6) --- The /etc/modprobe.d/blacklist file contains, amongst others, the following lines: -- blacklist hisax blacklist hisax_fcpcipnp blacklist 8139cp blacklist hfc4s8s_l1 -- The /etc/zaptel.conf file looks like this: - zaptel.conf - # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxsks=1 fxols=2 fxsks=3 fxols=4 # Span 2-5: Junghans span=2,1,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami bchan=5,6 dchan=7 bchan=8,9 dchan=10 bchan=11,12 dchan=13 bchan=14,15 dchan=16 # Global data loadzone= fr defaultzone = fr --- The lsmod | grep zap command gives the following: --- zaptel 182820 8 wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm crc_ccitt 6337 2 zaptel,irda --- The lspci -vv command returns the following info in relation to the Junghanns card: 00:0a.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST] Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at d400 [size=8] Region 1: Memory at e2001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- --- - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] don't want call to get answered
Arun Kumar wrote:
In my * box I've configured two queues and incoming number and whenever
any one calls those number call comes to my *box and it sends call to my
agents in queue. but if no agent is available it still answer the call.
Is there any why when my agents are not available I don't want call to
get answered. Here is my dialplan:
From:
http://voip-info.org/wiki/index.php?page=Asterisk+config+queues.conf
--
joinempty=strict
joinempty set to strict will keep incoming callers from being placed
in queues where there are no agents to take calls. The Queue()
application will return, and the dial plan can detemine what to do next.
--
You should also pick up a copy of Asterisk: The Future of Telephony.
pp.326-328 contain information on queues.conf.
exten = ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
exten = ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
exten = ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
exten = ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
exten = ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
exten = ,6,Goto(out-of-hours,5003,1)
exten = ,7,Answer()
exten = ,8,Playback(custom/next-avail-advisor)
exten =
,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID})
exten = ,10,Monitor(wav,${MONITOR_FILENAME},mb)
exten = ,11,Queue(kbsupport,t)
exten = ,12,Hangup()
You may need to put a handler after priority 11. Your current logic will
unconditionally hangup the channel if no agents are in the queue.
--
Jeff Davis
Netsource Consulting
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RE: [asterisk-users] TDM400P and Junghanns QuadBRI issue
I would check: Cat /proc/zaptel/ To make shure that the cards are activated in the order that you programmed them. Henk _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Backup e-mail Sent: maandag 30 april 2007 13:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM400P and Junghanns QuadBRI issue Hi List, I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns QuadBRI on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, libpri-1.2.4 and asterisk-1.2.17 When it's the time for ztcfg to do its job it complains with ZT_SPANCONFIG failed on span 2: No such device or address (6) I'm out of ideas what to do to make it to work. Your help is very much appreciated. The config files and results from various commands follow. Thanks, Costa. --- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXO Loopstart (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: D-channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: D-channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: D-channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) 16 channels configured. ZT_SPANCONFIG failed on span 2: No such device or address (6) --- The /etc/modprobe.d/blacklist file contains, amongst others, the following lines: -- blacklist hisax blacklist hisax_fcpcipnp blacklist 8139cp blacklist hfc4s8s_l1 -- The /etc/zaptel.conf file looks like this: - zaptel.conf - # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxsks=1 fxols=2 fxsks=3 fxols=4 # Span 2-5: Junghans span=2,1,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami bchan=5,6 dchan=7 bchan=8,9 dchan=10 bchan=11,12 dchan=13 bchan=14,15 dchan=16 # Global data loadzone= fr defaultzone = fr --- The lsmod | grep zap command gives the following: --- zaptel 182820 8 wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm crc_ccitt 6337 2 zaptel,irda --- The lspci -vv command returns the following info in relation to the Junghanns card: 00:0a.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST] Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at d400 [size=8] Region 1: Memory at e2001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- --- _ Ahhh...imagining that irresistible new car smell? Check out new http://us.rd.yahoo.com/evt=48245/*http:/autos.yahoo.com/new_cars.html;_ylc= X3oDMTE1YW1jcXJ2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDbmV3LWNhcnM- cars at Yahoo! Autos. ___ --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] SugarCRM, NO!, Foxpro, SI?
I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet abandoning SugarCRM, and opting to develop their own Visual FoxPro database/CRM? Please don't dump on me now, this is not my idea, I am just asking for comments, to see if my own initial thoughts are reasonably accurate. joe a. ___ --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] headsets for linksys/sipura phones?
Most of the headsets at http://preview.tinyurl.com/38ow27 should work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Per Jessen Sent: April 28, 2007 11:02 AM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] headsets for linksys/sipura phones? Nabeel Jafferali wrote: You can look for headsets made for Motorola cell phones. Also, Plantronics has some compatible models - I can dig up part numbers if you're interested. Yes, please - Plantronics is in my regular suppliers catalog, but still only with 3.5mm jacks. If you've got part#s or URLs, that would be very helpful. /Per Jessen, Zürich ___ --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] SugarCRM, NO!, Foxpro, SI?
Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet abandoning SugarCRM, and opting to develop their own Visual FoxPro database/CRM? Please don't dump on me now, this is not my idea, I am just asking for comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. ___ --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] TDM400P and Junghanns QuadBRI issue
The solution to this issue is to edit /etc/sysconfig/zaptel and add the
following line:
MODULES=$MODULES qozap# BRISTUFF driver
Costa
Henk Dick [EMAIL PROTECTED] wrote:
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
I would check:
Cat /proc/zaptel/
To make shure that the cards are activated in the order that you programmed
them.
Henk
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Backup e-mail
Sent: maandag 30 april 2007 13:26
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TDM400P and Junghanns QuadBRI issue
Hi List,
I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one
Junghanns QuadBRI
on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6).
I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16,
libpri-1.2.4
and asterisk-1.2.17
When it's the time for ztcfg to do its job it complains with
ZT_SPANCONFIG failed on span 2: No such device or address (6)
I'm out of ideas what to do to make it to work. Your help is very much
appreciated.
The config files and results from various commands follow.
Thanks,
Costa.
---
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Loopstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Loopstart (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: D-channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: D-channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: D-channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
16 channels configured.
ZT_SPANCONFIG failed on span 2: No such device or address (6)
---
The /etc/modprobe.d/blacklist file contains, amongst others, the following
lines:
--
blacklist hisax
blacklist hisax_fcpcipnp
blacklist 8139cp
blacklist hfc4s8s_l1
--
The /etc/zaptel.conf file looks like this:
- zaptel.conf -
# Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
fxsks=1
fxols=2
fxsks=3
fxols=4
# Span 2-5: Junghans
span=2,1,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
span=5,0,3,ccs,ami
bchan=5,6
dchan=7
bchan=8,9
dchan=10
bchan=11,12
dchan=13
bchan=14,15
dchan=16
# Global data
loadzone= fr
defaultzone = fr
---
The lsmod | grep zap command gives the following:
---
zaptel 182820 8 wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm
crc_ccitt 6337 2 zaptel,irda
---
The lspci -vv command returns the following info in relation to the
Junghanns card:
00:0a.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller
[HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST]
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr-
Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Interrupt: pin A routed to IRQ 5
Region 0: I/O ports at d400 [size=8]
Region 1: Memory at e2001000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA
PME(D0+,D1+,D2+,D3hot+,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
---
-
Ahhh...imagining that irresistible new car smell?
Check out new cars at Yahoo! Autos.
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-
Re: [asterisk-users] don't want call to get answered
Arun Kumar wrote:
In my * box I've configured two queues and incoming number and whenever
any one calls those number call comes to my *box and it sends call to my
agents in queue. but if no agent is available it still answer the call.
Is there any why when my agents are not available I don't want call to
get answered. Here is my dialplan:
I think that you want asterisk not to pick up the call when all of your
agents are busy, right? I guess that in this case you do not want to use
queues, but you want to build some dialgroup.
So you'll need to do a few things
1- have some extension that agents can call to log on. When they do,
append their account (Technology/resource) to a database record (or
global var).
2- have some extension that agents can call to log off. Reverse of the
above.
3- When somebody calls the extension '' from below, you use Dial on
the database entry or global var to call all of your agents. If the dail
fails, you check the dialstatus to see why and possibly retry after so
many seconds (for a limited amount of tries) and then maybe answer the
call to play an announcement that nobody is available and please try
again later, or would they want to wait an be placed in a queue.
exten = ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
exten = ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
exten = ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
exten = ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
exten = ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
exten = ,6,Goto(out-of-hours,5003,1)
exten = ,7,Answer()
exten = ,8,Playback(custom/next-avail-advisor)
exten =
,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID})
exten = ,10,Monitor(wav,${MONITOR_FILENAME},mb)
exten = ,11,Queue(kbsupport,t)
exten = ,12,Hangup()
thanks
arun
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--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. Thanks. joe a. ___ --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] Remodified Asterisk brute force blockers..
Top of the morning all... So I reworked the pseudo IDS/Brute Force Asterisk script for those who want to either use it, or use it as a baseline to build a better one... The script now does a few things... It logs those with password issues, and blocks them as well. This was done to ensure that a remote user who was blocked can be found in the log. E.g., Sally the homemaker keeps fiddling with her ATA or phone... Toasts her password... She will be blocked, and her username and IP address will be logged in the home directory of the admin running the script. This was done to ensure you don't go blowing away legitimate (01110011011101000111010101110110100101100100 / PEBKAC) users. It also double checks the entries to make sure no one is injecting false parameters into Asterisk which would log say... Your own domain... Some may need to tweak their columns under awk... Test before using on a production machine... Works fine for me under Debian and FC5, results may vary so test it on your own. If you have to ask about what it does, please don't use it... Comments on the awk/sed/grep nightmare... Fire away... It was started as a oneliner that spiraled out of control http://www.infiltrated.net/scripts/ashtray -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] SugarCRM, NO!, Foxpro, SI?
First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Have a look at vTiger then (fork of SugarCRM). Works with latest PHP and MySQL, easy to configure and is free : http://www.vtiger.com/ 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] SugarCRM, NO!, Foxpro, SI?
Time Bandit wrote: First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Have a look at vTiger then (fork of SugarCRM). Works with latest PHP and MySQL, easy to configure and is free : http://www.vtiger.com/ This is going somewhat OT, but we have pretty much had to shelve our vtiger project due lack of maturity - support for multiple countries/languages/currencies/VAT-rates is very poor. Or at least it was last time I looked. /Per Jessen, Zürich ___ --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] SugarCRM, NO!, Foxpro, SI?
Joe acquisto wrote: Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. ___ --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] Test
I love these :) - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 27, 2007 7:54 PM Subject: Re: [asterisk-users] Test Failed On 4/26/07, gc [EMAIL PROTECTED] wrote: ___ --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
Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?
I have heard of people rejecting Sugar for their existing CRM/ERP product based on VS Foxpro. I'm not a huge fan of Foxpro myself, but if the system already exist then a lot of people see little advantage in changing. On 4/30/07, Paul [EMAIL PROTECTED] wrote: Joe acquisto wrote: Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. ___ --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 -- Bruce Reeves Nortex 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
SV: [asterisk-users] Early audio(progress) and MOH
Hi, This does not work with early audio (the use of Progress() on a Zap channel before Dial(,20,m)). The caller will not need to pay anything before anyone answers(). But I want to play music or audio, while the call is progressing. Håkon _ Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Steve Totaro Sendt: 29. april 2007 19:06 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [asterisk-users] Early audio(progress) and MOH The m switch should play music just as r will generate a ring. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Håkon Nessjøen Sent: Sunday, April 29, 2007 12:28 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Early audio(progress) and MOH Hi, Is it possible to have MOH in early audio, while waiting for someone to pick up a Dial() call? (When using zap channels, I have early audio working with playback) Håkon Nessjøen Loopback Systems AS ___ --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] SugarCRM, NO!, Foxpro, SI?
I fully understand that but the OP says the programmer does not seem to be comfortable with SugarCRM, MySQL and php. That is quite different from It's easier to build on the code I already have. If I had resisted growth and change over the years, I might be looking for ways to integrate FORTRAN and RPG2 with asterisk today. How about punchcards for the CDR's? Bruce Reeves wrote: I have heard of people rejecting Sugar for their existing CRM/ERP product based on VS Foxpro. I'm not a huge fan of Foxpro myself, but if the system already exist then a lot of people see little advantage in changing. On 4/30/07, *Paul* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Joe acquisto wrote: Paul [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex 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 ___ --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] voicemail + Dynamic mailbox
HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as Extensions but I donot want to add mailboxes in Voicemail.conf Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee Mohammad Mirzaee ___ --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] Voicemail on Different Server
Noah Miller wrote: At the time I set this up, MySQL replication was really designed for one-way replication. Two way replication was possible, but required somewhat unorthodox methods. (Maybe this has changed, I don't know). Configuration is also a little tricky. It's not too bad to set it up between two machines, but 3 machines is more tricky, and 4 is even more tricky, etc, etc. This client had only 3 offices at the time, but I knew they would be expanding. They now have 6. Anyway, after getting everything working, I found that replication would periodically stop after some time. I'd have to re-create the setup, and then replication would work for a time, and then stop again later. This occurred across several different version of MySQL. I suppose I could have fixed this issue with persistence, but unfortunately this was only an annoyance compared to the major issue of data corruption. Your experience with database replication is not unique. I have seen this happen with many flavours of database, not just MySQL. At the critical sites where I've worked, database replication is not even on the table as an option for precisely the reasons you state above: I have yet to meet someone else who has had a positive experience with 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] SugarCRM, NO!, Foxpro, SI?
Paul wrote: Joe acquisto wrote: Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. original poster didn't say it was 'visual foxpro'. who knows, maybe this is that guy that wrote that http server in foxpro dos years ago. G as for the topic, most people tend to use what they know or *feel* will be easiest to integrate. (it is that simple) ___ --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] Viable using purchasing sip lines
Hi: I can try and answer some of your questions. [EMAIL PROTECTED] wrote: Hello All, We have been doing Asterisk and CME implementations recently but we almost always exlusively bring in analog lines and or PRI for PSTN access to our systems. I have known about providers providing SIP based lines and SIP trunks to end users for PSTN access. I am curious about the following: - How practical is this? The idea of terminating pstn calls to across the Internet which is an unguarenteed medium concerns me. Even if our access to it is quazi stable T1 data type of access. Do any of you do systems where this is soley the method used for incoming calls from the pstn? If this is done are there things to look for in a SIP provider, as in their presence on the Internet latency ..etc? You're absolutely right to be concerned. If you need critical service, leave the VoIP terminations for the time being. We *do* use it as an adjunct to primary PSTN lines; so, for example, we might dump a bunch of extra PSTN lines and replace them with SIP or IAX connections; in one case, we just forward on busy from the main PSTN number to the SIP numbers, and that works quite well. When it's working users even like the sound quality better. When it's working ;) It does work most of the time, but when it comes to phone service, user expectations are way different (hell, my expectations are way different, so I can understand). Here are the things I would recommend you pay close attention to when choosing a SIP provider: 1. There are numerous fly-by-night operations. Providing a stable, high availability service is not trivial and costs some money; you want a provider that offers as close to round-the-clock support as you can get. One of the best tests is just to call the provider directly and see what kind of experience you have. When calling some SIP carriers, I have had dropouts, chirping, clicking, calls terminating nowhere, etc. Those are the ones you want to avoid :) Also, if you are spending a long time in queue... be suspicious. There are outfits run by two guys in a basement -- and those two guys are often away skiing or windsurfing, depending on the season. Just be vigilant. 2. Ask for the IP of the PSTN POP for the provider and check the latency. I wouldn't tolerate anything higher than 75 ms, and shoot for something under 50 ms if you can manage it. Sometimes it's not possible and depends where you're located. - What are the major advantages? I know some places provide all you can eat plans which could be seen as a plus and some others provide really low rates. Are there others? The major advantage is that, in general, it is easier to support more channels (when you need them) and the cost per channel is, on average, lower than for a PSTN connection. For example, with certain providers, it's possible to get two channels for slightly more than the cost of a single business line with the PSTN provider. Another advantage: if you only need a few channels, and your PSTN lines are analog, then VoIP connections offer you some call progress detection, which can be useful when you're trying to do follow me ringing off the premises. -Stephen- ___ --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 430 , 501 and 550
Justin Hamade wrote: The 501 is more weird then that. The cat5 cable with the built in power injector is cool but to use it with a PoE (802.3af) switch you need a special cable (the pairs are just different you can probably look it up and make your own). Is this true? I read earlier on the list that there's some sort of logic unit in the cable -- but if it's just a matter of pin assignments, I'll make my own effing cable ;) Does anybody have a clear answer on this? -Stephen- ___ --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] SugarCRM, NO!, Foxpro, SI?
Richard Lyman wrote: Paul wrote: Joe acquisto wrote: Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. original poster didn't say it was 'visual foxpro'. Yes he did say that. Check the indent levels above to see OP is Joe acquisto. ___ --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 650
Klaverstyn, David C wrote: All, I have a Polycom 650 phone, when turned on displays “Checking application”. Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. 1. You need at least SIP 2.0.1 (2.1.0 recommended minimum, 2.1.1 now available) 2. Don't use CFG files from another phone. Follow the Polycom SIP Administrator's Guide (available on the website); make your changes in phone-specific config files and use the sip.cfg and phone1.cfg files in their default state. -Stephen- ___ --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] SugarCRM, NO!, Foxpro, SI?
Paul wrote: Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. What it *sounds* like is that's all he knows and he's too terrified (or unmotivated) to learn anything new. I have met people who are still programming in OS/360 (using an x86 emulation environment). Yes, you read that right. These people sell *software* to *customers*. There comes a time in the life of any technology when you have to take it behind the barn... -Stephen- ___ --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] Simple dial plan inquiry
Hi all,
This is a simple concept, however I'm not entirely comfortable with
available applications and functions available to me to make this happen.
I have a simple dialout macro such as the following:
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten = s,1,Set(CALLERID(number)=${ARG1})
exten = s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten = s,3,Set(ARG2=1${ARG2})
exten = s,4,Dial(${TRUNK}/${ARG2},,m)
exten = s,5,Congestion()exten = s,105,Busy()
This macro overrides one SIP endpoint which I use for personal usage and
do not wish to contain our default CID which is passed through arg1. Is
there anyway I can combine GotoIf/Goto to set it otherwise? I was
thinking in terms of pseudo code to do something similar to the following:
if ($arg1 = SIP/MyPersonal)
{
set caller ID to mypersonal
goto s,2
}
else
{
Set(CALLERID(number)=${ARG1}) ; leave as is
goto s,2 ; leave as is
}
Thanks for any insight.
- sf
___
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OT: Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?
Paul wrote: Richard Lyman wrote: Paul wrote: Joe acquisto wrote: Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. original poster didn't say it was 'visual foxpro'. Yes he did say that. Check the indent levels above to see OP is Joe acquisto. yep, i somehow skipped over that part. more coffee! ___ --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: headsets for linksys/sipura phones?
We are using VXI headsets with our Asterisk PBX as well as our legacy PBX's. A nice feature of these headsets is that you can use the same headset in either a USB port with their DSP translator cord or in a traditional rj-11 port with another cord. This adds some redundancy to your system if you want to enable both hard and softphones without buying all kinds of headsets. They work great with x-lite as well! http://www.vxicorp.com/ Paul ___ --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] SugarCRM, NO!, Foxpro, SI?
Stephen Bosch wrote: Paul wrote: Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. What it *sounds* like is that's all he knows and he's too terrified (or unmotivated) to learn anything new. I have met people who are still programming in OS/360 (using an x86 emulation environment). Yes, you read that right. These people sell *software* to *customers*. There comes a time in the life of any technology when you have to take it behind the barn... Do you mean a trip to the woodshed where I apply the board of education to some technologies I started out with? I don't have any tough love for things like FORTRAN. I'll just disinherit those children. :) ___ --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] Voicemail on Different Server
That's the way we want to go, but have been unable to divine the correct settings for getting it working with MS Exchange. CP Tim Panton wrote: If I were starting a project now, I'd take a look at the (newish) support for IMAP storage for voicemail. ___ --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: OT: Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?
Richard Lyman wrote: Paul wrote: Richard Lyman wrote: Paul wrote: Joe acquisto wrote: Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM: Joe acquisto wrote: I have dual posted this to the user and biz lists. Has anyone ever heard of someone running an Asterisk based system, yet Has abandoning SugarCRM, and opting to develop their own Visual FoxPro Has database/CRM? Please don't dump on me now, this is not my idea, I am just asking for Please comments, to see if my own initial thoughts are reasonably accurate. I'll answer it on the user list. I don't think the idea is developed enough to discuss on biz. First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Second - developing your own CRM is an ambitious undertaking. You need good reasons to go in that direction. Third - I have enough exposure to Visual FoxPro to quickly rule it out as a choice for anything new. The fact that somebody is proposing to use it might give you the idea that they don't know what they are talking about at all. BTW - my exposure to it did include things like access from linux apps using ODBC so I know enough to hate the product. It still makes me wonder why the programmer chooses Visual Foxpro. Sounds like he also rejects many other language and database options. original poster didn't say it was 'visual foxpro'. Yes he did say that. Check the indent levels above to see OP is Joe acquisto. yep, i somehow skipped over that part. more coffee! Sometimes more coffee is what makes me skip over things ___ --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] Send Variable in Dial
Hello to all I need send a data to sofphones screen when I use a Dial () . Thanks a lot Regards Andres Gomez ___ --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] ZAPTEL PROBLEM
Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the all circuits are busy now message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository voip:~# asterisk -rx 'zap show status' Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 voip:~# asterisk -rx 'zap show channels' Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 3from-pstn es 4from-pstn es I thought it could be an IRQ problem, but everything seems fine voip:~# cat /proc/interrupts CPU0 0: 118621819 XT-PIC timer 1:811 XT-PIC i8042 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd:usb3, via82cxxx 6: 5 XT-PIC floppy 7: 0 XT-PIC parport0 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:2879759 XT-PIC uhci_hcd:usb2, eth0 11:3048189 XT-PIC uhci_hcd:usb1, eth1 12: 474378440 XT-PIC ehci_hcd:usb4, wctdm 14:1074418 XT-PIC ide0 15:4239765 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 Hope you can help me with my problem. -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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] Send Variable in Dial
Andres Gomez wrote: I need send a data to sofphones screen when I use a Dial () . SendText()? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Confference function
I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. My problem right now is being able to disconnect the third party and keeping the caller on the line. Would this be a function of Asterisk or the SIP / IAX phone? Any comments would be appreciated. Thank you Ed Nuñez ___ --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] automatically close a meetme
I am looking for a way to automatically close a meetme conference when either a user hangs up or through an agi call? Some method that would automatically terminate the meetme. Is there a way to do that? Jerry ___ --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] Send Variable in Dial
Andres Gomez wrote: Hello to all I need send a data to sofphones screen when I use a Dial () . There is the applications SendText, SendImage or SendURL. Also, for SIP phones you could possibly use SipAddHeader... Thanks a lot Regards Andres Gomez ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Voicemail on Different Server
Anthony Rodgers wrote: That's the way we want to go, but have been unable to divine the correct settings for getting it working with MS Exchange. Just for laughs... what sort of problem do you have? (Stinky, stinky MS Exchange... worst IMAP support -- but hell, maybe we can find a solution) -Stephen- ___ --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 430 , 501 and 550
Stephen Bosch wrote: Justin Hamade wrote: The 501 is more weird then that. The cat5 cable with the built in power injector is cool but to use it with a PoE (802.3af) switch you need a special cable (the pairs are just different you can probably look it up and make your own). Is this true? I read earlier on the list that there's some sort of logic unit in the cable -- but if it's just a matter of pin assignments, I'll make my own effing cable ;) Does anybody have a clear answer on this? The IP 501 supports both Cisco and 802.11af with different cables. While there are pin assignments differences, there are also electrical differences in the discovery protocols. The special cable is an artifact of this. I don't know of anyone who was able to make the phone work without the cable, and the 501 is not designed to work without it. If it were just a matter of pin assignments, then people would be selling cables on eBay. I notice that now that 802.11af is THE standard, Polycom is supporting it on their new phones without any special cables. -- Jeff Davis Netsource Consulting Polycom Certified Reseller ___ --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: Anyone having trouble with claling US Domesticon Sellvoip?
On 2007-03-26 01:46:40 -0700, Salvatore Giudice [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. 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] Polycom 430 , 501 and 550
If it supports the old Cisco POE, you might be able to try this: 568b 1 OrWh 2 Or 3 GrWh 4 Bl 5 WhBl 6 Gr 7 BrWh 8 Br Phone Side 1 OrWh 2 Or 3 GrWh 4 BrWh 5 Br 6 Gr 7 Bl 8 WhBl (From voip-info.org wiki, Cisco POE) That config has allowed me to run 7940g's on a standard Dell POE switch. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Davis Sent: Monday, April 30, 2007 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 430 , 501 and 550 Stephen Bosch wrote: Justin Hamade wrote: The 501 is more weird then that. The cat5 cable with the built in power injector is cool but to use it with a PoE (802.3af) switch you need a special cable (the pairs are just different you can probably look it up and make your own). Is this true? I read earlier on the list that there's some sort of logic unit in the cable -- but if it's just a matter of pin assignments, I'll make my own effing cable ;) Does anybody have a clear answer on this? The IP 501 supports both Cisco and 802.11af with different cables. While there are pin assignments differences, there are also electrical differences in the discovery protocols. The special cable is an artifact of this. I don't know of anyone who was able to make the phone work without the cable, and the 501 is not designed to work without it. If it were just a matter of pin assignments, then people would be selling cables on eBay. I notice that now that 802.11af is THE standard, Polycom is supporting it on their new phones without any special cables. -- Jeff Davis Netsource Consulting Polycom Certified Reseller ___ --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] ZAPTEL PROBLEM
On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote: Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the all circuits are busy now message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) set verbose 3? Call from where? To where? Do you see the relevant channel as offhook in 'zap show channel N' ? Sanity check: asterisk -rx 'show channels' (hmm... asterisk -n -rx 'show channels'hangs for you as well?) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository asterisk-classic or asterisk-bristuff? voip:~# asterisk -rx 'zap show status' Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 voip:~# asterisk -rx 'zap show channels' Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 3from-pstn es 4from-pstn es I thought it could be an IRQ problem, but everything seems fine voip:~# cat /proc/interrupts CPU0 0: 118621819 XT-PIC timer 1:811 XT-PIC i8042 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd:usb3, via82cxxx 6: 5 XT-PIC floppy 7: 0 XT-PIC parport0 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:2879759 XT-PIC uhci_hcd:usb2, eth0 11:3048189 XT-PIC uhci_hcd:usb1, eth1 12: 474378440 XT-PIC ehci_hcd:usb4, wctdm ehci_hcd:usb4 does normally take all the USB interrupts. However this issue is probably not related to missed interrupts , if there are any. 14:1074418 XT-PIC ide0 15:4239765 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] IVR dictionary dial-plan
Does anyone know of an (E)AGI or program to develop a IVR dial-plan which will take a list of words and then do something when a unique branch has been found. i.e. Say there's 3 words demon deacon bishop On a phone they'd be represented as 33666 332266 247467 So if the user enters 2 we know they want bishop if they enter 336 they want demon and 332 they want deacon. Could run the dictionary through a script which could generate the dial-plan or do it via some script interactively. Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] Polycom 430 , 501 and 550
Jeff Davis wrote: The IP 501 supports both Cisco and 802.11af with different cables. While there are pin assignments differences, there are also electrical differences in the discovery protocols. The special cable is an artifact of this. I don't know of anyone who was able to make the phone work without the cable, and the 501 is not designed to work without it. If it were just a matter of pin assignments, then people would be selling cables on eBay. I notice that now that 802.11af is THE standard, Polycom is supporting it on their new phones without any special cables. The IEEE PoE standard -- for those who care -- is actually IEEE 802.3af, not 802.11af. The important one is the 802.3af-2003, because that introduced provisions for preventing ground loops between the PoE midspan and the switch. There's a white paper on the topic at Polycom's website which explains the need for non-standard cable (the newer version of which does in fact have some additional electronics in it): http://polycom.com/common/pw_item_show_doc/1,1276,2766,00.pdf I suspect that the 330, 430 and 550 phones have some sort of built-in ground loop detection and prevention. For anybody contemplating running the 501/301 phones with a straight Cat5 cable: in most cases, it won't work; in some cases, it will work, but you can bake your PoE injector, unless said injector has a ground loop prevention circuit built-in; all the newer ones are required to have this to be truly 802.3af compliant. Remember that a label indicating 802.3af compliance is no assurance, since the standard has been through numerous revisions in the last few years. The moral of the story -- it is probably safer to use the special cable if you have 301s or 501s. Thanks for the feedback! -Stephen- ___ --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] CDR and Billing Issue
Hi Guys, I am having an issue that I have been able to replicate and I want to know if anyone else has this. Extension 100 dials an external number. He speaks for 5 minutes and then transfers the call to extension 200. Extension 200 speaks for 1 hour. When we go through the call logs we see the five minute call to the external number from extension 100. We then see a call from extension 100 to extension 200 for 1 hour. The issue we are having is that we are billing the clients (100 and 200 are both the same client as ours) for calls only that hit the PSTN and not internal calls. The issue comes in that if the call is transfer from one extension to another since we see it as a call from one extension to another we assume that it is an internal call. Is there any way to fix asterisk so that it doesn't do this, am I doing some thing wrong or do all calls have to be attended transfers ? (We don't want to tell this to the clients because then they will figure out the loop hole). Thanks a lot. Dovid ___ --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] Simple dial plan inquiry
Hi Steve -
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten = s,1,Set(CALLERID(number)=${ARG1})
exten = s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten = s,3,Set(ARG2=1${ARG2})
exten = s,4,Dial(${TRUNK}/${ARG2},,m)
exten = s,5,Congestion()
exten = s,105,Busy()
This macro overrides one SIP endpoint which I use for personal usage and
do not wish to contain our default CID which is passed through arg1. Is
there anyway I can combine GotoIf/Goto to set it otherwise? I was
thinking in terms of pseudo code to do something similar to the following:
Looks like it should work. Does it? Dialplan logic is fairly terse.
I don't think you'll be able to clean it up much more than that. If
you're looking for something that looks prettier, you could always use
AEL/AEL2. Of course, in the end AEL code will compile down to
Dialplan code.
- Noah
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Re: [asterisk-users] ZAPTEL PROBLEM
2007/4/30, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote: Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the all circuits are busy now message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) set verbose 3? Call from where? To where? From PSTN to Asterisk and viceversa Do you see the relevant channel as offhook in 'zap show channel N' ? I'm not able to to see the channel anymore. voip*CLI zap show channel 3 Unable to find given channel 3 I found that this error happens every time i receive an inbound call: Apr 30 15:08:39 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED. Destroying channel 3 Sanity check: asterisk -rx 'show channels' voip*CLI show channels Channel Location State Application(Data) 0 active channels 0 active calls voip*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 4from-zaptel es (hmm... asterisk -n -rx 'show channels'hangs for you as well?) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository asterisk-classic or asterisk-bristuff? asterisk-classic voip:~# asterisk -rx 'zap show status' Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 voip:~# asterisk -rx 'zap show channels' Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 3from-pstn es 4from-pstn es I thought it could be an IRQ problem, but everything seems fine voip:~# cat /proc/interrupts CPU0 0: 118621819 XT-PIC timer 1:811 XT-PIC i8042 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd:usb3, via82cxxx 6: 5 XT-PIC floppy 7: 0 XT-PIC parport0 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:2879759 XT-PIC uhci_hcd:usb2, eth0 11:3048189 XT-PIC uhci_hcd:usb1, eth1 12: 474378440 XT-PIC ehci_hcd:usb4, wctdm ehci_hcd:usb4 does normally take all the USB interrupts. However this issue is probably not related to missed interrupts , if there are any. 14:1074418 XT-PIC ide0 15:4239765 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 Any help would be appreciated -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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] ADSL routers with integrated SIP QoS for other devices
Andrew Kohlsmith wrote: On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote: Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. Sangoma S518 (internal PCI) on a Linux box with iproute2/iptables/tc or BSD with pf. These are the best solutions, IMO. The latest Linux kernels also have SIP connection tracking/matching, so it should be possible to mark packets and prioritize based on iptables matching. I have not done this just yet, as the latest 2.6.20/2.6.21 kernels do not play nice with the wanrouter drivers. (note: there was a recent patch to 2.6.20.4 which apparently has much better SIP matching, and has been tested successfully with Asterisk. I have not tested it yet, and the iptables guys have rejected the patch as their direction for packet matching is shifting significantly in the near future. It can be found at http://thread.gmane.org/gmane.comp.security.firewalls.netfilter.devel/18860.) I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! Any chance we can get to see this as it sounds just what i'm looking for? If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. Forbidden You don't have permission to access /~andrew/rc.tc on this server. It's loosely based off of wondershaper, but works much better, IMO. It does host-based prioritization for VOIP, puts mail just underneath bulk traffic, and P2P beyond that (if you have the p2p connmark stuff set). I can completely saturate DSL links with the S518 with this config without appreciable VOIP degradation. Even without an S518, this script works well with external ADSL/cable modems. You may have to play with the upload rate; some cheap ADSL modems will start blocking your upstream traffic beyond as little as 50% of the upstream rate. -A. ___ --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 Thanks Bails ___ --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] Simple dial plan inquiry
Howdy Noah,
I just re-read my original inquiry and noticed my original purpose for
mailing the list was not simple to dig out of the message.
Ultimately, the dialout macro works fabulous. My issue is that I'd like
to be able to override one particular SIP endpoint with its own unique
callerID versus what is passed in $ARG1. So any exten that hits the
dialout macro will get set to the callerID in $ARG1. My one particular
SIP handset, for argument sake, SIP/123 .. should be set to CallerID = 234.
Does that clear up what I'm trying to accomplish some?
Thanks!
- sf
Noah Miller wrote:
Hi Steve -
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten = s,1,Set(CALLERID(number)=${ARG1})
exten = s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten = s,3,Set(ARG2=1${ARG2})
exten = s,4,Dial(${TRUNK}/${ARG2},,m)
exten = s,5,Congestion()
exten = s,105,Busy()
This macro overrides one SIP endpoint which I use for personal usage and
do not wish to contain our default CID which is passed through arg1. Is
there anyway I can combine GotoIf/Goto to set it otherwise? I was
thinking in terms of pseudo code to do something similar to the
following:
Looks like it should work. Does it? Dialplan logic is fairly terse.
I don't think you'll be able to clean it up much more than that. If
you're looking for something that looks prettier, you could always use
AEL/AEL2. Of course, in the end AEL code will compile down to
Dialplan code.
- Noah
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!DSPAM:1020,46364310262288221135878!
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Re: [asterisk-users] ZAPTEL PROBLEM
On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote: Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the all circuits are busy now message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository voip:~# asterisk -rx 'zap show status' Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 voip:~# asterisk -rx 'zap show channels' Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 3from-pstn es 4from-pstn es After some further digging: it turns out that when there' a problem, the output of the above command is actually: voip:~# asterisk -rx 'zap show channels' Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 4from-pstn es that is: channel 3 has been destroyed. The logs show: Apr 30 14:50:53 DEBUG[6003] chan_zap.c: Message status for 401 changed from -1 to 0 on 1 Apr 30 14:51:00 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED. Destroying channel 3 Apr 30 14:51:16 DEBUG[6004] chan_sip.c: Stopping retransmission on [snipped by Tzafrir] This seems to happen when you disconnect an incoming FXO call. ZT_EVENT_REMOVED is defined to be 20 . That package includes a patch that has been applied to trunk in rev 58321. Anybody encountered this in Asterisk/trunk with a wctdm card with an FXO module? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] ADSL routers with integrated SIP QoS for other devices
On Monday 30 April 2007 4:14 pm, bails wrote: I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! Any chance we can get to see this as it sounds just what i'm looking for? Once I find something, yes. :-) If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. Forbidden You don't have permission to access /~andrew/rc.tc on this server. Fixed. -A. ___ --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] Improving Asterisk's DNS support
Hello everyone, After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with iffy DNS servers: - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) - chan_sip will sometimes mark even local SIP peers as unreachable during/after any DNS problems - why? - dnsmgr doesn't support SIP (yikes!): http://bugs.digium.com/view.php?id=9153 - other randomness (please contribute your own experiences) What can we do about improving this situation? At the very least we need to extend DNS manager support to SIP. I'm willing to pay for this and any other Asterisk DNS improvements. Any other ideas? -- 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] ZAPTEL PROBLEM
On Mon, Apr 30, 2007 at 11:37:22PM +0300, Tzafrir Cohen wrote: The logs show: Apr 30 14:50:53 DEBUG[6003] chan_zap.c: Message status for 401 changed from -1 to 0 on 1 Apr 30 14:51:00 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED. Destroying channel 3 Apr 30 14:51:16 DEBUG[6004] chan_sip.c: Stopping retransmission on [snipped by Tzafrir] This seems to happen when you disconnect an incoming FXO call. ZT_EVENT_REMOVED is defined to be 20 . That package includes a patch that has been applied to trunk in rev 58321. oops. With one minor difference: the patch that went into trunk included an extra break that prevented polarity reversal events from falling through to the ZT_EVENT_REMOVED . In short: not a problem with trunk, only with my package. I'll upload fixed packages shortly. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Improving Asterisk's DNS support
Kristian Kielhofner wrote: Hello everyone, After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with iffy DNS servers: - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) wasn't there some setting for that? I run a dns caching deamon om my * box (speeds up enum lookups big time), but i seem to recall that some dns settings could be made - chan_sip will sometimes mark even local SIP peers as unreachable during/after any DNS problems - why? because your * can't resolve the names any more? - dnsmgr doesn't support SIP (yikes!): http://bugs.digium.com/view.php?id=9153 - other randomness (please contribute your own experiences) What can we do about improving this situation? At the very least we need to extend DNS manager support to SIP. I'm willing to pay for this and any other Asterisk DNS improvements. Any other ideas? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] automatically close a meetme
Jerry Geis ha scritto:
I am looking for a way to automatically close a meetme conference
when either a user hangs up or through an agi call?
Look at MeetMe docs.
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
create the MeetMe with the 'x' flag and then put inside it some marked
users ('A')
when the last marked user leaves the conference is closed
Hope it helps
Edoardo
Some method that would automatically terminate the meetme.
Is there a way to do that?
Jerry
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Tel: +39 011 678 100
Fax: +39 011 678 275
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Re: [asterisk-users] Confference function
Hi Ed Ed Nuñez ha scritto: I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service No problem since here 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. What is your Verification service ? A VoIP UA which is called for each received call ?? In this case you should kick it from the conference Here are my suggestion: - Use MeetM Web Control (http://www.voip-info.org/wiki/view/MeetMe-Web-Control) - Use MeetMe b option and write an AGI which react to DTMF pressed by the agent (Pay attention to it: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe#MoreonoptionbAGI_BACKGROUND) - Implement some dirty hack in app_meetme.c (you can define a key which kicks every user markned with 'A' option) Hope it helps Regards Edoardo Serra My problem right now is being able to disconnect the third party and keeping the caller on the line. Would this be a function of Asterisk or the SIP / IAX phone? Any comments would be appreciated. Thank you Ed Nuñez ___ --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 -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --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] 100 users - voip lan security and qos ?
This is a pretty common setup. Just make sure you have ACL's restricting traffic between your data and voice vlan's. Generally, we recommend more than two VLAN's for QoS and security. Usually customers setup the following: 1.) Voice VLAN's for Phones 2.) Data VLAN's for workstations 3.) Voice server VLAN's for IP telephony servers (anything that handles communications media) 4.) Data server VLAN's for intranet services 5.) converged communications VLAN's - Remote access VLAN's and workstation endpoints that have soft phones or IPTV clients fall into this category - 802.1p is recommended for these types of VLAN's 6.) wireless VLAN's - These are seldom built for QoS or streaming media, so they should be segmented and treated differently. All VLAN's should be properly segmented from each other. Ie. Data VLAN's should be restricted from accessing voice VLAN's. All network ingress/egress points should have appropriate SBC's and application layer gateways installed. The network should always be constructed to preserve voice services in the event of a network crisis. If you lose the data side of the network, 95% of large enterprises will always fall back on their telephone and conferencing systems for crisis management. Good luck. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Finkelstein Sent: Sunday, April 29, 2007 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 100 users - voip lan security and qos ? If you are using a cisco switch (2950, 3560, CE500, 4000, 6500, or 3750) then you will be able to setup the phone and have the computer daisy chained to it. I have a similar setup on mine. Here's how I configure my switch ports in order to achieve the desired effect: switchport access vlan 5 switchport voice vlan 6 auto qos voip cisco-phone This is assuming your data VLAN is configured as VLAN 5, and your VoIP VLAN is on VLAN 6. This will allow the phone to create a trunk port and facilitate both end nodes through one switch port. HTH - sf A_ Navone wrote: i have a customer that needs to plug the phones into the pc's using the pass-through rj45 available on most sip phones the question they are asking me is how to keep the data network separate from / secure from the voip network i understand they can set up vlans but i am hazy on a few details 1 since the phones are plugged into the pc's how will the phones be segmented into their own vlan ? 2 assuming the phone sends out a tos bit, how can we confirm that the customer's switch can read the tos bit and correctly prioritize it ? 3 to prioritize voip in the router (coming from the switch) we are looking at the wrtg54L and have found these 2 juicy websites http://openwrt.org and http://www.dd-wrt.com/dd-wrtv2/index.php has anyone downloaded and flashed the voip firmware ? does it give worthwhile advantages over the default firmware ? does the wrtg54L have any advantages over other routers ? any other advice to offer ? thank you so much in advance _ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglineapr il07 ___ --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 !DSPAM:1020,4634f9c388295209328925! ___ --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 1.4.4 VoiceMail ODBC Storage Help
Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] This warning occurs whenever a user leaves a message for an extension. It also occurs when someone dials in to listen to their messages when they hang up. These messages do actually exist within the database, and asterisk does extract them from the database when playing back or recording messages. Here is an example when someone leaves a message for someone: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, Caller VoiceMail Extension = 3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing '/usr/local/asterisk/var/spool/voicemail/users/3031/temp' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-intro' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /usr/local/asterisk/var/spool/voicemail/users/3031/tmp/hGkNG0 format: wav49, 0x90539c8 -- User ended message by pressing # -- SIP/bruce.mcalister-09051118 Playing 'auth-thankyou' (language 'en') [Apr 30 23:56:03] WARNING[30123]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3031/INBOX/msg0002.txt': Found Length is 20600 -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/bruce.mcalister-09051118, ) in new stack == Spawn extension (base-out, 170, 4) exited non-zero on 'SIP/bruce.mcalister-09051118' --- Here is an example when someone listens to their voicemail messages without deleting any: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, voicemail extension=3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing 'vm-password' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-youhave' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/19' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-and' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/20' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-Old' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-onefor' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-first' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-message' (language 'en') == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg.txt': Found -- SIP/bruce.mcalister-09051118 Playing 'vm-unknown-caller' (language 'en') -- SIP/bruce.mcalister-09051118 Playing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-message' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/2' (language 'en') == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0001.txt': Found -- SIP/bruce.mcalister-09051118 Playing 'vm-from-phonenumber' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/4' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/4' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/2' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/0' (language 'en')
[asterisk-users] Zaptel kernel module load order
Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which happens more often, nothing will initialize properly. When this happens, I have to unload all the zaptel modules, and re-load them over and over again, until the hardware comes up in the correct order. The order it is loaded is in no way related to what order I load the modules on the command line. This problems makes it unlikely that asterisk will start properly if the system is rebooted. Is there something I can do to ensure the modules get loaded in the correct order? Here's my config files, if they will help... # cat /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us span=2,1,0,esf,b8zs bchan=25-47 dchan=48 defaultzone=us loadzone=us fxoks=49-52 defaultzone=us loadzone=us # cat /etc/asterisk/zapata.conf [channels] language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=1 echocancel=yes resetinterval=never channel = 1-23 language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=3 echocancel=yes resetinterval=never channel = 25-47 signalling=fxo_ks usecallerid=yes callerid=Fidelity Reserves group=2 threewaycalling=no context=outgoing channel = 49-52 Thanks for any help, /Mitch ___ --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] Zaptel kernel module load order
mitch, not that I can answer your problem but is this ver 1.4.1? I had a similiar problem in that zapscan was updating the zaptel.conf and nothing would work until I mucked with zaptel.conf.zapscan... I might have the filename wrong as I have multiple files now :(... it has zapscan in the filename... daveC Mitch Jackson wrote: Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which happens more often, nothing will initialize properly. When this happens, I have to unload all the zaptel modules, and re-load them over and over again, until the hardware comes up in the correct order. The order it is loaded is in no way related to what order I load the modules on the command line. This problems makes it unlikely that asterisk will start properly if the system is rebooted. Is there something I can do to ensure the modules get loaded in the correct order? Here's my config files, if they will help... # cat /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us span=2,1,0,esf,b8zs bchan=25-47 dchan=48 defaultzone=us loadzone=us fxoks=49-52 defaultzone=us loadzone=us # cat /etc/asterisk/zapata.conf [channels] language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=1 echocancel=yes resetinterval=never channel = 1-23 language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=3 echocancel=yes resetinterval=never channel = 25-47 signalling=fxo_ks usecallerid=yes callerid=Fidelity Reserves group=2 threewaycalling=no context=outgoing channel = 49-52 Thanks for any help, /Mitch ___ --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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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] ADSL routers with integrated SIP QoS for other devices
Try the intertex gateways http://www.intertex.se/ Here their page outlining the their QoS settings: http://www.intertex.se/products/page.asp?iPageID=143 They have models with ADSL models and wireless access point components. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Saturday, April 28, 2007 11:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] ADSL routers with integrated SIP QoS for other devices Greetings list, Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which I'll never visit (minimum 3 boxes - ADSL modem, m0n0, WiFi AP). So, does anyone have any recommendations for a wireless ADSL router with integrated QoS for SIP/RTP? I've looked at some of the Draytek units (e.g. Vigor 2700V), but I can't find reference as to whether the integrated QoS applies only to the FXS ports in the router itself, or to all SIP traffic (most of the users will have separate SIP hardphones). These are all to be used in the UK, so the device in question needs to support PPPoA. Any suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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] Zaptel kernel module load order
It's generally not recommended to put an analog and digital card in the same box, however that being said Try this. Write a little hack in /etc/rc.local /sbin/modprobe wct4xxp sleep 5 /sbin/modprobe wct4xxp sleep 5 /sbin/ztcfg sleep 5 /sbin/modprobe wctdm sleep 5 /sbin/ztcfg /usr/sbin/safe_asterisk the rc.local script is loaded after all the others so it won't effect anything else, and we had some trouble with some low heat VIA motherboards so we did the modprobe twice for the PRI. Hope this helps. Cheers, Joel Hill Support Engineer Asterisk IT On Mon, 2007-04-30 at 19:14 -0500, Mitch Jackson wrote: Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which happens more often, nothing will initialize properly. When this happens, I have to unload all the zaptel modules, and re-load them over and over again, until the hardware comes up in the correct order. The order it is loaded is in no way related to what order I load the modules on the command line. This problems makes it unlikely that asterisk will start properly if the system is rebooted. Is there something I can do to ensure the modules get loaded in the correct order? Here's my config files, if they will help... # cat /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us span=2,1,0,esf,b8zs bchan=25-47 dchan=48 defaultzone=us loadzone=us fxoks=49-52 defaultzone=us loadzone=us # cat /etc/asterisk/zapata.conf [channels] language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=1 echocancel=yes resetinterval=never channel = 1-23 language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=3 echocancel=yes resetinterval=never channel = 25-47 signalling=fxo_ks usecallerid=yes callerid=Fidelity Reserves group=2 threewaycalling=no context=outgoing channel = 49-52 Thanks for any help, /Mitch ___ --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] Zaptel kernel module load order
On Mon, Apr 30, 2007 at 07:14:22PM -0500, Mitch Jackson wrote: Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. You have exactly two modules. either wctdm loads and registers its span first or wct4xxp loads first and registers its spans. There are three cases to consider: 1. system boot 2. Modules load in the zaptel init script 3. manual load It would make much sense to put the T1 card first, o it will automatically become the master, and you have indeed configured the cards so, The order in (1) is arbitrary but fixed. It is affected by the order of scanning of the PCI bus by udev or someone near-by. It may be modified by changing PCI slots and such. (2) is normally mostly meaningful on reastart: unloading and reloading of the modules. The order in the init.d script is set manually. The order is set by the value of MODULES, as set in /etc/sysconfig/zaptel . With (3) you do what you want. But this is no way for automation... When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which happens more often, nothing will initialize properly. When this happens, I have to unload all the zaptel modules, and re-load them over and over again, until the hardware comes up in the correct order. The order it is loaded is in no way related to what order I load the modules on the command line. This problems makes it unlikely that asterisk will start properly if the system is rebooted. Is there something I can do to ensure the modules get loaded in the correct order? Here's my config files, if they will help... # cat /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us span=2,1,0,esf,b8zs off-topic, but: span=2,2,0,esf,b8zs (give it lower priority), right? bchan=25-47 dchan=48 defaultzone=us loadzone=us fxoks=49-52 defaultzone=us loadzone=us You only need to put defaultzone and loadzone once (harmless, though). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] CDR and Billing Issue
We have the same problem, and it also showed up when clients made three+ way calls. The CDRs would show them making the same call three times simultaneously as the destination field for the second and third calls was still showing the first number they had dialed. One of our clients caught it and asked us how he could have been calling the same person at the same time as he was already calling them... Quite embarrassing. Not sure if issue has been fixed in subsequent release or not... Best, Jonathan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Monday, April 30, 2007 2:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR and Billing Issue Hi Guys, I am having an issue that I have been able to replicate and I want to know if anyone else has this. Extension 100 dials an external number. He speaks for 5 minutes and then transfers the call to extension 200. Extension 200 speaks for 1 hour. When we go through the call logs we see the five minute call to the external number from extension 100. We then see a call from extension 100 to extension 200 for 1 hour. The issue we are having is that we are billing the clients (100 and 200 are both the same client as ours) for calls only that hit the PSTN and not internal calls. The issue comes in that if the call is transfer from one extension to another since we see it as a call from one extension to another we assume that it is an internal call. Is there any way to fix asterisk so that it doesn't do this, am I doing some thing wrong or do all calls have to be attended transfers ? (We don't want to tell this to the clients because then they will figure out the loop hole). Thanks a lot. Dovid ___ --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: Voicemail on Different Server (MySQL Replication split thread)
Your experience with database replication is not unique. I have seen this happen with many flavours of database, not just MySQL. At the critical sites where I've worked, database replication is not even on the table as an option for precisely the reasons you state above: I have yet to meet someone else who has had a positive experience with it. I'm using MySQL replication for my realtime database, 1 master with 10+ slaves. I have not had any data corruption or problems since implementation 6 mos ago. To qualify that I must say, I only do one-way replication. Master and Slaves are on the same switch fabric, .7msec latency between hosts. I attended a MySQL HA class, Replication was covered in depth and discussed heavily, at no time did I receive cautionary information about possible data corruption or bad experiences from others in the class or from the instructors. I do daily backups of the master database and also regularly check bin file status between all servers to ensure no server is falling behind. I'm not trying to dispute or start a flame war, I'm sure replication is not perfect and 100% reliable for every instance. I'm sure there are many stories of failed replication or data corruption when replication is not implemented properly or setup in an environment not particularly suited well for replication. Just wanted to add my own experience. JR -- JR Richardson Engineering for the Masses ___ --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] Segfaults in 1.2.18
Hi, I have upgraded two production Asterisk servers to 1.2.18 from 1.2.17 in response to the security alert on April the 25th. Since the upgrade one server has seg faulted 4 times and the other 2, nothing else has changed on the two servers in the recent past. I use the safe_asterisk script so have core dumps from all the seg faults but running the backtrace only yeildsunresolved symbol messages so I have not been able to isolate which module the seg fault is occurring in. I did recompile Asterisk on one of the servers with the dont-optimize option and waited two days for the next seg fault but still did got the unresolved symbol messages, I'm probably not doing the build correctly. I'm going to downgrade to 1.2.17, which I had no seg faults with on either server in approximately 2 months, and take the risk that no malformed SIP responses will get sent to my servers. I run 90% of my SIP traffic through a SIP proxy anyway and the malformed SIP response would be rejected by it but as my Asterisk servers do use some SIP trunks for the other 10% of traffic I went ahead with the upgrade on the hope of avoding any crashes :(. If anyone else has noticed the same thing it would be worth posting to see if it's a code issue rather then a deployment issue on my side. Regards, The Grey Man Send instant messages to your online friends http://au.messenger.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] Re: Voicemail on Different Server (MySQL Replication split thread)
JR Richardson wrote: Your experience with database replication is not unique. I have seen this happen with many flavours of database, not just MySQL. At the critical sites where I've worked, database replication is not even on the table as an option for precisely the reasons you state above: I have yet to meet someone else who has had a positive experience with it. I'm using MySQL replication for my realtime database, 1 master with 10+ slaves. I have not had any data corruption or problems since implementation 6 mos ago. To qualify that I must say, I only do one-way replication. Master and Slaves are on the same switch fabric, .7msec latency between hosts. I attended a MySQL HA class, Replication was covered in depth and discussed heavily, at no time did I receive cautionary information about possible data corruption or bad experiences from others in the class or from the instructors. Having master and slave servers in the same switch fabric is the only situation in which I would consider replication. The cases that I described were with machines in separate subnets. Replication simply doesn't work that well when there is significant latency. Did they mention that in your HA class? I do daily backups of the master database and also regularly check bin file status between all servers to ensure no server is falling behind. I'm not trying to dispute or start a flame war, I'm sure replication is not perfect and 100% reliable for every instance. I'm sure there are many stories of failed replication or data corruption when replication is not implemented properly or setup in an environment not particularly suited well for replication. Just wanted to add my own experience. Thanks, -Stephen- ___ --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
