[asterisk-users] Asterisk Server : IDE HDD frequent crash
Hey guys,
Iam having a peculiar problem with my asterisk installation. The specs
are..
[EMAIL PROTECTED] ~]# asterisk -V
Asterisk 1.2.7.1
Wildcard: Digium Wildcard TE110P T1/E1
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS)
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1 FXO, 3 FXS)
Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) (12 FXO's - rest
empty)
Total 15 FX0's, 5 FXS out of which 5 to 6 FXO/FXS are being used. We have
about 300 active SIP accounts.
Queues, SIP extensions, Agents are in MySQL database using asterisk
realtime static.
CPU : Intel(R) Xeon(TM) CPU 3.06GHz with Hyper threading
RAM : 1G
Mobo : Intel SE7501HG2
The system is stable, however, the IDE disk crashes every 3/4 months. There
are DMA timeout errors for the IDE disk before it fails completely. The
same issue occured for the past three disks and I was doubting the
recommended hdparm setting
'hdparm -d 1 -X udma2 -c 3 /dev/IDE Device'
So, I removed this setting after the last crash and the system workd fine
for another 3 months. Yes'day, the disk failed again with same symptoms.
All the disks were seagate baraccuda IDE drives.
zttool doesnt show any IRQ misses even without the above hdparm setting and
there is no noticeable problem in asterisk with the PRI line etc. Below is
my /proc/interrupts as well as /dev/hda settings.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 24771857 24719039IO-APIC-edge timer
1:102 62IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 134159 135915IO-APIC-edge ide0
185: 32988610 16463264 IO-APIC-level wctdm
193: 22173177 27275710 IO-APIC-level wctdm
201: 21737611 27711650 IO-APIC-level wctdm24xxp
209: 22038077 27401613 IO-APIC-level wcte11xp
225: 18992311 0 IO-APIC-level eth1
233:1171166879 IO-APIC-level eth0
NMI: 0 0
LOC: 49493157 49493156
ERR: 0
MIS: 0
[EMAIL PROTECTED] ~]# hdparm -i /dev/hda
/dev/hda:
Model=ST340014A, FwRev=3.06, SerialNo=5JX96VFV
Config={ HardSect NotMFM HdSw15uSec Fixed DTR10Mbs RotSpdTol.5% }
RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=4
BuffType=unknown, BuffSize=2048kB, MaxMultSect=16, MultSect=16
CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=78165360
IORDY=on/off, tPIO={min:240,w/IORDY:120}, tDMA={min:120,rec:120}
PIO modes: pio0 pio1 pio2 pio3 pio4
DMA modes: mdma0 mdma1 mdma2
UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5
AdvancedPM=no WriteCache=enabled
Drive conforms to: ATA/ATAPI-6 T13 1410D revision 2:
* signifies the current active mode
I looked at the mailing lists and couldnt any such issues reported.
Please advice. Should i be using SCSI disks on RAID 1 or something ? Will
that help ?
Also, should i be looking at any other mobo then Intel SE7501HG2 ? Iam
planning to put in a another asterisk server as a failover and would
appreciate inputs abt the kind of hardware i should be using for the system
with the specs i mentioned.
Thanks
Dushyanth
___
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Re: [asterisk-users] Asterisk Server : IDE HDD frequent crash
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I would look at ventilation if I were you. Drive failures at the rate
you are talking about can usually be traced back to thermal failures.
Just a thought
Stu
Dushyanth wrote:
Hey guys,
Iam having a peculiar problem with my asterisk installation. The specs
are..
[EMAIL PROTECTED] ~]# asterisk -V
Asterisk 1.2.7.1
Wildcard: Digium Wildcard TE110P T1/E1
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS)
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1 FXO, 3 FXS)
Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) (12 FXO's - rest
empty)
Total 15 FX0's, 5 FXS out of which 5 to 6 FXO/FXS are being used. We have
about 300 active SIP accounts.
Queues, SIP extensions, Agents are in MySQL database using asterisk
realtime static.
CPU : Intel(R) Xeon(TM) CPU 3.06GHz with Hyper threading
RAM : 1G
Mobo : Intel SE7501HG2
The system is stable, however, the IDE disk crashes every 3/4 months. There
are DMA timeout errors for the IDE disk before it fails completely. The
same issue occured for the past three disks and I was doubting the
recommended hdparm setting
'hdparm -d 1 -X udma2 -c 3 /dev/IDE Device'
So, I removed this setting after the last crash and the system workd fine
for another 3 months. Yes'day, the disk failed again with same symptoms.
All the disks were seagate baraccuda IDE drives.
zttool doesnt show any IRQ misses even without the above hdparm setting and
there is no noticeable problem in asterisk with the PRI line etc. Below is
my /proc/interrupts as well as /dev/hda settings.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 24771857 24719039IO-APIC-edge timer
1:102 62IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 134159 135915IO-APIC-edge ide0
185: 32988610 16463264 IO-APIC-level wctdm
193: 22173177 27275710 IO-APIC-level wctdm
201: 21737611 27711650 IO-APIC-level wctdm24xxp
209: 22038077 27401613 IO-APIC-level wcte11xp
225: 18992311 0 IO-APIC-level eth1
233:1171166879 IO-APIC-level eth0
NMI: 0 0
LOC: 49493157 49493156
ERR: 0
MIS: 0
[EMAIL PROTECTED] ~]# hdparm -i /dev/hda
/dev/hda:
Model=ST340014A, FwRev=3.06, SerialNo=5JX96VFV
Config={ HardSect NotMFM HdSw15uSec Fixed DTR10Mbs RotSpdTol.5% }
RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=4
BuffType=unknown, BuffSize=2048kB, MaxMultSect=16, MultSect=16
CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=78165360
IORDY=on/off, tPIO={min:240,w/IORDY:120}, tDMA={min:120,rec:120}
PIO modes: pio0 pio1 pio2 pio3 pio4
DMA modes: mdma0 mdma1 mdma2
UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5
AdvancedPM=no WriteCache=enabled
Drive conforms to: ATA/ATAPI-6 T13 1410D revision 2:
* signifies the current active mode
I looked at the mailing lists and couldnt any such issues reported.
Please advice. Should i be using SCSI disks on RAID 1 or something ? Will
that help ?
Also, should i be looking at any other mobo then Intel SE7501HG2 ? Iam
planning to put in a another asterisk server as a failover and would
appreciate inputs abt the kind of hardware i should be using for the system
with the specs i mentioned.
Thanks
Dushyanth
___
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- --
Randomly Generated Fortune Tag:
Many pages make a thick book.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
iD8DBQFFJfmmK69Y+xPZrWYRAi5jAJ9z3DHMK0sWvjiomDj3Qw0o3CA3vwCeJeIZ
UtyXmqFJTTTQ6iWJCk/fOWI=
=vygm
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Re: [asterisk-users] Asterisk Server : IDE HDD frequent crash
Heat = #1 cause of disk failure. If they are roasting to the touch they will
fail in 2-3 months.
- Original Message -
From: Dushyanth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 10/05/2006 9:44 AM
Subject: [asterisk-users] Asterisk Server : IDE HDD frequent crash
Hey guys,
Iam having a peculiar problem with my asterisk installation. The specs
are..
[EMAIL PROTECTED] ~]# asterisk -V
Asterisk 1.2.7.1
Wildcard: Digium Wildcard TE110P T1/E1
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS)
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1 FXO, 3 FXS)
Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) (12 FXO's - rest
empty)
Total 15 FX0's, 5 FXS out of which 5 to 6 FXO/FXS are being used. We have
about 300 active SIP accounts.
Queues, SIP extensions, Agents are in MySQL database using asterisk
realtime static.
CPU : Intel(R) Xeon(TM) CPU 3.06GHz with Hyper threading
RAM : 1G
Mobo : Intel SE7501HG2
The system is stable, however, the IDE disk crashes every 3/4 months.
There
are DMA timeout errors for the IDE disk before it fails completely. The
same issue occured for the past three disks and I was doubting the
recommended hdparm setting
'hdparm -d 1 -X udma2 -c 3 /dev/IDE Device'
So, I removed this setting after the last crash and the system workd fine
for another 3 months. Yes'day, the disk failed again with same symptoms.
All the disks were seagate baraccuda IDE drives.
zttool doesnt show any IRQ misses even without the above hdparm setting
and
there is no noticeable problem in asterisk with the PRI line etc. Below is
my /proc/interrupts as well as /dev/hda settings.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 24771857 24719039IO-APIC-edge timer
1:102 62IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 134159 135915IO-APIC-edge ide0
185: 32988610 16463264 IO-APIC-level wctdm
193: 22173177 27275710 IO-APIC-level wctdm
201: 21737611 27711650 IO-APIC-level wctdm24xxp
209: 22038077 27401613 IO-APIC-level wcte11xp
225: 18992311 0 IO-APIC-level eth1
233:1171166879 IO-APIC-level eth0
NMI: 0 0
LOC: 49493157 49493156
ERR: 0
MIS: 0
[EMAIL PROTECTED] ~]# hdparm -i /dev/hda
/dev/hda:
Model=ST340014A, FwRev=3.06, SerialNo=5JX96VFV
Config={ HardSect NotMFM HdSw15uSec Fixed DTR10Mbs RotSpdTol.5% }
RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=4
BuffType=unknown, BuffSize=2048kB, MaxMultSect=16, MultSect=16
CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=78165360
IORDY=on/off, tPIO={min:240,w/IORDY:120}, tDMA={min:120,rec:120}
PIO modes: pio0 pio1 pio2 pio3 pio4
DMA modes: mdma0 mdma1 mdma2
UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5
AdvancedPM=no WriteCache=enabled
Drive conforms to: ATA/ATAPI-6 T13 1410D revision 2:
* signifies the current active mode
I looked at the mailing lists and couldnt any such issues reported.
Please advice. Should i be using SCSI disks on RAID 1 or something ? Will
that help ?
Also, should i be looking at any other mobo then Intel SE7501HG2 ? Iam
planning to put in a another asterisk server as a failover and would
appreciate inputs abt the kind of hardware i should be using for the
system
with the specs i mentioned.
Thanks
Dushyanth
___
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[asterisk-users] Asterisk CDR
Hi guys, i just want know how do i enable CDR in asterisk. and is it possible to get the time spent on each extension for a caller? for example time spent in a queue + time spent on agent exten + time spent on ivr so if its possible, how?-- RegardsRizwan HishamSoftware Engineer ___ --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: PoE IP Phone
On 05 Oct 2006 23:06:00 +0200, Benny Amorsen [EMAIL PROTECTED] wrote: Actually, does anyone make an IP phone which doesn't do PoE? It looks like the Linksys phone that resembles a traditional wall mount phone. I have seen no mention in the specs that it operates on PoE. That's a shame, because it would be perfect in breakrooms. ___ --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: Where is the PlayDTMF command?
So I patch my asterisk (version 1.2.12.1) with the patch given by Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch Thanks Moises. When I type in show manager command PlayDTMF it is their. With the show manager commands it is not within the list containing all the commands. When I execute the manager PlayDTMF action, the manager response says DTMF successfully queued. I don't hear anything on the phone, when I look at the CLI I see the following warning message. Its produced everytime I execute the PlayDTMF action. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Am I doing something wrong? Is this a bug? Please help, I need this to work as soon as possible... Thanks for all the help. PS: This reply will probably go under a new thread with the same subject. I receive the digest mode of the mails on this list, and replying to it breaks the thread. How can I avoid this in the future? Thanks. Regards, Jan. ___ --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] Where is the PlayDTMF command?
So I patch my asterisk (version 1.2.12.1) with the patch given by Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch Thanks Moises. When I type in show manager command PlayDTMF it is their. With the show manager commands it is not within the list containing all the commands. When I execute the manager PlayDTMF action, the manager response says DTMF successfully queued. I don't hear anything on the phone, when I look at the CLI I see the following warning message. Its produced everytime I execute the PlayDTMF action. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Am I doing something wrong? Is this a bug? Please help, I need this to work as soon as possible... Thanks for all the help. PS: This reply will probably go under a new thread with the same subject. I receive the digest mode of the mails on this list, and replying to it breaks the thread. How can I avoid this in the future? Thanks. Regards, Jan. ___ --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] DOA IAXy?
Eric I've had issues with Iaxy's that sound very similar, performing a full reset following these instructions and re-provisioning them solved any problems. http://www.voip-info.org/wiki/view/IAXy Bails Erik Anderson wrote: Greetings - I have recently purchased 2 IAXys. The documentation states that you should be able to plug in the network and phone, power it up, and it'll get an IP from the DHCP server. Neither of the IAXys that I have do this. I power them up, get link lights on the network interface, but after ~5 seconds, I get an amber flashing light on the front of the device. I've tried using the reset button many times, I've tried it with different DHCP servers to see if that was the issue, and I'm still seeing the same behavior. Before I return these guys, has anyone else seen this? If so, were you able to revive them? Thanks! -Erik ___ --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 Configuration Complete Newbie question
Hello Am starting on my Asterisk journey - am getting a single span Digium card to connect Asterisk to our Alcatel 4400 EPABX and install about 100 VoIP instruments. The Asterisk VoIP extensions and Alcatel digital extensions have to talk to each other. Am I right in understanding that IN ASTERRISK : I have to create a config with either all Asterisk and Alcatel extensions - which config files? extensions.conf for both with the two types of extensions in different contexts? Would that be the correct way? IN ALCATEL : List of Asterisk extensions and the PRI card to which the calls have to be delivered. Is that broadly correct? Thanks very much Best wishes Iyer ___ --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 gui sans live cd
is there a good and free asterisk gui that is not tight to a live cd? I like [EMAIL PROTECTED] but it looks like I need to install the livecd. I just want to run asterisk on my debian install. Is there a way to run [EMAIL PROTECTED] on debian? or anything similar? thanx in advance Pat ___ --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: OT: Polycom time sync - sorta
DF == Dave Fullerton [EMAIL PROTECTED] writes: DF Greetings I have a couple polycom phones (501 and 601) I'm messing DF around with and I've noticed something weird. Both phones DF synchronize their clocks to a central NTP server here on our DF network and both phones are 11 seconds slow. All of our servers, DF switches, routers and PCs also sync to this time source and are DF spot on. Even the budgetone 101 is spot on. Has anyone else DF experienced this? I know I'm being anal retentive but it's driving DF me nuts. The cumulative amount of leap seconds so far is 11 seconds, I believe. /Benny ___ --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] snom 360: how to make record button working ?
I cannot find this option in the snom firmware, the only thing I found is DTMF via SIP INFO: This sounds nice but I guess it will break stuff if you need DTMF tones to get through the menu of a remote PBX. Ideally * would need to interpret the SIP INFO message from the Snom as start recording. I looked at the patch someone mentioned earlier but to me this looks like re-inventing the wheel by starting the whole recording stuff all over again. All this is not necessary, * should simply treat the SIP INFO message the same as DTMF dialling *1 On Thu, 5 Oct 2006, Mojo with Horan Company, LLC wrote: We use SIP Polycom 501s, and their dtmfmode=rfc2833. The remote party can NOT hear the tones when you start recording. I suspect that if dtmfmode=inband, they WOULD be able to. Could be wrong here, that's just my current rudimentary understanding of the situation :) Moj Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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 !DSPAM:500,452494b8123922068143078! ___ --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] New Asterisk StumbleUpon Group
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Just thought I'd let people know that I've created a new StumbleUpon group for Asterisk sites. If you have a site that is related to Asterisk and is not listed, feel free to add it. Alternatively, if you're new to Asterisk and want to find out what sites are out there pop on over and have a look: http://asterisk.group.stumbleupon.com/ - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFJiRfS6d5vy0jeVcRAvMSAJ4iDOiCx/9wIsw4bUG8z1w5+eiRNgCePxzY P7CC17BwO/D/kZ/wO2DXzs0= =UyFr -END PGP 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] Asterisk Configuration Complete Newbie question
On 10/6/06, K Y Iyer [EMAIL PROTECTED] wrote: Is that broadly correct? Yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Configuration Complete Newbie question
Title: RE: [asterisk-users] Asterisk Configuration Complete Newbie question Thanks very much - let me see how far I can take it now. Best wishes Iyer -Original Message- From: [EMAIL PROTECTED] on behalf of Lacy Moore - Aspendora Sent: Fri 10/6/2006 03:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Configuration Complete Newbie question On 10/6/06, K Y Iyer [EMAIL PROTECTED] wrote: Is that broadly correct? Yes ___ --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: extensions.conf strangeness
On Thu, Oct 05, 2006 at 04:07:14PM +0200, Michael Neuhauser wrote: I've created and attached a one line patch (for 1.4 branch, r44464) that should give you the info you need (sort of). But be aware that I haven't tested it on 1.4 (only on 1.2, but things are different there). Only use this patch on a test system as it will generate massive amounts of output and will considerably slow down call handling. Thank you. I could have written the printf() myself, I just wouldn't have known where to put it :-) I have applied it to trunk (r44544) and it generates output. Unfortunately (or perhaps fortunately), now I'm running on trunk the problem has gone away. That is, with my dialplan of [internal] include = extensions include = outbound include = invalid include = test [from-sip] include = extensions include = outbound include = invalid include = test then both SIP phones and Zap phones work identically: dialling 611 gives I'm sorry, that's not a valid extension, presumably because 'invalid' is before 'test' (where 'invalid' matches _X!, and 'test' matches 611) So I can only guess this is a 1.2 issue which has been fixed in trunk - or else there was some uninitialised variable and the problem is now hidden. Many thanks, Brian. ___ --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] Newbie h/w Q, and confirming basic concepts
On Thu, Oct 05, 2006 at 07:22:16PM -0700, Mike Morris wrote: I'm preparing for my first asterisk install, and would like to ask a hardware question confirm my understanding of some basics: * The Q: I'm looking for 2 FXO ports to have asterisk answer 2 incoming lines. There are single FXO port cards for about $30... but dual cards, or the Digium 400 cards, are all several hundred dollars. Why is this? Are the chipsets so different, or am I missing something? FXS ports are a little bit more sophisticated - they have to provide voltage to ring the phone for example. However, the reason the FXO cards are so cheap is that they are basically WinModems (and hence obsolete consumer-grade gear being shifted out) You do have another alternative: buy an ATA (analogue telephone adaptor) which has, say, one FXO and two FXS ports, and connects to your LAN using ethernet. It talks to your Asterisk server using SIP. This probably works out cheaper than a TDM400P. You also get the advantage that it may reduce the CPU load on your box, since you can arrange for the media streams to run directly between the ATA and your local softphones (i.e. the Asterisk box handles signalling but not audio). This needs the ATA and your softphones to support reinvite, so that Asterisk can switch itself back into the audio stream when necessary (e.g. for conferencing, voicemail etc) The other advantage of ATAs is that they let you build simpler networks. If you don't want the features and complexity of a local Asterisk server, you can point your ATAs and VoIP phones all at an upstream SIP provider like sipgate. You can either give your phones separate accounts, or register them all with the same account (sipgate handles multiple registrations and forking, so that a call to your number will ring all the phones) If you're running behind NAT then you probably need to add a simple SIP proxy like siproxd, but that's something very simple and tiny compared to Asterisk. OTOH, if you do it that way, you deprive yourself of the experience of building and running your own softswitch :-) HTH, Brian. ___ --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] 2x* and realtime
Hi, can two * boxes use the same realtime database? I know they can in terms of connecting to the same db, but it is my understanding that the peers are created realtime as and when it registers, in other words even of the two boxes share the same db, the peer will only exist on the one it registers with? Is there a way to check if a peer is registered with the other box and forward the call there if a call comes in? Tx M ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to forward DID to another Server
Hi all i have Asterisk server I have IP authentication from provider when everi order some DID from him, he will forward to my Asterisk where i register the DID and works fine Now i have given access to one more office so i want to forward some of the DID from my asterisks to other Server how can i do that, and i need to give them access to calling out also Picture is like this USer-office1--InternetOffice1-internetProivider-network Ram ___ --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 act as a proxy ?
Hi can some one clarify does the aterisks act like a SER Ram ___ --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] snom 360: how to make record button working ?
Another way would be to set the dtmf option to speed dial and then add a speed dial number 1: *1 ___ --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: PoE IP Phone
No - at least not that I've been able to figure out. These phone's were made to be used with Cisco's Call manager software (Skinny?) and the SIP firmware doesn't seem to allow this. Softkey buttons (like hold, transfer, conference), seem to be static and you can't change them. You could always use asterisk feature codes of course (like *68, etc). There could be options that I've overlooked but the softkeys seem to be fixed. There IS a row of Line Appearance buttons on the side which you can either program as a line or as a speed dial. Unfortunately I haven't found a way to program the speeddials via tftp either. I've had to manually go in and program speed dials on each phone. I have heard that the 7970 is much more configurable via XML files. I don't know if this is true or not and I don't know how this phone compares to the 7960On 10/6/06, bilal ghayyad [EMAIL PROTECTED] wrote: Thanks a lot for your kindly reply.You can do button assignment, so u can assign for thebutton a function like call forward, call pickup, ...?Please advise me.RegardsBilal__ Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk act as a proxy ?
On 06/10/06, ram [EMAIL PROTECTED] wrote: Hi can some one clarify does the aterisks act like a SER http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie h/w Q, and confirming basic concepts
Look long and hard before purchase of a TDM400 It doesn't work with many motherboards, and Digium's anser is try another Motherboard Seriously consider the Sangoma A200 5 year Warranty and works with all motherboards. John Npvack Mike Morris wrote: I'm preparing for my first asterisk install, and would like to ask a hardware question confirm my understanding of some basics: * The Q: I'm looking for 2 FXO ports to have asterisk answer 2 incoming lines. There are single FXO port cards for about $30... but dual cards, or the Digium 400 cards, are all several hundred dollars. Why is this? Are the chipsets so different, or am I missing something? And here's my plan to teach myself using increasing complexity: * Starting point is standard PC, standard phone line * do trixbox install on PC connect to LAN (no special hardware yet) * then I should be able to setup IVR and voice mail boxes, and access them using a soft phone from any PC on the network, but onlylocally, using extension numbers, right? * I could also plug IP phones directly into the network and access the asterisk box locally, using extension numbers, right? * add FXO card(s), and connect incoming lines. Now I can also: o dialup the voice mailboxes from any standard phone worldwide o Use any of the IP and/or softphones to send and receive calls via PSTN, right? * add FXS card(s) and I can plug standard phones into the asterisk box and I can use them as extensions, to either access voicemail locally, or send/receive calls via PSTN ... right ??? * Now for VoIP I just have to order the service from a vendor like vonage, and I can use any of the soft phones, IP phones or standard handsets for VoIP... right ?? Do I have any major misconceptions about the above? Thanks a lot for your patience !!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: verbose logging to file in 1.4
2006/10/4, Benko [EMAIL PROTECTED]: Hello! How can i change the verbose logging level to a file in 1.4? In 1.2 i was used to set the verbose level via asterisk -Rx 'set verbose 5' but in 1.4 it is always reset to OFF again, so (nearly) nothing is logged to /var/lib/asterisk/verbose: seems the behaviour i was used to has been removed in 1.4, but it still works by uncommenting the verbose-option in asterisk.conf(which i didn't use before): [options] verbose = 5 ___ --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] no callerid from PSTN using TDM2400P
Thanks. My asterisk servers are in California, USA and the service provider is SBC (ATT).
Asterisk 1.2.8zaptel 1.2.6Hardware: digium TDM2422P
I get the following error messages in /var/log/asterisk/messages:
Oct 3 00:34:18 WARNING[16716] chan_zap.c: Ignoring signalling
Oct 3 00:34:18 WARNING[16716] chan_zap.c: Ignoring signalling
Oct 3 00:34:18 WARNING[16716] chan_zap.c: Ignoring signalling
And the following error messages in my Asterisk CLI:
-- Starting simple switch on 'Zap/3-1'
Oct 3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)...
Oct 3 22:53:16 ERROR[17948]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-22)
Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID feed failed: Success
Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID returned with error on channel 'Zap/3-1
My Configs are:
zapata.conf:
[channels]
;
usecallerid=yes
restrictcid=no
callerid=asreceived
cidsignalling=bell
cidstart=ring
hidecallerid=no
usecallingpres=yes
sendcalleridafter=2
ringtimeout=8000
callwaiting=no
usedistinctiveringdetection=no
callwaitingcallerid=yes
threewaycalling=no
transfer=no
canpark=no
cancallforward=no
callreturn=no
;callreturn=yes
faxdetect=no
echocancel=yes
echocancelwhenbridged=yes
callprogress=yes
busydetect=yes
musiconhold=default
useincomingcalleridonzaptransfer=yes
group=1
context=from-pstn
signalling=fxs_ks
channel = 3
;channel = 1-3
extensions.conf:
[from-pstn]
;
; Inbound calls from PSTN line
exten = s,1,NoOp(TIMESTAMP: ${TIMESTAMP})
exten = s,2,NoOp(CONTEXT: ${CONTEXT})
exten = s,3,NoOp(CALLERIDNUM: ${CALLERID(number)})
exten = s,4,NoOp(CALLERIDNAME: ${CALLERID(name)})
exten = s,n,Goto(main-ivr,start,1)Thanks.-- Forwarded message --
From:Eric \ManxPower\ Wieling [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comDate:Wed, 04 Oct 2006 19:32:30 -0500Subject:Re: [asterisk-users] no callerid from PSTN using TDM2400PNaija Man wrote: Hello all, Asterisk 1.2.8
zaptel 1.2.6 Hardware: digium TDM2422P I have a fully configured asterisk system with POTS line for PSTN access. I am not receiving the callerid for incoming calls from the PSTN. I get the
following error message. -- Starting simple switch on 'Zap/3-1' Oct 3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)... Oct 3 22:53:16 ERROR[17948]:
callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-22) Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID feed failed: Success Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID
returned with error on channel 'Zap/3-1It would be helpful to know what country you are in.
___
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Re: [asterisk-users] IP Phones
http://www.voipsupply.com/home.php On 10/4/06, Devraj Mukherjee [EMAIL PROTECTED] wrote: Nokia E series with proper firmware upgrade :) On 10/5/06, Steve Glaus [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I would like to know where I can find the IP Phones that can be used with Asterisk? Is there any link? Regards Bilal Ghayad Mobile: 00965 9849460 Office: 00965 2623174 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Any phone supporting SIP or IAX are good choices for asterisk. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] IP Phones
Grandsream IP phone Budge Tone 1001, 102 Softphone X-Lite Ekiga (Ubuntu) Etc Jose Diaz Forrest Beck wrote: http://www.voipsupply.com/home.php On 10/4/06, Devraj Mukherjee [EMAIL PROTECTED] wrote: Nokia E series with proper firmware upgrade :) On 10/5/06, Steve Glaus [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I would like to know where I can find the IP Phones that can be used with Asterisk? Is there any link? Regards Bilal Ghayad Mobile: 00965 9849460 Office: 00965 2623174 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Any phone supporting SIP or IAX are good choices for asterisk. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] How to forward DID to another Server
Hi Ram - so i want to forward some of the DID from my asterisks to other Server how can i do that, and i need to give them access to calling out also You need to connect your asterisk machine together. The most common ways to do this are either with IAX or SIP. To do this with IAX, you might want to read this: http://astrecipes.net/index.php?n=204 After you have your asterisk servers connected, you can direct calls from one server to the other like this: Dial(IAX2/peer name/exten on remote server) - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Server : IDE HDD frequent crash
On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote: Heat = #1 cause of disk failure. If they are roasting to the touch they will fail in 2-3 months. One word: smartd. I didn't know it existed, and I'm amazed I didn't. Everyone on this list should be running smartd, and know what it's saying. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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: Where is the PlayDTMF command?
On Fri, Oct 06, 2006 at 08:59:41AM +0200, Jan du Toit wrote: PS: This reply will probably go under a new thread with the same subject. I receive the digest mode of the mails on this list, and replying to it breaks the thread. How can I avoid this in the future? Switch out of digest mode. No, seriously. Digest mode is really only practical if all you do is read the traffic. Endless things won't work right if you're trying to participate instead. If that generates too much mail. you need to find a better mail program. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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 Server : IDE HDD frequent crash
I partitioned/formatted a new WD2500 with NTFS on a WinXP machine, filled it with data (mostly 10MB FLAC and SHN soundfiles). Then transferred it to an AAH Asterisk server box with a Digium TDM400P (1FXO/1FXS) and an Audigy2 soundcard. I installed it as hdb, booting off hda (no other drives). I mounted that drive with ntfs-fuse, and then remotely mounted it from another machine (Ubuntu) with sshfs. fuse doesn't fully work, so when I removed some files from the NTFS volume it failed to remove the last file specified for removal from some directories (and therefore their directories). I then opened several of the existing remote files from my local workstation. After about 6 hours, I got a CentOS kernel panic from the AAH server with the NTFS drive, indicating an IRQ conflict. When I rebooted, it continued to kernel panic. Until I rebooted with the Audigy2 soundcard removed, which forced CentOS to deinstall the driver. After which point I deleted the AC97 module for the motherboard soundchip, just to be safe, then shut down, reinserted the Audigy2, restarted, let CentOS automatically remove the AC97 configs, add the Audigy2 configs, and continue normally. Except the drive is now marked dirty, requiring chkdsk, which doesn't run on Linux, and has no Linux equivalent. The NTFS tools that come with fuse and fix the most basic state problems had no effect. But if I force mount, the drive mounts and reads files fine (I don't write to it in its dirty state). Then I shut down, added another WD2500 to the IDE as hdc, booted, and the kernel didn't find hdc when it probed the IDE, though it did see that there was a device on IDE1. I shut down, moved both WD2500s to IDE1, booted, and the kernel found neither hdc nor hdd. So I can't dd the NTFS drive to an ext3 (etc) Linux drive. Even when I removed the Audigy2, left the TDM400P, restored the AC97 module, the kernel is not finding the second IDE drive on probe, no matter where I install it on the IDE buses. I can recover the drive with chkdsk on the WinXP machine that formatted it, and either copy across the LAN or possibly mount in a USB enclosure locally to the Ubuntu machine, then copy across USB to a locally mounted Linux drive. But it looks like an IRQ conflict, or maybe DMA, or other conflict at that level, is interfering with the IDE. The conflict didn't happen with Audigy2 + TDM400P + IDE0/hda, but it does happen when adding hdb/c/d to the mix, unless I remove the soundcard. Maybe the Audigy2 conflicts with the TDM400P in a way that interferes with the IDE. This problem seems like it could destroy drives quicker than their MTBF, so I thought I'd throw it out there. On Fri, 2006-10-06 at 00:26 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 5 Oct 2006 16:44:10 + (UTC) From: Dushyanth [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk Server : IDE HDD frequent crash To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hey guys, Iam having a peculiar problem with my asterisk installation. The specs are.. [EMAIL PROTECTED] ~]# asterisk -V Asterisk 1.2.7.1 Wildcard: Digium Wildcard TE110P T1/E1 Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS) Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1 FXO, 3 FXS) Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) (12 FXO's - rest empty) Total 15 FX0's, 5 FXS out of which 5 to 6 FXO/FXS are being used. We have about 300 active SIP accounts. Queues, SIP extensions, Agents are in MySQL database using asterisk realtime static. CPU : Intel(R) Xeon(TM) CPU 3.06GHz with Hyper threading RAM : 1G Mobo : Intel SE7501HG2 The system is stable, however, the IDE disk crashes every 3/4 months. There are DMA timeout errors for the IDE disk before it fails completely. The same issue occured for the past three disks and I was doubting the recommended hdparm setting 'hdparm -d 1 -X udma2 -c 3 /dev/IDE Device' So, I removed this setting after the last crash and the system workd fine for another 3 months. Yes'day, the disk failed again with same symptoms. All the disks were seagate baraccuda IDE drives. zttool doesnt show any IRQ misses even without the above hdparm setting and there is no noticeable problem in asterisk with the PRI line etc. Below is my /proc/interrupts as well as /dev/hda settings. [EMAIL PROTECTED] ~]# cat /proc/interrupts CPU0 CPU1 0: 24771857 24719039IO-APIC-edge timer 1:102 62IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 134159 135915IO-APIC-edge ide0 185: 32988610 16463264 IO-APIC-level wctdm 193: 22173177 27275710 IO-APIC-level wctdm 201: 21737611 27711650 IO-APIC-level wctdm24xxp 209: 22038077 27401613 IO-APIC-level wcte11xp 225: 18992311
Re: [asterisk-users] No Dialtone
Hi Ed - 5. Digium TDM22 (TDM400P) 6. Analog phone plugged in port 3 Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Zaptel.conf : loadzone=us defaultzone=us fxoks=1,2 fxsks=3,4 Zapata.conf: ;FXS Modules signalling=fxo_ks channel = 1,2 ;FXO Modules signalling=fxs_ks channel = 3,4 Any suggestions? It looks like you may have the phone plugged into the wrong port. You have port 3 listed as an FXO module. You want to plug the phone into an FXS port. - Noah ___ --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] Call Center requirements
Todd,
Appreciate you have submitted to a non-commercial forum. One cannot but note
though that most of what you require is probably already available
off-the-shelf in commercially available packages and does not need to be
reinvented.
If you wish to know more of one such package, please contact me offline :
steve [at} bicomsystems{dot]com
Steve
- Original Message -
From: John Novack [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 05, 2006 8:12 PM
Subject: Re: [asterisk-users] Call Center requirements
HIPPA indeed needs to be considered in any medical application
Requirements are not unreasonable, but the client will suffer if data goes
where it shouldn't
I would also suggest that consideration be given to the Sangoma products.
They have a 5 year warranty, will work with ANY modern motherboard, and if
they don't, you will get top notch support, not the typical Digium answer
of try another motherboard
John Novack
BJ Weschke wrote:
On 10/5/06, Todd Houle Asterisk [EMAIL PROTECTED] wrote:
Hi Guys-
While I played a little with Asterisk a year or so ago, I'm getting
ready to
roll out a project now that I think is perfect for it. My friend with
with
a commercial solution he has been very unhappy with and is thinking of
replacing it with Asterisk. Below are his requirements. Anything here
jump
out as a problem? I'm thinking of purchasing a few Digiium card - not
sure
which we need yet... and finding a box to run it on. The only part I'm
not
sure is how to address is having the client record auto-appear on screen
when the call comes in. I did see plug ins for recording the calls...
Is
asterisk the best solution for this?
thanks
Todd
Begin forwarded message:
From: A. Pathuri [EMAIL PROTECTED]
Date: October 2, 2006 2:51:32 AM EDT
To: Todd Houle [EMAIL PROTECTED]
Subject: Call Center requirements
Todd,
Here is the brief doc you requested.
The process that we need is pretty simple...
We get a bunch of DID (Direct Inward Dialing) numbers from SBC.
As we get a client, we assign them a DID #.
They forward their existing phones to their DID number when their lines
are busy or after hours.
The DID # is programmed into the telephony system so we can program the
caller ID, and enter the appropriate script to pop up when that number
comes through.
When a call comes in, I would like to have all calls automatically
recorded without any of the call agents having to press a record button
for each call.
We also current have conference call functionality where we can connect
one caller to another caller (used when the ER needs to speak to a
doctor).
Ideally also, I would like the recorded calls to sort by client and
store in the appropriate clients folder, which then can be
automatically zipped and sent via email to the clients inbox at any
desired interval.
We are also developing a web-based app where the details of each call
can be entered ( a sort of call log) so the clients can also log into a
web interface and see the details of each call (currently, most clients
get their call logs via fax in the am and at midnight).
It would be great if somehow, the caller ID on the server/astericks can
automatically pull up the appropriate clients profile from our web app,
so the details can be entered into the correct profile. Otherwise, for
each call that comes in, the call agent has to pull up the clients
profile while the caller is on the phone, before s/he can take down the
details of the call.
This is really rough, but I hope it gives the basic idea. We can
discuss in further detail once you take a look at this.
Ofcourse, as well it would be great to be able to setup a co-location in
India utilizing the same infrastructure.
There are a number of ways to do this, but given the application it
appears to be (medical), and additional requirement not mentioned here
(and quite possibly the most important) is HIPPA compliance with
regard to security of who has access to what information.
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Re: [asterisk-users] No Dialtone
Did you set immediate=no in zapata.conf? Francesco Eddie Johnson Jr wrote: Hello, I have the following setup: 1. Ubuntu Dapper Server 6.06 plus latest patches 2. Asterisk 1.2.11 3. libpri 1.2.3 4. Zaptel 1.2.8 5. Digium TDM22 (TDM400P) 6. Analog phone plugged in port 3 7. The wctdm, zaptel modules load at startup, I type asterisk as root and it is activated. 8. I check the Channel Map and I have the following: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. I can ssh into the server and remotely connect to the server. Great! The card is not connected to an outside line as of yet but I have no dialtone on the phone. I spoke with a rep. at digium and was told a dialtone should be there. Zaptel.conf : loadzone=us defaultzone=us fxoks=1,2 fxsks=3,4 Zapata.conf: ;FXS Modules signalling=fxo_ks channel = 1,2 ; ;FXO Modules signalling=fxs_ks channel = 3,4 I made sure the card is not sharing an IRQ, I checked the hard drive and all is well. I load zttool and get the following: cat /proc/zaptel/* Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 Any suggestions? Ed ___ --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] IP Phones
Also http://www.enterux.com/ in Mumbai, India - very, very helpful people, indeed HTH Best wishes Iyer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jose diaz Sent: Friday, October 06, 2006 6:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IP Phones Grandsream IP phone Budge Tone 1001, 102 Softphone X-Lite Ekiga (Ubuntu) Etc Jose Diaz Forrest Beck wrote: http://www.voipsupply.com/home.php On 10/4/06, Devraj Mukherjee [EMAIL PROTECTED] wrote: Nokia E series with proper firmware upgrade :) On 10/5/06, Steve Glaus [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I would like to know where I can find the IP Phones that can be used with Asterisk? Is there any link? Regards Bilal Ghayad Mobile: 00965 9849460 Office: 00965 2623174 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Any phone supporting SIP or IAX are good choices for asterisk. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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 ___ --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] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i
On 10/6/06, Gareth Owen [EMAIL PROTECTED] wrote: Morten,Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system to see what is going on.Can you post the INVITE message that is being rejected? This INVITE results in a 488 from the phone: INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK42f78e77;rportFrom: 1011 sip:[EMAIL PROTECTED];tag=as3a35aa3aTo: sip:[EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70 Date: Fri, 06 Oct 2006 14:22:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 309 v=0o=root 4746 4746 IN IP4 192.168.10.2s=sessionc=IN IP4 192.168.10.2t=0 0m=audio 10066 RTP/AVP 8 0 3 101a=rtpmap:8 PCMA/8000 a=ptime:20a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:3 GSM/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=sendrecv And this INVITE works (only alaw is enabled): INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK3c04692a;rportFrom: 1011 sip:[EMAIL PROTECTED];tag=as39cd0724To: sip:[EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70 Date: Fri, 06 Oct 2006 14:23:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 238 v=0o=root 4762 4762 IN IP4 192.168.10.2s=sessionc=IN IP4 192.168.10.2t=0 0m=audio 10042 RTP/AVP 8 101a=rtpmap:8 PCMA/8000 a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=sendrecv Also, I know we've fixed a number of SDP related issues in 1.4.1, so if you haven't already you might want to try the 1.4.1 beta.Info on how to get the beta is available here:http://groups.google.com/group/Aastra-480i-Users/browse_frm/thread/8f6f0f3419ef396d I will try that and report back here.-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Hangups on PRI Interface
Hi Vicente, I solved my problem and now the PBX I set up can make and receive calls without any problem using Telecom (Italy) PRI lines. My zapata.conf is: context = telco_zap group = 1 immediate = no internationalprefix = 00 language = us nationalprefix = 0 pridialplan = unknown prilocaldialplan = unknown priindication = inband resetinterval = never signalling = pri_cpe switchtype = national usecallerid = yes callerid = asreceived overlapdial=yes relaxdtmf=yes usedistinctiveringdetection=yes channel = 1-15,17-31 for channels connecting Asterisk to telco lines and context = legacy_zap group = 2 immediate = no language = us priindication = inband resetinterval = never signalling = pri_net switchtype = euroisdn callerid = asreceived relaxdtmf=yes overlapdial=yes ; yes is mandatory usedistinctiveringdetection=yes channel = 32-46,48-62 for channels connecting Asterisk to legacy PBX. I noticed that overlap dial set to no couldn't make me call some old pbx. Hope this configuration may help. Giorgio Incantalupo Vicente Aguilar wrote: El jue, 28-09-2006 a las 11:06 +0200, Giorgio Incantalupo escribió: Do you have the same problem? If yes have you tried to call those bad numbers from legacy phones and from SIP/IAX? Haven't tried. We're still using mostly analog phones connected to the legacy PBX, and starting to play around with a couple of VoIP phones. But I've experienced other problems when both analog and SIP phones are involved in the same call (transfering calls from an analog phone to a SIP one usually fails). I'll try as soon as possible. ___ --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] Tutorial - avoiding queue_log file rotation
Hi list, I must be in tutorial writing mode this week, as I have prepared another tutorial on how to avoid queue_log file rotation on AAH/TrixBox and other systems. This is done automatically but it's quite an annoyance because it interferes with queue_log analyzers like QueueMetrics and ends up losing important business data if you run a call-center. The tutorial is here: http://www.astrecipes.net/index.php?n=205 Comments and updates are very well welcome. l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.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
[asterisk-users] defining trunks in sip.conf
I just upgraded an old Asterisk 1.0.xx to 1.2 but there are some changes in the trunk definitions of sip.conf All my trunks stopped working. Is the sintax someting like this? register=200:1000:[EMAIL PROTECTED]:5060/200 this is to user 200 (why do we need to put it 3 times???) with password 1000 and to register in domain.pt I already saw the manuals but the trunks arent still working :( Can someone help me? Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gui sans live cd
Patrick Aljord wrote: is there a good and free asterisk gui that is not tight to a live cd? I like [EMAIL PROTECTED] but it looks like I need to install the livecd. I just want to run asterisk on my debian install. Is there a way to run [EMAIL PROTECTED] on debian? or anything similar? You can install freepbx ([EMAIL PROTECTED]) on any linux box. Wheres the prob?! ___ --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] Tutorial - avoiding queue_log file rotation
Hi list, I must be in tutorial writing mode this week, as I have prepared another tutorial on how to avoid queue_log file rotation on AAH/TrixBox and other systems. This is done automatically but it's quite an annoyance because it interferes with queue_log analyzers like QueueMetrics and ends up losing important business data if you run a call-center. The tutorial is here: http://www.astrecipes.net/index.php?n=205 Comments and updates are very well welcome. l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.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
[asterisk-users] Voicemail and Forwarding
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
I tried.
exten = 2000,1,Voicemail([EMAIL PROTECTED])
this of course gives me a error that mailbox 2000 doesn't exist.
I also tried..
exten = 2000,1,Voicemail(${CALLERID(num)[EMAIL PROTECTED])
This gives the original caller his own mailbox.
Stumpped.
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Re: [asterisk-users] [EMAIL PROTECTED] problems
Thank you for your response. They are all connected to the LAN, and when they, out of the blue, go dead is that they loose their dial tone and so forth. Somethiing need to be changed in the Config, but I am affraid that if I start making changes, I can screw things even worst. Ed ___ --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] [EMAIL PROTECTED] problems
Thanks for your response. No, there;s no firewall and they are all correctly connected to the LAN. They work just fine, and then, one or two days later and out of the blue, they start having problems. Ed ___ --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 and Forwarding
Nevermind. Just decided to use:
exten = _22XXX,1,Voicemail(u${EXTEN:[EMAIL PROTECTED])
On 10/6/06, Forrest Beck [EMAIL PROTECTED] wrote:
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
I tried.
exten = 2000,1,Voicemail([EMAIL PROTECTED])
this of course gives me a error that mailbox 2000 doesn't exist.
I also tried..
exten = 2000,1,Voicemail(${CALLERID(num)[EMAIL PROTECTED])
This gives the original caller his own mailbox.
Stumpped.
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RE: [asterisk-users] No Dialtone
Yes, I have and I received the following: In zapata.conf your first two channels should be fxs_ks because the first two modules are FXO mdoules. Your last two channels should be fxo_ks because the second two modules are FXS modules. For the TDM400P(TDM 22) the FXS modules work with the phone. The 3 port is for the line. So I unplugged it from port 3, and plugged the analog phone in port 1, made the changes to the channels and set immediate=no, restart the server and activated asterisk. Nothing, my friend. Any more suggestions, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Francesconi Sent: Friday, October 06, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Dialtone Did you set immediate=no in zapata.conf? Francesco Eddie Johnson Jr wrote: Hello, I have the following setup: 1. Ubuntu Dapper Server 6.06 plus latest patches 2. Asterisk 1.2.11 3. libpri 1.2.3 4. Zaptel 1.2.8 5. Digium TDM22 (TDM400P) 6. Analog phone plugged in port 3 7. The wctdm, zaptel modules load at startup, I type asterisk as root and it is activated. 8. I check the Channel Map and I have the following: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. I can ssh into the server and remotely connect to the server. Great! The card is not connected to an outside line as of yet but I have no dialtone on the phone. I spoke with a rep. at digium and was told a dialtone should be there. Zaptel.conf : loadzone=us defaultzone=us fxoks=1,2 fxsks=3,4 Zapata.conf: ;FXS Modules signalling=fxo_ks channel = 1,2 ; ;FXO Modules signalling=fxs_ks channel = 3,4 I made sure the card is not sharing an IRQ, I checked the hard drive and all is well. I load zttool and get the following: cat /proc/zaptel/* Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 Any suggestions? Ed ___ --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] No Dialtone
Yes, I did. Still nothing. Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Francesconi Sent: Friday, October 06, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Dialtone Did you set immediate=no in zapata.conf? Francesco Eddie Johnson Jr wrote: Hello, I have the following setup: 1. Ubuntu Dapper Server 6.06 plus latest patches 2. Asterisk 1.2.11 3. libpri 1.2.3 4. Zaptel 1.2.8 5. Digium TDM22 (TDM400P) 6. Analog phone plugged in port 3 7. The wctdm, zaptel modules load at startup, I type asterisk as root and it is activated. 8. I check the Channel Map and I have the following: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. I can ssh into the server and remotely connect to the server. Great! The card is not connected to an outside line as of yet but I have no dialtone on the phone. I spoke with a rep. at digium and was told a dialtone should be there. Zaptel.conf : loadzone=us defaultzone=us fxoks=1,2 fxsks=3,4 Zapata.conf: ;FXS Modules signalling=fxo_ks channel = 1,2 ; ;FXO Modules signalling=fxs_ks channel = 3,4 I made sure the card is not sharing an IRQ, I checked the hard drive and all is well. I load zttool and get the following: cat /proc/zaptel/* Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 Any suggestions? Ed ___ --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] Voicemail and Forwarding
Hi Forrest -
I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
exten = 2000,1,Voicemail([EMAIL PROTECTED])
this of course gives me a error that mailbox 2000 doesn't exist.
exten = 2000,1,Voicemail(${CALLERID(num)[EMAIL PROTECTED])
This gives the original caller his own mailbox.
Why don't you try it a little differently. If a user wants to forward
all his/her calls to voicemail, you can give each user get a separate
voicemail extension. Maybe just add a '2' in front of the user's
normal extension number. So, if you have three digit extensions, you
can do that like this:
exten = _2XXX,1,Voicemail(u${EXTEN:[EMAIL PROTECTED])
What phones are you using, BTW? Many SIP phones have a Do Not
Disturb feature. Just press the DND button, and if the dialplan is
set up correctly, all calls to that extension will go directly to
voicemail.
- Noah
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Re: [asterisk-users] No Dialtone
Any more suggestions, Call Digium. They will get you to the point where the hardware will work. If it won't work (and there's nothing wrong with your system), they should exchange for a unit that will work. ___ --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] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i
Have you ever tried allow=alawulaw in the same line? just a tip...On 10/6/06, Morten Isaksen [EMAIL PROTECTED] wrote: On 10/6/06, Gareth Owen [EMAIL PROTECTED] wrote: Morten,Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system to see what is going on.Can you post the INVITE message that is being rejected? This INVITE results in a 488 from the phone: INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK42f78e77;rportFrom: 1011 sip:[EMAIL PROTECTED];tag=as3a35aa3aTo: sip:[EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70 Date: Fri, 06 Oct 2006 14:22:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 309 v=0o=root 4746 4746 IN IP4 192.168.10.2s=sessionc=IN IP4 192.168.10.2t=0 0m=audio 10066 RTP/AVP 8 0 3 101a=rtpmap:8 PCMA/8000 a=ptime:20a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:3 GSM/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=sendrecv And this INVITE works (only alaw is enabled): INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK3c04692a;rportFrom: 1011 sip:[EMAIL PROTECTED];tag=as39cd0724To: sip:[EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70 Date: Fri, 06 Oct 2006 14:23:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 238 v=0o=root 4762 4762 IN IP4 192.168.10.2s=sessionc=IN IP4 192.168.10.2t=0 0m=audio 10042 RTP/AVP 8 101a=rtpmap:8 PCMA/8000 a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=sendrecv Also, I know we've fixed a number of SDP related issues in 1.4.1, so if you haven't already you might want to try the 1.4.1 beta.Info on how to get the beta is available here: http://groups.google.com/group/Aastra-480i-Users/browse_frm/thread/8f6f0f3419ef396d I will try that and report back here.-- Morten Isaksen http://www.misak.dk/blog/ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --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] [EMAIL PROTECTED] problems
Ed,Do the phones lose their registration? If you run sip show peers when the phones are not working, do they show as being registered or not?AlexOn 10/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks for your response. No, there;s no firewall and they are all correctly connected to the LAN. They work just fine, and then, one or two days later and out of the blue, they start having problems. Ed ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom 360: how to make record button working ?
Remco Barendse wrote: I cannot find this option in the snom firmware, the only thing I found is DTMF via SIP INFO: This sounds nice but I guess it will break stuff if you need DTMF tones to get through the menu of a remote PBX. I'm pretty sure that when you AREN'T sending the DTMF inband, asterisk detects it, and if the keys pressed don't lead to any recording/transfer features, then it re-creates DTMF on the bridged channel. I mean to say, my called party can't hear me start recording or transfer them, but I don't have any trouble with outside IVRs. Ideally * would need to interpret the SIP INFO message from the Snom as start recording. I looked at the patch someone mentioned earlier but to me this looks like re-inventing the wheel by starting the whole recording stuff all over again. All this is not necessary, * should simply treat the SIP INFO message the same as DTMF dialling *1 On Thu, 5 Oct 2006, Mojo with Horan Company, LLC wrote: We use SIP Polycom 501s, and their dtmfmode=rfc2833. The remote party can NOT hear the tones when you start recording. I suspect that if dtmfmode=inband, they WOULD be able to. Could be wrong here, that's just my current rudimentary understanding of the situation :) Moj Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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 ___ --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:500,45261e8d254852002735277! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to forward DID to another Server
That's not a forward condition. As far as I know, you can't forward calls between Asterisk servers. A forward must complete on the Asterisk server the original call was serviced by. Doug. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Friday, October 06, 2006 7:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to forward DID to another Server Hi Ram - so i want to forward some of the DID from my asterisks to other Server how can i do that, and i need to give them access to calling out also You need to connect your asterisk machine together. The most common ways to do this are either with IAX or SIP. To do this with IAX, you might want to read this: http://astrecipes.net/index.php?n=204 After you have your asterisk servers connected, you can direct calls from one server to the other like this: Dial(IAX2/peer name/exten on remote server) - Noah ___ --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: AW: [asterisk-users] PoE IP Phone
On Oct 6, 2006, at 12:07 AM, Christian Stredicke wrote: Here comes the advertisement for snom phones: http://www.snom.com. CS And here the one for cisco phones: http://www.cisco.com/en/US/ products/sw/voicesw/products_category_buyers_guide.html#number_1 -- Michiel ___ --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: Getting Asterisk to work with GoogleTalk
I have followed this configuration to the letter but still no joy. Do I have to load some modules at start up like gtalk.so or jabber.so? Should my user show up as available on other peoples buddy list after asterisk starts? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bromont - Sent: Thursday, October 05, 2006 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE: Getting Asterisk to work with GoogleTalk It should work fine with 1.4Beta2 I use gtalk.conf instead of jingle.conf and this is what I would change in configurations (shown with the arrows): jabber.conf: [general] ;debug=yes ;autoprune=yes ;autoregister=yes [googletalk] type=client serverhost=talk.google.com [EMAIL PROTECTED]/Talk -- secret=gtpass port=5222 usetls=yes usesasl=yes [EMAIL PROTECTED] statusmessage=Voice Calls Only timeout=100 gtalk.conf: [general] context=from-gtalk allowguest=yes [guest] disallow=all allow=ulaw context=from-gtalk [google] [EMAIL PROTECTED]-- disallow=all allow=ulaw context=from-gtalk connection=googletalk -- extensions.conf: ;outgoing to GoogleTalk [to-gtalk] exten = 190,1,NoOp(Calling GoogleTalk user [EMAIL PROTECTED]) exten = 190,n,Dial(gtalk/googletalk/[EMAIL PROTECTED])-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 RT on Disk On Module Performance and Durability
Hi, Im doing some research for Disk on a Module (DOM)with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting. Does someone have working experience with this? Basically the Asterisk Realtime will be stored in MySQL and the DB will be stored in a Disk on a Module. I have read that the usual standard is 2,000,000 MTBF and 2,000,000 Read/Write Cycles. Is there an utility/section/procedure that can count/display the reads and writes a normal Linux system does? That result can be extrapolated to understand, in terms of days/week/months how much time a Disk on Module will last. Anyone with field experience? Thanks, -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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] Problem with 2 machines connected with IAX
Upgraded to 1.2.12.1 and the problem went away.. must have been an IAX bug. On 10/5/06, Matt [EMAIL PROTECTED] wrote: Interesting.. this is almost the same issue I'm having, except that I am sip before iax.. where as this guy is iax then sip. http://groups.google.com/group/Asterisk-users/browse_thread/thread/4fdc15e2acdb349a/e78466f6b1a5006b?lnk=stq=asterisk+%22one+way+audio%22+holdrnum=2hl=en#e78466f6b1a5006b On 10/5/06, Matt [EMAIL PROTECTED] wrote: I'm sorry, I don't understand the question. How is it? To add more information to the puzzle... if a call comes in: zapinterface--B--iax--C the customer can place the caller on local hold and there is no issue! On 10/5/06, Lenz [EMAIL PROTECTED] wrote: Hmmm... how is your IAX conf between the two boxes B and C? l. On Thu, 05 Oct 2006 20:55:51 +0200, Matt [EMAIL PROTECTED] wrote: Hi, I am purchasing minutes (800) from provider a (from now on A). My server is B, and my customer is C. When an 800 call comes in it goes: A---sip--B--iax--C and it sounds fine. If the customer at location C puts the caller on hold (local phone hold), when they pick the caller back up the caller can hear customer, but the customer can not hear the caller. If the customer at location C puts the caller on park (70), when they pick the caller back up everyone can hear everyone. Any thoughts? -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pop a web page with DID in url
Yes I would be interested in testing out your product. Does anyone have any other recommendations. A softphone would work for me. I would like something that had a chat feature like eyebeam does. I found another product called SNAP that will pop a web page, but it can only pass cid info not did. This is for an inbound call center project. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Time Bandit wrote: I'm looking to do this. When a call comes in to an agent in a queue, pop a web page like this http://www.mydomain.com/cgi-bin/script.cgi?did=952900 Where did is the number the caller dialed to reach the system in the first place. I know Hudlite can do this we caller ID, but the DID feature is not there yet. Does anyone have any other software they know of that can do this? Some softphones support handling URL when you pickup the call. You can set that URL to anything you want from the dialplan. shameless-plug My MediaX softphone (current beta version) support it. Let me know if you want to try it /shameless-plug 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 ___ --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] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
Thanks for the reply...zapta.comf[channels]group = 1language=encontext=incomingsignalling=fxs_ksswitchtype=nationalusecallerid=yeshidecallerid=nocallwaiting=yesmusiconhold=defaultusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechotraining=yesechocancelwhenbridged=yesrxgain=4txgain=-4channel = 1-4 original Post: Below is the text of my original post. I am not sure what Codec we are using. The "Codec Preferences" phone setting shows, in order of preference, G.711u, G.711A, G.729AB We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core 4-2.6.14-1.1656_FC4smp. It is installed on a Dell PE 2500 with 2x900 MHz processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium TDM400P card which is connected to 4 POTS lines. The server is also connected to a 100MB switched LAN where we have about 20 Polycom 501 phones with the latest firmware updates. Nothing else runs on the server except an ftp daemon which is never used except when a phone reboots.For about 20% of the calls to the outside world, the voice on the other end of an outside line is incredibly choppy. Enough to where we have to hang up and call on a cell phone. It is always the same numbers that are choppy. The funny thing is, if I press mute while talking on a choppy call, the choppiness goes away completely. I have tried: turning off ACPI, turning off APCI, moving the card to another PCI slot, changing the RX/TX gains. There are no shared IRQs. I have tested the lines by unplugging them from the asterisk server and plugging them directly into an analogue phone. Using "cat /proc/interrupts; sleep 10 ; cat /proc/interrupts" I see that there are about 1,000 interrupts per seconds between the card and the CPU. I do not think it is a network congestion problem as intra-office communications as well as voicemail retrieval are always perfect. The Voip does not go over any routers, just a max of 2 switches with a 1GB trunk. This happens even off-hours when the network isnt being used at all. There are never more than 2 people on the phone at the same time and it is definitely not an over-utilized processor. I have trying to figure this out for 2 months on and off with no success any help is appreciated. ThanksNoah Miller [EMAIL PROTECTED] wrote: Well I am using GSM as my main codec which seems to be very nice Polycom phones do not support GSM (GSM would not be necessary hereanyway, since all these phones are on a local LAN, so bandwidth doesnot need to be conserved). You can also change some settings in the zapta and zaptel config.. to reduce echo and interference on the line..This is the most important thing here - what does your zapata.conf look like?Other things:1. Update asterisk to a newer version. There have been MANY bugs thathave been fixed since 1.2.4.2. Update zaptel to a newer version. Not much has changed for the TDMcards since 1.2.7, but you should update anyway.- Noah___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Match Chat Author?
I stumble on this URL that is Chat Line script written by Steven L. Edwards called 'Match Chat' here: http://bugs.digium.com/file_download.php?file_id=11080type=bug But I can't seem to find any additional info on Author or Applications - I was wondering if you might know more about either? Bart ___ --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 MWI
I'd like to know if anyone has a suggested fix for this... You have a 'cluster' of Asterisk servers that use DUNDi etc for registration redundancy, finding other phones etc. You have a separate Asterisk box for voicemail. For voicemail deposit/retrieval you trunk the call over to the voicemail server. This all works fine. No issues there. What about MWI though? Your phones register with the cluster, not with the voicemail server, and therefore the voicemail server has no knowledge of where the phones are and therefore cannot send out SIP NOTIFY messages to phones. This is a general architectural problem with Asterisk. Has anyone solved it? Are the developers working on fixing problems like this for 1.6? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability
Erick Perez wrote: Hi, Im doing some research for Disk on a Module (DOM) with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting. Does someone have working experience with this? Basically the Asterisk Realtime will be stored in MySQL and the DB will be stored in a Disk on a Module. There are better ways than RealTime to configure asterisk, but that is a religious war, so I won't discuss it. I have read that the usual standard is 2,000,000 MTBF and 2,000,000 Read/Write Cycles. From my experience it is only writes that matter. Is there an utility/section/procedure that can count/display the reads and writes a normal Linux system does? That result can be extrapolated to understand, in terms of days/week/months how much time a Disk on Module will last. Anyone with field experience? Setup the system to mount a Ramdisk for the various standard locations (/var and /tmp and /). Symlink the standard files from a ROM partition to your mounted ramdisk / (root) partition. Then only write to the flash when absolutely necessary, like for system updates. Usually I charge for this kind of info, so consider yourself lucky since I am in a good mood today. (oddly enough) Jeremy McNamara ___ --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: verbose logging to file in 1.4
On Fri, Oct 06, 2006 at 02:42:41PM +0200, Benko wrote: 2006/10/4, Benko [EMAIL PROTECTED]: Hello! How can i change the verbose logging level to a file in 1.4? In 1.2 i was used to set the verbose level via asterisk -Rx 'set verbose 5' but in 1.4 it is always reset to OFF again, so (nearly) nothing is logged to /var/lib/asterisk/verbose: Try asterisk -Rx 'core verbose 5' ___ --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: failed registration
what about the interval of the registration? is 2 minutes too often? Dovid B [EMAIL PROTECTED] wrote: Timed out from what I have seen comes from either a poor internet connection or a problem with your ITSP.- Original Message - From: stan ford To: asterisk-users@lists.digium.com Sent: Friday, October 06, 2006 4:42 AM Subject: [asterisk-users] Re: failed registration stan ford [EMAIL PROTECTED] wrote: i have this issue with failed registrations with my sip provider. it doesn't happen often, but it does happen. it also happens with 2 different vsp providers, so i dont think its them. this happens maybe 8 times a day, but then that doesn't sound too bad considering it registers itself every 2 minutes. im using trixbox 1.1 and have grandsream 101 SIP phones. thanks alot.Oct 4 06:12:02 NOTICE[2831] chan_sip.c: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3)Oct 4 06:12:19 NOTICE[2831] chan_sip.c: -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #4) Stay in the know. Pulse on the new Yahoo.com. Check it out. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail. Get your email and more, right on the new 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] Asterisk RT on Disk On Module Performance and Durability
Jeremy McNamara wrote: Erick Perez wrote: Hi, Im doing some research for Disk on a Module (DOM) with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting. Does someone have working experience with this? Basically the Asterisk Realtime will be stored in MySQL and the DB will be stored in a Disk on a Module. There are better ways than RealTime to configure asterisk, but that is a religious war, so I won't discuss it. I have read that the usual standard is 2,000,000 MTBF and 2,000,000 Read/Write Cycles. From my experience it is only writes that matter. Is there an utility/section/procedure that can count/display the reads and writes a normal Linux system does? That result can be extrapolated to understand, in terms of days/week/months how much time a Disk on Module will last. Anyone with field experience? Setup the system to mount a Ramdisk for the various standard locations (/var and /tmp and /). Symlink the standard files from a ROM partition to your mounted ramdisk / (root) partition. Then only write to the flash when absolutely necessary, like for system updates. Usually I charge for this kind of info, so consider yourself lucky since I am in a good mood today. (oddly enough) Jeremy McNamara Erick, Or Just use AstLinux which kind of does what Jeremy described :) http://www.astlinux.org P.S. - I am the creator of AstLinux -- 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] Voicemail MWI
For us the voicemail server doesn't have to know what phones are registered where. We have an externnotify script that drops x number of msgx.txt files into the respective voicemail folders after any call that goes through the voicemail server. x in this would be the number of messages. Since our servers are all exactly the same, we just blanket the script across all our call servers, but if you have specific servers where the phones are registered at, you can modify the SIP channel to update the db with server information and have the voicemail server just run the script across servers that that particular phone is currently registered with. The scripts we use are relatively lightweight, but can probably be turned into some sort of listening service to remove some of the ssh overhead required. Not a problem for us, but you might not like it much :) On Fri, 2006-10-06 at 11:49 -0600, Douglas Garstang wrote: I'd like to know if anyone has a suggested fix for this... You have a 'cluster' of Asterisk servers that use DUNDi etc for registration redundancy, finding other phones etc. You have a separate Asterisk box for voicemail. For voicemail deposit/retrieval you trunk the call over to the voicemail server. This all works fine. No issues there. What about MWI though? Your phones register with the cluster, not with the voicemail server, and therefore the voicemail server has no knowledge of where the phones are and therefore cannot send out SIP NOTIFY messages to phones. This is a general architectural problem with Asterisk. Has anyone solved it? Are the developers working on fixing problems like this for 1.6? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Server : IDE HDD frequent crash
On 2006-10-06 06:31:48 -0700, Jay R. Ashworth [EMAIL PROTECTED] said: On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote: Heat = #1 cause of disk failure. If they are roasting to the touch they will fail in 2-3 months. One word: smartd. I didn't know it existed, and I'm amazed I didn't. Everyone on this list should be running smartd, and know what it's saying. SMART is useful, but not the be all and end all of disk drive care. Proper ventilation as already mentioned, is much more important then SMART status in my opinion... I have seen many drives that fail, while still reporting that everything is hunk dory as far as SMART is concerned. Still, you make a good point. 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
[asterisk-users] Asterisk Postgres Native support
Hello, I am trying to use Asterisk pull its configuration (Sip.conf, Extension.conf) from the Postgresql (ARA - Realtime). I am missing documentation regarding setting this up (connectivity portion) For example, in file extconfig.conf I need to add: sipusers = pgsql,asterisk,account sippeers = pgsql,asterisk,account extensions = pgsql,asterisk,extension I am aware that for MySql, you add following lines to extconfig.conf sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies voicemail = mysql,asterisk,voicemail_users extensions = mysql,asterisk,extensions Also you add the following to res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = asteriskuser dbpass = password dbport = 3306 and few more changes in sip.conf and extensions.conf makes it work for MySQL. BUT how this is done to connect to PGSQL. It says Native, I assume I do not need to recompile asterisk. Any help appreciated. Also please clarify that if Asterisk 1.2 has Native Support for Postgres. If so, that is even better. /doc/realtime.txt from Version 1.4 Beta Currently there are three realtime database drivers: * ODBC: Support for UnixODBC, integrated into Asterisk The UnixODBC subsystem supports many different databases, please check www.unixodbc.org for more information. * MySQL: Found in the asterisk-addons subversion repository on svn.digium.com * PostgreSQL: Native support for Postgres, integrated into Asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability
Kristian Kielhofner wrote: Erick, Or Just use AstLinux which kind of does what Jeremy described :) http://www.astlinux.org P.S. - I am the creator of AstLinux -- Kristian Kielhofner Sorry to reply to my own post, but there seems to have been some confusion in what I said here. To completely clear it up, Astlinux only writes to flash in these circumstances: 1) You update the configs. 2) You update AstLinux. 3) You are using voicemail and people leave voicemail. (most flash seems to last long enough given typical voicemail usage patterns) 4) If you have the PERSISTLOG option enabled, I will save syslogs to flash (not RAM - the default). Users are warned about this, and it is not the default. 5) astdb is stored in flash, so depending on your needs, SIP registrations and/or dundi keys may get written here periodically. I might make an option similar to PERSISTLOG to disable this. Also, you have the option of using a hard drive or alternate flash device for ALL writes. Boot from flash, run from HD. Do whatever works best for you and your application. -- 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] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
You can also change some settings in the zapta and zaptel
config.. to reduce
echo and interference on the line..
This is the most important thing here - what does your zapata.conf look
like?
zapta.comf
switchtype=national
This is not necessary in your case. It pertains to PRI lines, and not
the POTS lines you have.
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
You may want to turn each of these off, in turn, for testing,
especially the echocancewhenbridged.
You can also tune the echocancel setting in terms of taps (a tap is
one sample from the data stream per second). You can use the values:
16, 32, 64, 128, or 256 ('yes' just means 128).
- Noah
___
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[asterisk-users] AGI() in 1.2 and 1.4
I was experimenting with FastAGI in Asterisk 1.4 and wrote some code around it. I was using the AGISTATUS variable to determine if I had been able to connect to the fast agi server, and act accordingly. 1.2 appears to be different. It has no such AGISTATUS variable, but more importantly, it appears that if you fail to connect to your FastAGI server, all dial plan processing just stops dead. Is there a way around this? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AGI() in 1.2 and 1.4
There is a patch that allows a jump to N + 101. Thanks, Steve -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Friday, October 06, 2006 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AGI() in 1.2 and 1.4 I was experimenting with FastAGI in Asterisk 1.4 and wrote some code around it. I was using the AGISTATUS variable to determine if I had been able to connect to the fast agi server, and act accordingly. 1.2 appears to be different. It has no such AGISTATUS variable, but more importantly, it appears that if you fail to connect to your FastAGI server, all dial plan processing just stops dead. Is there a way around this? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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: Where is the PlayDTMF command?
JdT == Jan du Toit [EMAIL PROTECTED] writes: JdT PS: This reply will probably go under a new thread with the same JdT subject. I receive the digest mode of the mails on this list, and JdT replying to it breaks the thread. How can I avoid this in the JdT future? Thanks. Switch to a newsreader and use gmane.org... /Benny ___ --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] swap CID with DID
Does anyone have a way to send the DID in place of the CID number. I want pop a web page with the DID in the URL but all the software I have seen only supports putting the CID info in the URL. If I could swap the two I could just use the programs as is. The two programs I have looked at so far are SNAP and HUDlite. They both pop based on CID. SNAP works very well for that. Unless anyone knows of any software that can connect to asterisk, in any method, and pop a web page when a call comes in and pass the DID into the URL. -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --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] ChanIsAvail() in 1.2.12.1
Is there something wrong with the chanisavail() application in 1.2.12.1?
My dialplan has:
[syst_Route]
exten = _[*0123456789].,1,NoOp(*** Originated call ${CALLERID} - ${EXTEN})
exten = _[*0123456789].,n,NoOp(FOO1)
exten = _[*0123456789].,n,ChanIsAvail(SIP/${EXTEN})
exten = _[*0123456789].,n,NoOp(FOO2)
and the console is displaying...
*CLI -- Executing NoOp(SIP/3254101-0817a220, *** Originated call
Chocolate Chip 3254101 - 3254103) in new stack
-- Executing NoOp(SIP/3254101-0817a220, FOO1) in new stack
-- Executing ChanIsAvail(SIP/3254101-0817a220, SIP/3254103) in new stack
It never makes it past the call to ChanIsAvail(). Dialplan processing just
completely stops at this point.
What's up with that???
Doug.
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Re: [asterisk-users] swap CID with DID
Michael,You should be able to just do this: Set(CALLERID(num)=${DNID})... Though the VoIP-Info page is very vague about the DNID variable. You might try it out though.Best of luck!Alex
On 10/6/06, Michael Sampson [EMAIL PROTECTED] wrote:
Does anyone have a way to send the DID in place of the CID number. Iwant pop a web page with the DID in the URL but all the software I haveseen only supports putting the CID info in the URL. If I could swap the
two I could just use the programs as is. The two programs I have lookedat so far are SNAP and HUDlite. They both pop based on CID. SNAP worksvery well for that.Unless anyone knows of any software that can connect to asterisk, in any
method, and pop a web page when a call comes in and pass the DID intothe URL.--Michael SampsonInformation Systems ManagerCustomer Contact Services[EMAIL PROTECTED]
952-936-4000___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] ChanIsAvail() in 1.2.12.1
from http://www.asteriskguru.com/tutorials/chanisavail.html
If there is no available channel the ChanIsAvail application will
continue with the execution of the extension with priority n+101
Douglas Garstang wrote:
Is there something wrong with the chanisavail() application in 1.2.12.1?
My dialplan has:
[syst_Route]
exten = _[*0123456789].,1,NoOp(*** Originated call ${CALLERID} - ${EXTEN})
exten = _[*0123456789].,n,NoOp(FOO1)
exten = _[*0123456789].,n,ChanIsAvail(SIP/${EXTEN})
exten = _[*0123456789].,n,NoOp(FOO2)
and the console is displaying...
*CLI -- Executing NoOp(SIP/3254101-0817a220, *** Originated call Chocolate Chip
3254101 - 3254103) in new stack
-- Executing NoOp(SIP/3254101-0817a220, FOO1) in new stack
-- Executing ChanIsAvail(SIP/3254101-0817a220, SIP/3254103) in new stack
It never makes it past the call to ChanIsAvail(). Dialplan processing just
completely stops at this point.
What's up with that???
Doug.
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RE: [asterisk-users] ChanIsAvail() in 1.2.12.1
That's not how it appears to have worked before. Previously, I was able to call
it and then simply check the value of the ${AVAILCHAN} variable at n+1. The
docs imply that jumping to n+101 only occurs if j is supplied, and I'm not
passing a 'j'.
*CLI show application chanisavail
-= Info about application 'ChanIsAvail' =-
[Synopsis]
Check channel availability
[Description]
ChanIsAvail(Technology/resource[Technology2/resource2...][|options]):
This application will check to see if any of the specified channels are
available. The following variables will be set by this application:
${AVAILCHAN} - the name of the available channel, if one exists
${AVAILORIGCHAN} - the canonical channel name that was used to create the
channel
${AVAILSTATUS} - the status code for the available channel
Options:
s - Consider the channel unavailable if the channel is in use at all
j - Support jumping to priority n+101 if no channel is available
Doug.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Friday, October 06, 2006 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanIsAvail() in 1.2.12.1
from http://www.asteriskguru.com/tutorials/chanisavail.html
If there is no available channel the ChanIsAvail application will
continue with the execution of the extension with priority n+101
Douglas Garstang wrote:
Is there something wrong with the chanisavail() application
in 1.2.12.1?
My dialplan has:
[syst_Route]
exten = _[*0123456789].,1,NoOp(*** Originated call
${CALLERID} - ${EXTEN})
exten = _[*0123456789].,n,NoOp(FOO1)
exten = _[*0123456789].,n,ChanIsAvail(SIP/${EXTEN})
exten = _[*0123456789].,n,NoOp(FOO2)
and the console is displaying...
*CLI -- Executing NoOp(SIP/3254101-0817a220, ***
Originated call Chocolate Chip 3254101 - 3254103) in new stack
-- Executing NoOp(SIP/3254101-0817a220, FOO1) in new stack
-- Executing ChanIsAvail(SIP/3254101-0817a220,
SIP/3254103) in new stack
It never makes it past the call to ChanIsAvail(). Dialplan
processing just completely stops at this point.
What's up with that???
Doug.
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RE: [asterisk-users] Does a HST Saphir III ML PCI work with Asterisk?
James Harper schrieb: I tried one of these and pretty much got it working under visdn. If you do decide to try one, make sure you get the HFC version. Earlier ones used another chipset and definitely weren't supported using open sourced drivers. Please post back if you do get one and get it going though. Hi James, the card is working. I bought one over ebay. It is important to know, that the driver cannot be downloaded Linux Kernel 2.6.x - you have to ask for it at support AT hstnet.de ! :-/ I forgot there was a closed source binary driver for it :) I was trying to get the zaptel hfc driver working for it, and did get it going to the point where I could make a call, but only had it on loan and ran out of time... or maybe I'm thinking of the vISDN driver... Thanks for the update James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Codes negotiation problems betweenAsterisk1.4beta2 and Aastra 480i
The bad news is that the 1.4.1 beta firmware won't help solve your problem, the problem is being caused by the multiple ptime directives in the INVITE message. According to RFC2327 ptime is a media-level description and hence applies to all the codecs in the m=audio line, thus it is only valid to have one of these per stream. Because of this the phones parser is rejecting the SDP as being invalid and thus sending back a 488. I believe this new functionality has been added by the RTP Packetization work in 1.4 (see http://bugs.digium.com/view.php?id=5162) I'm going to raise a bug against asterisk on this, but at the same time I'll try and find a workaround on the phone-side. Gareth -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Morten Isaksen Sent: 06 October, 2006 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Codes negotiation problems betweenAsterisk1.4beta2 and Aastra 480i On 10/6/06, Gareth Owen [EMAIL PROTECTED] wrote: Morten, Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system to see what is going on. Can you post the INVITE message that is being rejected? This INVITE results in a 488 from the phone: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK42f78e77;rport From: 1011 sip:[EMAIL PROTECTED];tag=as3a35aa3a To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Oct 2006 14:22:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 309 v=0 o=root 4746 4746 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 10066 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv And this INVITE works (only alaw is enabled): INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK3c04692a;rport From: 1011 sip:[EMAIL PROTECTED];tag=as39cd0724 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 06 Oct 2006 14:23:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 4762 4762 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 10042 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv Also, I know we've fixed a number of SDP related issues in 1.4.1, so if you haven't already you might want to try the 1.4.1 beta. Info on how to get the beta is available here: http://groups.google.com/group/Aastra-480i-Users/browse_frm/thread/8f6f0f3419ef396d I will try that and report back here. -- Morten Isaksen http://www.misak.dk/blog/ ___ --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] commercial asterisk
anyone have experience with IntuitiveVoice's Asterisk system? Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] regexten regcontext broken for SIP?
Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, show dialplan xxx reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no SIP client is registered I get a CLI warning that my standard context tries to include an empty context - go figure... http://bugs.digium.com/view.php?id=7144 So, do I need to file a bug report, or is it working OK for others? Cheers, Philipp P.S.: Of course I am aware of this Wiki page: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext ___ --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 access Postgres for Realtime Configuration
Hello Comunity, How can I get Asterisk realtime working with Postgres? (without ODBC)? Thanks John /doc/realtime.txt in Version 1.4 Beta2 Currently there are three realtime database drivers: * ODBC: Support for UnixODBC, integrated into Asterisk The UnixODBC subsystem supports many different databases, please check www.unixodbc.org for more information. * MySQL: Found in the asterisk-addons subversion repository on svn.digium.com * PostgreSQL: Native support for Postgres, integrated into Asterisk ___ --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] A Call centre module on Asterisk
Hi, I'm a novice in asterisk. I'm just want to know if we can develop a Call centre application on an asterisk ? And if ok, have you some url link to help me or simple a open source application doing the job ? Thank you a lot. Imed Découvrez un nouveau moyen de poser toutes vos questions quel que soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. Cliquez ici. ___ --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] HTTP Connection Closed on 7960 SIP
Anyone know why I get HTTP Connection Closed on the display of a 7960 running a SIP image? Only seems to happen when registering against my Asterisk box from the WAN. I have 1:1 NAT happening on my firewall. Phones function perfectly otherwise. TFTP working fine across the firewall as well. Odd! Thanks in advance. -Rob. ___ --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] HTTP Connection Closed on 7960 SIP
This happens if you have a logo_url configured for your phone and the phone can't access it. I'm guessing you don't allow 80 through the firewall to the server that's serving the image. -- Aaron Daniel On Fri, October 6, 2006 20:13, Robert Goodyear wrote: Anyone know why I get HTTP Connection Closed on the display of a 7960 running a SIP image? Only seems to happen when registering against my Asterisk box from the WAN. I have 1:1 NAT happening on my firewall. Phones function perfectly otherwise. TFTP working fine across the firewall as well. Odd! Thanks in advance. -Rob. ___ --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] Options for moving to * friendly Business VSP
Hi all, I have a client whose business is currently running on [EMAIL PROTECTED] 2.6 with Cablevision' s (CV) Optimum Voice (OV) and 3 lines. There are going to be 4 additional trunks needed and I'd like to move/migrate them off of OV, to a better more flexible/open/supportive VSP. OV does not share SIP credentials and operates a closed system which required the use of digium tdm-400b card in order to get the trunks into * and limits what we can achieve. There are two parts to this plan. Here are some of the requirements for the first part. The current 3 lines are setup as a hunt group so there's only one published number. My client needs to (at least for the time being) retain that phone number and CV does NOT allow number's in exchange blocks they "own" to be ported out. Due to this fact, I was pondering keeping one of the OV trunks open (the main number from the hunt group), and set it to forward all calls to the new hunt group number on the new VSP. I'm not sure how something like this would function but my concern would be how the "hand-off" on the forward would behave. For example, can this scenario handle multiple incoming calls simultaneously or would one call be dumped off into OV's voicemail system? Also, once a call is forwarded to the new number, is the original OV trunk freed up to accept/forward more incoming calls? or is it tied to that call? Part two. Another business is merging in, bringing with it 4 lines of their own, one of which is an 800 TF number, all currently configured via Verizon POTS serivce. Ideally, I'd like to get those 4 trunks ported to a VSP also, keeping the TF 800 number and perhaps one of the "normal" phone numbers. The rquirement here is that the LNP be done atomically, without downtime (over a weekend for example). I don't know whether or not this is possible. Thanks in advance for any ideas, suggestions and advice you all provide. How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] Options for moving to * friendly Business VSP
previuos post mangled. Hi all, I have a client whose business is currently running on [EMAIL PROTECTED] 2.6 with Cablevision' s (CV) Optimum Voice (OV) and 3 lines. There are going to be 4 additional trunks needed and I'd like to move/migrate them off of OV, to a better more flexible/open/supportive VSP. OV does not share SIP credentials and operates a closed system which required the use of digium tdm-400b card in order to get the trunks into * and limits what we can achieve. There are two parts to this plan. Here are some of the requirements for the first part. The current 3 lines are setup as a hunt group so there's only one published number. My client needs to (at least for the time being) retain that phone number (business continuity) and CV does NOT allow number's in exchange blocks they own to be ported out. Due to this fact, I was pondering keeping one of the OV trunks open (the main number from the hunt group), and set it to forward all calls to the new hunt group number on the new VSP. This would be done until such time as the majority of customers are updated with the new phone number. I'm not sure how something like this would function but my concern would be how the hand-off on the forward would behave. For example, can this scenario handle multiple incoming calls simultaneously or would one call be dumped off into OV's voicemail system? Also, once a call is forwarded to the new number, is the original OV trunk freed up to accept/forward more incoming calls? or is it tied to that call? Part two. Another business is merging in, bringing with it 4 lines of their own, one of which is an 800 TF number, all currently configured via Verizon POTS serivce. Ideally, I'd like to get those 4 trunks ported to the same VSP also, keeping the TF 800 number and perhaps one of the normal phone numbers. The requirement here is that the LNP be done atomically, without downtime (over a weekend for example). I don't know whether or not this is possible. So overall, I'm trying to figure out what some of my options in achieving my goals here may be. I need to know which reliable, quality Business class VSP's can fit the bill. Preferably one that can handle hunt groups, or multiple channels so that more trunks can be simply added as the business grows. Thanks in advance for any ideas, suggestions and advice you all can provide. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Call centre module on Asterisk
Yes, you can easily use asterisk for a call center, start looking here http://www.voip-info.org/wiki/view/Asterisk+call+queues M Imed Imed wrote: Hi, I'm a novice in asterisk. I'm just want to know if we can develop a Call centre application on an asterisk ? ___ --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] astcc help-pleasssssseeee
Hi, I am wondering ifastcc has ever worked for someone because it always return 0 for answeredtime! I tracked every bit of informaion on google and wiki and finally found out that its because of asterisk returning to dial plan after executing Dial, so astcc.agi runs through the end without wating for call completion. Am I missing something crazy? please someone give me a hint. Thanks alot! ___ --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
