[Asterisk-Users] PCMCIA
Does anyone know of a PCMCIA FXO card or even a USB one? I'm looking at building an appliance out of a machine that has USB and PCMCIA but no PCI. Chris Tooley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 200 MWI Button
Please take a look at http://www.ietf.org/internet-drafts/draft-ietf-sipping-mwi-04.txt. The snom phone tries to use the Message-Account line, if its present; otherwise it will take the From header: Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 2/8 (0/2) Hope this helps, Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Oster Sent: Sunday, February 08, 2004 2:22 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 200 MWI Button I'm trying to get the MWI button to work with my Asterisk configuration. The snom is accepting and responding to the Message indications from *, but when I press the MWI button, it is dialing my extension (the one with the voice mail on it). I'm wondering if there is a way to specify what extension to dial to check email in the configuration, either the phone, or * itself. Asterisk Version 1/30/2003 checked out and compiled this evening Snom Version 2.03o (most recent auto-update) Any help would be greatly appreciated. At one point Mark had talked about adding a voicemail= directive in sip.conf on the mailing list at one point, however grepping the code doesn't reveal a feature like that at this time. Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you. Paul M. Oster
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Vic Cross wrote: On Sat, 7 Feb 2004, John Fraizer wrote: snip all the trace data Here are the configs: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 66.35.64.38 ; Address to bind to context = default ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls [100] type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes [228] type=friend username=228 secret=secret host=dynamic fromuser=228 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes [] type=friend username= secret=secret host=dynamic fromuser= context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes Remove fromuser= from your SIP statements. This overrides the caller-id data received with whatever is stated in fromuser -- Asterisk is doing exactly what you told it to. ;-) To explain a bit more: Fromuser= and fromdomain= is used to specify the caller when calling this device. This is used when we're calling a SIP proxy that requires a specific fromuser/domain in addition to an authentication. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 100 Code Recommendation
Hi Geert, Jason Ross wrote: G'Day, I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm having DMTF problems no matter what configuration I try. And as yet I haven't downgraded it to see if an earlier release makes a difference Just wondering if anyone can provide some guidance as to what the best release of code for this phone may be. I've been having busy problems with the 2.03x firmware versions, but no DTMF problems. I configured the phone for DTMF outband, with asterisk configured as dtmfmode=rfc2833. I'm running 2.02z. I've tried configuring the phone this way and haven't had much luck, but I'll give it another go. Cheers, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P
I have only 2 64 bits on my server's mobo :( I was wondering if it would work because the card's interface can fit into the 64 bits slot. I guess I have to change the motherboard. Regards, Soragan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Sunday, February 08, 2004 2:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P no why would it need to do that? If I put one in my 64 bit slots the machine won't boot. bkw On Sun, 8 Feb 2004, Soragan wrote: Hi all, Just a quick question Does digium x100p support 3.3v pci (64 bit 66mhz) ? Regards, Soragan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P
Soragan wrote: I have only 2 64 bits on my server's mobo :( I was wondering if it would work because the card's interface can fit into the 64 bits slot. I guess I have to change the motherboard. Regards, Soragan If you only need one analog channel infor your Asterisk box then you are probably wasting the power of your server, one channel will happily run on an old P2, then you can use the powerful system you have for something else.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P
I only have 1 server which is dual p3 1ghz. Its mobo only has 2 64 bits pci. :( Regards, Soragan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of WipeOut Sent: Sunday, February 08, 2004 10:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Soragan wrote: I have only 2 64 bits on my server's mobo :( I was wondering if it would work because the card's interface can fit into the 64 bits slot. I guess I have to change the motherboard. Regards, Soragan If you only need one analog channel infor your Asterisk box then you are probably wasting the power of your server, one channel will happily run on an old P2, then you can use the powerful system you have for something else.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P
I only have 1 server which is dual p3 1ghz. Its mobo only has 2 64 bits pci. :( Not good. Throw it away in my house's trash can and I send you 4 Pentium 2... more details in pvt. :-) Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] nortel i2004 ip phone
Hi All does anyone know anything about the nortel i2004 ip phone. It is very hard to find out if the unit is h323 or sip. Can anyone please tell me info regardsing this. All I can find on the net is manuals and what codecs it uses. but too many websites contradict each other. So I have no straight answer on what it is? regards Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with ATA's locking up..
Hi, -Original Message- Subject: [Asterisk-Users] Problems with ATA's locking up.. Anyone had any problems with ATA's running 3.0 software locking up? Nope, seems rock-solid here. Can you tell more about the circumstances this occurs with ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: SNOM 200 silence suppression
Hi Olle, you mail server thinks this is SPAM, so I resend it in the mailing-list... CS -Original Message- From: Christian Stredicke [mailto:[EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:54 PM To: 'Olle E. Johansson' Subject: RE: SNOM 200 silence suppression Hi Olle, we do silence suppression (slash half duplex) in hands free mode to kill the echo. If the handset is being used, it is definitely turned off. Did you try the latest Ethereal? It's able to generate .au files from G.711 RTP, usually a good first indicator what's going on on the cable. CS -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Sunday, February 08, 2004 9:55 AM To: Christian Stredicke Subject: SNOM 200 silence suppression Christian, I have a choppy sound from Asterisk to the Snom 200. Could it be that the SNOM supports silence suppression? If so, I can't find any setting in regards to this. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail/Playback Questions
We are using a SCSI based IBM eServer x300 for our PBX. In setting this unit up, we used a backup machine, which was IDE only. The problem that we are currently experiencing is that the voicemail prompts are coming out the system so fast that the words overlap each other, and sometimes are unintelligable. For instance: The person at extension 7-0-0-1 is unavailable might come out as The at 7-0-1 unavailable. This issue appears unique to the SCSI system, and did not occur with the IDE-only machine. It also is not limited to just voicemail, but all files run through Playback() /proc/cpuinfo on the SCSI machine indicates: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Celeron (Coppermine) stepping: 10 cpu MHz : 951.714 cache size : 128 KB ... and on the IDE machine: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 3 cpu MHz : 867.418 cache size : 256 KB The config files are the same across the two systems, and both are running the same version. Show version in the asterisk console reads: Asterisk CVS-01/30/04-19:07:39 Any help that you can provide would be appreciated. If there is any further information I am forgetting, please let me know. Thanks in Advance! Steve Ringwald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk pins CPU
Hello All.
Many times I've done a top and found that Asterisk is pinning the CPU,
even when Asterisk isn't being used (this is on a DEV box):
2044 root 15 0 5232 5228 2608 R98.6 8.5 626:58 0 asterisk
I'm running a recent build of Asterisk on Slackware (2.4.24 kernel):
www*CLI show version
Asterisk CVS-02/07/04-21:34:06 built by [EMAIL PROTECTED] on a i686 running Linux
A strace shows that it's looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1) = 0
ioctl(0, SNDCTL_TMR_STOP, {B38400 opost isig icanon echo ...}) = -1 EIO
(Input/o
utput error)
-end-
The machine does not have a soundcard and my /etc/asterisk/modules.conf has:
noload = chan_alsa.so
noload = chan_oss.so
If I restart asterisk all is well, for awhile. I'm not sure what
triggers this behvaior. Anyone else getting this behavior?
I wish the lists were searchable... :(
Thanks.
Cheers
Jason
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RE: [Asterisk-Users] Snom 200 MWI Button
-Original Message- From: Paul Oster [mailto:[EMAIL PROTECTED] Sent: Saturday, February 07, 2004 8:22 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 200 MWI Button\ I'm trying to get the MWI button to work with my Asterisk configuration. The snom is accepting and responding to the Message indications from *, but when I press the MWI button, it is dialing my extension (the one with the voice mail on it). I'm wondering if there is a way to specify what extension to dial to check email in the configuration, either the phone, or * itself. Asterisk Version 1/30/2003 checked out and compiled this evening Snom Version 2.03o (most recent auto-update) Any help would be greatly appreciated. At one point Mark had talked about adding a voicemail= directive in sip.conf on the mailing list at one point, however grepping the code doesn't reveal a feature like that at this time. Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you. Paul M. Oster Paul, We have ours working pretty well. If you would like to contact me off site I will try to explain what we have done and maybe we can work together on the Snow 200 issue. Thanks, Dustin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with ATA's locking up..
3.0.0 have some problems. Sometimes, ata answers to invite with Not found or Busy here. This is a strange behavior. I'm using now 2.16.2 Regards, Gus - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:56 AM Subject: [Asterisk-Users] Problems with ATA's locking up.. Anyone had any problems with ATA's running 3.0 software locking up? Thanks, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with ATA's locking up..
Hi, Citeren Billy Huddleston [EMAIL PROTECTED]: Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or anything, I am using re-invites. Pretty standard setup. When they lockup, you can't ping them, or get to the http interface, and I even think the IVR stops responding when you push the button. Yes, but is anything specific happening when they hang ? (What are you doing that seems to cause the hang ?) I have pretty similar setups, so I could try to recreate your scenario ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk pins CPU
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1. Ignoring the ioctl made
the CLI non-functional. Happy to get any help I can in this regard.
Mark
On Sun, 8 Feb 2004, Jason Becker wrote:
Hello All.
Many times I've done a top and found that Asterisk is pinning the CPU,
even when Asterisk isn't being used (this is on a DEV box):
2044 root 15 0 5232 5228 2608 R98.6 8.5 626:58 0 asterisk
I'm running a recent build of Asterisk on Slackware (2.4.24 kernel):
www*CLI show version
Asterisk CVS-02/07/04-21:34:06 built by [EMAIL PROTECTED] on a i686 running Linux
A strace shows that it's looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1) = 0
ioctl(0, SNDCTL_TMR_STOP, {B38400 opost isig icanon echo ...}) = -1 EIO
(Input/o
utput error)
-end-
The machine does not have a soundcard and my /etc/asterisk/modules.conf has:
noload = chan_alsa.so
noload = chan_oss.so
If I restart asterisk all is well, for awhile. I'm not sure what
triggers this behvaior. Anyone else getting this behavior?
I wish the lists were searchable... :(
Thanks.
Cheers
Jason
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Re: [Asterisk-Users] Problems with ATA's locking up..
Hi, Citeren CW_ASN [EMAIL PROTECTED]: 3.0.0 have some problems. Sometimes, ata answers to invite with Not found or Busy here. This is a strange behavior. I'm using now 2.16.2 Hm ? I have not seen this happening yet. 2.16 has alternative behaviour regarding flash transfers... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with ATA's locking up..
Could you share your 3.0.0 config? - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 2:10 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren CW_ASN [EMAIL PROTECTED]: 3.0.0 have some problems. Sometimes, ata answers to invite with Not found or Busy here. This is a strange behavior. I'm using now 2.16.2 Hm ? I have not seen this happening yet. 2.16 has alternative behaviour regarding flash transfers... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with ATA's locking up..
That's just it, I'm not doing anything.. Just normal use.. as far as I can tell, they end up locking up with or without anyone using them as far as I can tell.. Thanks, Billy - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:08 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren Billy Huddleston [EMAIL PROTECTED]: Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or anything, I am using re-invites. Pretty standard setup. When they lockup, you can't ping them, or get to the http interface, and I even think the IVR stops responding when you push the button. Yes, but is anything specific happening when they hang ? (What are you doing that seems to cause the hang ?) I have pretty similar setups, so I could try to recreate your scenario ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with ATA's locking up..
http://www.nxs.net/cisco_ata_186.htm - Original Message - From: CW_ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:40 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Could you share your 3.0.0 config? - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 2:10 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren CW_ASN [EMAIL PROTECTED]: 3.0.0 have some problems. Sometimes, ata answers to invite with Not found or Busy here. This is a strange behavior. I'm using now 2.16.2 Hm ? I have not seen this happening yet. 2.16 has alternative behaviour regarding flash transfers... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk pins CPU
Upon reading the bug I can confirm that I was doing:
asterisk -vvvc
through a ssh session/client (the Asterisk box is headless) and then for
whatever reason I would close that session/client and come back later
and start another ssh session/client and do a:
asterisk -r
and then probably a:
restart gracefully
triggered the behavior.
Thanks for the quick reply.
Cheers
Jason
Mark Spencer wrote:
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1. Ignoring the ioctl made
the CLI non-functional. Happy to get any help I can in this regard.
Mark
On Sun, 8 Feb 2004, Jason Becker wrote:
Hello All.
Many times I've done a top and found that Asterisk is pinning the CPU,
even when Asterisk isn't being used (this is on a DEV box):
2044 root 15 0 5232 5228 2608 R98.6 8.5 626:58 0 asterisk
I'm running a recent build of Asterisk on Slackware (2.4.24 kernel):
www*CLI show version
Asterisk CVS-02/07/04-21:34:06 built by [EMAIL PROTECTED] on a i686 running Linux
A strace shows that it's looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1) = 0
ioctl(0, SNDCTL_TMR_STOP, {B38400 opost isig icanon echo ...}) = -1 EIO
(Input/o
utput error)
-end-
The machine does not have a soundcard and my /etc/asterisk/modules.conf has:
noload = chan_alsa.so
noload = chan_oss.so
If I restart asterisk all is well, for awhile. I'm not sure what
triggers this behvaior. Anyone else getting this behavior?
I wish the lists were searchable... :(
Thanks.
Cheers
Jason
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Re: [Asterisk-Users] Problems with ATA's locking up..
I will test with TOS in a8b8. All other stuff are equal in my ata. Regards, Gus - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 2:51 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. http://www.nxs.net/cisco_ata_186.htm - Original Message - From: CW_ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:40 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Could you share your 3.0.0 config? - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 2:10 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren CW_ASN [EMAIL PROTECTED]: 3.0.0 have some problems. Sometimes, ata answers to invite with Not found or Busy here. This is a strange behavior. I'm using now 2.16.2 Hm ? I have not seen this happening yet. 2.16 has alternative behaviour regarding flash transfers... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nortel i2004 ip phone
yaboo wrote: Hi All does anyone know anything about the nortel i2004 ip phone. It is very hard to find out if the unit is h323 or sip. Neither. It uses a Nortel proprietary protocol. Can anyone please tell me info regardsing this. All I can find on the net is manuals and what codecs it uses. but too many websites contradict each other. So I have no straight answer on what it is? regards Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] short ringing
On Sunday 01 February 2004 08:13 am, Martin wrote:
Hello.
I'm a bit puzzled at the moment. I have a x100P and TDM400 with 4 modules
(extensions). Asterisk CVS-02/01/04-06:55:30
Part of my extensions.conf says this:
; Zap Phone #1
;
exten = 204,1,Dial(Zap/2,20) ; Ring for 20 seconds
exten = 204,2,Voicemail(u${EXTEN})
exten = 204,3,Hangup ; Unavail voicemail if extension
doesn't answer
exten = 204,102,Voicemail(b${EXTEN}) ; Busy Voicemail if extension is
busy
exten = 204,103,Hangup
;
;
; Zap Phone #2
;
exten = 120,1,Dial(Zap/3,20) ; Ring for 20 seconds
exten = 120,2,Voicemail(u${EXTEN})
exten = 120,3,Hangup ; Unavail voicemail if extension
doesn't answer
exten = 120,102,Voicemail(b${EXTEN}) ; Busy Voicemail if extension is
busy
exten = 120,103,Hangup
If I call from ext. 120 to ext. 204, it rings for 15 seconds. I call from
ext. 204 to ext. 120, it rings for 10 seconds. In both cases, it announces
the unavail and goes to voicemail.
Why different rings times and why not 20 seconds ?
Regards...Martin
Well, no answer so I assumed it was a simple question and had looked in
different areas, sites etc. for a hint, but I'm still none the wiser.
Can anyone point me in the right direction for this issue ?
Regards...Martin
--
Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html
Free MS-Office replacement for most platforms
http://www.openoffice.org/
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[Asterisk-Users] NanoBGA VIA Eden-N Processor
Hello. Anyone tried this newer via EDEN processor ased on a new streamlined Nehemiah core architecture released in October. http://www.via.com.tw/en/Digital Library/PR031014EdenN.jsp REgards...Martin -- Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Free MS-Office replacement for most platforms http://www.openoffice.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Panasonic KXTD - Vonage
Has anybody interfaced an asterisk system with a Panasonic KXTD switch? I have invested in a few digital phones for my house and would hate to throw them away. I'm thinking about interfacing with the KXTD using their voice mail integration. Only issue is that it is quite difficult to find any information about their protocol. Is the only solution to use their inband DTMVF signaling with an FXO card? 2nd question (should I use a separate email?) I have Vonage service. Is it possible to end the call directly in my asterisk system rather than in the Motorola V1005? Thanks Jacques
[Asterisk-Users] dialout redunancy.
Hi, How do I get asterisk to use an alternate outbound provider in the event my primary IAX provider goes down. I currently have an IAX provider that is having issues, so I signed up with a sip provider for a backup. I added the sip provider info into the extensions.conf file as the second outbound entry, but asterisk still tries to call the iax provider 1stand since the call is incomplete the end-user hangs up.Any ideas would behelpful. Thanks John Bittner Simlab.net
RE: [Asterisk-Users] dialout redunancy.
You will need to set priorities for each one.
For example:
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Playback(pstnallbusy)
exten =
_91NXXNXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _91NXXNXX,4,Congestion
Basically what happens here, is I try to put it out on the Verizon POTS
lines first, then if that doesn't work, I play a message saying all the
lines are busy, hold if the call is important (it's now billable), the
user holds, and it goes to voicepulse.
You could get rid of the All Busy message if you wanted, I just like to
know that the call is going to be billed (since I have unlimited LD on
my POTS lines). If that fails, It plays a fast busy.
You can also do a qualify in your iax.conf and sip entries to know
whether they are reachable before trying the call. Read up on qualify to
find out how to do it for your needs.
Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Bittner
Sent: Sunday, February 08, 2004 2:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialout redunancy.
Hi,
How do I get asterisk to use an alternate outbound provider in the event
my primary IAX provider goes down. I currently have an IAX provider that
is having issues, so I signed up with a sip provider for a backup. I
added the sip provider info into the extensions.conf file as the second
outbound entry, but asterisk still tries to call the iax provider
1st and since the call is incomplete the end-user hangs up. Any ideas
would be helpful.
Thanks
John Bittner
Simlab.net
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Re: [Asterisk-Users] Asterisk Panasonic KXTD - Vonage
2nd question (should I use a separate email?) I have Vonage service. Is it possible to end the call directly in my asterisk system rather than in the Motorola V1005? with any compatible FXO card, like digium X100P Youness ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 MWI Button
Dustin Knuttgen wrote: Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you. Read http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom The problem is well-known. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex == Screech using version 1.1.4
I just downloaded and built the latest beta version of Speex, and it appears to be the case that the 1.1.x versions do not work with asterisk. Just to be sure, I built the 1.1.4 (both with and without the fixed-point option) on two servers so I could test using the same codecs. All calls made that way yielded a horrid screeching, although by listening creatively the content was there, beneath the screeching. On a broader note, I would love to try to play with the very-low-bandwidth versions of Speex. I could have sworn I saw things on the bugtracker some weeks back on that topic, but I can't find them anymore. Anyone? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialout redunancy.
Dialout redundancy using this method works perfect. I've been using this method for
some time now. I currently have two IAX2 providers and plan to get another backup as
well (In addition to me getting my Digium cards tomorrow that'll be another backup.)
That's great for outgoing calls, but... I'm trying to figure out the best approach to
use for incoming calls.
I currently have a VP phone number, it's the only incoming number I have for the other
voip providers I have don't offer local termination (or any at all for that matter).
We have a POTS line from Verizon and we'd like to continue using that phone number.
Originally we were just going to forward that phone number to VP. But what happens if
VP goes down? I figure in that case (and we'd have to get in touch with VP if they
will forward to another number if they're done), to then forward to another voip /
pots line that we have.
Is there any other approach we can use to do this?
Possibly, a service that'll offer something like:
Transfer to 1609xxx but if busy, forward to 1609xxx, etc. and so on?
In addition does anyone know where I might be able to port my number to that supports
transferring instead of forwarding?
I currently have Verizon and they said we need a CustoFlex plan which will only
support 6 forwards so if 7 callers call in, the 7th will get a busy signal.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks
Sent: Sunday, February 08, 2004 3:15 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redunancy.
You will need to set priorities for each one.
For example:
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Playback(pstnallbusy)
exten =
_91NXXNXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _91NXXNXX,4,Congestion
Basically what happens here, is I try to put it out on the Verizon POTS
lines first, then if that doesn't work, I play a message saying all the
lines are busy, hold if the call is important (it's now billable), the
user holds, and it goes to voicepulse.
You could get rid of the All Busy message if you wanted, I just like to
know that the call is going to be billed (since I have unlimited LD on
my POTS lines). If that fails, It plays a fast busy.
You can also do a qualify in your iax.conf and sip entries to know
whether they are reachable before trying the call. Read up on qualify to
find out how to do it for your needs.
Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Bittner
Sent: Sunday, February 08, 2004 2:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialout redunancy.
Hi,
How do I get asterisk to use an alternate outbound provider in the event
my primary IAX provider goes down. I currently have an IAX provider that
is having issues, so I signed up with a sip provider for a backup. I
added the sip provider info into the extensions.conf file as the second
outbound entry, but asterisk still tries to call the iax provider
1st and since the call is incomplete the end-user hangs up. Any ideas
would be helpful.
Thanks
John Bittner
Simlab.net
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[Asterisk-Users] Re: Speex == Screech using version 1.1.4
Brian == Brian Capouch [EMAIL PROTECTED] writes: Brian On a broader note, I would love to try to play with the Brian very-low-bandwidth versions of Speex. I could have sworn I saw Brian things on the bugtracker some weeks back on that topic, but I Brian can't find them anymore. It is bug number: http://bugs.digium.com/bug_view_page.php?bug_id=149 And yes, * should be able to use all of the (narrow band) rates speex provides, with complexity, et al user-settable. But that will take (more than?) a bit of work to accomplish -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialout redunancy.
I got it working by configuring qualify in my iax.conf. I guess asterisk
didn't think the IAX provider was down until I added that line.
As for incoming I have an 800 number pointing to 2 local phone numbers. 1 on
voicepulse and 1 on voiceglo. This way if voicepulse is down it will route
the call to voiceglo. Hopefully as the voip providers get better they will
offer a forwarding feature. Vonage does.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew B Marlowe
Sent: Sunday, February 08, 2004 5:45 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redundancy.
Dialout redundancy using this method works perfect. I've
been using this method for some time now. I currently have
two IAX2 providers and plan to get another backup as well (In
addition to me getting my Digium cards tomorrow that'll be
another backup.)
That's great for outgoing calls, but... I'm trying to figure
out the best approach to use for incoming calls.
I currently have a VP phone number, it's the only incoming
number I have for the other voip providers I have don't offer
local termination (or any at all for that matter).
We have a POTS line from Verizon and we'd like to continue
using that phone number.
Originally we were just going to forward that phone number to
VP. But what happens if VP goes down? I figure in that case
(and we'd have to get in touch with VP if they will forward
to another number if they're done), to then forward to
another voip / pots line that we have.
Is there any other approach we can use to do this?
Possibly, a service that'll offer something like:
Transfer to 1609xxx but if busy, forward to 1609xxx,
etc. and so on?
In addition does anyone know where I might be able to port my
number to that supports transferring instead of forwarding?
I currently have Verizon and they said we need a CustoFlex
plan which will only support 6 forwards so if 7 callers
call in, the 7th will get a busy signal.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brent Franks
Sent: Sunday, February 08, 2004 3:15 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redunancy.
You will need to set priorities for each one.
For example:
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Playback(pstnallbusy)
exten =
_91NXXNXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _91NXXNXX,4,Congestion
Basically what happens here, is I try to put it out on the
Verizon POTS
lines first, then if that doesn't work, I play a message
saying all the
lines are busy, hold if the call is important (it's now billable), the
user holds, and it goes to voicepulse.
You could get rid of the All Busy message if you wanted, I
just like to
know that the call is going to be billed (since I have unlimited LD on
my POTS lines). If that fails, It plays a fast busy.
You can also do a qualify in your iax.conf and sip entries to know
whether they are reachable before trying the call. Read up on
qualify to
find out how to do it for your needs.
Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Bittner
Sent: Sunday, February 08, 2004 2:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialout redunancy.
Hi,
How do I get asterisk to use an alternate outbound provider
in the event
my primary IAX provider goes down. I currently have an IAX
provider that
is having issues, so I signed up with a sip provider for a backup. I
added the sip provider info into the extensions.conf file as
the second
outbound entry, but asterisk still tries to call the iax provider
1st and since the call is incomplete the end-user hangs up. Any ideas
would be helpful.
Thanks
John Bittner
Simlab.net
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RE: [Asterisk-Users] dialout redunancy.
You're toll-free number automatically forwards to the next number if one is busy?
Cool. I wasn't sure if it would do that. I know VP reports a fast busy. Don't know
what Voiceglo reports.
What toll-free provider do you have out of curiosity?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner
Sent: Sunday, February 08, 2004 6:12 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redunancy.
I got it working by configuring qualify in my iax.conf. I guess asterisk
didn't think the IAX provider was down until I added that line.
As for incoming I have an 800 number pointing to 2 local phone numbers. 1 on
voicepulse and 1 on voiceglo. This way if voicepulse is down it will route
the call to voiceglo. Hopefully as the voip providers get better they will
offer a forwarding feature. Vonage does.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew B Marlowe
Sent: Sunday, February 08, 2004 5:45 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redundancy.
Dialout redundancy using this method works perfect. I've
been using this method for some time now. I currently have
two IAX2 providers and plan to get another backup as well (In
addition to me getting my Digium cards tomorrow that'll be
another backup.)
That's great for outgoing calls, but... I'm trying to figure
out the best approach to use for incoming calls.
I currently have a VP phone number, it's the only incoming
number I have for the other voip providers I have don't offer
local termination (or any at all for that matter).
We have a POTS line from Verizon and we'd like to continue
using that phone number.
Originally we were just going to forward that phone number to
VP. But what happens if VP goes down? I figure in that case
(and we'd have to get in touch with VP if they will forward
to another number if they're done), to then forward to
another voip / pots line that we have.
Is there any other approach we can use to do this?
Possibly, a service that'll offer something like:
Transfer to 1609xxx but if busy, forward to 1609xxx,
etc. and so on?
In addition does anyone know where I might be able to port my
number to that supports transferring instead of forwarding?
I currently have Verizon and they said we need a CustoFlex
plan which will only support 6 forwards so if 7 callers
call in, the 7th will get a busy signal.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brent Franks
Sent: Sunday, February 08, 2004 3:15 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redunancy.
You will need to set priorities for each one.
For example:
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Playback(pstnallbusy)
exten =
_91NXXNXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _91NXXNXX,4,Congestion
Basically what happens here, is I try to put it out on the
Verizon POTS
lines first, then if that doesn't work, I play a message
saying all the
lines are busy, hold if the call is important (it's now billable), the
user holds, and it goes to voicepulse.
You could get rid of the All Busy message if you wanted, I
just like to
know that the call is going to be billed (since I have unlimited LD on
my POTS lines). If that fails, It plays a fast busy.
You can also do a qualify in your iax.conf and sip entries to know
whether they are reachable before trying the call. Read up on
qualify to
find out how to do it for your needs.
Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Bittner
Sent: Sunday, February 08, 2004 2:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialout redunancy.
Hi,
How do I get asterisk to use an alternate outbound provider
in the event
my primary IAX provider goes down. I currently have an IAX
provider that
is having issues, so I signed up with a sip provider for a backup. I
added the sip provider info into the extensions.conf file as
the second
outbound entry, but asterisk still tries to call the iax provider
1st and since the call is incomplete the end-user hangs up. Any ideas
would be helpful.
Thanks
John Bittner
Simlab.net
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[Asterisk-Users] OS X
Hi, I see there were a couple of posts regarding installing Asterisk on a OS X box. I have tried with no success. I am completely new to Asterisk but think it is very cool. If someone could please guide me through an install so that I can start to work with Asterisk, I would be very grateful. Thanks so much for the help. Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PCMCIA
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Tooley Sent: Sunday, 8 February 2004 18:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PCMCIA Does anyone know of a PCMCIA FXO card or even a USB one? I'm looking at building an appliance out of a machine that has USB and PCMCIA but no PCI. 2 possibilities: 1) AVM Fritz!Card PCMCIA - does EURO ISDN BRI with chan_capi (IIRC) 2) Digium S100U + FXO/FXS converter. 1st option is probably rock-solid, but you are probably in Seppo-land where there is no Euro-ISDN BRI. 2nd option I've used neither of these components and not heard of people putting them together either, but it might work for you :-). Cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
On Fri, Feb 06, 2004 at 08:18:21PM -0500, Andres wrote: Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response) So did it drop a few seconds into the call...like 5 - 15 seconds? If so then you are having a problem with call setup. I would guess it is the ACK that is not receiving a STATUS 200 OK so Asterisk cuts off the call. No, they drop at random points in the calls. Sometimes after 30 seconds, sometimes up to 5 minutes :( Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Motherboard and fxo suggestion.
What is the very best motherboard I should use to set up my new asterisk box? I plan on installing about 8 pots lines. And is X100P the only card available? I'm looking for multiple pots line cards. I'm trying to avoid irq conflicts as well as have a superstable box. Thanks George ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Motherboard and fxo suggestion.
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Geo_p15tt Sent: Monday, 9 February 2004 14:25 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Motherboard and fxo suggestion. What is the very best motherboard I should use to set up my new asterisk box? I plan on installing about 8 pots lines. Assuming US, since your email address has a state in the domain. There are other options for EURO ISDN countries. 8 is a lot for POTS lines, if your Telco can give you a fractional T1 for the same or a little bit more, you will be much better off, digital call handling, one T100P card ($595?) instead of 8 X100Ps ($792?). And is X100P the only card available? I'm looking for multiple pots line cards. If T1 isn't an option, move :-) If moving isn't an option... Digium has a 4 FXO card in the works, if you can wait a bit (several weeks?, few months?) Voicetronix Openline4 + Openswitch6/12 have Asterisk channel drivers (chan_vpb), don't know what features are missing. Various hardware boxes handle FXO-SIP (Mediatrix?) I'm trying to avoid irq conflicts as well as have a superstable box. The advantage of digital over POTS is that Asterisk is signalled when events happen (remote end pickup/hangup busy/ringing) rather than trying to work it out from tones/pulses, which is what humans have to do (and are a lot better at it). Cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer from a queue
I have set up call queue for incoming calls. However, when I try to transfer call after answering the queue to another station, the call is hung up. The agent login into Asterisk by AgentCallbackLogin(). When the agent's phone rings the agent pick up the call queue. Is it normal behaviour that transfer is not possible after call is picked from a call queue? -- David Kwok FWD#/IAXTEL# : 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
RE: [Asterisk-Users] X100P
On Sun, 8 Feb 2004, Soragan wrote: I only have 1 server which is dual p3 1ghz. Its mobo only has 2 64 bits pci. :( Ohh.. You better give that to me then. I'll send you my Pentium 133 w/ 16 megs of ram. It works great with the X100P. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registering SJPhone with Asterisk
I am trying to register SJPhone with my asterisk server but my SJPhone messages saying NON-INVITE transaction.. registration failed ... I dont have any FXO or FXS card installed. Just the asterisk server running on Linux and an SJPhone installed on my windows box. Some debug info: localhost*CLI sip show users Username Secret Authen Def.Context A/C 7211 abcdefg md5,plaintext sip No localhost*CLI sip show peers Name/username Host Mask Port Status 7211/7211 (Unspecified) (D) 255.255.255.255 0 Unmonitored mysipproxy.com 192.168.0.100 255.255.255.255 5060 Unmonitored sip.conf [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; All addresses on machine context=sip ; Default for incoming calls register = [EMAIL PROTECTED] [mysipproxy.com] type=peer host=192.168.0.100 fromuser=andy secret=mypassword fromdomain=mysipproxy.com [7211] type=friend username=7211 secret=abcdefg port=5060 reinvite=no context=sip host=192.168.0.101 dtmfmode=inband * extensions.conf [sip] exten = 100,1,dial(SIP/7211) exten = 7211,1,goto(100,1) ; To be able to dial with text, "mysjphone" Any clues why my SJPhone doesnt register ? If someone has successfully registered SJPhone with asterisk, can you send me the sip.conf extension.conf file and instructions on how to configure the SJPhone?Thanks. Yasir Find great local high-speed Internet access value at the MSN High-Speed Marketplace. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail/Playback Questions
Tilghman Lesher wrote: On Sunday 08 February 2004 06:27, Steven Ringwald wrote: We are using a SCSI based IBM eServer x300 for our PBX. In setting this unit up, we used a backup machine, which was IDE only. The problem that we are currently experiencing is that the voicemail prompts are coming out the system so fast that the words overlap each other, and sometimes are unintelligable. For instance: "The person at extension 7-0-0-1 is unavailable" might come out as "The at 7-0-1 unavailable". This issue appears unique to the SCSI system, and did not occur with the IDE-only machine. It also is not limited to just voicemail, but all files run through Playback() Sounds like one of your libraries is buffering output and is returning too soon. Are you running exactly the same distribution/version on each? Perhaps one got an online update and the other did not? That's the only thing I can think of that would cause this type of trouble. I wouldn't suspect hardware differences, as it sounds like you're using Digium hardware for both, where it matters. Yes. Fedora Core 1 on both systems. Same version of Asterisk on both machines. (I copied the source directories of one to create the other). I have also tried updating both to the same version of Asterisk 0.7.2 (CVS), with the same results. Yes, Digium hardware (X100) is in both systems. (Actually, the same card was in both systems). The card is on its own interrupt, also: [EMAIL PROTECTED] root]# cat /proc/interrupts CPU0 0: 16557048 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-uhci, usb-uhci 7: 165254978 XT-PIC wcfxo 8: 1 XT-PIC rtc 10: 4612230 XT-PIC eth0 11: 245282 XT-PIC aic7xxx 15: 1 XT-PIC ide1 NMI: 0 ERR: 0
[Asterisk-Users] Newbie - help
Hi, Is there a work around about Fax and Answering Machinedetection ?If not, where is the all process, at chan_zap.c ?Any site that could help ? Actually, how is this working? When we originate a call, * just recognize if the line is busy and then creates a record for that call at CDR ? or not ? Thanks, Marin Blu Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online
RE: [Asterisk-Users] X100P
Ohh.. You better give that to me then. I'll send you my Pentium 133 w/ 16 megs of ram. It works great with the X100P. LOL, can your Pentium do web server, mail server with spam and virus checking and ADSL router all together? If it can do without any performance loses compare with mine, I'd be happy to change it. ;p ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
