[Asterisk-Users] Static on ZAP channels
I have a TDM2400P with hardware echo cancel. We seem to have static on some calls but not others and the receive audio appears 'choppy'. Transmit side works fine and does not have any audio problems. I had to turn up the RX gain to 18 or the receive audio volume is too low. Can anyone shed some light? Thanks. TJ ___ --Bandwidth 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] TDM2400P problems
I am having issues with a TDM2400P. It appears when the ZAP channel dials out, it randomly chops the first digit off of the number. I have tried relaxdtmf=yes, turning up and down the txgain, turned off and on the echo cancellation, generated new zaptel (with updated spinlock.h)... I am at a loss. Can someone please offer some help? Thanks. TJ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
That's what I'm about to try, I keep getting pulled off of this project to go do other things. Thanks for the input. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim Jackson wrote: Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Someone has already pointed out that you might have ran into a network problem. What's the network setup between phone and the server? Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 I was unable to use Asterisk from latest CVS, I am using version from 12/02 CVS. I was getting authorization failed in CLI, and phone could not make calls with CVS-latest Asterisk. Might be something similar in your setup? Just copy /usr/src/asterisk from old server and try make install.. Please, someone, comment on latest changes in CVS for SIP configurations? Might enforced md5 passwords etc? Or anything like that? context=noawnser A typo, right? Andrei ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Polycom IP500
They were updated, to reflect the new card. And I can call in perfectly. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, January 06, 2005 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 How about your zapata.conf and zaptel.conf files? Were they updated for the new card? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid=Tim Jackson 101 secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 key 0106005724||*|00|Initial log entry. Current logging level 4 ssps 0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 sip 0106005724||*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang=) 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 1 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=angelinacounty.net, nonce=243b35d1 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk*CLI Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 2 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username=101, realm=angelinacounty.net, nonce=243b35d1, uri=sip:192.9.200.9:5060
RE: [Asterisk-Users] Polycom IP500
This isn't a dialplan issue, it's a SIP issue. The same dialplan and
sip.conf are working perfectly with the other server.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
FROM MY SIP.CONF
[1000]
type=friend
host=dynamic
context=local
allow=ulaw
secret=YESITIS
callerid=Front Desk 1000
[EMAIL PROTECTED]
dtmfmode=rfc2833
nat=0
FROM MY EXTENSION.CONF
[local]
include = mainmenu
include = parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld
include = trunkint
include = sip
YOURS
sip.conf:
[101]
type=friend
callerid=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
May as well just set allow=ulaw unless you are eally using something
else.
Does your extensions.conf have a context default which is set up with
something like...
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _9XXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9XXX,2,Congestion
exten = _9480NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9480NXX,2,Congestion
exten = _9602NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9602NXX,2,Congestion
exten = _9623NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9623NXX,2,Congestion
Where TRUNK is passed in from a global?
MINE GLOBALS
;Trunk Info
TRUNK=ZAP/g1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
On a guess, it seems like your context for incoming could be correct and
your context for out may be wrong.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
They were updated, to reflect the new card. And I can call in perfectly.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 2:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
How about your zapata.conf and zaptel.conf files? Were they updated for
the new card?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 12:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500
Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone
cards to a 1U IBM server with a TDM04B card. I finally got the card
working in the server, but I'm having issues with these Polycom IP500s
now. Using the exact same config from the old server I'm getting weird
errors. Dial a number on the phone and it gives you dialtone but no user
interaction (if that makes sense) then after about 35-40 seconds it
displays Line used remotely and hangs up. Inbound calls ring, but you
can't answer them, registration seems to be ok, but I'm at a loss.
sip.conf:
[101]
type=friend
callerid=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
OR
[101]
type=peer
callerid=Tim Jackson 101
secret=itsasekret
host=dynamic
dtmfmode=rfc2833
nat=yes
mailbox=101
canreinvite=no
context=default
disallow=all
allow=ulaw
app log from phone:
0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp,
errno 0xdd.
0106005724|net |*|00|Initial log entry. Current logging level 4 key
0106005724||*|00|Initial log entry. Current logging level 4 ssps
0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2
FS1, Version=1.1.2.0002 17-May-04 15:28
0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112.
0106005724|pps |*|00|Initial log entry. Current logging level 4 sip
0106005724||*|00|Initial log entry. Current logging level 4 cfg
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101]
0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting
lcl.ml.lang=)
0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005727|slog |*|00|Initial log entry. Current logging level 4
0106005927|cfg |4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
debug sip peer 101
Sip read:
REGISTER sip:192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6
RE: [Asterisk-Users] Polycom IP500
Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 running Linux [channels] echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=inbound-pots signalling=fxs_ks callerid=Unknown Caller group = 1 channel = 1-2 echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=noawnser signalling=fxs_ks callerid=Unknown caller group = 1 channel = 3-4 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, For what it's worth, from my working sip.conf for Polycoms: [2010] type=friend username=usr2010 callerid=MyName 2010 secret=nobodyknowswhatitis host=dynamic dtmfmode=inband context=admin defaultip=192.168.1.10 progressinband=no Notes: dtmfmode=inband and progressinband=no - that seems to be recommended from * sample sip.conf file for Polycoms. defaultip= setting helped with network issues, not only with Polycoms, with Cisco 7940 as well. Also in main sip.conf: [general] ... disallow=all ; Allow all codecs allow=ulaw,alaw maxexpirey=7200 defaultexpirey=3600 canreinvite=no Also, if you are not behind NAT, why nat=yes? And if NAT is in use, what is your network infrastructure? Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from October 2004). And of course: what is Asterisk and zaptel version? What is your zapata.conf (just curious)? Andrei Tim Jackson wrote: Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] TDM04B vs Dell
TDM400's use the wcfxs module to drive both FXO and FXS ports on them. I have an IBM xSeries 305 (1U P4 2.4ghz 1gb of RAM) server and I just picked up a TDM04B today, and I am getting the exact same problem. When I make calls to/from the TDM04B card I get this really really staticky sound. Calls show up however. Any resolutions to this? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, January 05, 2005 6:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM04B vs Dell I've struggled for several days trying to get a Digium TDM04B 4-port wxfco card working on a Dell 1U PowerEdge 750 machine running Fedora Core 1. I finally got a call back from Digium who indicated that there is a fundamental conflict between the card and the PowerEdge having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. That sounds a little hard to believe. The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages If you have a tdm04b, that says you have four fxo ports. Why are you trying to load wcfxs? I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. I don't use Fedora, but it seems those that do have had problems loading the drivers. Try the modprobe wcfxo then zaptel, then check your /proc/interrupts. If that doesn't work, try modprobe zaptel only. I think someone mentioned a readme in the src/zaptel directory for Fedora as well. Might look. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] TDM04B vs Dell
I dug around and found my newest UpdateXpress cd from IBM and ran it on this box and updated the BIOS and my problem went away. *shrugs* -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan Sent: Wednesday, January 05, 2005 7:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM04B vs Dell At 06:53 PM 1/5/2005 -0600, you wrote: I've struggled for several days trying to get a Digium TDM04B 4-port wxfco card working on a Dell 1U PowerEdge 750 machine running Fedora Core 1. I finally got a call back from Digium who indicated that there is a fundamental conflict between the card and the PowerEdge having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. That sounds a little hard to believe. I agree. Perhaps I have too much faith in Digium support. Does anyone else disagree with Digium's assessement? The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages If you have a tdm04b, that says you have four fxo ports. Why are you trying to load wcfxs? Actually, I tried modprobe wctdm which was supposed to load the correct TDM driver and this resulted in the same behavior described above (lights on, system locks.) In an attempt to figure out why the system locked up I subsequently issued a modprobe wcfxs to confirm that was causing the problem. I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. I don't use Fedora, but it seems those that do have had problems loading the drivers. Try the modprobe wcfxo then zaptel, then check your /proc/interrupts. If that doesn't work, try modprobe zaptel only. I think someone mentioned a readme in the src/zaptel directory for Fedora as well. Might look. Thanks for the advice. However, modprobe zaptel didn't do anything (that I could tell) and modprobe wcfxo returned the error. And, greping for Fedora in src/zaptel didn't turn up any matches. Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP500
Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid=Tim Jackson 101 secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 0106005724|key |*|00|Initial log entry. Current logging level 4 0106005724|ssps |*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 0106005724|sip |*|00|Initial log entry. Current logging level 4 0106005724|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 0106005724|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang=) 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 1 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=angelinacounty.net, nonce=243b35d1 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk*CLI Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 2 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username=101, realm=angelinacounty.net, nonce=243b35d1, uri=sip:192.9.200.9:5060, response=11f3478d812d35993018150f29fb5e81, algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP
RE: Re[2]: [Asterisk-Users] Swissvoice IP 10S VoIP Telephone
Has anyone gotten the Swissvoice IP110 to work w/ * ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Walker Sent: Friday, December 10, 2004 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re[2]: [Asterisk-Users] Swissvoice IP 10S VoIP Telephone Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? AB http://www.definitive-edge.com/index-2Swis.htm AB This would be interesting except it appears to be a bit pricey. AB Am looking for a nice quality SIP phone that supports Message Waiting AB Indicator (Grandstream are too Fisherprice for my liking). AB If anyone has experience of it and also knows somewhere for a good price AB (bulk buying is fine). The news release that I received from Swissvoice said sub $100. I will check with them and come back. Definitive Edge's price of Euro140/$185/£96 is well out of order. Adrian === This email has been scanned for Virus infection by MessageLabs For more information please contact [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] Polycom IP500
Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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
RE: [Asterisk-Users] Polycom IP500
Routing here isn't an issue. The router that is their gateway has all internal routes learned via OSPF. Connectivity to the * box is fine (all 100mbit, the interface the phones are on is a dot1q sub-interface though). I'm 100% confident that it's not an routing/nat problem (no NAT taking place). But I've given up. No real answers on the polycom issue, I've just taken the phones and the * box down. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ty Purcell Sent: Thursday, December 02, 2004 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Tim, I've experienced routing problems before where I could dial a phone on a different subnet and I could hear them, but they couldn't hear me. Is it possible that the phone learns the route, then loses it later? I ended up setting the default gateway on all of my phones to a router that knows about all of my subnets, instead of my internet gateway. Ty -Original Message- From: Tim Jackson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 02, 2004 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ 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
RE: [Asterisk-Users] Polycom IP500
I've already added nat=yes. Nothing fancy/special on the routing between these. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, December 02, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hello Tim, You are saying that: phone is on 10.24.102.0/24 and Asterisk resides on 10.24.100.0/24. Honestly, I see at least one hop forwarding here and possible network issues right away. At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work. Sincerely, Andrei Tim Jackson wrote: Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.p d f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies
RE: [Asterisk-Users] Polycom IP500
Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] SIP phones cutting out with Asterisk??
Ive had the same problem. I posted to the list earlier about the problem, and from what I can tell, its a Polycom issue (not a network issue as stated in the other post). It happens after the phones have been on for about 2-3 days from what I can tell. My solution to this was to use a script to reboot the phones every night at like 3am, and the problem has almost disappeared. If you find any other solutions, please post them to the list. -Tim From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Henderson Sent: Friday, November 26, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phones cutting out with Asterisk?? Importance: High Hi folks, I've got a very bizarre problem recurring when making calls with Polycom SoundPoint IP500 SIP phones and Asterisk. Sometimes when a call comes in to an IP500, one of the sides of the conversation is cut off (i.e. the caller can't hear the callee, or vice-versa). This isn't easily repeated, and rebooting the phone, or restarting Asterisk, doesn't seem to have an effect. Has anybody else experienced this sort of thing happening? I've seen this with both CVS-v1-0-10/27/04-21:54:17 and CVS-HEAD-09/02/04-22:57:21. Thanks for any insight, Dave Henderson Customer Service Manager The IT Department, Inc. ph: 613-523-2322x321 fx: 613-526-3949 ___ 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] MGCP
I havent found any recent information on this, but can Asterisk act as a MGCP UserAgent? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ 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] MGCP
Any other ideas for interacting with an MGCP provider? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Cox Sent: Tuesday, November 23, 2004 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MGCP I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent only. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels - Original Message - From: Tim Jackson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 2:48 PM Subject: [Asterisk-Users] MGCP I haven't found any recent information on this, but can Asterisk act as a MGCP UserAgent? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ 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] MGCP
The ILEC here is using VocalData for doing VoIP Centrex systems etc. My sales engineer preached SIP to me when he was talking about it, but I actually got a hold of an engineer today, and he told me they are using MGCP only for now. He seemed really interested in *, they are bringing out some demo units soon for us to beta-test for them and I told him I would show our * box to him. Maybe they might be interested in using MGCP with it, and would be willing to help out. I'll let you know. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stewart Nelson Sent: Tuesday, November 23, 2004 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP I haven't found any recent information on this, but can Asterisk act as a MGCP UserAgent? I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent only. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels Any other ideas for interacting with an MGCP provider? You could, of course, connect an MGCP ATA to FXO port(s) or device(s). That solution degrades quality, increases delay, may have echo problems, etc. However, it's an easy way to get started, e.g. if you have a spare ATA-186 that you can load some MGCP firmware into. I am seeking a proper solution to the same problem, as my ISP in France, Free Telecom, bundles MGCP service at very aggressive rates (including free calls to fixed phones anywhere in France) with their ADSL service. I have looked at some SIP - MGCP and H.323 - MGCP gateways, but they only talk the Call Agent side of the protocol. If you have found a solution, please let me know. If not, perhaps we could work together to write one. One possibility is enhancing MGCP support in * to allow it to act as a User Agent. Another is a stand-alone script, e.g. in perl, that would do SIP - MGCP. I'd be open to other suggestions, too. Thanks, Stewart ___ 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] Polycom Problems
We have Polycom IP500s, and just starting recently (after the broadvoice patch I might add) after about 1-2 days these phones ring, and answer, but we get no audio on the phones. The caller can hear us, but we cannot hear the caller. Its happened 4-5 times and is only intermittent. No errors on the console, using g.711u. Any ideas? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ 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] Headsets for Cisco 7940/7960
I have a Plantronics M12 amplifier and a bunch of interchangeable headsets. I haven't found anything that this won't work on yet. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Sunday, November 21, 2004 5:45 PM To: Shaun Ewing; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Headsets for Cisco 7940/7960 I have used the M175, which is a convertible(over the head or the ear loop) and has a volume control for the microphone and the earpiece and a mute switch for the mic. Lyle - Original Message - From: Shaun Ewing [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 21, 2004 5:30 PM Subject: Re: [Asterisk-Users] Headsets for Cisco 7940/7960 On Sun, 21 Nov 2004 18:01:07 -0500, Brian Pavane [EMAIL PROTECTED] wrote: What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to work, however these are both over-the-head type models. I was looking for an over-the-ear model, as I would like to be able to provide a variety of headsets depending on the individuals taste. I am not looking for a headset that requires an external amplifier, but rather a headset that can make use of the headset jack on the phone itself. We use the Plantronics H51 headsets with no problems. Unfortunately (for you), it's an over-the-head type model. -Shaun ___ 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] Routing between different interfaces
canreinvite=no ? http://www.voip-info.org/wiki-Asterisk+sip+canreinvite -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tracy R Reed Sent: Friday, November 19, 2004 6:56 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Routing between different interfaces I have an asterisk box with a public IP for people on the Internet to connect to. I also have a Lucent TNT on the same physical network but on a 10.0.0.0 subnet. It isn't safe to put the TNT on a public IP address and I never want it to talk to the net directly anyhow so this seemed like a good idea. However asterisk does not seem to properly route SIP calls between the interfaces. I tell the TNT to only allow connections from the ip of the asterisk box but the IP in the SIP headers comes through as that of the originating box, not the asterisk box. Is this how it is supposed to work? It would seem to make impossible what I want to do. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig ___ 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] Broadvoice
Anybody else having broadvoice problems? -- Executing SetAccount(SIP/101-d03b, LD) in new stack -- Executing Dial(SIP/101-d03b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 408 Request Timeout back from 147.135.0.128 == No one is available to answer at this time Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ 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] BroadVoice
Its working here, some issues tho. All outbound calls have no CID. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Saturday, November 13, 2004 1:16 PM To: Doug Shubert Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BroadVoice Same here... Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 13 Nov 2004, Doug Shubert wrote: yes.. started around 12:00 noon EST I get sip_reg_timeout: Registration for '[EMAIL PROTECTED] Does anyone know if this is related to the channels patch? Doug Gary White (Network Administrator) wrote: Anybody else having Broadvoice registration problems today? --- - ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-1 1-13%5Ccf62fbdc4a664e39b123d2ef9ce2d9a4C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Multiple NIC's on * box?
It's no issue to use more than one nic. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, November 11, 2004 7:29 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] Multiple NIC's on * box? Can * support a box with multiple nic cards correctly? Background: small isp operation in the US has a rather large wireless network covering multiple counties. The wireless net is an isolated network using private IP's and nat'ing (via Cisco 7206). Their dsl customers are on another isolated network using registered IP's out to the customer dsl modem (which then does nat'ing) on another Cisco 7206 interface. Will I need to dedicate an * system to each, or can I consider multiple nic's on a single system? (Traffic volumes will be rather low, so multiple machines are not thought to be a requirement now or in the future, unless multiple nic's are not reasonably supported.) Thoughts anyone? ___ 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] Broadvoice asterisk patch
I've applied the patch (after scanning over the file). No issues with *. BV still works, too. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Wilkins Sent: Wednesday, November 10, 2004 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broadvoice asterisk patch I was just about to ask a similar question having just received the message. I'm more concerned about someone trying to spread a virus or something like that. You have to admit that the URGENT, INSTALL THIS message with an attachment pretty much screams virus, even if its not. I tried calling Broadvoice support but they want me to leave a message for them to call me later. Can anyone comment on the validity of this message? thanks, Ryan Wilkins On Nov 10, 2004, at 2:54 PM, [EMAIL PROTECTED] wrote: Just received this from broadvoice, anyone know if this patch will become part of the CVS tree? ___ 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] Broadvoice asterisk patch
Nov 10 14:52:56 NOTICE[20579]: chan_sip.c:4041 sip_reregister:-- Re-registrm -- Responding to challenge, registration to domain/host name sip.broadvoicem Nov 10 14:52:56 NOTICE[20579]: chan_sip.c:6821 handle_response: Outbound Regist) I got this after applying the patch. I'm guessing this is normal? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Giagnocavo Sent: Wednesday, November 10, 2004 2:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Broadvoice asterisk patch They send patches out by email? Who thought of this brilliant idea? Hmm, let's teach our users not to be cautious. /me wonders when someone on linux is gonna install a patch that compromises their system cause some email said so -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 10, 2004 1:54 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Broadvoice asterisk patch Just received this from broadvoice, anyone know if this patch will become part of the CVS tree? -- THIS PATCH MUST BE APPLIED WITHIN 5 DAYS OF RECEIVING THIS E-MAIL OR YOU WILL RISK THE POSSIBLE SUSPENSION OF YOUR BROADVOICE SERVICE. WE APOLOGIZE FOR ANY INCONVENIENCE THIS MAY CAUSE BUT REQUIRE THIS PATCH IN ORDER TO MAINTAIN UNINTERRUPTED OPERATION. Dear Asterisk-Using BroadVoice Customer, BroadVoice has been working very hard in recent months to become a market leader in VoIP service.? As a part of that effort, we have made a concerted effort to facilitate interoperability with as many different SIP devices as possible -- including Asterisk.? While BroadVoice does not directly support Asterisk and will not be able to field specific question on your Asterisk set-up we are doing our best to assist. Unfortunately, the SIP channel in Asterisk has a number of serious issues which make it very difficult for BroadVoice to accommodate Asterisk.? One of these issues, a bug with the Asterisk registration system, is causing an unacceptable load our systems. BroadVoice has hired Olle Johansson and Steve Sokol (the AstriCon team) to work out a solution to the issue.? Attached is a patch that, when applied, will reduce the undue strain on the BroadVoice systems by properly handling registration for Asterisk servers located behind NAT gateways.? We ask that you take a few minutes and patch your server using the following instructions. This patch applies both to the current CVS Head and the Stable 1.0 versions of Asterisk.? If you are running an older version of Asterisk, please update your system to at least 1.0 prior to applying this patch (or you can hack the patch into place in the old chan_sip.c if you feel like it). Note that this patch will be incorporated into the Asterisk CVS at the earliest opportunity.? However, due to the serious nature of the issue we ask that you patch your servers immediately. -= Patch Instructions =- 1.? Copy the patch to /usr/src/asterisk/channels/ (or wherever you store your Asterisk source image. # cp /usr/bob/sip_patch.diff /usr/src/asterisk/channels/ 2.? Apply the patch using the following command: # cd /usr/src/asterisk/channels # patch chan_sip.c sip_patch.diff 3.? Re-compile the SIP channel by executing 'make' in the /etc/asterisk directory. # cd /usr/src/asterisk # make 4.? Install the newly compiled SIP channel with the 'make install' command. # make install 5.? Restart Asterisk to enable the patch as follows: # asterisk -rx restart when convenient This patch will update the Asterisk channel to cache and properly handle registration messages.? Please review the code and, if you have any suggestions, send comments to the author at [EMAIL PROTECTED] -= BroadVoice Configuration Notes =- Because Asterisk does not have outbound proxy support, you need to make a few other changes to make Asterisk work well with BroadVoice. 1.? Find the closest BroadVoice proxy using the 'ping' utility. proxy.dca.broadvoice.com??? 147.135.0.128 proxy.lax.broadvoice.com??? 147.135.8.128 proxy.mia.broadvoice.com??? 147.135.4.128 # ping proxy.lax.broadvoice.com PING proxy.lax.broadvoice.com (147.135.8.128) 56(84) bytes of data. 64 bytes from proxy.lax.broadvoice.com (147.135.8.128): icmp_seq=1 ttl=47 time=41 ms 64 bytes from proxy.lax.broadvoice.com (147.135.8.128): icmp_seq=2 ttl=47 time=31 ms 64 bytes from proxy.lax.broadvoice.com (147.135.8.128): icmp_seq=3 ttl=47 time=58 ms # ping proxy.dca.broadvoice.com PING proxy.dca.broadvoice.com (147.135.0.128) 56(84) bytes of data. 64 bytes from proxy.dca.broadvoice.com (147.135.0.128): icmp_seq=1 ttl=47 time=141 ms 64 bytes from proxy.dca.broadvoice.com (147.135.0.128): icmp_seq=2 ttl=47 time=312 ms 64 bytes from proxy.dca.broadvoice.com (147.135.0.128): icmp_seq=3 ttl=47 time=258 ms Which ever proxy is closer (has a shorter ping time) is the proxy you want to
RE: [Asterisk-Users] high-capacity systems / trouble with Tyan
From my experience the Tyan Tiger MPX is a great board. I've never used it with *, but I have been using it as a high volume samba server for over a year and its never even hicupped. 16:24:30 up 197 days, 20:45, 2 users, load average: 0.94, 0.92, 0.89 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Wednesday, November 10, 2004 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] high-capacity systems / trouble with Tyan Hello, I've had a Tyan dual Athlon MP(2800) machine for a year now and have had several lockups for strange reasons on stock redhat kernel and on custom compiled kernel off of Slackware. I've tried every combination of BIOS settings and changed out all assiciated hardware and found the problem: It's the Tyan. I've also had issues with a couple of SCSI RAID cards when I tried using them with the Tyan card. This all would have really upset me if the Athlon MP platform performed better than the Intel platform, but it doesn't. This Dual Athlon MP system actually handles LESS total Asterisk load than a single P4 3.2 GHz, and the P4 has a lot more Motherboard options and cost much less. This is just my experience, I'm sure I am using Asterisk a little differently than you, I don't have 3 Quad T1 cards in any of my machines, but if that's what you're looking for, I'd suggest the PowerPC(Mac) platform. Asterisk installs just fine right on top of Yellow Dog Linux and the bus speed of a Mac mops the floor with most x86 motherboards, meaning more bandwidth for those bus-hungry Digium boards. MATT--- -Original Message- From: Jim Gottlieb [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 10, 2004 5:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] high-capacity systems / trouble with Tyan On 2004-10-29 at 20:49, Chris A. Icide ([EMAIL PROTECTED]) wrote: The culprit is the RedHat kernel. I don't know what redhat does with their kernel or sources. But If you build your own kernel from non-redhat source, asterisk will compile perfectly. I did as instructed and recompiled a kernel from kernel.org and rebuilt asterisk. However, the problem remains. I can run one or two cards with no problem. But once I enable the third card, the system locks up within a few minutes. I tried getting Athlon MP motherboards other than the Tyan S2466, but no one has any anymore. ___ 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] * does not listen to DTMF during wait ?
Instead of Wait use Background and play silence: exten = s,3,Background(custom/menu) exten = s,4,Background(loligo/silence/10) -Tim On Sat, 2004-11-06 at 13:03 -0700, Damon Estep wrote: My incoming auto attendant plays a prompt, waits for 5 seconds, and the plays the prompt again giving the user a chance to respond. The exten = s,2,Wait,5 prevents users from being able to make a selection during the wait interval. DTMF is only processed during the background prompt playback interval. Is this by design? Is there another way to do the same thing? Damon ___ 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] Linux and Windows
Not to say that all Govt's are like this, but we employ a LOT of open source. It comes down to a money issue. But besides that, we still use Windows, and YES it does have its place. I don't think that this thread belongs on this list. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling Sent: Monday, November 01, 2004 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linux and Windows As far as I understand, corporation, and Govt's like commercial products because of the issue of liability. That is why the Commercial market and Govt market don't accept open source solutions. What is perplexing about the whole situation is that the licensing agreements with commercial software 99.9% of the time indemnify the vendor of any and all liability. So, what we have is the feeling of security... Perhaps Linus should convince the various entities that distribute Linux to include a nice fluffy security blanket with the licensing agreement embroidered on it? That way the attorneys can get that warm fuzzy feeling they so desire. I speak from much experience regarding this matter... The U.S. Govt won't accept an open-source solution even if it is the only option to cover their ass. They'd rather leave their cheese out in the wind than cover it with an open source solution. Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? At 05:59 PM 11/1/2004, you wrote: Jay Milk wrote: Why are you so angry? At the risk of throwing oil on the fire, I would submit that Benjamin was *kidding* at the beginning of that mail, and trolling at the end. I agree with him about the quit anytime I want, but I digress. . . It does appear his flamebait was eagerly pounced upon. Why would the Window$ user$ mind the nose-tweaking? They've got 90%+ of the market. 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] Linux and Windows
I can't speak for the US Govt, but I speak for a local government. We use open source everywhere. My departments PBX is Asterisk, Fileservers, Webservers, we use Linux everywhere. In my dealings with the State of TX they are adopting open source for some very mission critical applications. If you wonder about opensource and Govt go read GCN. They talk all about it. There's a place for both Windows and opensource. If you can't do both, or work around either one on either platform you are too narrow minded. I sit here sending you this e-mail on my laptop running Windows XP, through my Exchange server running Windows 2000, that goes to my Linux mail gateway running Postfix to relay the mail outbound. Be more open minded, Windows isn't going away, and neither is open source software. Learn to deal with it. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling Sent: Monday, November 01, 2004 7:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Linux and Windows At 06:51 PM 11/1/2004, you wrote: [snip for brevity[ So the U.S. Govt has never used linux anywhere? Wow. Not in most installations, and definitely not in DoD facilities. The Office of Inspector General has deemed open source to be Verboten. That's going to become an interesting situation when Solaris goes open source... http://www.eweek.com/article2/0,1759,1647198,00.asp Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? Again making the mistake that open source equates non-commercial. Once again... The Office of Inspector General has deemed (any and all) open-source to be forbidden. Whether it be commercial of non-commercial open-source software it's forbidden. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] MeetMe
I've got a problem with MeetMe. I dial the extension that dynamically creates the new conf, but it just hangs up on me after telling me I'm the only person in the conference. Here's my extensions.conf and what its doing: -- Executing Answer(SIP/101-74c0, ) in new stack -- Executing Wait(SIP/101-74c0, 1) in new stack -- Executing MeetMe(SIP/101-74c0, |Dx) in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getpin' (language 'en') -- Created MeetMe conference 1023 for conference '100' -- Playing 'conf-onlyperson' (language 'en') -- Hungup 'Zap/pseudo-874077465' == Spawn extension (default, 800, 3) exited non-zero on 'SIP/101-74c0' [meetme-int] exten = 800,1,Answer exten = 800,2,Wait(1) exten = 800,3,MeetMe(|Dx) Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ 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] MeetMe
I figured it out. I don't need the 'x' in there. 'x' - close the conference and hangup on all others when last marked user exits I'm an idiot, sorry. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Wednesday, October 27, 2004 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MeetMe Tim Jackson wrote: I've got a problem with MeetMe. I dial the extension that dynamically creates the new conf, but it just hangs up on me after telling me I'm the only person in the conference. Here's my extensions.conf and what its doing: -- Executing Answer(SIP/101-74c0, ) in new stack -- Executing Wait(SIP/101-74c0, 1) in new stack -- Executing MeetMe(SIP/101-74c0, |Dx) in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getpin' (language 'en') -- Created MeetMe conference 1023 for conference '100' -- Playing 'conf-onlyperson' (language 'en') -- Hungup 'Zap/pseudo-874077465' == Spawn extension (default, 800, 3) exited non-zero on 'SIP/101-74c0' [meetme-int] exten = 800,1,Answer exten = 800,2,Wait(1) exten = 800,3,MeetMe(|Dx) Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile Tim, Even though the conference is dynamic, (I think) you still need to define a room number: exten = 800,3,MeetMe(800|Dx) Try that and let us know. -- Kristian Kielhofner ___ 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] polycom IP 500/600
Use something like ProFTPD or something that is supported under their manual (These are better FTP daemons anyway). The default username/pass is PlcmSpIp btw. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Knight Sent: Tuesday, October 26, 2004 7:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] polycom IP 500/600 Kristian Kielhofner wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? As a polycom user, it is the default username that is the issue. It is mixed case, something like Polycom. I think the good old tty drivers still support upper case only terminals, so as soon as it sees the capital P, it will turn on folding. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Broadvoice
We just got setup with Broadvoice yesterday for LD. This isnt something I REALLY need (No local numbers avail so we just got a Houston number), but Im just curious. I can make outbound calls to Broadvoice and they work great, but I cant do inbound. I have bvs voicemail turned off so all I get is a busy signal when I call our bv number. Ive tried this with both type=peer and type=friend and I get the same results, any ideas? context=default recordhistory=yes realm=angelinacounty.net port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw dtmfmode=inband tos=reliability register = 7134810061:[EMAIL PROTECTED] [Broadvoice] type=friend username=7134810061 fromuser=7134810061 secret=[password] host=sip.broadvoice.com context=inbound-pots fromdomain=sip.broadvoice.com nat=yes canreinvite=no dtmfmode=inband ___ 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 and Broadvoice, no incoming voice
I'm having the same issue, and I'm not behind NAT. Maybe this is a BV issue? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry Evans Sent: Saturday, October 23, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router since someone had a problem with that, but still no dice. I've had my Asterisk server running fine for a few months, but this is the first time I've tried a VOIP service with it. I just downloaded and installed the lastest CVS and the problem is still there also. Here's some of my configuration information: sip.conf (I've tried with nat=no and it didn't help) [general] context=from-sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) maxexpirey=3600 defaultexpirey=120 callerid=No CallID tos=lowdelay; 0x18 ; reliabile before dtmfmode=inband srvlookup=yes ;progressinband=no nat=yes notifymimetype=text/plain [broadvoice] type=friend username=801527 (hid real number) fromuser=801527 (hid real number) secret= (hid real password) fromdomain=sip.broadvoice.com host=sip.broadvoice.com canreinvite=no dtmfmode=inband context=broadvoice-inbound nat=yes (tried nat=never also) disallow=all allow=ulaw insecure=very I have the following ports forwarded to my linux server (it's behind a NAT router): 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of those have both TCP and UDP forwarded for now. I've tried several different combinations from different posts, including splitting the broadvoice section up into parts for incoming and outgoing, but it still didn't work. Anyone have any ideas? Let me know if traces, etc. will help and I'll capture and post some. Thanks, Terry ___ 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] Sipura or X100P Option
I'm using 3 X100Ps with no problems in an old IBM machine. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks Sent: Tuesday, October 19, 2004 11:31 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura or X100P Option Hello, Our client currently has two X100P's running in an HP box that has been running for almost a year now with no problems. They have found however that two phone lines are not enough and are bringing in a third phone line. I wouldn't expect this line to be used very often as there are only two employees in the office. I am curious which route to head. I am hesitant to throw another X100P in the box and create the potential for problems, or should I use a Sipura as an FXO device. Has anyone had any experience with Sipura as an FXO? Are there any issues I should know about? Thanks in advance, Brent D. Franks Mindworks Internet Services ___ 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] VoIP over 1xRTT
Anybody ever tried doing voice over Sprint/Verizon 1xRTT cell service? 10-15KB/sec downloads/uploads with 400-1200MS latency is what I usually see on my service. Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile ___ 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] Re: Advice on OS Choice
The horse has been dead for a long while. Please stop beating it. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Saturday, October 16, 2004 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Advice on OS Choice Joe, could we stop this now? It's obvious that if you go to a GPL project and start slinging mud at the GPL, you are in the wrong place. I would recommend that you head over to a Microsoft mailing list where I'm sure you will find an abundance of fodder for your outdated methodologies. When was the last time you actually worked in the industry? I think you'll find if you get back out there that things have changed a lot since the 80's (JK). But seriously. This thread is getting a little silly. Can't we just agree to disagree? The longer you continue this, the more people you will involve from this list. Anyway -1 Flamebait. (also muted in playerlist) Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Cheap, Highquality IP Phones
Best bang for the buck out there are Polycom SoundPoint IP phones. We use IP500s. Pros: Pricetag (Cheaper than Cisco ~$180/phone) Quality (Built really well) Features (3 lines, XML Directory, DND, MWI, etc etc) Fairly straight-forward provisioning (Once you get the hang of it) Very very very configurable Cons: Confusing XML configurations No direct support from Polycom for Asterisk users No XML minibrowser on the IP500 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Friday, October 15, 2004 3:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cheap, Highquality IP Phones I know that there is a list of phones on the wiki, but most of them are now out of date by months if not a year. Our whole office is using Cisco 7960s. Nice phones. Works great with asterisk. However, $300 each. If people could send the phone they use with asterisk, a quick pros/cons and its price, it would be appreciated. Basically, I am looking for a high quality $100 2-line SIP phone that supports g729 and works well with asterisk. Much appreciated, Matthew ___ 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] GPL Violations (Was: Advice on OS Choice)
Break down and learn Debian, its more GNU than you. You'll never go back. Also worth mentioning is Ubuntu Linux which is a Debian offshoot. http://www.ubuntulinux.org/ http://www.debian.org -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Thompson Sent: Thursday, October 14, 2004 7:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GPL Violations (Was: Advice on OS Choice) Brian West wrote: Not to aruge one way or the other, but there are a number of free RH Enterprise work-alike distributions http://www.taolinux.org/ http://www.whiteboxlinux.org/ etc. Jim James H. Thompson [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] Asterisk 1.0 released
Ditto. I'll provide a mirror as well. -Tim -Original Message- From: William Suffill [mailto:[EMAIL PROTECTED] Sent: Thursday, September 23, 2004 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.0 released If anyone who got the 1.0 tar's would be able to get them to me I'd be more than willing to donate traffic toward the effort by mirroring it on some bandwidth. ___ 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] 1.0 Mirrors
Got the 1.0 tarball up, anything else that needs to be mirrored? http://mirrors.angelinacounty.net/asterisk/ ftp://mirrors.angelinacounty.net/asterisk/ -Tim -Original Message- From: Benjamin on Asterisk Mailing Lists [mailto:[EMAIL PROTECTED] Sent: Thursday, September 23, 2004 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.0 Mirrors On Thu, 23 Sep 2004 10:21:06 -0400 (EDT), Greg Boehnlein [EMAIL PROTECTED] wrote: Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz I am happy to provide another mirror (on a 100Mbit fiber link) but I would rather do it for the complete package. Where is the tarball for Zaptel? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] 1.0 Mirrors
BTW, That machine is on 100mbit. Should be able to rape it pretty bad, as long as you don't go over my 1600gigs/month. -Tim -Original Message- From: Tim Jackson Sent: Thursday, September 23, 2004 10:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 1.0 Mirrors Got the 1.0 tarball up, anything else that needs to be mirrored? http://mirrors.angelinacounty.net/asterisk/ ftp://mirrors.angelinacounty.net/asterisk/ -Tim -Original Message- From: Benjamin on Asterisk Mailing Lists [mailto:[EMAIL PROTECTED] Sent: Thursday, September 23, 2004 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.0 Mirrors On Thu, 23 Sep 2004 10:21:06 -0400 (EDT), Greg Boehnlein [EMAIL PROTECTED] wrote: Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz I am happy to provide another mirror (on a 100Mbit fiber link) but I would rather do it for the complete package. Where is the tarball for Zaptel? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Organization wide
After our department went to using *, Ive had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now were using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have 29 POTS lines going into the NEC system. At another location we have a single PRI, and at a lot of the other locations we have just analog phones. Cisco has approached us about using all Cisco equipment, but their idea is going to be costly. Is it wise to use Asterisk on something this big? I am not a PBX/Voice guy, I just do IP up here right now. Any tips, pointers, design guides, or advice to give? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ 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] Organization wide
How can we save? J -Tim -Original Message- From: Brandon Patterson (peering) [mailto:[EMAIL PROTECTED]] Sent: Friday, September 10, 2004 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Organization wide Cisco = Big $ email me direct and I will explain how you can save. Brandon - Original Message - From: Tim Jackson To: [EMAIL PROTECTED] Sent: Friday, September 10, 2004 4:03 PM Subject: [Asterisk-Users] Organization wide After our department went to using *, Ive had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now were using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have 29 POTS lines going into the NEC system. At another location we have a single PRI, and at a lot of the other locations we have just analog phones. Cisco has approached us about using all Cisco equipment, but their idea is going to be costly. Is it wise to use Asterisk on something this big? I am not a PBX/Voice guy, I just do IP up here right now. Any tips, pointers, design guides, or advice to give? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ 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] Organization wide
Woops, wasn't supposed to go to the list ;) -Tim -Original Message- From: Tim Jackson Sent: Friday, September 10, 2004 5:16 PM To: Brandon Patterson (peering); Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Organization wide How can we save? H -Tim -Original Message- From: Brandon Patterson (peering) [mailto:[EMAIL PROTECTED] Sent: Friday, September 10, 2004 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Organization wide Cisco = Big $ email me direct and I will explain how you can save. Brandon - Original Message - From: Tim Jackson To: [EMAIL PROTECTED] Sent: Friday, September 10, 2004 4:03 PM Subject: [Asterisk-Users] Organization wide After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have 29 POTS lines going into the NEC system. At another location we have a single PRI, and at a lot of the other locations we have just analog phones. Cisco has approached us about using all Cisco equipment, but their idea is going to be costly. Is it wise to use Asterisk on something this big? I am not a PBX/Voice guy, I just do IP up here right now. Any tips, pointers, design guides, or advice to give? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ 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] Organization wide
My current asterisk box is a Quad Xeon 450 (2mb cache) IBM Netfinity 7000. About how many SIP extensions (normal usage) would this machine handle? What about redundancy? How would I implement an auto-failover Asterisk box at a remote location, or could I? Thanks, Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 10, 2004 5:40 PM To: Tim Jackson Subject: Re: [Asterisk-Users] Organization wide Tim Jackson [EMAIL PROTECTED] writes: After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have 29 POTS lines going into the NEC system. At another location we have a single PRI, and at a lot of the other locations we have just analog phones. Cisco has approached us about using all Cisco equipment, but their idea is going to be costly. Is it wise to use Asterisk on something this big? I am not a PBX/Voice guy, I just do IP up here right now. Any tips, pointers, design guides, or advice to give? Cisco hardware and software is amazingly expensive. You can save lots of money by using asterisk, digium hardware, and possibly Cisco phones. I recently helped our ~200 person, 10 location company migrate away from an entire Cisco solution to one using asterisk with Cisco handsets. Not only is it vastly cheaper, but it is a much easier system to manage and maintain. I highly recommend you look further into an asterisk system. The only thing that stands out that might not work so well is the 29 pots lines in a single location. Ideally you could install a PRI in this location, but if not you'll need some other less common hardware to handle all those lines. On the IP side, the calls don't actually use up that much bandwidth, probably 30kbits/sec/call if you use ILBC. The only thing you need to do is make sure that all the RTP packets are delivered with a higher priority. Either custom queuing or bandwidth reservation or both will make everyone's life better. -- Matt Ranney - [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
RE: [Asterisk-Users] Re: RE: Organization wide
Well, 1GB is what it has now, I can up it to 4GB but I think that's over kill ;). The coolest part about this machine is 3 PCI busses. 2 64bit and 1 32bit. I'm assuming that this would make IP through the machine quite a bit more robust. Since these machines can be had for $500-600 refurbed it is what I was looking at using for the PBX machines. 3x power supplies in them, no RAID as of now, but it's a matter of putting in a card (hot-swap drives already) I'm assuming the best way for failover is to have identical dialplans on the machines (using IAX trunks?). What about synchronizing SIP extensions between machines? Use MySQL? Like you mentioned about primary and secondary sip proxies on phones, what's the best way to notify the phone that the proxy has changed? Use DNS with some sort of scripting? -Tim -Original Message- From: Jason Kawakami [mailto:[EMAIL PROTECTED] Sent: Friday, September 10, 2004 8:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: RE: Organization wide - Original Message - My current asterisk box is a Quad Xeon 450 (2mb cache) IBM Netfinity 7000. About how many SIP extensions (normal usage) would this machine handle? think about concurrent connections not how many extensions. there are no hard and fast rules but i would think that the box you have described could support +- 70 concurrent connections. up the RAM to the max the box can handle and you could squeeze out a few more maybe 10-15. What about redundancy? How would I implement an auto-failover Asterisk box at a remote location, or could I? you should think about having an * server at each distinct location and using IAX trunks to tie everything together. i wouldnt think about doing this from one box. also, i am not sure about whether you can have a primary and secondary SIP proxy with any openly available phones out there. snip The only thing that stands out that might not work so well is the 29 pots lines in a single location. Ideally you could install a PRI in this location, but if not you'll need some other less common hardware to handle all those lines. i agree with matt here. probably have so many pots because the system in place didnt support t-1/pri. just a guess but a t-1/pri is most likely less expensive from a MRC perspective. On the IP side, the calls don't actually use up that much bandwidth, probably 30kbits/sec/call if you use ILBC. The only thing you need to do is make sure that all the RTP packets are delivered with a higher priority. Either custom queuing or bandwidth reservation or both will make everyone's life better. echo here Jason Kawakami ___ 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] Polycom IP500 vs Cisco 7940
A local vendor here carries IP500s for sub $200. Right now they are out of stock, but he has more coming in. If you want his contact info msg me off list. -Tim -Original Message- From: Ty Purcell [mailto:[EMAIL PROTECTED] Sent: Thursday, September 09, 2004 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 In July I bought one from CDW for $280.75. Ty -Original Message- From: hank smith [mailto:[EMAIL PROTECTED] Sent: Thursday, September 09, 2004 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 what is the price range in us dollars? - Original Message - From: Jody N. Rudolph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:32 AM Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ 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] which distro for asterisk?
I'm using it on Debian Stable (Woody), works great, using it with the backports.org 2.6.7-1-686-smp kernel, zaptel drivers compile fine with their headers. I think it's all a matter of personal preference. I prefer Debian, so I use it, use whatever you like best :) -Tim -Original Message- From: Deon Rodden [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] which distro for asterisk? We discussed this earlier and I believe the general consensus was that it's personal choice. I've personally used Asterisk on Redhat 9.0, Fedora Core 1 and Gentoo 2004.2 Each has required some minor securing and cleaning up, but Redhat/Fedora tended to need more babying as far as securing default configs and speeding up certain things. But after a little bit of time with it, I had it running efficient and Asterisk loved it. The main problem I found with Redhat/Fedora was the default kernel, the sources are all messed up and the zaptel/libpri drivers didn't compile quite right. I simply downloaded the latest kernel, compiled it (and set it to my specific hardware to conserve memory) and libpri/zaptel compiled fine. Personally I'm now using Gentoo. What I did with Gentoo was emerge asterisk-0.9.0 or whatever, and it handled all the dependencies for more. When I was done with that, I did a emerge cvs and once I had cvs I downloaded, compiled and upgraded to the latest libpri/zaptel/asterisk, just did a make upgrade. Gentoo tends to be a faster OS and more efficient with resources, it gives you bare minimum. Which sometimes is good, for security and such, but also a pain when it comes to standard interaction. ie by default they don't include ftp or telnet and traceroute just commands I'm used to having, nothing I can't emerge though. In the end, it's personal choice. For now I've gone with Gentoo. Asterisk really isn't that resource intensive, it seems to like memory a lot, but other than that I don't see it putting heavy loads on my systems, and the speed difference between two identical machines, one with with Fedora Core 1 and one with Gentoo, is almost imperceivable when it comes to working with Asterisk, (ie, the IVR and playback of messages, and interacting with the voicemail system, etc.). Tzafrir Cohen wrote: Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS maintinance task. I really wouldn't like to start worrying about upgrading sshd due to some stupid secuirty hole, and to worry what will it break on my system. I expect my distro to do that for me. I'd also like to have solid astrisk packages that won't break unnecessarily when the sshd package is updated next time. Hopefully also some sort of integration of zaptel in the distro's kernel package. I saw numerous complaints about unofficial RPM packages of asterisk. Besides them, the following free distros include asterisk packages: 1. Debian: http://packages.debian.org/asterisk . 2. Gentoo: Current package seems to be version 0.9.0 from 10-May-2004 3. The DAG repository for RH/Fedora: http://dag.wieers.com/packages/asterisk/ I have some experince with Debian, Mandrake and RedHat/Fedora. I'm unfamiliar with Gentoo and I have no good/bad experince with DAG packages with respect to quality and stability. Any recommendations, relevant experince and other learned opinions? thx ___ 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] Re: New to Asterisk and a question
I recently dug into this, from what I've seen, the best bang for the buck out there is going to be Polycom's. A local vendor has Polycom IP500 phones for $174 shipped to me. IP500 would be comparable to a 7940G I'm assuming. I ran into the same problem with pricing, don't want grandstreams, but can't afford the nice Ciscos. Check out the Polycom's. -Tim -Original Message- From: Brad Stockdale [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 7:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: New to Asterisk and a question Greetings all, I have been watching Asterisk for a while now, but haven't had the nerve to jump in and start playing until now... I'm fed up with our phone system (or lack thereof) at my office, so I decided to start seriously looking at Asterisk... Mostly as a plaything to get my hands on it, but with the goal of making our phone system here somewhat bearable... We have four incoming POTS lines, which I am going to purchase a TDM400P with four FXO modules to handle... I saw a bundle on Digiums website that I think will suit this requirement nicely... I want to use some Cisco IP phones for each of our desks (there's four of us here)... This is where my question comes in... Does anyone have any recommendations for a model of Cisco IP phone to use? I cannot afford anything expensive -- we're a small company, and to be honest this whole project will be coming out of my personal pocket (I'm part owner of the company, so I can do that and not feel too bad about it. ;) )... Been watching Ebay and saw that Cisco 7940G phones, while expensive, aren't totally out of the question... The next rung down looks ok too (7912G)... Does anyone have any firm objections or praise for these two models? Is there some other make and model that I should look at with similar functionality that may work better with Asterisk? I'm assuming that if the Cisco phones are OK to use, I will have to make them work with SIP? Just looking for opinions on the interoperability of Cisco IP phones and Asterisk. Thanks! Brad ___ 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] xlite Problems
-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and asterisk crashes. I'm running the CVS from yesterday. Any ideas? Here's the sip.conf 1009 is identical: [101] type=friend callerid=Tim Jackson 100 host=dynamic dtmfmode=rfc2833 nat=no; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT context=default disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ 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] SMP Performance
25 should be the max ever. This machine used to be my testbed server. I may end up swapping it out later for a 1U IBM, but I just wanted to make sure that in the meantime it'd be able to handle what we are doing with it. We bought it refurbished for $600 about a year ago. I was just wondering about the SMP part, I've been told that it doesn't work well with SMP, and then I've been told it works fine. I just wanted a 2nd or 3rd opinion before I went ahead and implemented this. Another dumb question, I've gotten the idea that the best phones out there are the Cisco 7960s, any other good phones out there that are decently priced? Nortel? 3Com? -Tim -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SMP Performance There is nothing wrong with running Asterisk on SMP. It runs quite well actually. I'm assuming you just have the Quad Xeon 450mhz sitting around because you can't buy them new anymore, so it probably isn't costing you anything to use it. In which case it isn't a waste. If you are paying more than $800 for it, save it and just buy a new P4 for less. A $200 machine may not be able to handle 25 concurrent conversations, and may have some used or sub-standard parts in it, so that may not be the best choice. You should be able to have upto 25 channels running on this machine no problem, How many maximum conversations do you forsee running concurrently at one time on this system? MATT--- -Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SMP Performance Meaning Asterisk won't/can't take advantage of the four CPU's? Or it's overkill for this scenario? -Original Message- From: joachim [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 12:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SMP Performance Send me the quad and i'll send you a 200$ pc to do this job. The quad is heavily overpowered. Joachim. At 22:00 24/08/2004, you wrote: content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C48A15.130BF232 We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ 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 ___ 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] SMP Performance
Were looking at implementing Asterisk in our department in the near future, were looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. Ive heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ 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] Dell PowerEdge 750 rackmount
What I have found are the best servers, are IBM xSeries machines. You can obtain a Refurb xSeries 305 P4 2.4ghz for around $750-800. I have never had the first problem out of these boxes. SuperMicro is also good, but the IBMs tend to be a little smaller (a lot less deep) than SuperMicro 1U machines. -Tim -Original Message- From: Steve Szmidt [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 1:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge 750 rackmount -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 24 August 2004 01:02 pm, Steven Critchfield wrote: The big thing to look into is what PCI busses the machine supports. We where very surprised with our Dell when it came with a PCIX slot and a 66mhz 64bit slot. The included ethernet card wasn't directly supported by our install disks and we couldn't install a cheap card to get the install bootstrapped due to the incompatible slots. So after building a special boot disk, all was better except the T100P card I owned wouldn't work in it again due to slot incompatibility. That prompted our TE410P card purchase. After using various Dell's for a few years I'm very weary if I need something not bland. They are very very good at cutting corners at all sorts of places. Their intended public is mostly run of the mill machines. They are the pro's in cutting pennies and turn it into BIG savings. Too many times have I discovered something odd or less than I would have expected. SuperMicro on the other hand have striken a good balance of price and quality. They are really are going after the market with some very solid hardware and not being the cheapest either. Which is really less important in a business environment anyway. Support and quality being more important. It's really true that you get what you pay for. If you can afford quality it's cheaper in the long run. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBK45fljK16xgETzkRAtG+AJ9dStuHpzMSxiQFafCD1SSToRF+TACgknUl yL4OmWEaSPs+abhFM3i1S2Y= =lIxc -END PGP SIGNATURE- ___ 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
