Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information
Sigma Networks wrote: I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones composed of 7970, 7960 and 7906 phones. I've upgraded all of these to SIP 8-3-3SR2S and the basic functions are working. My current questions are: 1. How to remotely reboot 7970s. I have both web access and SSH access to the phones. The instructions I have for SSH are to use (1) user/pass (or whatever is in the confg) and then (2) debug/debug. Surprisingly reset is not a valid command to restart the phone. There doesn't appear to be a reset on the web page, maybe there's a hidden URL? 2. BusyLampField? Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We have about 200 79x1's running SIP w/ asterisk and we are very pleased despite some of the non-standard things Cisco does. In answer to question 1 the only way we have found to reboot the phone remotely is shutdown the port on the POE switch. This will drop the PC's network as well if it is plugged into the phone. Question 2 I would like to know the answer to myself. I would be curious to know if it works with the SIP image in call manager. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which phones to use ??
On Sun, Mar 2, 2008 at 4:26 AM, C F [EMAIL PROTECTED] wrote: 1. Way to slow to boot 2. Lack of features, can't reconfigure the buttons to show something decent, like BLF, and the buttons you could configure are limited even though they are soft buttons. Compare that to the Aastra 480i I'd like to, maybe Aastra will loan me one? In the meantime, I hate the whole Polycom menu and boot slowness and general interface, but yeah, once booted, they sound good and are solid phones. I really like my Sipura, but I hear good things from owners of both Snom and Aastra, I just haven't worked with either yet. This said, and I have given a few speeches on this already to small business people, buy the phone that your users want. The one that will make them most productive and comfortable. If it's just for you, buy the phone you can afford. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which phones to use ??
On Sun, Mar 2, 2008 at 4:26 AM, C F [EMAIL PROTECTED] wrote: 1. Way to slow to boot 2. Lack of features, can't reconfigure the buttons to show something decent, like BLF, and the buttons you could configure are limited even though they are soft buttons. Compare that to the Aastra 480i I'd like to, maybe Aastra will loan me one? In the meantime, I hate the whole Polycom menu and boot slowness and general interface, but yeah, once booted, they sound good and are solid phones. I really like my Sipura, but I hear good things from owners of both Snom and Aastra, I just haven't worked with either yet. This said, and I have given a few speeches on this already to small business people, buy the phone that your users want. The one that will make them most productive and comfortable. If it's just for you, buy the phone you can afford. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID number
hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. I am experimenting with asterisk, for home use. thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
On Sun, 2 Mar 2008, Mike wrote: hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. I am experimenting with asterisk, for home use. Telling people what country you're in will really help here. If you're in the UK, I'll give you a free number. (But it won't be free to call ;) Most UK and European ITSPs will give out free inbound numbers, geographic or non-geo (which are often revenue generating for the ITSP) or maybe you need to put a few pounds/euros of call credit in your account first. Just google... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information
Sean Dennis ha scritto: Sigma Networks wrote: ... My current questions are: 1. How to remotely reboot 7970s. I have both web access and SSH access to the phones. The instructions I have for SSH are to use (1) user/pass (or whatever is in the confg) and then (2) debug/debug. Surprisingly reset is not a valid command to restart the phone. There doesn't appear to be a reset on the web page, maybe there's a hidden URL? 2. BusyLampField? ... We have about 200 79x1's running SIP w/ asterisk and we are very pleased despite some of the non-standard things Cisco does. In answer to question 1 the only way we have found to reboot the phone remotely is shutdown the port on the POE switch. This will drop the PC's network as well if it is plugged into the phone. Question 2 I would like to know the answer to myself. I would be curious to know if it works with the SIP image in call manager. Same here. We have about 500 phones, from both 79x1 and 79x0 series; I posted the same two questions twice some time ago but never got an answer: I do reboot phones by power cycling them too, while I've been able to use blf with sccp images only. Furthermore, XML Services on 7940/7960 seem to be broken or at least to behave in different way than the one described in the sdk documentation. I needed the reboot feature to implement extension mobility but I wasn't able to find a clean way. Power cycling is not always an usable method, as many phones are powered by the AC adaptor. I think I will able to put my hands on an UCM6.1 box very soon to try that out and eventually grab the xml profiles. As soon as I get the info I'll surely post it on this ML and on voip-info too. Alberto. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
try sipgate.co.uk M On Sun, Mar 2, 2008 at 8:21 PM, Mike [EMAIL PROTECTED] wrote: hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. I am experimenting with asterisk, for home use. thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- regards, Mark P. Edwards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7965g and asterisk
On Fri, Feb 29, 2008 at 02:12:18PM +0100, Patrick wrote: On Fri, 2008-02-29 at 11:09 +, Nuno Fernandes wrote: Hi, We've just bought a new cisco 7965g and web are trying to connect it to asterisk. I've bought smartnet and downloaded [snip] How can i install the sip firmware? You need to setup a tftp server, put the 8 sip firmware files and the configuration files in the tftp server directory so the Cisco phone can pick them up when it boots. I've setup an tftp server (the same one i used for other cisco phones). Do i require some other file(s)? The Cisco phone can be very picky about the configuration files. With the slightest error the phone will refuse to boot so make sure you have got it all right. I don't have configuration files for the 7965 so ask the company where you bought the phone or google around. If you can't find them for the 7965 please note that the 7940/7960 configuration files will not work for a 7965. Maybe the 7941/7961 configuration files will. I'm not sure. Does anyone have some example of a conf file for this phone? Thanks Nuno Fernandes If you are mainly using Asterisk (SIP) then I recommend you buy Polycom, Aastra or Snom phones next time. The Polycom phones have the best sound quality and imho are the best SIP phones you can buy. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nuno Pais Fernandes Eurotux Informática, SA Tel: +351 253257395 Fax: +351 253257396 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] override/redefine asterisk DB function
Hi.
Is it possible to override the standard DB function in
Asterisk?
My dialplan contains a lot of calls to Set(DB(...))
and ${DB(...)} which of course use astdb to
store/read data. I would like to stop using astdb and
switch to Clustered MySQL (I don't suppose clustered
astdb exists?).
So instead of rewriting extensions.conf and replacing
the DB calls with MYSQL calls, would it be possible to
just user-define the DB function so that I can leave
the extensions file intact but make the appropriate
MYSQL function calls?
Thanks.
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Re: [asterisk-users] manager ignore my settings
On Wed, Feb 20, 2008 at 08:39:07PM +0200, ik wrote: Hello, I have the following settings for manager on two Asterisk 1.2.24 (that have installed over a year ago): [user] secret = password deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 write = call,command On one server, Asterisk only react as you would expect - sending a command without having any verbose on anything. On the other machine, I have verbose like I enabled everything, including the read option. So, what other differences are there between those two servers? What do you have in the [general] section of manager.conf in both? Do both use the same platform? Same version of Asterisk? Another issue I have on the weird machine is that I have unexplained crash (where safe asterisk return asterisk to life), the core dump each time is different, but one thing is in common: it all fails on a free command. Any ideas what might cause this issues, and what should I be looking for ? This is kind of meaningless. Can you post some backtraces? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - CEBIT next week!
Any Asterisk people going to Cebit ? Let's meet! If you go and would like to go for a drink and meet some others from the voip business, please add your name to the list below Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com - wednesday / thursday. Tan Aksoy - Telappliant - wednesday / thursday skyler??? - Digium [add name here:P] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] override/redefine asterisk DB function
On Sunday 02 March 2008 05:33:49 Vieri wrote: Is it possible to override the standard DB function in Asterisk? No. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] real zaptel call durations
Thanx alot for reply I mean i have to use the fxo to connect to the pstn line and i do not know if there is any asterisk functions ,Application, options that could help to know what is the real call duration [ how to deal with pstn line signaling how to detect the pstn ringing tone or pstn auto-machinese voice message in case if the user did not answer the call ] , i saw some billing software and i am not sure if they are calculating the bills using cdr in case of using fxo. thank u in advance ayman _ Helping your favorite cause is as easy as instant messaging. You IM, we give. http://im.live.com/Messenger/IM/Home/?source=text_hotmail_join___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 - register with NAT phone
continuing discussions of 79xx issues. i've seen referenced and am
experiencing difficulty getting a 7970 to work behind NAT to a public
asterisk server. i am successful with 7960s.
1. SIP load is 70.8-3-3SR2S
2. config works fine if 7970 is connecting to an asterisk server a
local LAN (same subnet)
3. when debugging it in a NAT'd environment I see the register and
OK to the phone from the public asterisk server, but the phone
continues to show the phone as unregistered.
any thoughts would be appreciated.
device xsi:type=axl:XIPPhone ctiid=203849429
uuid={96f8508b-10ef-f98c-d20d-0471777ec725}
fullConfigtrue/fullConfig
deviceProtocolSIP/deviceProtocol
sshUserIduser/sshUserId
sshPassword/sshPassword
devicePool uuid={a755aa55-089c-2b47-9603-c7d51b9ca4b5}
nameDallas 5.0 Beta/name
dateTimeSetting uuid={9ec4850a-7748-11d3-bdf0-00108302ead1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZonePacific Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
name5.0 Beta/name
tftpDefaulttrue/tftpDefault
members
member priority=0
callManager
nameccm-beta-5-1/name
descriptionCallManager 5.0 Beta Pub - 5.0.1.032/description
ports
ethernetPhonePort2000/ethernetPhonePort
sipPort5060/sipPort
securedSipPort5061/securedSipPort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeNameccm-beta-5-1/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={cd241e11-4a58-4d3d-9661-f06c912a18a3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1206.80.94.20/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
sipIpAddr1206.80.94.20/sipIpAddr1
sipPort15060/sipPort1
sipIpAddr2/sipIpAddr2
sipPort25060/sipPort2
sipIpAddr3/sipIpAddr3
sipPort35060/sipPort3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
sipProfile
sipProxies
backupProxyx.x.x.x/backupProxy
backupProxyPort5060/backupProxyPort
emergencyProxyx.x.x.x/emergencyProxy
emergencyProxyPort5060/emergencyProxyPort
outboundProxyz.z.z.z/outboundProxy
outboundProxyPort5060/outboundProxyPort
registerWithProxytrue/registerWithProxy
/sipProxies
sipCallFeatures
cnfJoinEnabledtrue/cnfJoinEnabled
callForwardURIx-cisco-serviceuri-cfwdall/callForwardURI
callPickupURIx-cisco-serviceuri-pickup/callPickupURI
callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI
callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI
meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI
abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI
rfc2543Holdfalse/rfc2543Hold
callHoldRingback2/callHoldRingback
localCfwdEnabletrue/localCfwdEnable
semiAttendedTransfertrue/semiAttendedTransfer
anonymousCallBlock2/anonymousCallBlock
callerIdBlocking2/callerIdBlocking
dndControl0/dndControl
remoteCcEnabletrue/remoteCcEnable
/sipCallFeatures
sipStack
sipInviteRetx6/sipInviteRetx
sipRetx10/sipRetx
timerInviteExpires180/timerInviteExpires
timerRegisterExpires3600/timerRegisterExpires
timerRegisterDelta5/timerRegisterDelta
timerKeepAliveExpires120/timerKeepAliveExpires
timerSubscribeExpires120/timerSubscribeExpires
timerSubscribeDelta5/timerSubscribeDelta
timerT1500/timerT1
timerT24000/timerT2
maxRedirects70/maxRedirects
remotePartyIDtrue/remotePartyID
userInfoNone/userInfo
/sipStack
autoAnswerTimer1/autoAnswerTimer
autoAnswerAltBehaviorfalse/autoAnswerAltBehavior
autoAnswerOverridetrue/autoAnswerOverride
transferOnhookEnabledfalse/transferOnhookEnabled
enableVadfalse/enableVad
preferredCodecnone/preferredCodec
dtmfAvtPayload101/dtmfAvtPayload
dtmfDbLevel3/dtmfDbLevel
dtmfOutofBandavt/dtmfOutofBand
alwaysUsePrimeLinefalse/alwaysUsePrimeLine
alwaysUsePrimeLineVoiceMailfalse/alwaysUsePrimeLineVoiceMail
kpml3/kpml
phoneLabelTest2/phoneLabel
stutterMsgWaiting2/stutterMsgWaiting
callStatsfalse/callStats
offhookToFirstDigitTimer15000/offhookToFirstDigitTimer
silentPeriodBetweenCallWaitingBursts10/silentPeriodBetweenCallWaitingBursts
disableLocalSpeedDialConfigtrue/disableLocalSpeedDialConfig
startMediaPort16384/startMediaPort
stopMediaPort32766/stopMediaPort
sipLines
line button=1
featureID9/featureID
featureLabellabel/featureLabel
proxyz.z.z.z/proxy
port5060/port
namename/name
displayNameKerry/displayName
autoAnswer
autoAnswerEnabled2/autoAnswerEnabled
/autoAnswer
callWaiting3/callWaiting
authNamezz/authName
authPassword555/authPassword
sharedLinefalse/sharedLine
messageWaitingLampPolicy3/messageWaitingLampPolicy
messagesNumber*97/messagesNumber
ringSettingIdle4/ringSettingIdle
ringSettingActive5/ringSettingActive
contact7b452e87-4496-4762-e11f-b26751a1884b/contact
forwardCallInfoDisplay
callerNametrue/callerName
callerNumberfalse/callerNumber
redirectedNumberfalse/redirectedNumber
dialedNumbertrue/dialedNumber
/forwardCallInfoDisplay
/line
line
Re: [asterisk-users] which phones to use ??
On Sat, 1 Mar 2008 22:26:18 -0500, C F wrote: On Sat, Mar 1, 2008 at 9:31 AM, Michael Graves [EMAIL PROTECTED] wrote: When in doubt there is only one sure answerPolycom. Without a doubt the best functionality, performance and reliabilityeven in the lower cost models. Although the lesser models are still over $100. While I agree with you that once they have booted and are configured the way one wants, they are the best. But here are the downsides on them: 1. Way to slow to boot 2. Lack of features, can't reconfigure the buttons to show something decent, like BLF, and the buttons you could configure are limited even though they are soft buttons. Compare that to the Aastra 480i Here are the upsides which in my opinion makes it still the best SIP phone, and the best for your money even though it's the most expensive: 1. Boots reliable. 2. LAN side Ethernet does NOT go down on a reboot. 3. Very reliable and good sound quality, with no tricks attached, just works without playing with configuration files to adjust volume etc. 4. Extremely easy to use, you only have to teach an end user what the difference between a blind and attd xfer is, and they know how to use it. 5. They stay on, unless you reboot them manually. Unlike ANY other phone out there that I tried, it's the only one that didn't reboot or froze when not asked to. The following phones did freeze or reboot out of the blue: Aastra, Cisco, Sipura, GS and Snom. Although I have seen on certain firmwares that it's not responsive to a certain command under certain conditions, I have never seen them lock up completely. As far as the config files go, at this point I don't consider them hard, since what I'm trying to do is doable and they have great documentation. The XML files are just confusing because of the layout, searching within vi is not that hard. However, it's not something that is easy to teach oneself overnight. Compare that to Cisco or Aastra config files and you have a nightmare. For the longest time I didn't even bother with config files for my Polycom phones. I only had a few so I used the web interface. And I hated it. Still do. However, once I came into a situation that demanded that I use the config files I found the phone was much easier to deal with. Using the confg files was worth ethe effort. It allowed me to appraciate the phones more. I have some Aastra phones are a really like them, especially the 480iCT. I've used some older snom's...they're cool, too. I'm not sure why people dwell on the boot time so much. I only reboot the phones VERY rarely. I can't recall the last time I had to. Unlike some others, the manage to do it on their own periodically. I've never used Cisco phones for simple cost reasons although the colour backlit LCD is enticing. Nor Linksys, but I should get a couple just to play with. The 942 has a colour screen too right? I'm not really a Polycom fanboy. I'm very into the snom m3 for portablility. I just think that when you need someting that works, reliably, all the time...it's a short list. And for me Polycom is on the top of it. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need the least expensive way to do this
On Sat, 01 Mar 2008 18:34:36 -0600, Timothy C Litwiller wrote: Everyone seems to think a used proprietary system would be better - I looked on ebay for over a week and never found something that indicates that you can conference and have voicemail. In this day and age how can you get along without that. and for a small school without a diedicated person answering the phone I would think that an IVR would be critical. I didn't find anything on ebay that had that for less than 3000. For the kind of small arrangement you've described you could find that the Panason KX-TG4500 is suitable. It's a two or four line system. The main phone has the FXO ports. The rest of the phones are cordless 5.8GHz extensions. It can have up to 8 extensions. They come in handheld handset ans desktop handset form factors. The basic system with one portable is around $400 new. The extra handsets are around $130 each. The base unit has built-in battery backup. Tr here: http://www.101phones.com/cat/1971/Panasonic-4-Line-Multi-Handset-Phone-S ystems.html I've owned one of these for a long time. It's easy to administer. Has on-board VM, hold, conference, transfer, etc. It's an all-in-one solutioon for small operations. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] real zaptel call durations
On Sun, Mar 02, 2008 at 08:30:17AM -0500, aymen warfalli wrote: Thanx alot for reply I mean i have to use the fxo to connect to the pstn line and i do not know if there is any asterisk functions ,Application, options that could help to know what is the real call duration [ how to deal with pstn line signaling how to detect the pstn ringing tone or pstn auto-machinese voice message in case if the user did not answer the call ] , Again: does the PSTN provider provide you that information? through polarity reversal at call start and end? By any other means? If not: no. Guessing this is very tricky, and Asterisk does not attempt to. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information (was: Re: asterisk-users Digest, Vol 44, Issue 3)
The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount of experts in this asterisk-users list. And there is a fair amount of info in lots of different voip-info wiki pages (and elsewhere, including the Cisco website labyrinth). If people in this discussion could update the wiki pages with current and more complete info, others (like me, with less info to contribute but willing to edit for usability) could revise the wiki pages to be more accessible and less redundant/isolated. Announcing wiki revisions on this -users list also makes them easier to find, especially since it gives Google two pages to find. The wiki pages (there are more which at least mention some detail relevant to a 79xx phone; please link them together if you add info to a different one): Cisco http://www.voip-info.org/wiki/view/Cisco Cisco phones http://www.voip-info.org/wiki/index.php?page=Cisco+Phones Cisco Phones Table http://www.voip-info.org/wiki/view/Cisco+Phones+Table Cisco Phone Headsets http://www.voip-info.org/wiki/view/Cisco+Phone+Headsets Asterisk phones http://www.voip-info.org/wiki/index.php?page=Asterisk%20phones cisco 79xx http://www.voip-info.org/wiki/index.php?page=cisco+79xx Asterisk phone cisco 79xx http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx Covad Voip phone Cisco 79xx http://www.voip-info.org/wiki/index.php?page=Covad+Voip+phone+cisco+79xx Configuring Cisco 12SP phones with Asterisk http://www.voip-info.org/wiki/view/Configuring+Cisco+12SP+phones+with +Asterisk Cisco 7905/7912 IP Phones http://www.voip-info.org/wiki/view/Cisco+7905%252F7912+IP+Phones Cisco 7940 7960 Single Step Upgrade http://www.voip-info.org/wiki/index.php?page=Cisco+7940+7960+Single+Step +Upgrade Cisco 7940 7960 upgrade to version 7.x http://www.voip-info.org/wiki/view/Cisco+7940-7960+upgrade+to+version +7.x Firmware issues on 7940 - 7960 http://www.voip-info.org/wiki/index.php?page=Firmware+issues+on+7940 +-+7960 Setup SiP on 7940 - 7960 http://www.voip-info.org/wiki/index.php?page=Setup+SiP+on+7940+-+7960 Asterisk phone cisco 7970 SIP http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970 +SIPview_comment_id=11255 Asterisk phone cisco 7970 SIP http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP Cisco 7940-60 disassembly http://www.voip-info.org/wiki/view/Cisco+7940-60+disassembly SCCP http://www.voip-info.org/wiki/view/SCCP SCCP HOWTO2 http://www.voip-info.org/wiki/index.php?page=SCCP-HOWTO2 Asterisk SCCP channels http://www.voip-info.org/wiki/index.php?page=Asterisk+SCCP+channels chan sccp2 http://www.voip-info.org/wiki/view/chan_sccp2 Asterisk Skinny channels http://www.voip-info.org/wiki/edit.php?page=Asterisk+Skinny+channels Asterisk Cisco 79XX XML Services http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services Cisco 79XX XML Push http://www.voip-info.org/wiki/view/Cisco+79XX+XML+Push Asterisk phone cisco 79x1 xml configuration files for SIP http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml +configuration+files+for+SIP Cisco 7940-7960 auto-answer config http://www.voip-info.org/wiki/view/Cisco+7940-7960+auto-answer+config Cisco 7940-7960 Daylight Savings http://www.voip-info.org/wiki/view/Cisco+7940-7960+Daylight+Savings Asterisk Linksys NSLU2 http://www.voip-info.org/wiki/view/Asterisk+Linksys+NSLU2 Asterisk Manager AM-WEB http://www.voip-info.org/wiki/view/Script+to+page+mixed+SIP+%252F+SCCP +system Script to page mixed SIP / SCCP system http://www.voip-info.org/wiki/view/Script+to+page+mixed+SIP+%252F+SCCP +system QoS Cisco - voip-info.org http://www.voip-info.org/wiki/view/QoS+Cisco Standalone Cisco 7941/7961 without a local PBX http://www.voip-info.org/wiki/view/Standalone+Cisco+7941%252F7961 +without+a+local+PBX Asterisk Cisco CallManager Express Integration - voip-info.org http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express +Integration Exposing the Cisco Call Manager License Scam http://www.voip-info.org/wiki/view/Exposing+the+Cisco+Call+Manager +License+Scam Cisco POE http://www.voip-info.org/wiki/index.php?page=Cisco+POE SSH with PuTTY to Cisco IP Phone http://www.voip-info.org/wiki/view/SSH+with+PuTTY+to+Cisco+IP+Phone+ On Sun, 2008-03-02 at 07:30 -0600, [EMAIL PROTECTED] wrote: Date: Sun, 02 Mar 2008 11:38:20 +0100 From: Alberto Pastore [EMAIL PROTECTED] Subject: Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Sean
Re: [asterisk-users] TDM400P dialout problem
Have the some symptoms mentioned in http://bugs.digium.com/view.php?id=12099 After upgrade to Zaptel 1.4.9.2 I can't dial out at all, with 1.4.8 there were just random dial-out problems. -- Executing [EMAIL PROTECTED]:1] Answer(SIP/210-081e9968, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/210-081e9968, Zap/3) in new stack -- Called 3 [2008-03-02 09:43:17] WARNING[5051]: chan_zap.c:4114 zt_handle_event: Detected alarm on channel 3: No Alarm -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/210-081e9968, ) in new stack == Spawn extension (Martin, *903, 3) exited non-zero on 'SIP/210-081e9968' -- Executing [EMAIL PROTECTED]:1] Answer(SIP/210-081e9968, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/210-081e9968, Zap/3) in new stack -- Called 3 [2008-03-02 09:44:17] WARNING[5055]: chan_zap.c:4114 zt_handle_event: Detected alarm on channel 3: No Alarm -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/210-081e9968, ) in new stack == Spawn extension (Martin, *903, 3) exited non-zero on 'SIP/210-081e9968' -- Executing [EMAIL PROTECTED]:1] Answer(SIP/210-081e9968, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/210-081e9968, Zap/4) in new stack -- Called 4 [2008-03-02 09:44:29] WARNING[5056]: chan_zap.c:4114 zt_handle_event: Detected alarm on channel 4: No Alarm -- Hungup 'Zap/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/210-081e9968, ) in new stack == Spawn extension (Martin, *904, 3) exited non-zero on 'SIP/210-081e9968' -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 29. února 2008 08:59 Subject: Re: [asterisk-users] TDM400P dialout problem Anthony Messina wrote: i'm looking forward to 1.4.9.2, but am also concerned about http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and 1.4.9.1 on both platforms. The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of Asterisk will no longer generate them, and once my battery_alarms branch has been merged into Zaptel 1.4 (scheduled to be part of the 1.4.10 release) then Zaptel will stop generating spurious battery alarm events. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which phones to use ??
I'll admit in my case, the main reason the boot time of the Polycom drives me nuts is that I don't see the phones unless I'm installing them for the first time or supporting them when something isn't quite right. It's for this reason that I very much appreciate a phone that either (a) boots fairly quickly or (b) doesn't feel the need to reboot for everything except lifting the handset. Both the Linksys and Polycom phones score poorly on this point - both of them restart to some degree with just about every config change, no matter how minor - but more to the point, reboot without asking you first. If I've got half a dozen small changes to make or if you're experimenting with a couple of settings, this gets old /real/ quick. The Snoms are nice here - on those rare occasions the phone needs to restart, it prompts you first - allowing you to continue to make changes if you want. Michael Graves wrote: I'm not sure why people dwell on the boot time so much. I only reboot the phones VERY rarely. I can't recall the last time I had to. Unlike some others, the manage to do it on their own periodically. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
On Sun, Mar 2, 2008 at 3:21 AM, Mike [EMAIL PROTECTED] wrote: Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. Hey Mike - give IPKall a try: http://www.ipkall.com/ They'll give you a free Washington state DID along with free SIP to your asterisk server. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
On Sun, Mar 2, 2008 at 10:21 AM, Mike [EMAIL PROTECTED] wrote: hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. If you are in the USA, see http://www.IPKall.com it's free, works great. Another idea, if they still do this is http://freeworlddialup.com which would provide a FWD nulber and that number can be reached via numerous cities worldwide. You have to enter a bunch of codes, but it's free (local call) and works ok for experimenting. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. That's almost certainly your problem. When you run sidecars with the Polycom 601, you can't rely on PoE - there isn't enough power supplied. Connect your powerpack to the phone and the problem /should/ go away. Semi random reboots are not uncommon on the 601 with sidecars if you're running it on PoE. That makes sense but in my case the 601 w/3 sidecars did not reboot at all and it is run from POE. The 650 just seems to perform much better. JR --- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
Kevin P. Fleming wrote: ... The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of Asterisk will no longer generate them, Using today's svn 3915: .. Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/2375678) in new stack -- Called 4/2375678 [Mar 2 11:15:35] WARNING[2320]: chan_zap.c:4424 zt_handle_event: Detected alarm on channel 4: Red Alarm -- Hungup 'Zap/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(Zap/2-1, ) in new stack [Mar 2 11:15:35] NOTICE[2320]: chan_zap.c:7514 handle_init_event: Alarm cleared on channel 4 == Spawn extension (internal, 2375678, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' zap show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
On Sun, Mar 02, 2008 at 11:36:03AM -0500, sean darcy wrote: Kevin P. Fleming wrote: ... The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of Asterisk will no longer generate them, Using today's svn 3915: .. Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/2375678) in new stack -- Called 4/2375678 [Mar 2 11:15:35] WARNING[2320]: chan_zap.c:4424 zt_handle_event: Detected alarm on channel 4: Red Alarm -- Hungup 'Zap/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(Zap/2-1, ) in new stack [Mar 2 11:15:35] NOTICE[2320]: chan_zap.c:7514 handle_init_event: Alarm cleared on channel 4 == Spawn extension (internal, 2375678, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' zap show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) This shows alarms on the whole span (the card). The alarms in question are alarms on specific channels. The whole interface makes more sense in the other meduims where the span usually corresponds to one physical meduim. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
Tzafrir Cohen wrote: On Sun, Mar 02, 2008 at 11:36:03AM -0500, sean darcy wrote: Kevin P. Fleming wrote: ... The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of Asterisk will no longer generate them, Using today's svn 3915: .. Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/2375678) in new stack -- Called 4/2375678 [Mar 2 11:15:35] WARNING[2320]: chan_zap.c:4424 zt_handle_event: Detected alarm on channel 4: Red Alarm -- Hungup 'Zap/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(Zap/2-1, ) in new stack [Mar 2 11:15:35] NOTICE[2320]: chan_zap.c:7514 handle_init_event: Alarm cleared on channel 4 == Spawn extension (internal, 2375678, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' zap show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) This shows alarms on the whole span (the card). The alarms in question are alarms on specific channels. The whole interface makes more sense in the other meduims where the span usually corresponds to one physical meduim. I'm a little confused. Are you responding to my including the zap show status command ( which I did just for background), or to the call description? If you are responding to the call description, doesn't the alarm show up specifically on channel 4? In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
sean darcy wrote: In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
Kevin P. Fleming wrote: sean darcy wrote: In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. Sorry, I hadn't seen this mentioned. I'll try it asap. thanks for the lead. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
Kevin P. Fleming wrote: sean darcy wrote: In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. Got it. No more Red Alarms. Which is great. But...now I keep getting incomplete number messages from the co. No trouble on the console: Starting simple switch on 'Zap/2-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/6981000) in new stack -- Called 4/6981000 -- Zap/4-1 answered Zap/2-1 -- Native bridging Zap/2-1 and Zap/4-1 -- Hungup 'Zap/4-1' which shows the correct local number, which can be dialed from a plain telephone. It's as though the Dial command just didn't send some of the digits correctly. Thanks for all the help. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] override/redefine asterisk DB function
Tilghman Lesher [EMAIL PROTECTED] writes: On Sunday 02 March 2008 05:33:49 Vieri wrote: Is it possible to override the standard DB function in Asterisk? No. Is it permitted to modify the astdb outside Asterisk, while Asterisk is running? It is a SQLite file, right? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] override/redefine asterisk DB function
On Sun, Mar 02, 2008 at 11:17:20PM +0100, Benny Amorsen wrote: Tilghman Lesher [EMAIL PROTECTED] writes: On Sunday 02 March 2008 05:33:49 Vieri wrote: Is it possible to override the standard DB function in Asterisk? No. Is it permitted to modify the astdb outside Asterisk, while Asterisk is running? It is a SQLite file, right? Nope. Berkeley DB. Of an old vintage (1.86? the last one before the license change to the sleepycat license). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_ss7 0.10
Thanks for the update. I have Sangoma A104D and wanted to use ss7 signalling. I came accross chan_ss7 but found sifira is not in active development. But is this chan_ss7 stable and can be used in production server implementation. We are going to have 2 to 3 boxes with ss7 signalling using sangoma but I am looking for open source ss7 implementation which is chan_ss7. so need to know about stability and recommendation for using on production server. long term supported solution is libss7 from digium. but this depends on asterisk 1.6 which is not officialy stable chan_ss7 is now developed by www.dicea.dk. http://www.dicea.dk/company/downloads it's used on production servers. it is very stable solution --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
sean darcy wrote: Kevin P. Fleming wrote: sean darcy wrote: In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. Got it. No more Red Alarms. Which is great. But...now I keep getting incomplete number messages from the co. No trouble on the console: Starting simple switch on 'Zap/2-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/6981000) in new stack -- Called 4/6981000 -- Zap/4-1 answered Zap/2-1 -- Native bridging Zap/2-1 and Zap/4-1 -- Hungup 'Zap/4-1' which shows the correct local number, which can be dialed from a plain telephone. It's as though the Dial command just didn't send some of the digits correctly. And incoming calls aren't answered. No ring event. No nothing. svn-3915 incoming calls were answered, but generated a Red Alarm. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speex: complexity, VBR, ABR, CBR, quality
Hi All; If someone used speex and has experience with its settings, then who can help to explain the following: 1) When it is recommended to use VBR (vbr = true)? 2) If there relation between setting the vbr = true and the abr value (for example to be 0 or 1 or 10) and the relation between this value and abr (true / false). 3) Any relation between the quality value and the abr value? 4) Is there any setting that I can use cbr and what will be the vbr setting in that case? 5) What is the difference between the complexity and the quality? 6) What is the relation between the complexity and the vbr? And the effection if abr was determined? Sorry for all these questions, but I readed some webs about speex at C:\0Bilal\0Afkarona\Supplier\PBX\Asterisk\Codec\Speex - Wikipedia, the free encyclopedia.mht and others, but in the end, I was feel still the things are not clear. Any help? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speex: complexity, VBR, ABR, CBR, quality
First hit for speex asterisk settings in Google answers all of your questions. http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf On Sun, Mar 2, 2008 at 5:54 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; If someone used speex and has experience with its settings, then who can help to explain the following: 1) When it is recommended to use VBR (vbr = true)? 2) If there relation between setting the vbr = true and the abr value (for example to be 0 or 1 or 10) and the relation between this value and abr (true / false). 3) Any relation between the quality value and the abr value? 4) Is there any setting that I can use cbr and what will be the vbr setting in that case? 5) What is the difference between the complexity and the quality? 6) What is the relation between the complexity and the vbr? And the effection if abr was determined? Sorry for all these questions, but I readed some webs about speex at C:\0Bilal\0Afkarona\Supplier\PBX\Asterisk\Codec\Speex - Wikipedia, the free encyclopedia.mht and others, but in the end, I was feel still the things are not clear. Any help? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie on VoIP
Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who's calling overseas. 2. How do I secure it? Co'z I have to open it via Public IP. Can I know the port asterisk used assuming I'll use SIP. 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? This questions might annoyed experts. Please bear with me... Thank you. nocph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
My incoming calls work just fine with 1.4.9.2 and TDM400P but I'm completely unable to dial out the line. even this doesn't work: exten=*903,1,Answer() exten=*903,n,Dial(Zap/3) exten=*903,n,Hangup() I use this to pick up the line shared with analog phone. Zap/3 and Zap/4 are connected to PSTN. Sometimes I have luck with Zap/3 but not a single call through Zap/4 -- Executing [EMAIL PROTECTED]:1] Answer(SIP/210-081e9968, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/210-081e9968, Zap/3) in new stack -- Called 3 [2008-03-02 09:43:17] WARNING[5051]: chan_zap.c:4114 zt_handle_event: Detected alarm on channel 3: No Alarm -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) The alarm issue is not just annoyance What about downgrade Zaptel drivers...? Are they corrupted since 1.4.8? Im only afraid older versions won't compile with latest kernels (2.6.24 in my case) ... Sometimes its hard to find working combination of kernel and zaptel :-) I will try the fix mentioned below first Martin - Original Message - From: sean darcy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 2. brezna 2008 17:29 Subject: Re: [asterisk-users] TDM400P dialout problem In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although How can I download this, do I need SVN installed? one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. Got it. No more Red Alarms. Which is great. But...now I keep getting incomplete number messages from the co. No trouble on the console: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
By default cdr is stored in a delimited fille, but can easily be put into a mysql table. Securing it is a good question. Asterisk is secure on its own, and I would secure the rest like ssh by changing the default ports and such. I don't know about bsd. You will fall in love with asterisk though, enjoy. Sent from my Verizon Wireless BlackBerry -Original Message- From: NOC Ph [EMAIL PROTECTED] Date: Mon, 3 Mar 2008 10:14:02 To:[EMAIL PROTECTED], asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie on VoIP Hi Guys, I’m new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I’m planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who’s calling overseas. 2. How do I secure it? Co’z I have to open it via Public IP. Can I know the port asterisk used assuming I’ll use SIP. 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? This questions might annoyed experts. Please bear with me... Thank you. nocph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
Martin wrote: . In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although How can I download this, do I need SVN installed? yes. install svn. Then: svn co http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms trunk good luck. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-bsd] Newbie on VoIP
I just finished downloading and install the asterisknow. It's pretty cool GUI... How do I see my CDR on asterisknow? Thank you... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Neese Sent: Monday, March 03, 2008 10:39 AM To: Asterisk on BSD discussion Subject: Re: [Asterisk-bsd] Newbie on VoIP On March 2, 2008 06:14:02 pm NOC Ph wrote: Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who's calling overseas. 2. How do I secure it? Co'z I have to open it via Public IP. Can I know the port asterisk used assuming I'll use SIP. 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? This questions might annoyed experts. Please bear with me... Thank you. nocph ok I have a answer for you yes it does cdr. you can use a gui interface to look at them yes it runs on bsd and if its for your company then you would want a gui for ease of use. you have basicly 2 major choices right now freepbx or thirdlane. freepbx be it is a ok project requires sql and is a resource hog in the long run. thirdlane does not require a sql but you can implement a sql for cdr storage and recall via other gui interfaces. -- Welcome to the World. An the World gets smaller. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- Asterisk-BSD mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-bsd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
On Mon, 3 Mar 2008 10:14:02 +0800, NOC Ph [EMAIL PROTECTED] wrote: This questions might annoyed experts. Please bear with me... The journey of a thousand miles begins with a single step. Lao Tzu. Free PDF of Asterisk: The Future of Telephony, Second Edition http://downloads.oreilly.com/books/9780596510480.pdf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
A simple inrease of DEFAULT_BATT_DEBOUNCE to 32 (originally 4) work for me... Probably a nasty hack but allows me to dial (I also need DEFAULT_RING_DEBOUNCE increased to 256 to minimize problems with caller ID, which is right above) Martin My incoming calls work just fine with 1.4.9.2 and TDM400P but I'm completely unable to dial out the line. even this doesn't work: exten=*903,1,Answer() exten=*903,n,Dial(Zap/3) exten=*903,n,Hangup() I use this to pick up the line shared with analog phone. Zap/3 and Zap/4 are connected to PSTN. Sometimes I have luck with Zap/3 but not a single call through Zap/4 -- Executing [EMAIL PROTECTED]:1] Answer(SIP/210-081e9968, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/210-081e9968, Zap/3) in new stack -- Called 3 [2008-03-02 09:43:17] WARNING[5051]: chan_zap.c:4114 zt_handle_event: Detected alarm on channel 3: No Alarm -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get a clean, basic configuration?
On Tue, 26 Feb 2008 01:19:23 +0200, Atis Lezdins [EMAIL PROTECTED] wrote: To help you on your way of minimizing modules, here's some basic setup that generally works Thanks much for sharing your modules.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_ss7 0.10
Hi marek, gr8. I am working on chan_ss7 now.. Regards, Joel - Original Message - From: marek cervenka [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 03, 2008 3:55 AM Subject: Re: [asterisk-users] chan_ss7 0.10 Thanks for the update. I have Sangoma A104D and wanted to use ss7 signalling. I came accross chan_ss7 but found sifira is not in active development. But is this chan_ss7 stable and can be used in production server implementation. We are going to have 2 to 3 boxes with ss7 signalling using sangoma but I am looking for open source ss7 implementation which is chan_ss7. so need to know about stability and recommendation for using on production server. long term supported solution is libss7 from digium. but this depends on asterisk 1.6 which is not officialy stable chan_ss7 is now developed by www.dicea.dk. http://www.dicea.dk/company/downloads it's used on production servers. it is very stable solution --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
On Mon, Mar 03, 2008 at 10:14:02AM +0800, NOC Ph wrote: Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. Welcome, 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who's calling overseas. Sure. To a CSV file, and also , optionally, to several databases. 2. How do I secure it? Co'z I have to open it via Public IP. Can I know the port asterisk used assuming I'll use SIP. It's not only Asterisk you have to secure. You install Asterisk on a system. It is that system you have to secure. You'll probably expose some sort of management interface to the world (a web interface, ssh, whatever). 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? Asterisk is well known to build and run on FreeBSD and OpenBSD. But if you're interested in that, I suppose you can ask further questions in the asterisk-bsd mailing list. http://lists.digium.com/mailman/listinfo/asterisk-bsd This questions might annoyed experts. Please bear with me... To save you asking question, try: http://voip-info.org/ http://www.oreilly.com/catalog/asterisk -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
Hi Tilghman, Thanks for taking interest in my problem. I just want to send a http post request to my website without changing the dial plan. So I have added slightly modified http post code and some other code to channel.c got from curl/curl.h. After adding the code I compiled the asterisk code and got the error: channel.o(.text+0x): channel.c:: undefined reference to 'curl_global_init' Thanks Regards, Prashant Sharma On Friday 29 February 2008 08:10:40 Prashant Sharma wrote: * When I try to add CURL code to file channel.c we get an error - undefined ** reference to curl_easy_init. ** I've added #include curl/curl.h so the code compiles fine. ** this error is generated by the linker, even though func_curl.c is compiled ** and linked with no errors ** My asterisk machine have curl and curl-devel 7.12 installed. ** Asterisk version i am using is 1.4.17. * Let's start with, why are you adding curl code to channel.c? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
It's not only Asterisk you have to secure. You install Asterisk on a system. It is that system you have to secure. You'll probably expose some sort of management interface to the world (a web interface, ssh, whatever). [NOCPH] I have to open the SIP port and web. Another question, the SIP port is 5060 UDP, how about the conference? Does it use the same port also? Thansk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, March 03, 2008 2:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie on VoIP On Mon, Mar 03, 2008 at 10:14:02AM +0800, NOC Ph wrote: Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. Welcome, 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who's calling overseas. Sure. To a CSV file, and also , optionally, to several databases. 2. How do I secure it? Co'z I have to open it via Public IP. Can I know the port asterisk used assuming I'll use SIP. It's not only Asterisk you have to secure. You install Asterisk on a system. It is that system you have to secure. You'll probably expose some sort of management interface to the world (a web interface, ssh, whatever). 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? Asterisk is well known to build and run on FreeBSD and OpenBSD. But if you're interested in that, I suppose you can ask further questions in the asterisk-bsd mailing list. http://lists.digium.com/mailman/listinfo/asterisk-bsd This questions might annoyed experts. Please bear with me... To save you asking question, try: http://voip-info.org/ http://www.oreilly.com/catalog/asterisk -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
On Mon, Mar 3, 2008 at 12:40 AM, NOC Ph [EMAIL PROTECTED] wrote: [NOCPH] I have to open the SIP port and web. Another question, the SIP port is 5060 UDP, how about the conference? Does it use the same port also? That's a good start, but you'll also need to open the RTP ports as well - these usually fall in the 10k-20k udp range. 5060/udp is used for call signalling only, the actual voice data can use a variety of ports, depending on how you're set up. You can specify what RTP ports you want to use in your rtp.conf. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
On Mon, Mar 03, 2008 at 02:40:03PM +0800, NOC Ph wrote: It's not only Asterisk you have to secure. You install Asterisk on a system. It is that system you have to secure. You'll probably expose some sort of management interface to the world (a web interface, ssh, whatever). [NOCPH] I have to open the SIP port and web. Another question, the SIP port is 5060 UDP, how about the conference? Does it use the same port also? There are also the RTP ports, as mentioned above. But there is more to securing than opening / closing ports. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
I have asterisknow 1.0.1 install on my box. Seems working now... but how can I see or monitor the calls and CDR is important? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, March 03, 2008 2:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie on VoIP On Mon, Mar 03, 2008 at 10:14:02AM +0800, NOC Ph wrote: Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. Welcome, 1. Does asterisk generate CDR? If yes how do I see it or generate it? Because I have to monitor people who's calling overseas. Sure. To a CSV file, and also , optionally, to several databases. 2. How do I secure it? Co'z I have to open it via Public IP. Can I know the port asterisk used assuming I'll use SIP. It's not only Asterisk you have to secure. You install Asterisk on a system. It is that system you have to secure. You'll probably expose some sort of management interface to the world (a web interface, ssh, whatever). 3. If it run on linux, it run will on BSD but I read from google that it has specific version for BSD. Can I know what version are for FreeBSD? Asterisk is well known to build and run on FreeBSD and OpenBSD. But if you're interested in that, I suppose you can ask further questions in the asterisk-bsd mailing list. http://lists.digium.com/mailman/listinfo/asterisk-bsd This questions might annoyed experts. Please bear with me... To save you asking question, try: http://voip-info.org/ http://www.oreilly.com/catalog/asterisk -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
