[asterisk-users] cisco 7970 multiple lines with asterisk
Hi I have a problem that I can't pass. I have asterisk and cisco 7970 phones with 8.0.3 sip firmware. I registered two extensions: Line1: 260 Line2: 160 Regardless of which extension I call, always Line 1 on cisco is blinking. This makes impossible to recognize which extension is calling. Also, I've set Line 2 to be automatically answered with speaker phone (intercom). Even though I call extension 160 from Line 2 it is never automatically answered. Can anybody help me with this issue? I've been searching Internet to find clue on what I do wrong for few days. No success. Anybody had problem like this? Any hint would be appreciated. Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 Not registering
Hi All, I ' m using Cisco 7970 IP Phone and Asterisk 1.6.0.10-FONCORE-r40 (Tirxbox). My problem is that I upgrade my phone to SIP image but now this phone is not registering. The error likes this : SIP/2.0 403 Forbidden (Bad auth) The phone and Trixbox are in the same network. There arenot any NAT rules. Can you help me please? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 SIP endless ringing...?
Anyone know what would cause an endless ringing situation? I have a snom360 and cisco 7970 (sip 8.5.3). I have an incoming trunk which dials both phones: [gp710] exten = _[*1-9].,1,Dial(SIP/li...@cisco7970SIP/li...@snom360,60) exten = _[*1-9].,n,Hangup If a call comes in, I can answer the call on the cisco no problem. However if I answer on the snom360, the cisco never stops ringing. -Dan ___ -- 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 7970, CTLSEPmac.tlv
Jason Parker wrote: I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. Responding to myself... When I initially sent this, I had made several false assumptions. The biggest of which, was that the 'tftpd' package in Debian was no longer maintained (upstream hadn't made a release in 8 years, and Debian hadn't made a release in 3 years - I think it was a fairly reasonable one). Well, the maintainer of this package, Alberto, emailed me to let me know that somebody pointed him to this post, and that less than 24 hours later, he had fixed this bug (I've confirmed this) and made a new release - 0.17-16 - which is currently in Sid, and will hopefully be put into Lenny. This can be downloaded from http://packages.debian.org/search?keywords=tftpd Also, as Alberto correctly pointed out - I violated one of the most important rules of Open Source Software. If I may quote him: You had perfectly traced the problem, you perfectly described it, god! you even gave a reference to the RFC. You had the perfect bug report, but it was never going to make it to me arrrggg :) Such a great loss!! I failed to complete one critical step - reporting a bug. It ended up working out, but only because somebody else took the time to report the bug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970, CTLSEPmac.tlv
I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970, CTLSEPmac.tlv
On 15:02, Fri 01 Aug 08, Jason Parker wrote: I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. Thanks for the write-up. I tried with the latest 7960 firmware, and it did work with the default debian tftpd (had to install a new VM) For googleable stuff: The default tftpd on OpenBSD works fine ;) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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
[asterisk-users] Cisco 7970 BLF/Presence
I have been trying to get the 7970 (running SIP firmware) to display presence information about other extensions. Thus far, I have been unsuccessful. Does anyone have BLF working on the SIP-loaded 7941/7961/7970/7971? I have been using the following as a guide for my work: http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones I have successfully implemented every step (although there are a few minor errors in the code on that page that I have corrected) except for the part where the 7970 has to connect to Asterisk via SIP/TCP. I have chan_sip.c patched with the SIP/TCP patch to allow for connectivity, but I cannot seem to force the 7970 to connect that way. I have read several posts that indicated that the SIP firmware does not support presence information as of yet, but supposedly, whoever wrote that article above has it working somehow. Does anyone know how to make the phone connect over TCP? Or, better yet, does anyone have a working method that they would be willing to share? These are great phones but in the environment that we're in they are almost useless if we don't know who's on a call when. I'd rather not go the SCCP route unless I absolutely have to. 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
[asterisk-users] Cisco 7970 with skinny on * 1.4.x
Sorry bringing it up again Meanwhile switched to asterisk 1.4.3 on fbsd-6.2 but still no luck getting my 7970G to run via skinny... It registers fine with *: Adding button: 9, 1 Device capability set to '268' asterisk*CLI skinny show devices Name DeviceId IP TypeR NL --- --- - -- ciscoSEP00175A872053 xx.xx.xxx.xx7970Y 1 But on the phone I just see displayed the time and date but no linelabel... My skinny.conf is: [general] bindaddr=xx.xx.xxx.xx ; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=D.M.Y; M,D,Y in any order (5 chars max) keepalive=30 disallow=all allow=all ; see doc/rtp-packetization for framing options [cisco] device=SEP00175A872053 model=7970 nat=1 callerid=Richard Klingler 995 mailbox=995 callwaiting=yes transfer=yes threewaycalling=yes context=klingler linelabel=phonelab line = 995 any ideas left? Using now cmterm-7970_7971-sccp.8-2-2SR1 cheers rick ___ --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] Cisco 7970 with skinny on * 1.4.x
A little with skinny debug set to on shows during register: Device SEP00175A872053 is attempting to register Requesting capabilities Buttontemplate requested Adding button: 9, 1 Sending 30006 template to cisco Received SoftKey Template Request Received SoftKeySetReq RECEIVED UNKNOWN MESSAGE TYPE: c Received CapabilitiesRes Adding codec capability '0 (25)' Adding codec capability '4 (4)' Adding codec capability '8 (2)' Adding codec capability '0 (15)' Adding codec capability '0 (16)' Adding codec capability '0 (11)' Adding codec capability '256 (12)' Adding codec capability '256 (12)' Adding codec capability '0 (257)' Device capability set to '268' RECEIVED UNKNOWN MESSAGE TYPE: 49 RECEIVED UNKNOWN MESSAGE TYPE: 49 RECEIVED UNKNOWN MESSAGE TYPE: 4a RECEIVED UNKNOWN MESSAGE TYPE: 9 Received Time/Date Request Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :92 It also show this message when going offhook: RECEIVED UNKNOWN MESSAGE TYPE: 49 Setting ringer mode to '1'. skinny_new: tmp-nativeformats=268 fmt=4 Attempting to Clear display on Skinny [EMAIL PROTECTED] Clearing Display Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :85 Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :11 Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :9a Received Alarm Message: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S : Invalid SCCP message! : ID :82 Looks to me that chan_skinny doesn't understand many important messages. Any previous 7970G SCCP firmware that might work? cheers rick Richard Klingler schrieb: Sorry bringing it up again Meanwhile switched to asterisk 1.4.3 on fbsd-6.2 but still no luck getting my 7970G to run via skinny... It registers fine with *: Adding button: 9, 1 Device capability set to '268' asterisk*CLI skinny show devices Name DeviceId IP TypeR NL --- --- - -- ciscoSEP00175A872053 xx.xx.xxx.xx7970Y 1 But on the phone I just see displayed the time and date but no linelabel... My skinny.conf is: [general] bindaddr=xx.xx.xxx.xx ; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=D.M.Y; M,D,Y in any order (5 chars max) keepalive=30 disallow=all allow=all ; see doc/rtp-packetization for framing options [cisco] device=SEP00175A872053 model=7970 nat=1 callerid=Richard Klingler 995 mailbox=995 callwaiting=yes transfer=yes threewaycalling=yes context=klingler linelabel=phonelab line = 995 any ideas left? Using now cmterm-7970_7971-sccp.8-2-2SR1 cheers rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1
Hmm..interestingly no one answered if chan_skinny works with 7970G on * 1.4.x (o; I know that CIsco phones are bad with NAT and SIP...old story (o; THat's why I use local Cisco phones with SIP and local * which then connects to outside * vis IAX... cheers rick Hermann Wecke schrieb: Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1
Richard Klingler wrote: Hmm..interestingly no one answered if chan_skinny works with 7970G on * 1.4.x (o; I know that CIsco phones are bad with NAT and SIP...old story (o; THat's why I use local Cisco phones with SIP and local * which then connects to outside * vis IAX... I've a 7912G running with 1.4.x and chan_skinny, and seems to be working just fine (better than 1.2 anyway, the 7912G is not the 'heavy usage' phone at home, but still..) I tried twice to acquire the proper license to upgrade the 7912G to SIP, but the order got 'dropped' by the reseller after 2 weeks of 'shipping' :-), since 1.4 seems to be handling it just fine, I've moved this to the lower priority TODO list. Hermann Wecke schrieb: Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --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] Cisco 7970 with skinny on * 1.4.1
If you have 7970 right configured to point to asterisk server, you should be able to see some skinny debug on console, or look what report skinny show devices I haven't any 7970, so can't help so much, I'm using only 7920 wifi phone with chan_skinny and 1.4trunk, it's usable, basic functionality is working, but don't expect too much, btw, if you have money to buy this highend phone with proprietary signaling, why don't connect to callmanager? asterisk will never support all features available in proprierary system as good as original ;-) PJ Richard Klingler wrote: Hmm..interestingly no one answered if chan_skinny works with 7970G on * 1.4.x (o; I know that CIsco phones are bad with NAT and SIP...old story (o; THat's why I use local Cisco phones with SIP and local * which then connects to outside * vis IAX... cheers rick Hermann Wecke schrieb: Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1
I was able to register to * 1.4.1 via skinny...and it showed up on the lines and devices show output.. On the phone, however, no lines were displayed nor could it phone out or receive any calls... Anyone able to share some snippets of their skinny.conf? I just used the examples and modified the MAC line and extension line config...but seems something else is missing... cheers rick Pavel Jezek schrieb: If you have 7970 right configured to point to asterisk server, you should be able to see some skinny debug on console, or look what report skinny show devices I haven't any 7970, so can't help so much, I'm using only 7920 wifi phone with chan_skinny and 1.4trunk, it's usable, basic functionality is working, but don't expect too much, btw, if you have money to buy this highend phone with proprietary signaling, why don't connect to callmanager? asterisk will never support all features available in proprierary system as good as original ;-) PJ Richard Klingler wrote: Hmm..interestingly no one answered if chan_skinny works with 7970G on * 1.4.x (o; I know that CIsco phones are bad with NAT and SIP...old story (o; THat's why I use local Cisco phones with SIP and local * which then connects to outside * vis IAX... cheers rick Hermann Wecke schrieb: Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1
my simple, but working config for 7920... Dial(Skinny/[EMAIL PROTECTED]) [general] bindaddr=193.179.38.20; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=D-M-Y; M,D,Y in any order (5 chars max) keepalive=30 disallow=all allow=alaw [PJ] device=SEPxxx linelabel=xxx context = xxx nat=1 callwaiting=1 transfer=1 threewaycalling=1 line = 324 Richard Klingler wrote: I was able to register to * 1.4.1 via skinny...and it showed up on the lines and devices show output.. On the phone, however, no lines were displayed nor could it phone out or receive any calls... Anyone able to share some snippets of their skinny.conf? I just used the examples and modified the MAC line and extension line config...but seems something else is missing... cheers rick Pavel Jezek schrieb: If you have 7970 right configured to point to asterisk server, you should be able to see some skinny debug on console, or look what report skinny show devices I haven't any 7970, so can't help so much, I'm using only 7920 wifi phone with chan_skinny and 1.4trunk, it's usable, basic functionality is working, but don't expect too much, btw, if you have money to buy this highend phone with proprietary signaling, why don't connect to callmanager? asterisk will never support all features available in proprierary system as good as original ;-) PJ Richard Klingler wrote: Hmm..interestingly no one answered if chan_skinny works with 7970G on * 1.4.x (o; I know that CIsco phones are bad with NAT and SIP...old story (o; THat's why I use local Cisco phones with SIP and local * which then connects to outside * vis IAX... cheers rick Hermann Wecke schrieb: Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7970 with skinny on * 1.4.1
last chan_sccp was released a year ago, Sergio, main developer, gone away minimal activity in forum, chan_sccp.org, unoficial chan_sccp site, is for sale this are reasons, why I also considering chan_sccp as death project. Bill Hackensack wrote: chan_sccp is far from dead and it works with 1.4. more fud being spread... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1
Bill Hackensack schrieb: On 3/21/07, *Richard Klingler* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... chan_sccp is far from dead and it works with 1.4. more fud being spread... Maybe not dead...but hust won't compile on FBSD (o; cheers rick ___ --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] Cisco 7970 with skinny on * 1.4.1
Evnin' (o; As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... The Cisco 7970 registers and is being acknowledged by * but that's it... I see no lines on the 7970 display configured and it is not reachable or it can't make any outboudn calls... The docs are pretty non-existent for skinny and the sample configuration are of no help... Has any1 got their 7970 to work with * 1.4.x ? cheers rick ___ --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] Cisco 7970 with skinny on * 1.4.1
Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check http://preview.tinyurl.com/345fmj ___ --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] Cisco 7970 with skinny on * 1.4.1
On 3/21/07, Richard Klingler [EMAIL PROTECTED] wrote: As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... chan_sccp is far from dead and it works with 1.4. more fud being spread... ___ --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] Cisco 7970 Unprovisioned
Hi everyone! I just want to thank everybody. My phone works now and just a little hint: set qualify=no in sip.conf of your phone's extension. Best regards Mihaela MJ On 1/21/07, Token PBX [EMAIL PROTECTED] wrote: On 1/20/07, Pavel Jezek [EMAIL PROTECTED] wrote: you have probably something wron in config file and phone refuses to configure, here is my minimalistic file for 7941/61, you can try... device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPasswordadmin/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentpserver/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameasteriskserver/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg729a/preferredCodec natEnabled0/natEnabled phoneLabelSIP/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 999/featureLabel proxyasteriskserver/proxy name999/name displayNameyourname/displayName authName999/authName authPasswordxxx/authPassword messagesNumber/messagesNumber /line line button=2 featureID21/featureID featureLabelHelpdesk/featureLabel speedDialNumber5880/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePasswordadmin/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp directoryURL/directoryURL servicesURL/servicesURL /device ___ --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 Hi! Here's my configuration file: device xsi:type=axl:XIPPhone fullConfigtrue/fullConfig deviceProtocolSIP/deviceProtocol sshUserIduser/sshUserId sshPasswordpass/sshPassword devicePool nameDefault/name dateTimeSetting nameCMLocal/name dateTemplateD.M.Y/dateTemplate timeZoneW. Europe Standard/Daylight Time/timeZone /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort /ports processNodeNameMy Asterisk IP/processNodeName /callManager /member /members /callManagerGroup srstInfo nameEnable/name srstOptionEnable/srstOption userModifiabletrue/userModifiable ipAddr1My Asterisk IP/ipAddr1 port12000/port1 ipAddr2/ipAddr2 port22000/port2 ipAddr3/ipAddr3 port32000/port3 /srstInfo mlppDomainId-1/mlppDomainId mlppIndicationStatusDefault/mlppIndicationStatus preemptionDefault/preemption /devicePool commonProfile phonePassword/phonePassword backgroundImageAccesstrue/backgroundImageAccess callLogBlfEnabled2/callLogBlfEnabled /commonProfile loadInformation/loadInformation vendorConfig disableSpeakerfalse/disableSpeaker disableSpeakerAndHeadsetfalse/disableSpeakerAndHeadset forwardingDelay1/forwardingDelay pcPort0/pcPort settingsAccess1/settingsAccess garp0/garp voiceVlanAccess0/voiceVlanAccess videoCapability1/videoCapability autoSelectLineEnable0/autoSelectLineEnable webAccess1/webAccess daysDisplayNotActive1,7/daysDisplayNotActive displayOnTime08:30/displayOnTime displayOnDuration11:30/displayOnDuration displayIdleTimeout01:00/displayIdleTimeout spanToPCPort1/spanToPCPort loggingDisplay1/loggingDisplay /vendorConfig versionStamp1136931633-57191cee-5ffc-4342-b286-4246b4991890/versionStamp userLocale nameEnglish_United_States/name uid1/uid langCodeen_US/langCode version1.0.0.0-1/version winCharSetiso-8859-1/winCharSet /userLocale networkLocaleUnited_States/networkLocale networkLocaleInfo nameUnited_States/name uid64/uid version1.0.0.0-1/version /networkLocaleInfo deviceSecurityMode1/deviceSecurityMode idleTimeout120/idleTimeout authenticationURL/authenticationURL directoryURL/directoryURL idleURL/idleURL informationURL/informationURL messagesURL/messagesURL proxyServerURL/proxyServerURL
[asterisk-users] Cisco 7970 Unprovisioned
Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! MihaelaMJ ___ --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] Cisco 7970 Unprovisioned
Are you setting the TFTP server address in the DHCP? Are you checking the TFTP log to see what files the phone is requesting and not finding? Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Token PBX [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, 20 January, 2007 1:01:25 PM Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! MihaelaMJ ___ --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 ___ New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --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] Cisco 7970 Unprovisioned
On 1/20/07, Jon Farmer [EMAIL PROTECTED] wrote: Are you setting the TFTP server address in the DHCP? Are you checking the TFTP log to see what files the phone is requesting and not finding? Regards Jon Jon Farmer Telford, Shropshire, UK Hi Jon! Yes I checked log, and phone requested and loaded all required files and then some: It also requested file: CTLSEP-MAC.tlv, that has something to do with license. Since it couldn't find it returned error and continued to load SEP- MAC.cnf.xml . Phone booted with SIP firmware but did not load any of the settings from SEP-MAC.cnf.xml. I checked that from phone's display. None of the settings were loaded, no sip proxy address, phone label, SIP lines etc.. All was blank. Just some dynamically assigned settings were set like DHCP address, phone's IP and such. I followed instructions from wiki voip-info when building SEP- MAC.cnf.xml. Please help and thanks. Mihaela MJ - Original Message From: Token PBX [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, 20 January, 2007 1:01:25 PM Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! MihaelaMJ ___ --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 -- To help you stay safe and secure online, we've developed the all new Yahoo! Security Centrehttp://us.rd.yahoo.com/mail/uk/taglines/default/security_centre/*http://uk.security.yahoo.com/ . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7970 Unprovisioned
Sounds like you need to dig into the documentation for the 7970 and perhaps even contact Cisco TAC if that doesn't help. It doesn't sound like your problem is related to Asterisk. The Cisco IP phone won't register with asterisk until it's been provisioned. Those 7900 series cisco phones are very finicky. Best of luck! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Token PBX Sent: Saturday, January 20, 2007 6:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! Mihaela ___ --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] Cisco 7970 Unprovisioned
you have probably something wron in config file and phone refuses to configure, here is my minimalistic file for 7941/61, you can try... device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPasswordadmin/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentpserver/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameasteriskserver/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg729a/preferredCodec natEnabled0/natEnabled phoneLabelSIP/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 999/featureLabel proxyasteriskserver/proxy name999/name displayNameyourname/displayName authName999/authName authPasswordxxx/authPassword messagesNumber/messagesNumber /line line button=2 featureID21/featureID featureLabelHelpdesk/featureLabel speedDialNumber5880/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePasswordadmin/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp directoryURL/directoryURL servicesURL/servicesURL /device ___ --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] Cisco 7970 Unprovisioned
On 1/20/07, Pavel Jezek [EMAIL PROTECTED] wrote: you have probably something wron in config file and phone refuses to configure, here is my minimalistic file for 7941/61, you can try... device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPasswordadmin/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentpserver/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameasteriskserver/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg729a/preferredCodec natEnabled0/natEnabled phoneLabelSIP/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 999/featureLabel proxyasteriskserver/proxy name999/name displayNameyourname/displayName authName999/authName authPasswordxxx/authPassword messagesNumber/messagesNumber /line line button=2 featureID21/featureID featureLabelHelpdesk/featureLabel speedDialNumber5880/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePasswordadmin/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp directoryURL/directoryURL servicesURL/servicesURL /device ___ --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 Hi! Here's my configuration file: device xsi:type=axl:XIPPhone fullConfigtrue/fullConfig deviceProtocolSIP/deviceProtocol sshUserIduser/sshUserId sshPasswordpass/sshPassword devicePool nameDefault/name dateTimeSetting nameCMLocal/name dateTemplateD.M.Y/dateTemplate timeZoneW. Europe Standard/Daylight Time/timeZone /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort /ports processNodeNameMy Asterisk IP/processNodeName /callManager /member /members /callManagerGroup srstInfo nameEnable/name srstOptionEnable/srstOption userModifiabletrue/userModifiable ipAddr1My Asterisk IP/ipAddr1 port12000/port1 ipAddr2/ipAddr2 port22000/port2 ipAddr3/ipAddr3 port32000/port3 /srstInfo mlppDomainId-1/mlppDomainId mlppIndicationStatusDefault/mlppIndicationStatus preemptionDefault/preemption /devicePool commonProfile phonePassword/phonePassword backgroundImageAccesstrue/backgroundImageAccess callLogBlfEnabled2/callLogBlfEnabled /commonProfile loadInformation/loadInformation vendorConfig disableSpeakerfalse/disableSpeaker disableSpeakerAndHeadsetfalse/disableSpeakerAndHeadset forwardingDelay1/forwardingDelay pcPort0/pcPort settingsAccess1/settingsAccess garp0/garp voiceVlanAccess0/voiceVlanAccess videoCapability1/videoCapability autoSelectLineEnable0/autoSelectLineEnable webAccess1/webAccess daysDisplayNotActive1,7/daysDisplayNotActive displayOnTime08:30/displayOnTime displayOnDuration11:30/displayOnDuration displayIdleTimeout01:00/displayIdleTimeout spanToPCPort1/spanToPCPort loggingDisplay1/loggingDisplay /vendorConfig versionStamp1136931633-57191cee-5ffc-4342-b286-4246b4991890/versionStamp userLocale nameEnglish_United_States/name uid1/uid langCodeen_US/langCode version1.0.0.0-1/version winCharSetiso-8859-1/winCharSet /userLocale networkLocaleUnited_States/networkLocale networkLocaleInfo nameUnited_States/name uid64/uid version1.0.0.0-1/version /networkLocaleInfo deviceSecurityMode1/deviceSecurityMode idleTimeout120/idleTimeout authenticationURL/authenticationURL directoryURL/directoryURL idleURL/idleURL informationURL/informationURL messagesURL/messagesURL proxyServerURL/proxyServerURL servicesURL/servicesURL dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices dscpForCm2Dvce96/dscpForCm2Dvce capfAuthMode0/capfAuthMode capfList capf phonePort3804/phonePort processNodeNameccm-beta-5-1/processNodeName /capf /capfList certHash/certHash encrConfigfalse/encrConfig sipProfile
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Hi Is NAT set to NO? It needs to be set to NO in 8.0.3 or it just sits there at registering as you say Thanks - Original Message - From: Pavel Jezek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 9:08 AM Subject: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2) Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
there is any way to configure a 7970 without using the display, I have my LCD broken so I cannot see what I'm doing :) but the phone works fine. 2006/12/13, Paul A Brown [EMAIL PROTECTED]: Hi Is NAT set to NO? It needs to be set to NO in 8.0.3 or it just sits there at registering as you say Thanks - Original Message - From: Pavel Jezek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 9:08 AM Subject: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2) Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Hi Pavel, Thanks for the config! I modified mine so it was more minimal like yours, and it registers just fine now. So much nicer without those big red X's! MG On 13/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Matt Gibson wrote: Hi Pavel, Thanks for the config! I modified mine so it was more minimal like yours, and it registers just fine now. So much nicer without those big red X's! MG This modified config works sweet!! Any tricks to get the MWI working? Mark ___ --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] Cisco 7970 + New Firmware (8.2)
Hi All, Found out Cisco has some newer firmware available for the 7970 series of phones. New sip images are at version level 8.2 (instead of 8.0.2,8.0.3,8.0.4), posted Dec 10, 2006. This major jump in version numbers has fixed a few bugs (time zone not updating properly), but hasn't figured what some would consider to be showstoppers (registration not fully working, and mwi still not working). Just thought I would let you all know there's new firmware to mess around with! Also, to note, to get the phone to actually take this upgrade, and you're running your tftp server on a linux box, then you will need to rename one of the files for it to find it properly. # cd tftpdroot # mv jar70sip.8-2-0-55.sbn Jar70sip.8-2-0-55.sbn Calls in and Out work, though the phone still shows that dreaded red x next to the extension saying it's not registered. MWI is also still not working with 3 or 1 in the MWI indicator slot in the .xml file. And no, I won't email you the firmware, you need a cisco login to get one, so get a friend, or join cisco yourself! :) Happy Testing! Matt G http://www.voipphreak.ca ___ --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] Cisco 7970 + New Firmware (8.2)
I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ Matt Gibson wrote: Hi All, Found out Cisco has some newer firmware available for the 7970 series of phones. New sip images are at version level 8.2 (instead of 8.0.2,8.0.3,8.0.4), posted Dec 10, 2006. This major jump in version numbers has fixed a few bugs (time zone not updating properly), but hasn't figured what some would consider to be showstoppers (registration not fully working, and mwi still not working). Just thought I would let you all know there's new firmware to mess around with! Also, to note, to get the phone to actually take this upgrade, and you're running your tftp server on a linux box, then you will need to rename one of the files for it to find it properly. # cd tftpdroot # mv jar70sip.8-2-0-55.sbn Jar70sip.8-2-0-55.sbn Calls in and Out work, though the phone still shows that dreaded red x next to the extension saying it's not registered. MWI is also still not working with 3 or 1 in the MWI indicator slot in the .xml file. And no, I won't email you the firmware, you need a cisco login to get one, so get a friend, or join cisco yourself! :) Happy Testing! Matt G http://www.voipphreak.ca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --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] Cisco 7970 SIP upgrade issues
Hi Ok I have the right version many thanks However I am still a tad stuck (Sorry) I have all the configs to upgrade from SCCP to SIP but what config files do I need just to upgrade the sccp to the 7.0-3 version. I am assuming I need to have a file in the tftp dir that tells the phone to load a specific image. Thanks - Original Message - From: Lacy Moore - Aspendora To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 01, 2006 10:29 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues On 11/29/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( You apparently downloaded the wrong version. I don't know what version you downloaded. You need the zip version of cmterm-7970-7971-sccp-7.0-3. Unzip it to your tftp directory. There is no setup file. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
On 11/29/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( You apparently downloaded the wrong version. I don't know what version you downloaded. You need the zip version of cmterm-7970-7971-sccp-7.0-3. Unzip it to your tftp directory. There is no setup file. ___ --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] Cisco 7970 SIP upgrade issues
Hi Thanks for the advice but it really is more fundamental. I have an old (v5) sccp phone. I need to upgrade it to v7 sccpbefore I can load the Sip image. I downloaded the V7 sccp file from the cisco website but it seems to want call manager to load. Does anyone have any experience of upgrading a V5 7970? Please please :-) - Original Message - From: Alfred Nagl [EMAIL PROTECTED] To: Paul [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 2:26 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Paul writes: I am having problems putting a SIP image on a 7970. Hi! Two weeks ago I loaded a recent SIP Image, SIP70.8-0-4SR1S, on a 7970, but I started from a relatively new SCCP Image. ( the phone has Boot Load ID 7970_64060118.bin) I did the following: .) configured a tftp server on the phone, to unlock I had to type star star numbersign (**#), and then I could save that configuration .) Got cmterm-7970_7971-sip.8-0-4SR1.zip from cisco website and unzipped it in tftp Directory .) Created file SEPMAC.cnf.xml with the following entry: loadInformationSIP70.8-0-4SR1S/loadInformation Most of the content of my SEPMAC.cnf.xml is from the follwing webpage http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIPview_comment_id=12387 and also from http://www.reub.net/files/cisco-7941/SEP-my-mac.cnf.xml If you are in a hurry, I could try to send you a sanitized / shorted working version of my SEPMAC.cnf.xml. regards, --alfred P.S.: I have tried to find some Documentation about the Meaning of all these XML Tags in the cnf.xml file, but was only partly successfull: http://www.voip-info.org/wiki-Asterisk+Cisco+79XX+XML+Services PP.S: there is a docoment about converting from SCCP to SIP and back (but it does not mention the 7970) http://www.cisco.com/warp/public/788/voip/handset_to_sip.html -- Alfred Nagl ([EMAIL PROTECTED]) Fax +43 (1) 31336-904811 University of Economics, A-1090 Vienna, Austria, EUROPE ___ --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] Cisco 7970 SIP upgrade issues
Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 11:26 AM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7970 SIP upgrade issues
Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 11:26 AM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi Its not even at the tftp stage. When I run the image file from Chisco and attempt to run setup I get a registry error. I am assuming its because its expecting a call manager. How do I upgrade the firmware? The image I have is only for callmanager cmterm-7970_7971-sccp.7-0-2SR1 Anyone know of a standalone image? - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 12:41 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 11:26 AM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7970 SIP upgrade issues
Hi believe that you nead a standalone image. Would you consider use SIP image, that could be possible to find on the net. //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Its not even at the tftp stage. When I run the image file from Chisco and attempt to run setup I get a registry error. I am assuming its because its expecting a call manager. How do I upgrade the firmware? The image I have is only for callmanager cmterm-7970_7971-sccp.7-0-2SR1 Anyone know of a standalone image? - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 12:41 PM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 11:26 AM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Thanks for all the help guys. I cannot load the new SIP image straight on as the SCCP image is very old. i read the FAQs posted on the lists and it tells me I need to upgrade the SCCP image to at least 7 before I can load the SIP image. This is the problem I am having. I cannot load SIP until I have at least V7 of SCCP. I downloaded the SCCP image but when you run setup it comes back with a registry error making me think it needs a call manager. Has anyone EVER managed to load the SIP image onto a 7970 that had V5 code? Thanks - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 2:15 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi believe that you nead a standalone image. Would you consider use SIP image, that could be possible to find on the net. //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Its not even at the tftp stage. When I run the image file from Chisco and attempt to run setup I get a registry error. I am assuming its because its expecting a call manager. How do I upgrade the firmware? The image I have is only for callmanager cmterm-7970_7971-sccp.7-0-2SR1 Anyone know of a standalone image? - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 12:41 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 11:26 AM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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] Cisco 7970 SIP upgrade issues
Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards___ --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] Cisco 7970 - versionStamp
If I put versionStamp in cnf.xml file, how do I check it on the phone? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7970 strange Xml , but upgrade success.
When I try to upgrade 7970 phone to sip 8.0.4SR1, Im getting this error all time: Read request for file .loads. Mode octet [16/10 15:14:12.187] File .loads : error 2 in system call CreateFile The system cannot find the file specified. [16/10 15:14:12.187] But I found this inside SEP(MAC).cnf.xml : loadInformationSIP70.8-0-4SR1S./loadInformationcare for . When I add .(dot) at the end of version information ; upgrade started and successfully finished. I hope this help. Best Regards. ___ --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] Cisco 7970 SIP won't update?
I am experiencing the same issue. However, I have not tried the VersionStamp field and will do so tomorrow. If you find an answer please post it to the list. On 10/13/06, Tim Connolly [EMAIL PROTECTED] wrote: Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Any suggestions? ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 SIP won't update?
Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Any suggestions? ___ --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] Cisco 7970 SIP won't update?
On Fri, 2006-10-13 at 11:53 -0500, Tim Connolly wrote: Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Any suggestions? The status screen should have errors if the config file is invalid. -Greg ___ --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] Cisco 7970 SIP won't update?
Tim Connolly wrote: Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Sounds like there is probably an error in the config. I had this problem with my 7970's. If there was an error in the config, they would download it but not apply it. The best thing to do is use a cisco console adapter and a regular RJ11 phone cord and hook it up to the aux port on the phone (normally used for sidecar). Then open a serial console session 9600-8,N,1. It will output debug messages as it boots and you can see exactly what lines it is choking on. I had problems with name fields being too many characters and other such minor things that were not well documented. Don't know what I'd have done without the serial debug output. -Evan ___ --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] Cisco 7970 Unbootable After FW Upgrade
I tried upgrading a used Cisco 7970 from the image it shipped with to SIP 8.0.2 SR1 but didn't have any luck so I followed the procedures to do a factory reset on the phone. The phone is grabbing an IP and attempting to grab my term70.default.loads file but not moving any further. The phone screen no longer shows anything. Has anyone else had the same problem? All of my other 7970s upgraded with no problems. Since our 7970s are all used I couldn't tell what image they shipped with or what the default is. I've tried grabbing a much older SCCP image version and placing that image in my tftp server hoping it would like that but still no success. Does anyone have any suggestions as to how I can at least get this phone to boot some default SCCP image? As of right now this phone is unuseable. I get the feeling that if I can figure out what the default image is for one of these I may be able to get it to boot to that. Thanks! Jeremiah ___ --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] Cisco 7970 Unbootable After FW Upgrade
When you do a factory reset on a 41/61/70/71, it actually deletes ALL of the firmware except the bootloader from the phone. You would have to have all of the 70s firmware files that come with them in order to boot them. The term70.default.loads tells the phone what version of software to tftp. Does the phone actually try to receive the file from your tftp server? What does your tftp log say? -Greg On Mon, 2006-10-09 at 13:23 -0500, Jeremiah Millay wrote: I tried upgrading a used Cisco 7970 from the image it shipped with to SIP 8.0.2 SR1 but didn't have any luck so I followed the procedures to do a factory reset on the phone. The phone is grabbing an IP and attempting to grab my term70.default.loads file but not moving any further. The phone screen no longer shows anything. Has anyone else had the same problem? All of my other 7970s upgraded with no problems. Since our 7970s are all used I couldn't tell what image they shipped with or what the default is. I've tried grabbing a much older SCCP image version and placing that image in my tftp server hoping it would like that but still no success. Does anyone have any suggestions as to how I can at least get this phone to boot some default SCCP image? As of right now this phone is unuseable. I get the feeling that if I can figure out what the default image is for one of these I may be able to get it to boot to that. Thanks! Jeremiah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 - DTMF
In sip.conf for one friend (Cisco 7970 phone) I have define this dtmfmode=inband And in xml.conf of that phone I have preferredCodecnone/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandnone/dtmfOutofBand But DTMF doesn't work for that phone. Phone establishes call using g711 alaw codec. How should I configure phone and sip.conf to make DTMF work? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7970 - DTMF
Tomislav Parčina wrote: In sip.conf for one friend (Cisco 7970 phone) I have define this dtmfmode=inband And in xml.conf of that phone I have preferredCodecnone/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandnone/dtmfOutofBand But DTMF doesn't work for that phone. Phone establishes call using g711 alaw codec. How should I configure phone and sip.conf to make DTMF work? In asterisk sip.conf, use dtmfmode=rfc2833 for that extension, and in the SIPDefault.cnf boot file for the cisco, include: dtmf_inband: 1 dtmf_outofband: avt dtmf_db_level: 3 (you'll need to translate the above 7960 parameters into the 7970 xml parameters since I don't have a 7970 to play with.) Taking a wild-ass guess, you might be able to get by simply using the dtmfmode=rfc2833 parameter in asterisk without touching the phone. Try it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7970 behind NAT
Jeremiah wrote: Does anyone have this working? I have a Cisco 7970 with the 8-0-2-SR1S firmware loaded on it. I can get the phone to register with * just fine when I place my asterisk server on the same subnet and do no NAT. When I give my asterisk server a static public IP and put the phone behind a NAT to connect to the server registration fails. I turn on sip debugging and see that the phone is trying to register but it gets 401 Unauthorized. The same phone config is being used with only modifications to the IPs of the proxy and some NAT settings. I've adjusted NAT settings in two places (phone config and sip.conf). The problem is that the 7970 phones by default are listening for replies to their register requests on port 5060. Unfortunately, the phone sends them out from random ports. So, if you have nat=yes on the sip peer in asterisk then the asterisk will send the reply to the port the request came from and not 5060. The only deployment we have done of these phones with NAT involved was for 2 executives at a branch office. In order to get the phones working we had to set the XML configs for the phones to send the external IP address of the firewall (you'll need a static IP for this to work) and to request replies on a custom port other than 5060. We then gave the phones DHCP reservations so they would always get the same private IP and mapped the custom sip ports through the firewall to each of the 2 phones. The sip peers in asterisk then had nat=no. Kind of a kludge but since there were only two 7970 phones it was manageable. The other cisco phones don't seem to have this problem. Perhaps somebody out there knows a way to make the 7970 phones accept SIP responses back to the originating port. I wasted several hours but couldn't figure it out. -Evan ___ --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] Cisco 7970 behind NAT
Shortly after I sent this e-mail I got it figured out. In sip.conf I had to put nat=no. The phone config also need to have all NAT features turned off. It was strange because I was sniffing the packets for the registration and saw no authentication information coming from the phone (with a really high source port number I might add), then I turned off NAT in sip.conf and did a reload and all of a sudden the phone was registered. This is the opposite of what I do for my 7960s running the 7.4 SIP image. After I got the 7970 working I had a 7961 running the 8.0.2SR1 unified image and had to do the same thing. The config files and settings for phones running the newer Cisco SIP software all require these parameters. Just an F.Y.I. Jeremiah The problem is that the 7970 phones by default are listening for replies to their register requests on port 5060. Unfortunately, the phone sends them out from random ports. So, if you have nat=yes on the sip peer in asterisk then the asterisk will send the reply to the port the request came from and not 5060. The only deployment we have done of these phones with NAT involved was for 2 executives at a branch office. In order to get the phones working we had to set the XML configs for the phones to send the external IP address of the firewall (you'll need a static IP for this to work) and to request replies on a custom port other than 5060. We then gave the phones DHCP reservations so they would always get the same private IP and mapped the custom sip ports through the firewall to each of the 2 phones. The sip peers in asterisk then had nat=no. Kind of a kludge but since there were only two 7970 phones it was manageable. The other cisco phones don't seem to have this problem. Perhaps somebody out there knows a way to make the 7970 phones accept SIP responses back to the originating port. I wasted several hours but couldn't figure it out. -Evan -- __ Rock River InternetJeremiah Millay 202 W. State St, 8th Floor [EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 Ext. 2202 USA fax 968-6888 ___ --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] Cisco 7970 behind NAT
Does anyone have this working? I have a Cisco 7970 with the 8-0-2-SR1S firmware loaded on it. I can get the phone to register with * just fine when I place my asterisk server on the same subnet and do no NAT. When I give my asterisk server a static public IP and put the phone behind a NAT to connect to the server registration fails. I turn on sip debugging and see that the phone is trying to register but it gets 401 Unauthorized. The same phone config is being used with only modifications to the IPs of the proxy and some NAT settings. I've adjusted NAT settings in two places (phone config and sip.conf). Example: sip.conf change nat=never to nat=yes Phone config: change natEnabled0/natEnabled natAddress/natAddress to natEnabled1/natEnabled natAddress/natAddress Does anyone have a similar setup with a 7970 behind NAT to an asterisk server that is not behind NAT? Any help or thoughts would be greatly appreciated. Jeremiah ___ --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] Cisco 7970 behind NAT
Since the phone is the one behind a NAT, and the registration is done only with SIP packages, setting or not the "nat" is not an issue (ONLY for registration purposes). You can see this since Asterisk is receiving the registration. Why is it denying it?... wel, that's something that will most likely has to do with the registrationn parameters (user-passwd), but certainly not with the network configuration.Alyed Return-Path: [EMAIL PROTECTED] Wed Sep 20 13:35:46 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 20 Sep 2006 13:35:46 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) Does anyone have this working? I have a Cisco 7970 with the 8-0-2-SR1S firmware loaded on it. I can get the phone to register with * just fine when I place my asterisk server on the same subnet and do no NAT. When I give my asterisk server a static public IP and put the phone behind a NAT to connect to the server registration fails. I turn on sip debugging and see that the phone is trying to register but it gets 401 Unauthorized. The same phone config is being used with only modifications to the IPs of the proxy and some NAT settings. I've adjusted NAT settings in two places (phone config and sip.conf).Example:sip.confchange "nat=never" to "nat=yes"Phone config:change0to1Does anyone have a similar setup with a 7970 behind NAT to an asterisk server that is not behind NAT? Any help or thoughts would be greatly appreciated.Jeremiah___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 directories and services xml
According to this thread http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3 Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco 7970 IP Phone doesn't show phone directory or services. It seams there is the same problem with SIP 8.0.3 firmware. Has anybody find any solution to this? Or all we can do is to wait new SIP firmware (8.0.4 can't register with Asterisk). -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7970 directories and services xml
Tomislav Parčina schrieb: According to this thread http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3 Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco 7970 IP Phone doesn't show phone directory or services. It seams there is the same problem with SIP 8.0.3 firmware. Has anybody find any solution to this? Or all we can do is to wait new SIP firmware (8.0.4 can't register with Asterisk). My 7970G running 8.0.2 SIP firmware works perfectly with the Open XML 79xx directory frontend... Also can can push XML alarm messages to the phone from nagios system. For me all other SIP version won't register with * 1.2.9 (o; - Do you have access to the webserver logs? - can you telnet to your webserver port and look on the console if something is returned? (telnet x.x.x.x 80 and do a manual get) - Can you point your phone to some other URLs mentioned on voip-info.org? cheers rick ___ --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] Cisco 7970 8.0.4 SIP firmware
Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7970 8.0.4 SIP firmware
I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
Does the 8.0.3 image has the same flaws as 8.0.4? Wasn't even able to register with * at all since most configuration examples from voip-info.org wouldn't work... Do you have any example config for me to try with SIP image on 7970G? Only tried 8.0.3 on my 7970G and had to switch to SCCP image...which is now 8.0.4 cheers rick Aaron Daniel schrieb: I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
Aaron, was the MWI working for you on 8.0.2? I've got a 7970 and 7961 sitting on a shelf because the MWI doesn't work. On the 8.0.4, it never registered, but I was able to make calls with it. I didn't try calling it, since I never saw it register. It appeared it was authenticating for outgoing calls. On 8/31/06, Aaron Daniel [EMAIL PROTECTED] wrote: I tried that image for about 5 minutes.Kept getting errors in asteriskfrom the phone and it wouldn't stay registered.Rolled back to 8.0.2and that works fine for us for now.On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Aaron DanielComputer Systems TechnicianSam Houston State University [EMAIL PROTECTED](936) 294-4198___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy MooreAspendora, Inc. ___ --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] Cisco 7970 8.0.4 SIP firmware
MWI has been working on our (2) 7970's, as far as I can tell. My boss usually complains when his doesn't work, so it seems to be working fine as far as that's concerned. The 8.0.4 firmware attempted to register, but asterisk threw an error on a response it got back from the phone (I don't remember exactly which one), but I could make calls from it, just not to it. Aaron On Thu, 2006-08-31 at 14:33 -0500, Lacy Moore - Aspendora wrote: Aaron, was the MWI working for you on 8.0.2? I've got a 7970 and 7961 sitting on a shelf because the MWI doesn't work. On the 8.0.4, it never registered, but I was able to make calls with it. I didn't try calling it, since I never saw it register. It appeared it was authenticating for outgoing calls. On 8/31/06, Aaron Daniel [EMAIL PROTECTED] wrote: I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970
Is 8.0.2.SR1 still the latest firmware? I still haven't managed to do anything useful with that weary expensive phone. It still only receives and places calls, nothing else. Is there any exciting feature that can work with asterisk and SIP firmware? Has anybody managed to do anything of the following: - my screensaver - picture of calling person - External directory - dialplan.xml - How to setup hinting (Multiple Call Appearance) - How to login true ssh? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7970 problems
I did not get this back from the list so I'm not sure if thishit the list last week or not so I'm sending it again. Sorry if this is a duplicate post! --- Has anyone had problems with a Cisco 7970 running sip image SIP70.8.0-2SR1S hanging up zap channels? Calls to SIP and IAX are fine. Just when the call goes out via the zap channels I have some Cisco 7960 running SIP and they work fine. Any ideas? Thanks-Eric Hall ___ --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] Cisco 7970 problems
Hall, Eric M. wrote: I did not get this back from the list so I'm not sure if this hit the list last week or not so I'm sending it again. Sorry if this is a duplicate post! --- Has anyone had problems with a Cisco 7970 running sip image SIP70.8.0-2SR1S hanging up zap channels? Calls to SIP and IAX are fine. Just when the call goes out via the zap channels I have some Cisco 7960 running SIP and they work fine. A classic cause of this is callprogress=yes or busydetect=yes in zapata.conf -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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: Spam? Re: [Asterisk-Users] Cisco 7970 problems
I don't see that anywhere. Here is my zapata.conf This is only happing on my 7970 all other phone are working without trouble. [channels] context=pri signalling=pri_cpe switchtype=dms100 group=1 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=incoming musiconhold=default echocancel=yes echocancelwhenbridged=yes channel = 1-23 context=Fax switchtype=national signalling=pri_net group=2 overlapdial=yes usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=no musiconhold=default channel = 25-47 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Saturday, May 13, 2006 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 problems Hall, Eric M. wrote: I did not get this back from the list so I'm not sure if this hit the list last week or not so I'm sending it again. Sorry if this is a duplicate post! -- -- --- Has anyone had problems with a Cisco 7970 running sip image SIP70.8.0-2SR1S hanging up zap channels? Calls to SIP and IAX are fine. Just when the call goes out via the zap channels I have some Cisco 7960 running SIP and they work fine. A classic cause of this is callprogress=yes or busydetect=yes in zapata.conf -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 problems
Has anyone had problems with a Cisco 7970 running sip image SIP70.8.0-2SR1S hanging up zap channels? Calls to SIP and IAX are fine. Just when the call goes out via the zap channels I have some Cisco 7960 running SIP and they work fine. Any ideas? Thanks-Eric Hall ___ --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] Cisco 7970 running SIP question
Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm unable to find the correct wording when searching Google to find that post again. Can anyone help me out here. How can I remove the asterisk servers IP from the phone number? Also I'm unable to get the time zone correct on the phone. It is in UTC and I'm in EST I see in the file where it looks like it goes but what I have tried has not worked as of yet. Here is what it looks like dateTimeSetting dateTemplateM/D/Y/dateTemplate timeZoneEST/timeZone /dateTimeSetting Thanks again for all your help!!! ___ --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] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having
the IP address on there, but it happens on ours too. It shouldn't cause
any problems with making outgoing calls from the directory, it's just
annoying to see it pop up.
As for the date time settings... this is what we have in ours:
dateTimeSetting uuid={9ec4850a-7748-11d3-bdf0-00108302ead1}
nameCMLocal/name
dateTemplateM/D/YA/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
I'm guessing you should be able to change it to say Eastern instead of
Central
On Fri, 5 May 2006, Hall, Eric M. wrote:
Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when searching Google to find that
post again. Can anyone help me out here. How can I remove the asterisk
servers IP from the phone number?
Also I'm unable to get the time zone correct on the phone. It is in UTC
and I'm in EST I see in the file where it looks like it goes but what I
have tried has not worked as of yet. Here is what it looks like
dateTimeSetting
dateTemplateM/D/Y/dateTemplate
timeZoneEST/timeZone
/dateTimeSetting
Thanks again for all your help!!!
___
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron
Yes it is very annoying!
Thanks for the date time settings. That worked GREAT!!!
Thanks
- Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, May 05, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having
the IP address on there, but it happens on ours too. It shouldn't cause
any problems with making outgoing calls from the directory, it's just
annoying to see it pop up.
As for the date time settings... this is what we have in ours:
dateTimeSetting uuid={9ec4850a-7748-11d3-bdf0-00108302ead1}
nameCMLocal/name
dateTemplateM/D/YA/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
I'm guessing you should be able to change it to say Eastern instead of
Central
On Fri, 5 May 2006, Hall, Eric M. wrote:
Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when searching Google to find that
post again. Can anyone help me out here. How can I remove the asterisk
servers IP from the phone number?
Also I'm unable to get the time zone correct on the phone. It is in
UTC and I'm in EST I see in the file where it looks like it goes but
what I have tried has not worked as of yet. Here is what it looks like
dateTimeSetting
dateTemplateM/D/Y/dateTemplate
timeZoneEST/timeZone
/dateTimeSetting
Thanks again for all your help!!!
___
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron
Any idea how to change it from 24hr to 12hr ?
Thanks again!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, May 05, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron
Yes it is very annoying!
Thanks for the date time settings. That worked GREAT!!!
Thanks
- Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, May 05, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having
the IP address on there, but it happens on ours too. It shouldn't cause
any problems with making outgoing calls from the directory, it's just
annoying to see it pop up.
As for the date time settings... this is what we have in ours:
dateTimeSetting uuid={9ec4850a-7748-11d3-bdf0-00108302ead1}
nameCMLocal/name
dateTemplateM/D/YA/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
I'm guessing you should be able to change it to say Eastern instead of
Central
On Fri, 5 May 2006, Hall, Eric M. wrote:
Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when searching Google to find that
post again. Can anyone help me out here. How can I remove the asterisk
servers IP from the phone number?
Also I'm unable to get the time zone correct on the phone. It is in
UTC and I'm in EST I see in the file where it looks like it goes but
what I have tried has not worked as of yet. Here is what it looks like
dateTimeSetting
dateTemplateM/D/Y/dateTemplate
timeZoneEST/timeZone
/dateTimeSetting
Thanks again for all your help!!!
___
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
The A at the end of the dateTemplate sets that.
Should read M/D/YA instead of M/D/Y.
Aaron
On Fri, 5 May 2006, Hall, Eric M. wrote:
Aaron
Any idea how to change it from 24hr to 12hr ?
Thanks again!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, May 05, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron
Yes it is very annoying!
Thanks for the date time settings. That worked GREAT!!!
Thanks
- Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, May 05, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having
the IP address on there, but it happens on ours too. It shouldn't cause
any problems with making outgoing calls from the directory, it's just
annoying to see it pop up.
As for the date time settings... this is what we have in ours:
dateTimeSetting uuid={9ec4850a-7748-11d3-bdf0-00108302ead1}
nameCMLocal/name
dateTemplateM/D/YA/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
I'm guessing you should be able to change it to say Eastern instead of
Central
On Fri, 5 May 2006, Hall, Eric M. wrote:
Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when searching Google to find that
post again. Can anyone help me out here. How can I remove the asterisk
servers IP from the phone number?
Also I'm unable to get the time zone correct on the phone. It is in
UTC and I'm in EST I see in the file where it looks like it goes but
what I have tried has not worked as of yet. Here is what it looks like
dateTimeSetting
dateTemplateM/D/Y/dateTemplate
timeZoneEST/timeZone
/dateTimeSetting
Thanks again for all your help!!!
___
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
___
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
___
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Re: [Asterisk-Users] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. It's so the phone routes the call to the correct server especially in a multiple server environment (ex: dialing a missed call) _ Mobilcom http://www.mobilcom.net ___ --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] Cisco 7970 SIP - few questions
Restarting the 7970 is like unlocking it twice, *-*-# to unlock, *-*-# to reboot. I don't believe hint functionality works on the SIP firmware for the 7970. Omar A. Sabek On 4/18/06, Tomislav Parčina [EMAIL PROTECTED] wrote: - How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is line button=4 featureID9/featureID ... For speeddial is line button=5 featureID2/featureID featureLabel341/featureLabel speedDialNumber341/speedDialNumber /line How to define hinting? - How to login true ssh? I have setup username and password, and when I try to log in it sends me challenge!?! login as: root [EMAIL PROTECTED]'s password: login: root challenge: YDXWGXMTpassword: Invalid Username/Password Entry. login: That is all, for now :)) -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 SIP - few questions
- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is line button=4 featureID9/featureID ... For speeddial is line button=5 featureID2/featureID featureLabel341/featureLabel speedDialNumber341/speedDialNumber /line How to define hinting? - How to login true ssh? I have setup username and password, and when I try to log in it sends me challenge!?! login as: root [EMAIL PROTECTED]'s password: login: root challenge: YDXWGXMTpassword: Invalid Username/Password Entry. login: That is all, for now :)) -- Tomislav Parcina tparcina#lama.hr ___ --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] Cisco 7970 SIP
I have upgrade Cisco 7970 on SIP using configuration file that was sent on the list. Now, phone tries to register on Asterisk but always fails. I have sniffed for packets with ethereal, and this is what I have found out. First, 7970 tries to register with *. * reply's that it's trying * reply's 401 - unauthorized 7970 tries again to register with * * reply's that it's trying * reply's 403 - forbidden I think that problem could be in way that 7970 is sending password. Can anybody help me on this? -- Tomislav Parcina tparcina#lama.hr ___ --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] Cisco 7970 SIP Config
Does anyone have a SEPMAC.cnf.xml file that works with asterisk? I have the SIP firmware loaded on my Cisco 7970 but the status log shows errors parsing the config. I copied a config that was posted to the list but it didn't seem to work. Any help would be appreciated. Jeremiah -- __ Rock River InternetJeremiah Millay 202 W. State St, 8th Floor [EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 Ext. 2202 USA fax 968-6888 ___ --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] Cisco 7970 SIP
Hi, Does anybody has a working SEPxx.cnf.xml SIP configuration for the Cisco 7970 with SIP 8-0-2 image Asterisk tanks Armand winmail.dat___ --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] Cisco 7970
Hello, can I use same settings and config files with Cisco IP Phone 7910 ? :)On 3/24/06, jason justman [EMAIL PROTECTED] wrote: Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/1.) setup your /etc/asterisk/sccp.conf with something like:[devices]type= 7970; device type (see below) autologin = 30,31,; lines list. You can add an empty line for anempty button (7960, 7970, 7940, 7920)description = jj7970; internal description. Notimportanttzoffset= -9transfer = on ; enable or disable the transfer capability. It does remove the transfer softkeypark = on ; take a look to thecompile howto. Park stuff is not compiled by defaultspeeddial = ; you can add an empty speedial if you want an empty button (7960, 7970, 7920)speeddial = *97,voicemail,cfwdall = off ; activate the callforward stuffand softkeyscfwdbusy = offdtmfmode = inband ; inband or outofband. outofband is the native cisco dtmf tone play.; Some phone model doesnot play dtmf tones while connected (bug?), so the default is inbandimageversion = P00405000700 ; useful to upgrade old firmwares (the ones that do not load *.xml from the tftp server)deny=0.0.0.0/0.0.0.0; Same as generalpermit= 192.168.1.90/255.255.255.255 ; This device can register onlyusing this ip addressdnd = on; turn on the dndsoftkey for this device. Valid values are off, on (busy signal), reject (busy signal), silent (ringer = silent)trustphoneip = no ; The phone has a ipaddress. It could be private, so if the phone is behind NAT; we don't have to trust the phone ip address, but the ip address of the connection;earlyrtp = none; valid options: none,offhook, dial, ringout. default is none.; The audio strem will be open in the progress and connected state.private = on; permit the private functionsoftkey for this devicemwilamp = on; Set the MWI lamp style when MWI active to on, off, wink, flash or blinkmwioncall = off ; Set the MWI on call.device = SEP00131A1F6366 ; device name SEPMAC[lines]id= 30; future use pin = 1234; future uselabel = 30; button line label (7960, 7970,7940, 7920)description = Line 30 ; top diplay descriptioncontext = from-internal ; sccp incominglimit = 2 ; more than 1 incomingcall = call waiting.transfer = on ; per line transfer capability.on, off, 1, 0mailbox = 30; voicemail.conf (syntax:[EMAIL PROTECTED]:folder])vmnum = *97 ; speeddial forvoicemail administration, just a number to dialcid_name = JJJ; caller id name cid_num = 30trnsfvm = 1000; extension to redirect thecaller (e.g for voicemail)secondary_dialtone_digits = 9 ; digits for the secondarydialtone (max 9 digits)secondary_dialtone_tone = 0x21; outside dialtone musicclass=default; Sets the default music on holdclasslanguage=en ; Default language setting;accountcode=79501; accountcode to ease billing rtptos = 184; sets the the rtp packets TOSfor this lineechocancel = on ; sets the phone echocancel forthis linesilencesuppression = off; sets the silence suppression for this line;callgroup=1,3-4; We are in callergroups 1,3,4. Valid for this line;pickupgroup=1,3-5; We can do call pick-p for callgroup 1,3,4,5. Valid for this line ;amaflags = ; Sets the default AMA flag codestored in the CDR record for this lineline = 30(do the same for line 31)2.)setup lines 30/31 as a custom extension in astersik (i used amp) and had it dial SCCP/30 and SCCP/31 as needed3.)setup /tftpboot config for SEPMAC.xmldevicexsi:type=axl:XIPPhonedevicePoolnameDefault/name dateTimeSettingnameCMLocal/namedateTemplatey-M-D/dateTemplatetimeZoneW. Europe Standard/Daylight Time/timeZone/dateTimeSettingcallManagerGroup membersmemberpriority=0callManagerportsethernetPhonePort2000/ethernetPhonePort/portsprocessNodeName(ASTERISK IP HERE)/processNodeName /callManager/member/members/callManagerGroupsrstInfonameEnable/namesrstOptionEnable/srstOptionuserModifiabletrue/userModifiable ipAddr1(ASTERISK IP HERE)/ipAddr1port12000/port1ipAddr2/ipAddr2port22000/port2ipAddr3/ipAddr3port32000/port3 /srstInfomlppDomainId-1/mlppDomainIdmlppIndicationStatusDefault/mlppIndicationStatuspreemptionDefault/preemption/devicePoolloadInformation/loadInformation vendorConfigdisableSpeakerfalse/disableSpeakerdisableSpeakerAndHeadsetfalse/disableSpeakerAndHeadsetforwardingDelay1/forwardingDelaypcPort0/pcPort settingsAccess1/settingsAccessgarp0/garpvoiceVlanAccess0/voiceVlanAccessvideoCapability1/videoCapabilityautoSelectLineEnable0/autoSelectLineEnable webAccess1/webAccessdaysDisplayNotActive1,7/daysDisplayNotActivedisplayOnTime08:30/displayOnTimedisplayOnDuration11:30/displayOnDurationdisplayIdleTimeout01:00/displayIdleTimeout /vendorConfigversionStamp/versionStampuserLocalename/nameuid1/uidlangCodeen/langCodeversion4.0(1)/version winCharSetiso-8859-1/winCharSet/userLocalenetworkLocale/networkLocalenetworkLocaleInfoname/nameuid64/uidversion 4.0(1)/version/networkLocaleInfodeviceSecurityMode1/deviceSecurityModeidleTimeout120/idleTimeoutauthenticationURL/authenticationURLdirectoryURL
[Asterisk-Users] Cisco 7970
I have search wiki, asteriskguru, chan_sccp and some other site's for information's how to upgrade, and make Cisco 7970 IP phone to work with asterisk on SCCP firmware. I'm sure that there are users on this group that have working Cisco 7970 phone. Please send me some information's how to do that. -- Tomislav Parcina tparcina#lama.hr ___ --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] Cisco 7970
Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/ 1.) setup your /etc/asterisk/sccp.conf with something like: [devices] type= 7970 ; device type (see below) autologin = 30,31, ; lines list. You can add an empty line for an empty button (7960, 7970, 7940, 7920) description = jj7970; internal description. Not important tzoffset = -9 transfer = on ; enable or disable the transfer capability. It does remove the transfer softkey park = on ; take a look to the compile howto. Park stuff is not compiled by default speeddial = ; you can add an empty speedial if you want an empty button (7960, 7970, 7920) speeddial = *97,voicemail, cfwdall = off ; activate the callforward stuff and softkeys cfwdbusy = off dtmfmode = inband ; inband or outofband. outofband is the native cisco dtmf tone play. ; Some phone model does not play dtmf tones while connected (bug?), so the default is inband imageversion = P00405000700 ; useful to upgrade old firmwares (the ones that do not load *.xml from the tftp server) deny=0.0.0.0/0.0.0.0; Same as general permit=192.168.1.90/255.255.255.255 ; This device can register only using this ip address dnd = on; turn on the dnd softkey for this device. Valid values are off, on (busy signal), reject (busy signal), silent (ringer = silent) trustphoneip = no ; The phone has a ip address. It could be private, so if the phone is behind NAT ; we don't have to trust the phone ip address, but the ip address of the connection ;earlyrtp = none; valid options: none, offhook, dial, ringout. default is none. ; The audio strem will be open in the progress and connected state. private = on; permit the private function softkey for this device mwilamp = on; Set the MWI lamp style when MWI active to on, off, wink, flash or blink mwioncall = off ; Set the MWI on call. device = SEP00131A1F6366 ; device name SEPMAC [lines] id = 30; future use pin = 1234 ; future use label = 30; button line label (7960, 7970, 7940, 7920) description = Line 30 ; top diplay description context = from-internal ; sccp incominglimit = 2 ; more than 1 incoming call = call waiting. transfer = on ; per line transfer capability. on, off, 1, 0 mailbox = 30; voicemail.conf (syntax: [EMAIL PROTECTED]:folder]) vmnum = *97 ; speeddial for voicemail administration, just a number to dial cid_name = JJJ ; caller id name cid_num = 30 trnsfvm = 1000 ; extension to redirect the caller (e.g for voicemail) secondary_dialtone_digits = 9 ; digits for the secondary dialtone (max 9 digits) secondary_dialtone_tone = 0x21 ; outside dialtone musicclass=default ; Sets the default music on hold class language=en ; Default language setting ;accountcode=79501 ; accountcode to ease billing rtptos = 184; sets the the rtp packets TOS for this line echocancel = on ; sets the phone echocancel for this line silencesuppression = off; sets the silence suppression for this line ;callgroup=1,3-4; We are in caller groups 1,3,4. Valid for this line ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5. Valid for this line ;amaflags = ; Sets the default AMA flag code stored in the CDR record for this line line = 30 (do the same for line 31) 2.) setup lines 30/31 as a custom extension in astersik (i used amp) and had it dial SCCP/30 and SCCP/31 as needed 3.) setup /tftpboot config for SEPMAC.xml device xsi:type=axl:XIPPhone devicePool nameDefault/name dateTimeSetting nameCMLocal/name dateTemplatey-M-D/dateTemplate timeZoneW. Europe Standard/Daylight Time/timeZone /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort /ports processNodeName(ASTERISK IP HERE)/processNodeName /callManager /member /members /callManagerGroup srstInfo nameEnable/name srstOptionEnable/srstOption userModifiabletrue/userModifiable ipAddr1(ASTERISK IP HERE)/ipAddr1 port12000/port1 ipAddr2/ipAddr2
[Asterisk-Users] Cisco 7970 SIP Image - hint lines
Hello I patche dmy 7970 with the current SIP image i have 2 lines on it via sip and 6 hint speeddials but it seems thats only a speeddial no infos about busy status or so comes to the speddial button. somebody can help me? ___ --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] Cisco 7970 SIP Image - hint lines
On Thu, 23 Mar 2006, René Enskat [Teamware GmbH] wrote: I patche dmy 7970 with the current SIP image i have 2 lines on it via sip and 6 hint speeddials but it seems thats only a speeddial no infos about busy status or so comes to the speddial button. somebody can help me? cisco 7970 sip images explicitly do not support hints. only speeddial is supported. it's sorta like cisco is punishing us for choosing to use sip over sccp. -Dan___ --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] Cisco 7970 SIP Image
Hi, I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-) Any pointers would be appreciated ___ --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] Cisco 7970 SIP Image
On Wed, 2006-03-22 at 11:52 +0100, Paul Brown wrote: Hi, I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-) Any pointers would be appreciated http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/ip-7900ser/cmterm-7970_7971-sip.8-0-2-0.copapp=Tablebuildstatus=showC2A ___ --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] Cisco 7970 SIP Image
It's in the NON-SIP section of the site, you'll find it on the page somewhere under the 7970 SCCP images... They're harping that this release is for their new CCM, so although it's SIP, it kinda sucks. Aaron On Wed, 22 Mar 2006, Paul Brown wrote: Hi, I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-) Any pointers would be appreciated ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 problems
Hi Guys cant seem to find a more relevant place to ask this, and since it is slightly Asterisk-related, I figured Id ask here. I was tinkering with my Cisco 7970 and getting the chan_sccp setup to run on my Asterisk box. Things were working fine, sort of I went home for the night, came back, and the 7970 had suddenly lost its firmware. I dont have Call Manager. When the phone boots, it just goes to the TFTP server looking for TERM70.5-0-0-6S.loads and dies. Am I poop out of luck with this phone? Anyone have any suggestions? I do have access to some versions of Cisco software, but it looks like they dont have a plain ZIP file of just the firmware files for 5-0-0-6S any help would be greatly appreciated! Andy -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 267.14.22/238 - Release Date: 1/23/2006 ___ --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] Cisco 7970
Thank you Kerry. I was able to download the firmware. Does anybody know what files need to reside on the tfpt server. If someone is willing to help get my 7970 phone functional again, I would really appreciate it. -John You have to have a login to the Cisco site to download the firmware. -Kerry -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Riek Sent: Tuesday, November 29, 2005 2:02 PM To: asterisk user list Subject: [Asterisk-Users] Cisco 7970 I have the same problem after doing a factory reset. Does anybody have the website link to download firmware for the Cisco phones? Thanks, John Riek I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on your server you can see what file its getting stuck on. This is how I figured out what it is looking for: tcpdump -i eth1 port tftp -vv It will output what file the phone is looking for. Have my 7970 working great with *. Hope this helps. Jeremiah On Nov 7, 2005, at 10:24 AM, asterisk-users-request at lists.digium.com wrote: Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine dan at cytexone.com 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --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] Cisco 7970
I have the same problem after doing a factory reset. Does anybody have the website link to download firmware for the Cisco phones? Thanks, John Riek I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on your server you can see what file its getting stuck on. This is how I figured out what it is looking for: tcpdump -i eth1 port tftp -vv It will output what file the phone is looking for. Have my 7970 working great with *. Hope this helps. Jeremiah On Nov 7, 2005, at 10:24 AM, asterisk-users-request at lists.digium.com wrote: Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine dan at cytexone.com 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7970
You have to have a login to the Cisco site to download the firmware. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Riek Sent: Tuesday, November 29, 2005 2:02 PM To: asterisk user list Subject: [Asterisk-Users] Cisco 7970 I have the same problem after doing a factory reset. Does anybody have the website link to download firmware for the Cisco phones? Thanks, John Riek I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on your server you can see what file its getting stuck on. This is how I figured out what it is looking for: tcpdump -i eth1 port tftp -vv It will output what file the phone is looking for. Have my 7970 working great with *. Hope this helps. Jeremiah On Nov 7, 2005, at 10:24 AM, asterisk-users-request at lists.digium.com wrote: Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine dan at cytexone.com 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7970
Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine [EMAIL PROTECTED] 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
